Table Of Contents
New Hardware Features Supported in Cisco IOS Release 12.2(15)T
1 Port Enhanced ATM Port Adapter with Support for 8K VCs
1 and 2-port T1/E1 Multiflex Voice/WAN Interface Card
1- and 2-Port V.90 Modem WICs for Cisco 1720, 1751 and 1760 Routers
Catalyst 4500 Access Gateway Module 16-port RJ21 FXS Module (WS-U4604-16FXS)
Catalyst 4500 AGM Voice/WAN Bundle (WS-X4604-VOICE)
Gigabit Ethernet Network Module
SDH/STM-1 Trunk Card for Cisco AS5850 Universal Gateway
New Software Features in Cisco IOS Release 12.2(15)T
Any Transport over MPLS (AToM)
Asynchronous Call Queueing by Role
BGP Increased Support of Numbered AS-Path Access Lists to 500
BGP Nonstop Forwarding (NSF) Awareness
BGP Restart Session After Max-Prefix Limit
BGP Route-Map Policy List Support
Certificate Security Attribute-Based Access Control
Cisco Easy VPN Remote Enhancements
Cisco IOS Firewall Stateful Inspection of ICMP
Cisco IOS Firewall Support for SIP
Cisco IOS Firewall Websense URL Filtering
Cisco IOS Software Feature Removal—Phase II
Cisco IOS Telephony Service Version 2.1
Cisco Mobile Networks—Priority Home Agent Assignment
Cisco Mobile Networks—Static Collocated Care-of Address
Cisco Mobile Networks—Tunnel Templates for Multicast
Cisco Survivable Remote Site Telephony Version 2.1
Class-Based Policer for the DiffServ AF PHB
Clear Channel T3/E3 with Integrated CSU/DSU
DHCP Secured IP Address Assignment
DHCP Server Import All Enhancement
DHCP Server—ODAP Support for Non-MPLS VPN Pools
EIGRP Nonstop Forwarding (NSF) Awareness
Enhanced Debug Capabilities for Cisco Voice Gateways
Expanded Scope for Cause-Code-Initiated Call Establishment Retries
Exporting and Importing RSA Keys
Fax and Modem Pass-Through over VoIP
Firewall Intrusion Detection System Signature Enhancements
Firewall Support of HTTPS Authentication Proxy
Frame Relay Voice-Adaptive Traffic Shaping
G.732 Support for the Integrated Signaling Link Terminal
Gatekeeper Management Statistics
GLBP: Gateway Load Balancing Protocol
H.323v4 Gateway Zone Prefix Registration Enhancements
Integrated IS-IS Multi-Topology Support for IPv6
Integrated IS-IS Nonstop Forwarding (NSF) Awareness
Integrated Voice and Data WAN on T1/E1 Interfaces Using the AIM-ATM-VOICE-30 Module
IP Access List Entry Sequence Numbering
IPv6 Provider Edge Router over MPLS
ISDN Generic Transparency Descriptor (GTD) for Setup Message
ISDN Progress Indicator Support for SIP Using 183 Session Progress
L2TP Dial-Out Load Balancing and Redundancy
L2TP Large-Scale Dial-Out per-User Attribute via AAA
Malicious Caller Identification (MCID) Invocation Support for Enterprise Networks
Measurement-Based Call Admission Control for SIP
MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
MGCP Based Fax (T.38) and DTMF Relay
MGCP Basic CLASS and Operator Services
MGCP VoIP Call Admission Control
Mobile IP—Home Agent Accounting
MPLS VPN Support for EIGRP Between Provider Edge (PE) and Customer Edge (CE)
Multicast Subsecond Convergence
Multiple OPC Support for the Cisco Signaling Link Terminal
NAT Support for IPSec ESP—Phase II
Network-Based Application Recognition Protocol Discovery Management Information Base
OSPF Forwarding Address Suppression in Translated Type-5 LSAs
OSPF Inbound Filtering Using Route Maps with a Distribute List
OSPF Nonstop Forwarding (NSF) Awareness
OSPF Shortest Path First Throttling
OSPF Support for Fast Hello Packets
Per-User QoS via AAA Policy Name
RADIUS Support of 56-Bit Acct Session-Id
RADIUS Timeout Set During Pre-Authentication
RSVP Support for RTP Header Compression, Phase 1
SIP Call Transfer and Call Forwarding Supplementary Services
SIP—Configurable PSTN Cause Code Mapping
SIP Diversion Header Implementation for Redirecting Number
SIP—DNS SRV RFC2782 Compliance
SIP Gateway Support for Third Party Call Control
SIP Gateway Support of RSVP and TEL URL
SIP INVITE Request with Malformed Via Header
SIP: ISDN Suspend/Resume Support
SIP—Session Initiation Protocol for VoIP Enhancements
Source Interface Selection for Outgoing Traffic with Certificate Authority
Support for Bridged RFC 1483 Encapsulated Traffic over ATM SVCs
Support for IUA with SCTP for Cisco Access Servers
T1 Channel Associated Signaling (CAS)
Tunneled GR-303 for the Cisco Cable Modem
UDP Forwarding Support of IP Redundancy Virtual Router Group (VRG)
V.92 and V.44 Support for Digital Modems
XML Interface to Syslog Messages
New Hardware Features Supported in Cisco IOS Release 12.2(13)T
Catalyst 4224 Access Gateway Switch
Cisco 3631 Router Enhanced Functionality
Cisco 3725 Router, Cisco 3745 Router, Cisco 2691 Router Enhanced Functionality
Cisco 7401 ASR-BB and Cisco 7401 ASR-CP
Content Engine Network Module for Caching and Content Delivery
Unchannelized support for PA-MC-2T3+ port adapter
Update to the Enhancements for the Cisco Voice Gateway 200
New Software Features in Cisco IOS Release 12.2(13)T
Advanced Encryption Standard (AES)
Analog DID (Direct Inward Dial)
ATM Multilink PPP Support on Multiple VCs
ATM Policing by Service Category for SVC/SoftPVC
Automatic Protection Switching (APS)
BGP 4 MIB Support for per-Peer Received Routes
Bisync-to-IP Conversion for Automated Teller Machines
Call Admission Control for H.323 VoIP Gateways
Call Release Source Reporting in Gateway-Generated Call Accounting Records
CEF and Distributed CEF Switching for IPv6
Cisco Conferencing and Transcoding for Voice Gateway Routers
Cisco IOS Software Feature Removal
Cisco IOS Telephony Service (ITS) Version 2.02
Cisco Mobile Networks—Asymmetric Link
Cisco Mobile Networks—Dynamic Network Support
Cisco Survivable Remote Site Telephony Service V2.02
Class-Based RTP and TCP Header Compression
Connection-Oriented Media (Comedia) Enhancements for SIP
Dial-Peer Support for Data Calls
Distributed IPv6 for Cisco IOS software
DLR Enhancements: PGM RFC-3208 Compliance
Dual Serial Line Management to Interface Lucent 5ESS
Dynamic Multipoint VPN (DMVPN)
Dynamic Subscriber Bandwidth Selection
Enhanced Features for Local and Advanced Voice Busyout
Enhanced ITU-T G.168 Echo Cancellation
Enhancements for the Cisco Voice Gateway 200
Exterior Gateway Protocol (EGP)
File System Check and Repair for PCMCIA ATA Disks
Frame Relay PVC Bundles with IP and MPLS QoS Support
Frame Relay Queueing and Fragmentation at the Interface
H.323 Call Redirection Enhancements
H.323 Dual Tone Multifrequency Relay Using Named Telephone Events
H.323 Scalability and Interoperability Enhancements
H.323 Support for Virtual Interfaces
Interim Local Management Interface (ILMI)
Interim-Interswitch Signaling Protocol (IISP)
Interim Update at Call Connect
Interior Gateway Routing Protocol (IGRP)
IPv6 ADSL and Dial Deployment Support
IPv6 Extended Access Control Lists
Low Latency Queueing (LLQ) for IPSec
LZ Software with Hardware Encryption
Manual Certificate Enrollment (TFTP and Cut-and-Paste)
MGCP 1.0 and TGCP 1.0 Profiles
MGCP Gateway Support for the Bind Command
Mobile IP—Challenge/Response Extensions
Mobile IP—Fastswitching Support on Foreign Agent
Mobile IP—Generic NAI Support and Home Address Allocation
Mobile IP Home Agent Policy Routing
Mobile IP —IPsec for Home Agent to Foreign Agent Tunnel
Mobile IP—MIB Support for NAI and HA Redundancy
Mobile IP—Private Addressing Support
Mobile IP—Support for FA Reverse Tunneling
Modular QoS CLI (MQC)-Based Frame-Relay Traffic Shaping
Modular QoS CLI (MQC) Three-Level Hierarchical Policer
Modular QoS CLI (MQC) Unconditional Packet Discard
MPLS Label Distribution Protocol (LDP) MIB
MPLS VPN—Carrier Supporting Carrier—IPv4 BGP Label Distribution
MPLS VPN—Inter-AS—IPv4 BGP Label Distribution
Multicast-VPN—IP Multicast Support for MPLS VPNs
NAT Integration with MPLS VPNs
NAT Stateful Failover of Network Address Translation
NAT-Support of H.323 v2 Call Signaling
NetWare Link Services Protocol (NLSP)
Next Hop Resolution Protocol (NHRP) for IPX
OSPF Support for Multi-VRF on CE Routers
Packet Classification Based on Layer 3 Packet Length
Packet Classification Using the Frame Relay DLCI Number
Percentage-Based Policing and Shaping
RADIUS Attribute 52 and Attribute 53 Gigaword Support
RADIUS Centralized Filter Management
RSVP Refresh Reduction and Reliable Messaging
Show Command Output Redirection
Simple Multicast Routing Protocol (SMRP) for AppleTalk
SIP and H.323 Fax Enhancements
SIP—Call Transfer Enhancements Using the Refer Method
SIP Enhanced 180 Provisional Response Handling
SIP Extensions for Caller Identity and Privacy
SIP Gateway Compliance to RFC2543-bis-04
SIP Redirect Processing Enhancements
Subscriber Service Switch (SSS)
Support for IPsec ESP Through NAT
Terminal Line Security for PAD Connections
Update to the Interworking of Cisco MGCP Voice Gateways and Cisco CallManager Version 3.2 Feature
Update to the playout-delay Command
Virtual Router Redundancy Protocol (VRRP)
X.25 Suppression of Security Signaling Facilities
New Hardware Features Supported in Cisco IOS Release 12.2(11)T1
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(11)T
16-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series
Cisco AS5350 Universal Gateway
Cisco AS5850 Universal Gateway
Cisco Signaling Link Terminal (SLT) Dual Ethernet
New Software Features in Cisco IOS Release 12.2(11)T
Accounting of VPDN Disconnect Cause
ACL Authentication of Incoming rsh and rcp Requests
Analog Centralized Automatic Message Accounting E911 Trunk
ATM Service Level Monitoring (SLM)
Barge-In and Busy Line Verify Operator Services
Basic Service Relationships (H.225 Annex G)
BGP Conditional Route Injection
BGP Hide Local-Autonomous System
BGP Multipath Load Sharing for Both eBGP and iBGP in an MPLS-VPN
BGP Prefix-Based Outbound Route Filtering
Call Admission Control Based on CPU Utilization
Call Admission Control for H.323 VoIP Gateways
Call Status Tracking Optimization
Call Tracker show Commands Extensions
Certificate Enrollment Enhancements
Circuit Interface Identification Persistence for SNMP
Cisco Gateway Management Agent
Cisco H.323 Multizone Enhancements
Cisco IOS Telephony Service Version 2.0
CISCO-BULK-FILE-MIB Enhancements
CISCO-SIP-UA-MIB Enhancements Providing Functional Parity to SIP related CLI
Configuring a Gatekeeper to Provide Nonavailability Information for Terminating Endpoints
Connect-Info RADIUS Attribute 77
Customer Profile Idle Timer Enhancements for Interesting Traffic
DF Bit Override Functionality with IPSec Tunnels
DHCP Client—Dynamic Subnet Allocation API
DHCP Relay Agent Support for Unnumbered Interfaces
DHCP Server—On-Demand Address Pool Manager
DHCP Server—Option to Ignore All BOOTP Requests
Distributed Management Event and Expression MIB Persistence
Distributed Management Event MIB Conformance to RFC 2981
DTMF Events Through SIP Signaling
DTMF Relay for SIP calls Using Named Telephone Events
Enable Multilink PPP via RADIUS for Preauthentication User
Encrypted Vendor-Specific Attributes
Enhanced Codec Support for SIP Using Dynamic Payloads
Enhanced Debug Capabilities for Cisco Voice Gateways
Enhancements for the Cisco VG200 Voice Gateway
Enhancing Raw Buffer Management: Audit and Prepopulation for Channel-Associated Signaling
Fax and Modem Pass-Through over VoIP
Fax Detection (Single-number Voice and Fax)
Fax Relay Packet Loss Concealment
G.Clear, GSMFR, and G.726 Codecs and Modem and Fax Pass-Through for Cisco Universal Gateways
Gatekeeper Endpoint Control Enhancements
Gatekeeper-to-Gatekeeper Authentication
Generic Routing Encapsulation (GRE) Tunnel Keepalive
Globalized Cadence and Tone for Cisco IOS Gateways
GTD for GKTMP using SS7 Interconnect version 2.0
H.323 Call Redirection Enhancements
H.323 Dual Tone Multifrequency (DTMF) Relay Using Named Telephone Events
IGMP MIB Support Enhancements for SNMP
Integrated Signaling Link Terminal
Integrated IS-IS Point-to-Point Adjacency Over Broadcast Media
Interactive Voice Response Version 2.0 on VoIP Gateways
Inter-Domain Gatekeeper Security Enhancement
Interface Alias Long Name Support
Internal Cause Code Consistency Between SIP and H.323
Interworking of Cisco MGCP Voice Gateways and Cisco CallManager Version 3.2
Interworking Signaling Enhancements for H.323 and SIP VoIP
ip dhcp-client default-router distance value Command
IPSec VPN High Availability Enhancements
ISDN and V.120 Support for NextPort DSPs
ISDN-NFAS with D Channel Backup
IVR: Configuring Dynamic Prompts
IVR: Customizing Accounting Templates
IVR: Enhanced Multilanguage Support
Location Confirmation (LCF) Enhancements for Alternate Endpoints
Low Latency Queueing with Priority Percentage Support
Media Gateway Control Protocol-Based Fax (T.38) and Dual Tone Multifrequency (IETF RFC 2833) Relay
MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
MGCP Basic CLASS and Operator Services
MGCP Generic Configuration Support for Call Manager (IP-PBX)
MGCP Line Package Enhancements for Loop Current Feed Open (LCFO)
MGCP PRI Backhaul and T1-CAS Support for Call Manager (IP-PBX)
MGCP Voice on Cisco AS5850 Universal Gateway
MGCP VoIP Call Admission Control
Modem Relay Support on VoIP Platforms
Modem Script and System Script Support in Large-Scale Dial-Out
Monitoring Voice and Fax Services on the Cisco AS5350 and Cisco AS5400 Universal Gateways
MPLS Label Distribution Protocol (LDP)
Multicast Music on Hold Support for Call Manager (IP-PBX)
NetFlow Multiple Export Destinations
NetFlow ToS-Based Router Aggregation
Network Access Server (NAS) Package for MGCP
Network Side ISDN PRI Signaling, Trunking, and Switching
Nonblocking Gatekeeper AAA Interface
OSPF Sham-Link Support for MPLS VPN
OSPF Stub Router Advertisement
OSPF Update Packet-Pacing Configurable Timers
PIAFS Wireless Data Protocol Version 2.1 for Cisco MICA Modems
PIM MIB Extension for IP Multicast
Preauthentication with ISDN PRI and Channel-Associated Signalling Enhancements
PRI Backhaul Using the Stream Control Transmission Protocol and the ISDN Q.921 User Adaptation Layer
PRI/Q.931 Signaling Backhaul for Call Agent Applications
R2 and ISUP Transparency and R2-to-ISUP Interworking Enhancements
RADIUS Attribute 66 (Tunnel-Client-Endpoint) Enhancements
RADIUS Attribute 82: Tunnel Assignment ID
RADIUS Attribute Value Screening
RADIUS Number Translation VSAs for VoIP
RADIUS Packet Suppression for VoIP GW Rotary Dial-Peer Attempts
RADIUS Preauthentication for H.323 and SIP Voice Calls
RADIUS Tunnel Preference for Load Balancing and Fail-over
Reverse Path Forwarding - Source Exists Only
Route Switch Controller (RSC) Handover Redundancy
Router-Shelf Redundancy for the AS5800 Series
SGCP RSIP and AUEP Enhancement
Shell-Based Authentication of VPDN Users
SIP—Call Transfer Using Refer Method
SIP Carrier Identification Code
SIP—Configurable PSTN Cause Code Mapping
SIP—DNS SRV RFC2782 Compliance
SIP Diversion Header Implementation for Redirecting Number
SIP—Enhanced Billing Support for Gateways
SIP Gateway Support for the Bind Command
SIP Gateway Support for Third Party Call Control
SIP Gateway Support of RSVP and TEL URL
SIP INFO Method for DTMF Tone Generation
SIP INVITE Request with Malformed Via Header
SIP—Session Initiation Protocol for VoIP
SIP—Session Initiation Protocol for VoIP Enhancements
SIP T.37 Store and Forward Fax
Speech Recognition and Synthesis for Voice Applications
Static Cache Entry for IPv6 Neighbor Discovery
Survivable Remote Site Telephony Version 2.0
TCL IVR 2.0 Call Initiation and Callback
TCL IVR Disconnect Cause-Code Manipulation
TCL-Enabled Signaling Parameter Mapping
Timer and Retry Enhancements for L2TP and L2F
Universal Port Resource Pooling for Voice and Data Services
V.44 LZJH Compression for Cisco AS5300 and Cisco AS5800 Universal Access Servers
V.92 Modem on Hold for Cisco AS5300 and Cisco AS5800 Universal Access Servers
V.92 Quick Connect for Cisco AS5300 and Cisco AS5800 Universal Access Servers
VoAAL2 Profile 9 Support for Broadband Loop Emulation Services Specification Interoperability
Voice Application Access To SS7 Signaling
Voice DSP Control Message Logger
Voice over IP Q.SIG Network Transparency
VoiceXML SS7 ISUP Session Variables
VoiceXML Media Volume and Rate Controls
VoiceXML Transfer Enhancements
VoiceXML Voice Store and Forward
VoIP Call Admission Control using RSVP
VoIP Interoperability with Cisco Express Forwarding and Policy Based Routing
VoIP Gatekeeper Trunk and Carrier Based Routing Enhancements
VoIP Gateway Trunk and Carrier Based Routing Enhancements
VoIP Outgoing Trunk Group Identification and Carrier ID for Gateways
VPN Routing Forwarding (VRF) Framed Route (Pool) Assignment via PPP
WRED Enhancement—Explicit Congestion Notification (ECN)
X.25 Record Boundary Preservation for Data Communications Networks
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(8)T1
36-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series
New Software Features in Cisco IOS Release 12.2(8)T1
MPLS Label Switch Controller and Enhancements
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(8)T
1- and 2-Port V.90 Modem WICs for Cisco 2600 and 3600 Series
16-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series
AIM-ATM, AIM-VOICE-30, and AIM-ATM-VOICE-30 on the Cisco 2600 Series and Cisco 3660
Analog Station Interface (ASI) Cards
Cisco 806 Broadband Gateway Router
Cisco 3725 Application Service Router
Cisco 3745 Application Service Router
Cisco High-Density Analog Voice and Fax Network Module
Cisco IOS Voice Features on IGX 8400 Series Universal Router Module
Digital J1 Voice Interface Card
Multichannel STM-1 Port Adapter
NM-AIC-64, Contact Closure Network Module
New Software Features in Cisco IOS Release 12.2(8)T
ACL Authentication of Incoming rsh and rcp Requests
Asynchronous Serial Traffic Over User Datagram Protocol (UDP)
ATM PVC Bundle Enhancement—MPLS EXP-Based PVC Selection
ATM Software Segmentation and Reassembly (SAR)
ATM SVC Troubleshooting Enhancements
BGP Hide Local-Autonomous System
BIP—BSC to IP Conversion for Automated Teller Machines
Call Admission Control for H.323 VoIP Gateways
CDP and ODR Support for ATM PVCs
Cisco Discovery Protocol (CDP)— IPv6 Address Family Support for Neighbor Information
CEF-Switched Multipoint GRE Tunnels
Certificate Enrollment Enhancements
CISCO-BULK-FILE-MIB Enhancements
Cisco Gateway Management Agent (CGMA) Phase 2
Cisco IOS Firewall Performance Improvements
Cisco IOS Telephony Service Version 2.0
Cisco Service Assurance Agent Support for the Cisco 820 Series and SOHO 70 Series
Class-Based Weighted Fair Queueing (CBWFQ)
Configurable PSTN Cause Code to SIP Response Mapping
DHCP Client—Dynamic Subnet Allocation API
DHCP Server—On-Demand Address Pool Manager
DHCP Server—Option to Ignore All BOOTP Requests
DHCP Server Options Import and Autoconfiguration
Dialer Map VRF-Aware for an MPLS VPN
Diff-Serv-aware MPLS Traffic Engineering
DistributedDirector Boomerang Support
DistributedDirector Cache Auto Refresh
DistributedDirector Configurable Cache
DistributedDirector MIB Support
Distributed LFI/dQoS over Leased Lines
Distributed Multilink Point-to-Point Protocol
DNS Client AAAA Record Lookups over IPv6
Dual Tone Multifrequency (DTMF) Relay for SIP Calls Using Named Telephone Events
Enabling Fax Rate on POTS to POTS Fax Calls
Encrypted Vendor-Specific Attributes
Enhanced Billing Support for SIP Gateways
Fax Detection for Cisco 2600 Series and Cisco 3600 Series Routers
Gatekeeper Transaction Message Protocol Interface Resiliency Enhancement
Generic Routing Encapsulation (GRE) Tunnel Keepalive
GKTMP Security Token Enhancement
IGMP Version 3—Explicit Tracking of Hosts, Groups, and Channels
Integrated IS-IS Point-to-Point Adjacency Over Broadcast Media
Integrated IS-IS Support for IPv6
Interactive Voice Response Version 2.0 on VoIP Gateways
IPSec VPN High Availability Enhancements
Large-Scale Dial-Out (LSDO) VRF Aware
Media Gateway Control Protocol Based Fax (T.38) and Dual Tone Multifrequency (IETF RFC 2833) Relay
MGCP VoIP Call Admission Control
MPLS Label Distribution Protocol (LDP)
MPLS Over ATM: Virtual Circuit (VC) Merge
MPLS Traffic Engineering (TE) MIB
MPLS VPN Carrier Supporting Carrier
Multiprotocol BGP (MP-BGP) Support for CLNS
Network-Based Application Recognition RTP Payload Type Classification
Nonstop Forwarding Enhanced FIB Refresh
OSPF Sham-Link Support for MPLS VPN
Policer Enhancement—Multiple Actions
Secure Shell (SSH) Support over IPv6
Secure Shell (SSH) Version 1 Server Support
Session Initiation Protocol (SIP) for VoIP
Simple Network-Enabled Auto-Provisioning for Cisco IAD2420 Series IADs
SIP Gateway Support for the Bind Command
SIP Gateway Support of RSVP and TEL URL
SIP INVITE Request with Malformed Via Header
SIP T.37 Store and Forward Fax
SIP—Call Transfer Using Refer Method
SIP—DNS SRV RFC2782 Compliance
SNMP IF-MIB Support for VLAN (ISL, 802.1Q) Subinterfaces
Static Cache Entry for IPv6 Neighbor Discovery
Stream Control Transmission Protocol (SCTP) Release 2
Survivable Remote Site Telephony Version 1.0
Survivable Remote Site Telephony Version 2.0
T.37 Store-and-Forward Fax for Cisco 1751 Modular Access Routers
T.37 Store-and-Forward Fax for the Cisco 2600 Series and Cisco 3600 Series Routers
Unspecified Bit Rate Plus (uBR+) and ATM Enhancements for Service Provider Integrated Access
Update to the MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
VoAAL2 Profile 9 Support for BLES Interoperability
Voice Support for Japan on Cisco 800 Series Routers, Phase 2
VPN Routing Forwarding (VRF) Framed Route (Pool) Assignment via PPP
WRED Enhancement—Explicit Congestion Notification (ECN)
X.25 Record Boundary Preservation for Data Communications Networks
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(4)T
1-Port ADSL WAN Interface Card
1-Port T1/E1 Digital Voice Port Adapters for Cisco 7200 and Cisco 7500
8-Port Mix-Enabled T1/E1/PRI PA
Cisco uBR925 Cable Access Router
Cisco CVA122 Cable Voice Adapter
Cisco CVA122E Cable Voice Adapter
New Software Features in Cisco IOS Release 12.2(4)T
Ability to Disable Xauth for Static IPSec Peers
Accounting of VPDN Disconnect Cause
Adaptive Frame Relay Traffic Shaping for Interface Congestion
ATM SNMP Trap and OAM Enhancements
AutoInstall over Frame Relay-ATM Interworking Connections
Automatic Bandwidth Adjustment for MPLS Traffic Engineering Tunnels
BGP Conditional Route Injection
BGP Multipath Load Sharing for Both eBGP and iBGP in an MPLS-VPN
BGP Prefix-Based Outbound Route Filtering
Call Admission Control for H.323 VoIP Gateways
Circuit Interface Identification Persistence for SNMP
Cisco H.323 Scalability and Interoperability Enhancements
Crashinfo Support for Cisco 3600 Series
DFP Support in DistributedDirector
Diff-Serv-aware Traffic Engineering
Distinguished Name Based Crypto Maps
Distributed Link Fragmentation and Interleaving
DistributedDirector Enhancements
Distributed Management Event and Expression MIB Persistence
DNS Server Support for NS Records
Enhancements to H.323 Call Statistics
Four SS7 Link Support on the Cisco Signaling Link Terminal
ICMP ECHO-Based RTT Probing by DRP Agents
IGMP MIB Support Enhancements for SNMP
Inter-Domain Gatekeeper Security Enhancement
Interesting Traffic PPP and Customer Profile Idle Timer
IP to ATM Class of Service Mapping for SVC Bundles
IPSec MIB Support for VPN Management
ISIS: Allows BGP to Control the Configuration of the Overload Bit
Leased and Switched BRI Interfaces for ETSI NET3
Location Confirmation Enhancements for Alternate Endpoints
MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
MGCP Voice Gateway Interoperability with Cisco CallManager
Mobile IP MIB Support for SNMP
MPLS Label Switch Controller and Enhancements
MPLS Traffic Engineering (TE)—Automatic Bandwidth Adjustment for TE Tunnels
MPLS Traffic Engineering (TE)—IP Explicit Address Exclusion
Multiservice Interchange (MIX) Support
NAT—Ability to Use Route Maps with Static Translations
NAT—Static Mapping Support with HSRP for High Availability
NAT—Translation of External IP Addresses Only
NetFlow Multiple Export Destinations
NetFlow ToS-Based Router Aggregation
Offload Server Accounting Enhancement
OSPF Stub Router Advertisement
OSPF Update Packet-Pacing Configurable Timers
PIM MIB Extension for IP Multicast
PPPoA/PPPoE Autosense for ATM PVCs
PRI Backhaul Using the Stream Control Transmission Protocol and the ISDN Q.921 User Adaptation Layer
PRI/Q.931 Signaling Backhaul for Call Agent Applications
PSTN Fallback for Cisco 7200 and 7500 Series Routers
RADIUS Attribute 82: Tunnel Assignment ID
RADIUS Tunnel Preference for Load Balancing and Fail-Over
RSVP Support for Low Latency Queueing
SS7 Four-Link Support for Cisco Signaling Link Terminal
Stream Control Transmission Protocol (SCTP), Release 1
T.38 Fax Services for Cisco 1750 Access Routers
Timer and Retry Enhancements for L2TP and L2F
Using 31-Bit Prefixes on IPv4 Point-to-Point Links
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(2)T
1-Port ADSL WAN Interface Card
Cisco uBR905 Cable Access Router
Small Office, Home Office ADSL Router
WT-2750 Multipoint Broadband Wireless System
New Software Features in Cisco IOS Release 12.2(2)T
56K CSU Support for the Cisco Signaling Link Terminal
Analog DID for Cisco 2600 and Cisco 3600 Series Routers
ATM PVC Range and Routed Bridge Encapsulation Subinterface Grouping
Circuit Interface Identification Persistence for SNMP
Cisco High-Performance Gatekeeper
Cisco IOS Server Load Balancing
Cisco Signaling Link Terminal G.732 Support
Cisco Quality of Service Device Manager 2.0 Support for Cisco 1700 Series Routers
Classifying VoIP Signaling and Media with DSCP for QoS
DF Bit Override Functionality with IPSec Tunnels
DFP Support in DistributedDirector
DHCP Option 82 Support for Routed Bridge Encapsulation
Distributed Time-Based Access Lists
DNS Server Support for NS Records
Enhanced Multilingual Support for Cisco IOS Integrated Voice Response
Firewall Feature Set for Cisco 820 Series Routers
Frame Relay Discard Eligibility Bit Setting
Frame Relay Point-Multipoint Wireless
Functionality Changed for the tunnel mpls traffic-eng autoroute metric Command
FXO Answer and Disconnect Supervision
H.323 Call Redirection Enhancements
Interactive Voice Response Version 2.0 on Cisco VoIP Gateways
Interface Alias Long Name Support
IP Header Compression Enhancement—PPPoATM and PPPoFR Support
IPSec and 3DES Feature Set for Cisco 820 Series Routers
Low Latency Queueing with Priority Percentage Support
MGCP CAS PBX and PRI Backhaul on Cisco 7200 Series Routers
MGCP CAS PBX and AAL2 PVC with Basic CLASS and Operator Services
MGCP VoIP Signaling for 1750 Series
Mobile IP MIB Support for SNMP
Modem Script and System Script Support in Large-Scale Dial-Out
MPLS Label Distribution Protocol
MPLS Label Distribution Protocol MIB
MPLS Label Switching Router MIB
MPLS QoS Multi-VC Mode for PA-A3
NetFlow Multiple Export Destinations
Network-Based Application Recognition
Preauthentication with ISDN PRI and Channel-Associated Signaling Enhancements
Prefix Dial for 800 Series Routers
Quality of Service for Virtual Private Networks
RADIUS Attribute 66 (Tunnel-Client-Endpoint) Enhancements
SA Agent Support for Application Monitoring, Frame Relay, VoIP, and MPLS VPN
Secure Shell Terminal-Line Access
Shell-Based Authentication of VPDN Users
SIP Diversion Header Implementation for Redirecting Number
SIP Gateway Support for Third-Party Call Control
SNMP Trap Support for the Virtual Switch Interface Master MIB
Supplementary Telephone Services for the Euro-ISDN Switch
TCL IVR disconnect cause-code Manipulation
Trimble Palisade NTP Synchronization Driver for the Cisco 7200 Series Routers
Using 31-bit Prefixes on IPv4 Point-to-Point Links
Voice over ATM with AAL2 Trunking on Cisco 7200 Series Routers
X.25 Annex G Session Status Change Reporting
Deprecated and Replacement MIBs
SNMP Version 1 BGP4-MIB Limitations
Important Notes for Cisco IOS Release 12.2(15)T9
Cisco Images Deferred Because of Caveat CSCec46250
Cisco Images Deferred Because of Caveat CSCec46250
Important Notes for Cisco IOS Release 12.2(15)T8
Cisco Images Deferred Because of Caveats CSCec46250 and CSCin50865
Cisco Images Deferred Because of Caveat CSCec46250
Cisco Images Deferred Because of Caveat CSCec46250
Important Notes for Cisco IOS Release 12.2(15)T5
Cisco Images Deferred Because of Caveat CSCea91464
Important Notes for Cisco IOS Release 12.2(15)T4
Images Deferred Because of Caveats CSCea21186, CSCeb07534, CSCeb07595, and CSCeb10053
Important Notes for Cisco IOS Release 12.2(15)T3
Important Notes for Cisco IOS Release 12.2(15)T1
Images Deferred Because of Caveat CSCin40652
Important Notes for Cisco IOS Release 12.2(15)T
Cisco Images Deferred Because of Caveat CSCdy01600
Important Notes for Cisco IOS Release 12.2(13)T1
Cisco 1600 Series Router Images Deferred Because of Caveat CSCdz38371
Important Notes for Cisco IOS Release 12.2(13)T
Configuring MD5 Authentication for BGP Peering Sessions
Cisco 1600 Series Router Images Deferred Because of Caveat CSCdz38371
Cisco 3620 Series Router Images Deferred Because of Caveat CSCdz45923
Cisco AS5800 Images Deferred Because of Caveats CSCdz04856, CSCdz09639, CSCdz26779, and CSCdy87529
Cisco Catalyst 4000 Access Gateway Module Images Deferred Because of Caveat CSCdz27525
Cisco Images Deferred Because of Caveat CSCdy01600
Important Notes for Cisco IOS Release 12.2(11)T9
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Important Notes for Cisco IOS Release 12.2(11)T8
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Important Notes for Cisco IOS Release 12.2(11)T6
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Important Notes for Cisco IOS Release 12.2(11)T5
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Important Notes for Cisco IOS Release 12.2(11)T3
Cisco IAD2420 Images Deferred Because of Caveat CSCdz62759
Important Notes for Cisco IOS Release 12.2(11)T2
Update to the mgcp fax t38 Command
Important Notes for Cisco IOS Release 12.2(11)T
Cisco Catalyst 4000 Access Gateway Module Images Deferred Because of Caveat CSCdy17203
Cisco H.235 Accounting and Security Enhancements for Cisco Gateways
Cisco Images Deferred Because of Caveat CSCdy01600
Detecting Carrier Sense Errors on the Cisco uBR905 and Cisco uBR925 Cable Access Routers
Displaying Alarm Settings on the Cisco AS5800
Fine-Grain Address Segmentation in Dial Peers
Gatekeeper Alias Registration and Address Resolution Enhancements
MICA and NextPort Modem Tech-Support Commands for the AS5xxx Platforms
OSP Client Performance Improvement
SS7 Interconnect to Lucent 1AESS Switches
Important Notes for Cisco IOS Release 12.2(8)T2
Use 12.2(8)T1 Version of c7200-kboot-mz Image
Important Notes for Cisco IOS Release 12.2(8)T1
Cisco IGX 8400 Series URM Images Deferred Because of Caveat CSCdx41149
Cisco 7200 Series Router Limitation
Important Notes for Cisco IOS Release 12.2(8)T
Changes to Feature Support with Cisco IOS Release 12.2(8)T
Cisco IGX 8400 Series URM Images Deferred Because of Caveat CSCdx41149
Cisco Images Deferred Because of Caveat CSCdy01600
Enhanced Gigabit Ethernet Interface Processor Support on Cisco 7500/RSP Series
MPLS Defects in Cisco IOS Release 12.2(8)T
Important Notes for Cisco IOS Release 12.2(4)T
Cisco 7500 Series Images Deferred Because of Caveat CSCdu01272
Cisco 15104 Optical Networking System Image Deferred
Cisco Images Deferred Because of Caveat CSCdy01600
MPLS VPN with TE and MPLS InterAS Advisory on Cisco IOS Software
Important Notes for Cisco IOS Release 12.2(2)T
Addition of the squeeze Command for Cisco 2600 and Cisco 3600 Series Routers
Changes to the output attenuation Command
Cisco 820 and SOHO 70 Router Images Deferred Because of Caveat CSCds69577
Cisco Catalyst 4000 Gateway Images Deferred Because of Caveats CSCdu59093 and CSCdu63022
Cisco Images Deferred Because of Caveat CSCdy01600
Caveats for Cisco IOS Release 12.2T
New and Changed Information
The following is a list of the new features that are supported in Cisco IOS Release 12.2 T. For additional information regarding the features supported in Cisco IOS Release 12.2T, refer to the new feature documentation at the following location:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/index.htm
Determining Platform Support Through Cisco Feature Navigator
Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated information regarding platform support for the new features listed in Cisco IOS Release 12.2 T, access Cisco Feature Navigator. Cisco Feature Navigator is regularly updated as new platform support is added for features.
Cisco Feature Navigator is a web-based tool that enables you to determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:
Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
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MPLS Class of Service is now referred to as MPLS Quality of Service. This transition reflects the growth of MPLS to encompass a wider meaning and highlight the path toward Any Transport over MPLS.
New Hardware Features Supported in Cisco IOS Release 12.2(15)T
The following new hardware features are supported in Cisco IOS Release 12.2(15)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
1 Port Enhanced ATM Port Adapter with Support for 8K VCs
The PA-A6 is a series of single-width, single-port, ATM port adapters for Cisco 7200 series and Cisco 7401ASR routers. With advanced ATM features, the PA-A6 supports broadband aggregation, WAN aggregation, and campus/MAN aggregation.
1 and 2-port T1/E1 Multiflex Voice/WAN Interface Card
1- and 2-port T1/E1 Multiflex Voice/WAN interface cards provide basic structured and unstructured service for T1 or E1 networks. The card provides fractional data service and channelized voice services and TDM drop and insert (voice/data integration) services.
1- and 2-Port V.90 Modem WICs for Cisco 1720, 1751 and 1760 Routers
The one- and two-port V.90 Modem WICs expand the extensive range of WICs currently available on these routers. The modem WIC cards provide cost-effective basic telephone service connectivity to allow remote router management, asynchronous Dial-on-Demand routing (DDR) and dial back-up, and low-density remote access server (RAS) services.
Catalyst 4500 Access Gateway Module 16-port RJ21 FXS Module (WS-U4604-16FXS)
The 16-Port RJ21 FXS module for the Catalyst 4500 Access Gateway Module is a high density analog phone and fax interface. By providing service to analog phones and fax machines, the sixteen Foreign Exchange Station (FXS) ports emulate a PSTN central office (CO) or PBX.
Catalyst 4500 AGM Voice/WAN Bundle (WS-X4604-VOICE)
The Cisco Catalyst 4500 AGM Voice/WAN bundle provides integrated telephony and routing services to the Cisco Catalyst 4000 series and Cisco Catalyst 4500 series switches. The Cisco Catalyst 4500 AGM Voice/WAN bundle consists of the following products:
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Gigabit Ethernet Network Module
The Gigabit Ethernet (GE) network module provides gigabit connectivity. The throughput of the interface depends on the platform. The network module has one GBIC slot to carry any standard copper or optical Cisco GBIC, including CWDM. The GE network module optimizes the performance for branch office customers by offering a high-speed uplink to both existing and new LAN or WAN environments. The extended reach of the provided fiber connectivity allows customers the option of interconnecting branch offices with Gigabit Ethernet and avoids expensive leased serial lines. Metro area service providers now have additional options when connecting their customers in branch offices to MANs.
The Gigabit Ethernet network module is supported on the following platforms: Cisco 2691, Cisco 3660, Cisco 3725, and Cisco 3745.
MRP300
The Multiservice Route Processor 300 (MRP300) is a voice-and-data-capable router that can carry voice traffic over an IP network and that can link small-to-medium-size remote Ethernet LANs to central offices over WAN links. The MRP300 has a slot for expanding flash memory; two slots that support WICs, VWICs, and VICs; two PVDM slots for adding DSPs; and a DIMM slot for upgrading DRAM.
MRP3-8FXS
The MRP3-8FXS contains an 8-port Foreign Exchange Station (FXS) module and a slot for any VIC, WIC, or VWIC module that supports digital and analog voice trunks and WAN routing interfaces. The MRP3-8FXS is similar to the analog station interface 81 card (ASI81), with the exception that the ASI81 does not have onboard Flash memory.
MRP3-16FXS
The MRP3-16FXS contains a 16-port Foreign Exchange Station (FXS) module. The MRP3-16FXS is similar to the analog station interface 161 card (ASI160), except that the ASI160 does not have onboard Flash memory.
NPE-G1
The NPE-G1 is the first network processing engine for the Cisco 7200 VXR routers to provide the functionality of both a network processing engine and an I/O controller. If used without an I/O controller, an I/O blank panel must be in place.
Although its design provides I/O controller functionality, it can also work with any I/O controller supported in the Cisco 7200 VXR routers. The NPE-G1, when installed with an I/O controller, provides the primary input/out functionality; that is, the NPE-G1 input/out functionality enhances that of the existing I/O controller. However, when both the I/O controller and NPE-G1 are present, the functionality of the auxiliary port and console port are on the I/O controller.
The NPE-G1 maintains and executes the system management functions for the Cisco 7200 VXR routers and also holds the system memory and environmental monitoring functions.
The NPE-G1 consists of one board with multiple interfaces. It is keyed so that it can be used only in the Cisco 7200 VXR routers.
RPM-XF Card for the MGX 8850
The RPM-XF card is a next-generation, high-performance model of the RPM for the MGX 8850 platform, using PXM45 processor modules. It is a router module based on an RM7000A MIPS processing engine.
The RPM-XF hardware provides forwarding technology for packet switching capabilities in excess of 2-million pps. The forwarding engine is packet based and is interfaced to the midplane of the system through a combination of switch interface technologies.
SDH/STM-1 Trunk Card for Cisco AS5850 Universal Gateway
Channelized STM-1 provides a high speed remote access aggregation solution with 63 E1s and 1890 DSO channels. The SDH/STM-1 trunk card is a high density mux/demux card that takes in an STM-1 (SDH) pipe, used to transport up to 1890 DS0 channels. The SDH/STM-1 trunk card provides an ingress connection between the Cisco AS5850 universal gateway and external networks. The SDH/STM-1 trunk card has a 155-mbps channelized SDH physical interface in a standard dial feature card (DFC) format. The SDH interface supports channelization to 64 kbps and connects to single mode fiber optic supporting intermediate reach PPP applications.
New Software Features in Cisco IOS Release 12.2(15)T
The following new features are supported in Cisco IOS Release 12.2(15)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
ADSL over ISDN
Cisco 826 routers connect corporate telecommuters and small offices via Internet service providers (ISPs) over asymmetric digital subscriber lines (ADSLs) to corporate LANs and the Internet. The router can provide bridging and multiprotocol routing between LAN and WAN ports. Cisco 826 routers provide connectivity to an ISDN network through an ADSL port.
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Any Transport over MPLS (AToM)
Any Transport over MPLS (AToM) transports Layer 2 packets over a Multiprotocol Label Switching (MPLS) backbone. AToM enables service providers to connect customer sites with existing data link layer (Layer 2) networks, by using a single, integrated, packet-based network infrastructure—a Cisco MPLS network. Instead of separate networks with network management environments, service providers can deliver Layer 2 connections over an MPLS backbone. AToM provides a common framework to encapsulate and transport supported Layer 2 traffic types over an MPLS network core. AToM supports the following transport types:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/atomt/index.htm
ARP Optimization
The Address Resolution Protocol (ARP) is used to map a Layer 3 IP address to a Layer 2 MAC address. A Cisco router stores this mapped information in an ARP table. The ARP table provides MAC rewrite information when the router is forwarding a packet using Cisco Express Forwarding (CEF) or other IP switching technologies.
In previous versions of Cisco IOS software, the ARP table was organized for easy searching on an entry based on the IP address. However, there are cases such as interface flapping on the router and a topology change in the network in which all related ARP entries need to be refreshed for correct forwarding. This situation could consume a significant amount of CPU time in the ARP process to search and clean up all the entries. The ARP Optimization feature improves ARP performance by reducing the ARP searching time by using an improved data structure.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s22/arpoptim.htm
Asynchronous Call Queueing by Role
The Asynchronous Call Queueing by Role feature allows priority users who are making Telnet connection requests to busy asynchronous rotary groups to be placed at the head of the queue when asynchronous rotary line queueing is enabled. If a second priority user makes a Telnet connection request, this user will be placed behind the first priority user at the head of the queue. This feature allows a priority user to access the first available line. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftasyncq.htm
AutoQoS - VoIP
The AutoQoS - VoIP feature allows you to automate the delivery of quality of service (QoS) on your network, and provides a means for simplifying the implementation and provisioning of QoS for voice over IP (VoIP) traffic. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftautoq1.htm
BGP Hybrid CLI Support
The BGP Hybrid CLI Support feature allows the network operator to configure the Border Gateway Protocol (BGP) using the Network Layer Reachability Information (NLRI) format for IPv4 unicast commands and the address-family identifier (AFI) format for address family commands, such as IPv6, VPNv4, and Connectionless Network Service (CLNS) protocol commands. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftbhycli.htm
BGP Increased Support of Numbered AS-Path Access Lists to 500
The BGP Increased Support of Numbered AS-Path Access Lists to 500 feature is an enhancement for Border Gateway Protocol (BGP) autonomous system access lists. This enhancement increases the maximum number autonomous system access lists from 199 to 500. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftiaaspa.htm
BGP Nonstop Forwarding (NSF) Awareness
Nonstop Forwarding (NSF) awareness allows a router to assist NSF-capable neighbors to continue forwarding packets during a switchover operation or during a well-known failure condition. The BGP Nonstop Forwarding Awareness feature allows an NSF-aware router that is running BGP to forward packets along routes that are already known for a router that is performing a switchover operation or is in a well-known failure mode. This capability allows the BGP peers of the failing router to retain the routing information that is advertised by the failing router and continue to use this information until the failed router has returned to normal operating behavior and is able to exchange routing information. The peering session is maintained throughout the entire NSF operation.
Cisco Nonstop Forwarding (NSF) works with the Stateful Switchover (SSO) feature in Cisco IOS software. SSO is a prerequisite of Cisco NSF. NSF works with SSO to minimize the amount of time a network is unavailable to its users following a switchover. The main objective of Cisco NSF is to continue forwarding IP packets following a Route Processor (RP) switchover. NSF/SSO is configured in the core of your network, and NSF awareness is configured on iBGP peers in the core and the edge of the network.
BGP Restart Session After Max-Prefix Limit
The BGP Restart Session After Max-Prefix Limit feature enhances the capabilities of the neighbor maximum-prefix command with the introduction of the restart keyword. This enhancement allows the network operator to configure the time interval at which a peering session is reestablished by a router when the number of prefixes that have been received from a peer has exceeded the maximum prefix limit. The restart keyword has a configurable timer argument that is specified in minutes. The time range of the timer argument is from 1 to 65535. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftbrsamp.htm
BGP Route-Map Policy List Support
The BGP Route-Map Policy List Support feature introduces new functionality to Border Gateway Protocol (BGP) route maps. This feature adds the capability for a network operator to group route-map match clauses into a named list called a policy list. A policy list functions like a macro within a route map. When the policy list is referenced within a route map with the match policy-list command, all match statements in the policy list are executed. Policy lists can be used for all applications of a route map and for redistribution between routing protocols. Policy lists can coexist with configured match and set clauses within the same subblock. Policy lists, however, do not support set statements, and policy lists are not supported by IP routing policy. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftbgprpl.htm
BRI QSIG Protocol
BRI QSIG is the QSIG support over BRI interface. QSIG protocol support allows Cisco voice gateways to connect PBXs, key telephone systems (KTS), and central office switches that communicate by using the QSIG protocol.
Certificate Security Attribute-Based Access Control
Under the IP Security (IPSec) protocol, certification authority (CA) interoperability permits Cisco IOS devices and a CA to communicate so that the Cisco IOS device can obtain and use digital certificates from the CA. Certificates contain several fields that are used to determine whether a device or user is authorized to perform a specified action. The Certificate Security Attribute-Based Access Control feature adds fields to the certificate that allow specifying an access control list (ACL) to create a certificate-based ACL. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftcrtacl.htm
Cisco Easy VPN Remote Enhancements
The Cisco Easy VPN Remote Enhancements feature improve the capabilities of the Cisco Easy VPN Client feature first delivered in Cisco IOS Release 12.2(4)YA. Additional capabilities include the following:
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These enhancement were introduced in Cisco IOS Release 12.2(8)YJ to support Cisco 806, Cisco 826, Cisco 827, and Cisco 828 routers; Cisco 1700 series routers; and Cisco uBR905 and Cisco uBR925 cable access routers. This release is adding support for Cisco 2600, Cisco 3600, and Cisco 3700 series routers. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftezvpnr.htm
Cisco IOS Firewall Stateful Inspection of ICMP
The Cisco IOS Firewall Stateful Inspection of ICMP feature addresses the limitation of qualifying Internet Control Management Protocol (ICMP) messages into either a malicious or benign category by allowing the Cisco IOS firewall to use stateful inspection to "trust" ICMP messages that are generated within a private network and to permit the associated ICMP replies. Thus, network administrators can debug network issues without needing to block ICMP messages from entering the network because of possible intruders.
Cisco IOS Firewall Support for SIP
The Cisco IOS Firewall Support for SIP feature integrates Cisco IOS firewalls, the Voice over IP (VoIP) protocol, and Session Initiation Protocol (SIP) within a Cisco IOS based platform, enabling better network convergence.
Cisco IOS Firewall Websense URL Filtering
The Cisco IOS Firewall Websense URL Filtering feature enables your Cisco IOS firewall (also known as Cisco Secure Integrated Software [CSIS]) to interact with the Websense URL filtering software, thereby allowing you to prevent users from accessing specified websites on the basis of some policy. The Cisco IOS Firewall feature works with the Websense server to know whether a particular URL should be allowed or denied (blocked). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122yu11/ftwebsen.htm
Cisco IOS Software Feature Removal—Phase II
The Cisco IOS Software Feature Removal feature is an engineering project to permanently remove selected legacy features (or components) from the Cisco IOS code. These features will not be available in future releases of Cisco IOS software. The legacy features that have been removed as of Cisco IOS Release 12.2(15)T are as follows:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
Cisco IOS Telephony Service Version 2.1
Cisco IOS Telephony Service (ITS) offers an entry-level IP telephony solution integrated directly into Cisco IOS software. Customers can now deploy voice, data, and IP telephony on a single platform for their small offices. ITS offers a core set of phone features that customers commonly require for their everyday business needs, and leverages the wide array of voice capabilities that are available in Cisco IOS software to provide a very robust IP telephony offering for the small office environment.
Cisco ITS version 2.1 provides support for the following new features:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/itsv21/index.htm
Cisco Mobile Networks—Priority Home Agent Assignment
The mobile router currently preconfigures home agents with different priorities, registering with only the highest priority home agent. However, there are situations in which the mobile router roams to an area where a closer home agent is more desirable to register with. The Cisco Mobile Networks—Priority Home Agent Assignment feature allows a mobile router to register with the closer home agent using the existing home agent priority configurations on the mobile router and care-of address access lists configured on the home agent. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftdynaha.htm
Cisco Mobile Networks—Static Collocated Care-of Address
The Cisco Mobile Networks—Static Collocated Care-of Address feature allows a mobile router to roam to foreign networks where foreign agents are not deployed. Before the introduction of this feature, the mobile router was required to use a foreign agent care-of address when roaming. Now a roaming interface with a static IP address configured on the mobile router itself works as the collocated care-of address (CCoA).
Cisco Mobile Networks—Tunnel Templates for Multicast
The Cisco Mobile Networks—Tunnel Templates for Multicast feature allows the configuration of multicast on statically created tunnels to be applied to dynamic tunnels brought up on the home agent and mobile router. A tunnel template is defined and applied to the tunnels between the home agent and the mobile router. The mobile router can now roam carrying multicast sessions to its mobile networks. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftmultic.htm
Cisco Survivable Remote Site Telephony Version 2.1
The Cisco Survivable Remote Site Telephony (SRST) feature offers enterprises a reliable mechanism for providing continuous IP telephony services to small branch offices in the event of an outage. SRST enables enterprises to build large IP telephony networks using centralized call processing resources.
SRST Version 2.1 provides support for the Cisco IP Phone Extension Module 7914, Unity Voice Mail integration, additional languages for Cisco IP Phone 7940 and Cisco IP Phone 7960 display, higher directory number (DN) maximums, and a new command for creating global prefixes. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/srst21/index.htm
Note
Class-Based Policer for the DiffServ AF PHB
The Class-Based Policer for the DiffServ AF PHB feature is based on RFC 2697 A Single Rate Three Color Marker. The packet stream is metered and packets are marked "conform," "exceed," or "violate." Marking is based on a Committed Information Rate (CIR) and two associated burst sizes, a Committed Burst Size (CBS) and an Excess Burst Size (EBS). A packet is marked "conform" if it does not exceed the CBS, "exceed" if it exceeds the CBS but not the EBS, and "violate" otherwise.
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Clear Channel T3/E3 with Integrated CSU/DSU
Nonchannelized (Clear Channel) T3/E3 service is delivered as a T3/E3 pipe with the bandwidth being 28x24x64k for T3 or 16x32x64k for E3. Clear Channel T3/E3 service is generally used in point-to-point applications (one customer sending data to one remote site). Any subdivision of bandwidth is performed at each customer site rather than at the central office. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122yt/122yt11/ft_te3nm.htm
Clear IPC Statistics
This existing feature provides a way to clear and reset the interprocess communications (IPC) statistics. When debugging IPC problems, the ipc stat counters are clearable, making it easier to diagnose the problem.
DHCP Accounting
The DHCP Accounting feature introduces authentication, authorization, and accounting (AAA) and Remote Authentication Dial-In User Service (RADIUS) support for Dynamic Host Configuration Protocol (DHCP) configuration. The introduction of AAA and RADIUS support improves public wireless LAN (PWLAN) security by sending secure START and STOP accounting messages. The configuration of this feature adds a layer of security that allows DHCP lease assignment and termination to be triggered for the appropriate RADIUS START and STOP accounting records so that the session state is properly maintained by upstream devices, such as a Service Selection Gateway (SSG). The additional security provided by this feature can help to prevent unauthorized clients or hackers from gaining illegal entry to the network by spoofing authorized DHCP leases. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftdhcpac.htm
DHCP ODAP Server Support
The DHCP ODAP Server Support feature introduces the capability to configure an IOS Dynamic Host Configuration Protocol (DHCP) server (or router) as a subnet allocation server. This capability allows the IOS DHCP server to be configured with a pool of subnets for lease to On-Demand Address Pool (ODAP) clients. Subnet pools can be configured for global ODAP clients or Multiprotocol Label Switched (MPLS) Virtual Private Network (VPN) ODAP clients on a per-client basis. The DHCP subnet allocation server creates bindings for the subnet leases and stores these leases in the DHCP database. This feature also supports database agents for subnet lease recovery. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftodapss.htm
DHCP Secured IP Address Assignment
The DHCP Secure IP Address Assignment feature introduces the capability to secure ARP table entries to Dynamic Host Configuration Protocol (DHCP) leases in the DHCP database. This feature secures and synchronizes the MAC address of the client to the DHCP binding, preventing unauthorized clients or hackers from spoofing the DHCP server and taking over a DHCP lease of an authorized client. When this feature is enabled and the DHCP server assigns an IP address to the DHCP client, the DHCP server adds a secure ARP entry to the ARP table with the assigned IP address and the MAC address of the client. This ARP entry cannot be updated by any other dynamic ARP packets, and this ARP entry will exist in the ARP table for the configured lease time or as long as the lease is active. The secured ARP entry can be deleted only by an explicit termination message from the DHCP client or by the DHCP server when the DHCP binding expires. This feature can be configured for a new DHCP network or used to upgrade the security of an existing network. The configuration of this feature does not interrupt service and is not visible to the DHCP client. The configuration of this feature does not interrupt service and is not visible to the DHCP client. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftdsiaa.htm
DHCP Server Import All Enhancement
When the import all DHCP pool configuration command is used, the DHCP Server Import All Enhancement feature allows options imported by one subsystem to coexist with options imported from another subsystem. When the session is terminated or the lease is released, the imported options are cleared from the DHCP server database.
DHCP Server—ODAP Support for Non-MPLS VPN Pools
The DHCP Server—On-Demand Address Pool Manager is a feature in which pools of IP addresses can be dynamically increased or reduced in size depending on the address utilization level. On-demand address pools (ODAPs) support address assignment using the Dynamic Host Configuration Protocol (DHCP) for customers using private addresses. Each ODAP is configured and associated with a particular Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN).
The DHCP Server—ODAP Support for Non-MPLS VPN Pools feature enhances the existing feature to provide support for non-MPLS VPN pools. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftondhcp.htm
E1 R2 Signaling
R2 signaling is an international signaling standard that is common to channelized E1 networks. The E1 R2 Signaling feature was introduced in Cisco IOS Release 11.3(2)T and is now supported on Cisco 1751 and Cisco 1760 platforms in Cisco IOS Release 12.2(15)T
EIGRP Nonstop Forwarding (NSF) Awareness
Nonstop Forwarding (NSF) awareness allows a router to assist NSF-capable neighbors to continue forwarding packets during a switchover operation or during a well-known failure condition. The EIGRP Nonstop Forwarding Awareness feature allows an NSF-aware router that is running EIGRP to forward packets along routes that are already known for a router that is performing a switchover operation or is in a well-known failure mode. This capability allows the EIGRP peers of the failing router to retain the routing information that is advertised by the failing router and continue to use this information until the failed router has returned to normal operating behavior and is able to exchange routing information. The peering session is maintained throughout the entire NSF operation.
Enhanced Debug Capabilities for Cisco Voice Gateways
The enhanced debugging capability for Cisco voice gateways provides improvements to the debugging output in order to identify and track a specific call in a multiple-call environment. Before the implementation of this feature, it was difficult to correlate call information between gateways or to identify specific debug messages associated with a single call, when multiple voice calls were simultaneously active. The output was unstructured and presented in a free form.
This feature adds a standardized header to the debug outputs of multiple voice modules, such as voice telephony service provider (VTSP), call control application program interface (CCAPI), session application (SSAPP), and interactive voice response (IVR). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_dbgs2.htm
Enhanced Object Tracking
Prior to the introduction of the Enhanced Object Tracking feature, the Hot Standby Router Protocol (HSRP) had a simple tracking mechanism that allowed you to track the interface line protocol state only. If the line protocol state of the interface went down, the HSRP priority of the router was reduced, allowing another HSRP router with a higher priority to become active. The Enhanced Object Tracking feature separates the tracking mechanism from HSRP and creates a separate standalone tracking process that can be used by any other process as well as by HSRP. This feature allows tracking of other objects in addition to the interface line protocol state.
A client process, such as HSRP, Virtual Router Redundancy Protocol (VRRP), or Gateway Load Balancing Protocol (GLBP), can now register with the tracking service, its interest in tracking a particular object, such as an interface or a route, and then be notified when the tracked object changes state. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/fthsrptk.htm
Expanded Scope for Cause-Code-Initiated Call Establishment Retries
The Expanded Scope for Cause-Code-Initiated Call Establishment Retries feature enables the gateway to reattempt calls when a disconnect message is received from the public switched telephone network (PSTN) without maintaining extra dial peers. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_ccu.htm
Exporting and Importing RSA Keys
The Exporting and Importing RSA Keys feature allows you to transfer security credentials between devices by exporting and importing RSA keys.
The Exporting and Importing RSA Keys feature allows you to share the private RSA key pair of a router with standby routers, therefore transferring the security credentials between networking devices. The key pair that is shared between two routers will allow one router to immediately and transparently take over the functionality of the other router. If the main router were to fail, the standby router could be dropped into the network to replace the failed router without the need to regenerate keys, reenroll in certification authority (CA), or manually redistribute keys.
You can also use the Exporting and Importing RSA Keys feature to place the same RSA key pair on multiple routers, so that all management stations that use SSH can be configured with a single public RSA key. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_key.htm
Fax and Modem Pass-Through over VoIP
Fax and modem pass-through are now supported on the Cisco 1750 and Cisco 1761 platforms beginning in Cisco IOS Release 12.2(15)T.
Note
On detection of the fax or modem tone on an established VoIP call, the gateways switch into modem fax or pass-through mode: the voice codec and configuration is suspended and the pass-through parameters are loaded for the duration of the fax or modem session. This changes the bandwidth needed for the call to the equivalent of G.711.
With pass-through, the fax or modem traffic is carried between the two gateways in RTP packets, using an uncompressed format resembling the G.711 codec. Packet redundancy may be used to mitigate the effects of packet loss in the IP network. Even so, fax and modem pass-through remain susceptible to packet loss, jitter and latency in the IP network. The two endpoints must be clocked synchronously for this type of transport to work predictably.
The Fax and Modem Pass-Through feature is also known as Voice Band Data (VBD) by the International Telecommunication Union (ITU). VBD refers to the transport of fax or modem signals over a voice channel through a packet network with an encoding appropriate for fax or modem signals. The minimum set of coders for VBD mode is G.711 ulaw and alaw with VAD disabled. For modem transport, Echo cancellation is also be disabled.
Firewall Intrusion Detection System Signature Enhancements
Before the Firewall Intrusion Detection System Signature Enhancements, the Cisco Intrusion Detection System (IDS) contained 59 signatures, which was only a small subset of the signatures supported by Cisco Secure IDS. Firewall Intrusion Detection System (IDS) Signature Enhancements introduces 42 additional IDS signatures to Cisco IOS IDS that are supported by other Cisco products, such as PIX; these newly added signatures are categorized as follows:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122yu11/ft_fwids.htm
Firewall N2H2 Support
The Cisco IOS Firewall N2H2 Support feature provides users with an additional option when choosing the URL filter vendor. Just like the Websense URL filtering server, N2H2 interacts with your Cisco IOS firewall (also known as Cisco Secure Integrated Software [CSIS]) to allow you to prevent users from accessing specified websites on the basis of some policy. The Cisco IOS firewall works with the N2H2 Internet Filtering Protocol (IFP) server to know whether a particular URL should be allowed or denied (blocked). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122yu11/ft_n2h2.htm
Firewall Support of HTTPS Authentication Proxy
The Firewall Support of HTTPS Authentication Proxy feature allows a user to encrypt the change of the username and password between the HTTP client and the Cisco IOS router via Secure Socket Layer (SSL) when authentication proxy is enabled on the Cisco IOS firewall, thereby ensuring confidentiality of the data that is passing between the HTTP client and the Cisco IOS router. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122yu11/ftfwhttp.htm
Frame Relay Voice-Adaptive Traffic Shaping
The Frame Relay Voice-Adaptive Traffic Shaping feature enables a permanent virtual circuit (PVC) to adjust the rate of traffic on the basis of the presence of packets in the priority queue or H.323 call setup signaling packets. This feature also introduces voice-adaptive fragmentation. Frame Relay voice-adaptive fragmentation allows fragmentation to be turned on when packets are detected in the priority queue or H.323 signaling packets are present and to be turned off when priority queue traffic and signaling packets are not present. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_vats.htm
G.732 Support for the Integrated Signaling Link Terminal
The G.732 Support for the Integrated Signaling Link Terminal feature ports the existing International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.732 bit error rate (BER) detection and alarm processing functionality from the Cisco Signaling Link Terminal (SLT) onto the Cisco AS5350 and Cisco AS5400 network access server (NAS) platforms. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftg7325x.htm
Gatekeeper Management Statistics
The Gatekeeper Management Statistics feature adds support for gatekeeper performance management parameters that provide statistics that may be used to monitor and troubleshoot a network. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_gms.htm
GLBP: Gateway Load Balancing Protocol
The Gateway Load Balancing Protocol feature provides automatic router backup for IP hosts configured with a single default gateway on an IEEE 802.3 LAN. Multiple first-hop routers on the LAN combine to offer a single virtual first-hop IP router while sharing the IP packet forwarding load between them. Other routers on the LAN may act as redundant GLBP routers that will become active if any of the existing forwarding routers fail.
This feature was originally introduced in Cisco IOS Release 12.2(14)S. This release is porting the feature into the Cisco 1700 series, Cisco 2600 series, Cisco 3640, and Cisco 3660 platforms. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_glbp.htm
H.323v4 Gateway Zone Prefix Registration Enhancements
The H.323v4 Gateway Zone Prefix Registration Enhancements feature provides support for two capabilities included in H.323 version 4: additive registration and dynamic zone prefix registration. Additive registration allows a gateway to add to or modify a list of aliases contained in a previous registration without first unregistering from the gatekeeper. Dynamic zone prefix registration allows a gateway to register actual public switched telephone network (PSTN) destinations served by the gateway with its gatekeeper. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftgwzpre.htm
Hot Standby MAC Address
The Hot Standby MAC Address (HSMA) feature achieves redundancy and fault tolerance and avoids a single point of failure of Cisco Channel Interface Processors (CIPs) or Channel Port Adapters (CPAs). This feature also ensures that multiple devices on the Ethernet can have a common MAC address.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123newft/123_1/ft_hsma.htm
HTTP 1.1 Client
This feature implements support for HTTP clients within Cisco IOS software compliant with the HTTP 1.1 standard (RFC 2616). The HTTP 1.1 Client allows the network device to contact a remote web server and obtain content or interact with remote applications. The HTTP 1.1 Client is enabled by default on supported platforms.
HTTP 1.1 Web Server
The HTTP 1.1 Web Server feature provides a consistent interface for users and applications by implementing the HTTP 1.1 standard (RFC 2616). Prior to this release, Cisco software supported only a partial implementation of HTTP 1.0. The integrated HTTP Server API supports server application interfaces. When combined with the HTTPS and HTTP 1.1 Client features, the HTTP 1.1 Web Server feature provides a complete, secure solution for HTTP services to and from Cisco devices. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/fthttp1s.htm
HTTPS-HTTP with SSL 3.0
The HTTPS-HTTP with SSL 3.0 feature provides integrated Secure Socket Layer (SSL) 3.0 support for the HTTP 1.1 Server and Client in Cisco IOS software. SSL provides encryption to allow secure HTTP communications. HTTP with SSL (HTTPS) allows for encrypted HTTP communications with Cisco devices.
IGMP State Limit
The IGMP State Limit feature provides protection against denial of service (DoS) attacks caused by Internet Group Management Protocol (IGMP) packets. The new command-line interface (CLI) introduced by this feature allows you to configure a limit on the number of IGMP states that results from IGMP, IGMP Version 3 lite (IGMP v3lite), and URL Rendezvous Directory (URD) membership reports on a per-interface or global basis. Membership reports in excess of the configured limits will not be entered in the IGMP cache, and traffic for those excess membership reports will not be forwarded. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_igmps.htm
Implementing OSPF for IPv6
The Open Shortest Path First (OSPF) Version 3 for IPv6 (RFC 2740) feature expands on OSPF to provide support for IPv6 routing prefixes. In OSPF for IPv6, the commands used to customize OSPF are in interface configuration mode rather than router configuration mode. When using a nonbroadcast multiaccess (NBMA) interface in OSPF for IPv6, users must manually configure the router in order to detect neighbors.
Integrated IS-IS Multi-Topology Support for IPv6
The Integrated IS-IS Multi-Topology Support for IPv6 feature provides support for routing IPv6 prefixes in Intermediate System-to-Intermediate System (IS-IS) using a multi-topology solution.
Integrated IS-IS Nonstop Forwarding (NSF) Awareness
The Integrated IS-IS Nonstop Forwarding (NSF) Awareness feature allows customer premises equipment (CPE) routers that are NSF-aware to help NSF-capable routers perform nonstop forwarding of packets. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/isnsfawa.htm
Integrated Voice and Data WAN on T1/E1 Interfaces Using the AIM-ATM-VOICE-30 Module
The Integrated Voice and Data WAN on T1/E1 Interfaces Using the AIM-ATM-VOICE-30 Module feature provides configuration enhancements for the AIM-ATM-VOICE-30 digital signaling processor (DSP) card on the Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745. This feature provides a migration path to higher bandwidth without the need to change transport facilities and provides a voice processing (termination) solution with AIM-ATM-VOICE-30 without consuming a network module slot. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftbckaim.htm
IP Access List Entry Sequence Numbering
Users can apply sequence numbers to permit or deny statements and also reorder, add, or remove such statements from a named IP access list. This feature makes revising IP access lists much easier. Prior to this feature, users could add access list entries to the end of an access list only; therefore needing to add statements anywhere except the end required reconfiguring the entire access list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122s/122snwft/release/122s14/fsaclseq.htm
IPMROUTE-STD-MIB
This feature introduces support for the IPMROUTE-STD-MIB in Cisco IOS software. IPMROUTE-STD-MIB, as defined in RFC 2932, is a module for management of IP multicast routing in a manner independent of the specific multicast routing protocol in use. Support for this MIB replaces the draft form of the IPMROUTE-MIB.
The IPMROUTE-STD-MIB supports all the MIB objects of the IPMROUTE-MIB and in addition supports the following four new MIB objects:
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Note
IPSec VPN Accounting
The IPSec VPN Accounting feature allows for a session to be accounted for by indicating when the session starts and when it stops. Additionally, session identifying information and session usage information will be passed to the RADIUS server via RADIUS attributes and vendor-specific attributes (VSAs). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_evpna.htm
IPv6 ISATAP Tunnel Support
The Intra-Site Automatic Tunnel Addressing Protocol (ISATAP) is an automatic overlay tunneling mechanism that uses the underlying IPv4 network as a nonbroadcast multiaccess (NBMA) link layer for IPv6. The IPv4 address is encoded in the last 32 bits of the IPv6 address, enabling automatic IPv6-in-IPv4 tunneling within an IPv4 network. ISATAP tunnels allow individual IPv4/IPv6 dual-stack hosts within a site to connect to an IPv6 network using the IPv4 infrastructure. ISATAP uses a normal global IPv6 prefix (/64), that can be used with both local and global unicast IPv6 prefixes, enabling IPv6 routing on the Internet. For additional information, refer to the following document:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/ipv6_c/sa_tunv6.htm
IPv6 MIB Support
IPv6 MIBs are now available for managing IPv6 traffic. Supported MIBs include the CISCO-IETF-IP-MIB and CISCO-IETF-IP-FORWARDING-MIB.
IPv6 Provider Edge Router over MPLS
The IPv6 Provider Edge Router over MPLS (Cisco 6PE) feature allows service providers that are running an MPLS/IPv4 infrastructure to offer IPv6 services on an Multiprotocol Label Switching (MPLS) network. A Cisco 6PE-enabled backbone allows IPv6 domains to communicate with each other over an MPLS IPv4 core network. A Cisco 6PE implementation requires no backbone infrastructure upgrades and no reconfiguration of core routers, because forwarding is based on labels rather than on the IP header itself.
Additionally, the inherent Virtual Private Network (VPN) and Traffic Engineering (TE) services available within an MPLS environment allow IPv6 networks to be combined into VPNs or extranets over an infrastructure that supports IPv4 VPNs and MPLS-TE.
The provider edge (PE) routers at each end of the MPLS network must be IPv6-enabled. The PE routers apply an appropriate label for the address in the packet to reach the other side of the MPLS backbone. This is similar to tunneling because it allows IPv6 traffic to be transported over MPLS without the routers in the backbone being aware of the IPv6 traffic. An MPLS packet enters and exits the MPLS network on different routers, and each router must be IPv6- and 6PE-enabled.
For more information about the IPv6 Provider Edge Router over MPLS (Cisco 6PE) feature, refer to the following document:
http://www.cisco.com/application/pdf/en/us/guest/products/ps6553/c1161/cdccont_0900aecd80311df4.pdf
ISDN Generic Transparency Descriptor (GTD) for Setup Message
The ISDN Generic Transparency Descriptor for Setup Message feature provides support for mapping ISDN information elements (IEs) to corresponding GTD parameters. Supported IEs and GTD parameters include the following:
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This feature allows networks to do the following:
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Cisco implements this feature on Cisco IOS gateways by providing a mechanism to allow creating and passing the Q931 Setup message and its parameters in a GTD format. The Setup message, sent by the gateway to initiate call establishment, is mapped to the GTD Initial Address Message (IAM). Generic transparency descriptors represent parameters within signaling messages and enable transport of signaling data in a standard format across network components and applications. The GTD mechanism allows them to share signaling data and achieve interworking between different signaling types.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftgtdisd.htm
ISDN PRI-SLT
The ISDN PRI-SLT feature allows you to release the ISDN PRI signaling time slot for Redundant Link Manager (RLM) configurations and for Signaling System 7 (SS7) applications in integrated Signaling Link Terminal (SLT) configurations. This feature supports the use of DS0 time slots for SS7 links and allows the coexistence of SS7 links and PRI voice and data bearer channels on the same T1 or E1 controller span. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_8/ftprislt.htm
ISDN Progress Indicator Support for SIP Using 183 Session Progress
The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature adds the SIP 183 Session Progress and Ringing messages to better map to the ISDN/CAS messages.
The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was previously released in Cisco IOS Release 12.1(5)T. This feature has been added on the Cisco 1751 and the Cisco 1760 in Cisco IOS Release 12.2(15)T.
L2TP Dial-Out Load Balancing and Redundancy
The L2TP Dial-Out Load Balancing and Redundancy feature enables an L2TP network server (LNS) to dial out to multiple L2TP access concentrators (LACs). When the LAC with the highest priority goes down, it is possible for the LNS to failover to another lower priority LAC. The LNS can also load-balance the sessions between multiple LACs that have the same priority settings. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftl2tlbr.htm
L2TP Large-Scale Dial-Out per-User Attribute via AAA
The L2TP Large-Scale Dial-Out per-User Attribute via AAA feature enhances Layer 2 Tunneling Protocol (L2TP) to support per-user attributes using authentication, authorization, and accounting (AAA) for large-scale dial-out. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftl2taaa.htm
Malicious Caller Identification (MCID) Invocation Support for Enterprise Networks
The Malicious Caller Identification (MCID) Invocation Support for Enterprise Networks feature enables a called party inside an enterprise network to use a configurable sequence of digits to notify the local law enforcement agency of a malicious call. MCID uses Tool Command Language (TCL) and interactive voice response (IVR) to trigger the gateway to send calling number information to the authorities.
The feature is platform independent; uses dual tone multifrequency (DTMF) tones to generate the trigger; and operates in both H.323 and Session Initiation Protocol (SIP) voice gateways and on all phones. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftmcid.htm
Measurement-Based Call Admission Control for SIP
The Measurement-Based Call Admission Control for SIP feature implements support within Session Initiation Protocol (SIP) to monitor IP network capacity and check the availability of router and interface resources, and to decide if adequate resources are available to carry a successful Voice over IP (VoIP) session. This feature also implements a mechanism to prevent calls that arrive from the IP network from entering the gateway when required resources are not available to process the call. This feature also provides the ability to support measurement-based call admission control processes as well as check for resource availability. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftcacsip.htm
MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
The MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles feature implements the following Media Gateway Control Protocol (MGCP) protocols on the supported Cisco media gateways:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_24mg1.htm
Note
MGCP Based Fax (T.38) and DTMF Relay
This feature adds support for T.38 fax relay and dual tone multifrequency (DTMF) relay with Media Gateway Control Protocols (MGCP). This feature provides two modes of implementation for each component: gateway (GW)-controlled mode and call agent (CA)-controlled mode. In GW-controlled mode, GWs negotiate DTMF and fax relay transmission by exchanging capability information in Session Description Protocol (SDP) messages. That transmission is transparent to the CA. GW-controlled mode allows use of the MGCP-Based Fax (T.38) and DTMF (IETF RFC 2833) Relay feature without upgrading the CA software to support the feature. In CA-controlled mode, CAs use MGCP messaging to instruct GWs to process fax and DTMF traffic. For MGCP T.38 Fax Relay, the CAs can also instruct GWs to revert to GW-controlled mode if the CA is unable to handle the fax control messaging traffic; for example, in overloaded or congested networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmgcpfx.htm
The MGC Protocol Based Fax (T.38) and DTMF (IETF RFC 2833) Relay feature was previously released in Cisco IOS Release 12.2(8)T on the Cisco 3600 series and Cisco MC3810, and in Cisco IOS Release 12.2(11)T on the Cisco AS5300, Cisco AS5400, and Cisco AS5850. This feature has been added on the Cisco 1751 and the Cisco 1760 in Cisco IOS Release 12.2(15)T.
MGCP Basic CLASS and Operator Services
The Media Gateway Control Protocol (MGCP) Basic CLASS and Operator Services feature provides CLASS and 3-way calling functionality using the Simple Gateway Control Protocol (SGCP) and MGCP protocols.
MGCP VoIP Call Admission Control
The MGCP VoIP Call Admission Control (CAC) feature determines if calls can be accepted on the IP network on the basis of available network resources. Before this release, Media Gateway Control Protocol (MGCP) Voice over IP (VoIP) calls were established regardless of the available resources on the gateway or network. The gateway had no mechanism for gracefully refusing calls if resources were not available to process the call. New calls would fail with unexpected behavior and in-progress calls would experience quality-related problems.
The MGCP VoIP Call Admission Control feature provides three CAC mechanisms to address the need for improved quality and predictable gateway behavior. The first mechanism is local/system CAC, which provides the ability to gracefully refuse calls on the basis of the availability of local gateway call processing resources such as CPU utilization and memory. The second CAC mechanism provides synchronization with Resource Reservation Protocol (RSVP) and reports the reservation request to the call agent. The third mechanism provides network congestion detection to gracefully refuse calls on the basis of a measured level of congestion.
The MGCP VoIP Call Admission Control feature was previously released in Cisco IOS Release 12.2(8)T and is now supported on the Cisco 1751 and Cisco 1760 platforms.
Mobile IP—Home Agent Accounting
In Cisco IOS Mobile IP, the home agent keeps track of the location of the mobile node as it roams away from its home network and forwards all traffic destined to the mobile node to its new location on the Internet. The Mobile IP—Home Agent Accounting feature allows the home agent to generate the following three new accounting messages that are forwarded to the Service Selection Gateway (SSG):
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The SSG acts as the proxy server for the authentication, authorization, and accounting (AAA) server and acknowledges the accounting messages sent by the home agent. The accounting records generated by the home agent can be stored on the AAA server and used by Internet service providers (ISPs) for billing, capacity planning, and operations. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/fthaacct.htm
MPLS VPN—MIB Support
The MPLS VPN—MIB Support feature provides Simple Network Management Protocol (SNMP) agent support in Cisco IOS software for Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN) management, as implemented in the draft MPLS/BGP Virtual Private Network Management Information Base Using SMIv2 (draft-ietf-ppvpn-mpls-vpn-mib-03.txt). The Provider-Provisioned VPN (PPVPN)-MPLS-VPN MIB provides access to VPN routing/forwarding instance (VRF) information, interfaces included in the VRF, and other configuration and monitoring information.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftvnmb15.htm
The MPLS VPN—MIB Support feature was introduced in Cisco IOS Release 12.0(21)ST. The PPVPN-MPLS-VPN MIB notifications were supported in Cisco IOS Release 12.2(13)T. The PPVPN-MPLS-VPN MIB tables were integrated into Cisco IOS Release 12.2(15)T.
MPLS VPN Support for EIGRP Between Provider Edge (PE) and Customer Edge (CE)
The MPLS VPN Support for EIGRP Between Provider Edge (PE) and Customer Edge (CE) feature provides the Enhanced Interior Gateway Routing Protocol (EIGRP) with the capability to redistribute routes through a Border Gateway Protocol (BGP) Virtual Private Network (VPN) cloud. This feature is configured only on PE routers, requiring no upgrade or configuration changes to customer equipment. This feature also introduces EIGRP support for Multiprotocol Label Switching (MPLS) and BGP extended community attributes. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/fteipece.htm
Multicast Subsecond Convergence
The Multicast Subsecond Convergence feature comprises a comprehensive set of features and protocol enhancements that provide for improved scalability and convergence in multicast-based services. This feature set provides for the ability to scale to larger service levels and to recover multicast forwarding after service failure in subsecond time frames.
Multicast subsecond convergence allows you to send Protocol Independent Multicast (PIM) router-query messages (PIM hellos) every few milliseconds. In earlier releases, you could send the PIM hellos every few seconds. By enabling a router to send PIM hello messages more often, this feature allows the router to discover unresponsive neighbors more quickly. As a result, the router can implement failover or recovery procedures more efficiently.
The scalability enhancements improve on the efficiency of handling increases (or decreases) in service users (receivers) and service load (sources or content). Scalability enhancements in this release include the following:
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The scalability enhancements provide the following benefits:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122s/122snwft/release/122s14/fs_subcv.htm
Multiple OPC Support for the Cisco Signaling Link Terminal
Multiple OPC Support for the Cisco Signaling Link Terminal (SLT) feature allows Cisco SLTs to access multiple Signaling System 7 (SS7) point codes (PCs) on a media gateway controller (MGC).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftsltopc.htm
NAT Support for IPSec ESP—Phase II
The NAT Support for IPSec ESP—Phase II feature allows multiple concurrent IP Security (IPSec) Encapsulating Security Payload (ESP) tunnels or connections through a Cisco IOS Network Address Translation (NAT) device configured in overload or Port Address Translation (PAT) mode. The IPSec ESP deployment does not need to use wrapper techniques that typically use the User Datagram Protocol (UDP) to pass through the NAT router. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftsecnat.htm
Network-Based Application Recognition Protocol Discovery Management Information Base
The existing Network-Based Application Recognition (NBAR) feature is used to identify protocols so that traffic can be classified appropriately for quality of service purposes. NBAR also contains a protocol discovery feature that displays for the user any NBAR-supported protocol traffic that is traversing an interface.
The NBAR Protocol Discovery MIB expands the capabilities of NBAR protocol discovery by providing the following new protocol discovery functionality through simple network management protocol (SNMP):
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Configure thresholds based on traffic of particular NBAR-supported protocols or applications that report breaches and send notifications when these thresholds are crossed. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftpdmib.htm
No Service Password-Recovery
The No Service Password-Recovery feature disables password-recovery capability for better console security.
OSPF Forwarding Address Suppression in Translated Type-5 LSAs
The OSPF Forwarding Address Suppression in Translated Type-5 LSAs feature causes a not-so-stubby area (NSSA) area border router (ABR) to translate Type-7 link state advertisements (LSAs) to Type-5 LSAs, but use the address 0.0.0.0 for the forwarding address instead of that specified in the Type-7 LSA. This feature causes routers that are configured not to advertise forwarding addresses into the backbone to direct forwarded traffic to the translating NSSA ABRs. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftoadsup.htm
OSPF Inbound Filtering Using Route Maps with a Distribute List
Users can define a route map to prevent Open Shortest Path First (OSPF) routes from being added to the routing table. In the route map, the user can match on any attribute of the OSPF route.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s24/routmap.htm
OSPF Nonstop Forwarding (NSF) Awareness
The OSPF Nonstop Forwarding (NSF) Awareness feature allows customer premises equipment (CPE) routers that are NSF-aware to help NSF-capable routers perform nonstop forwarding of packets. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftosnsfa.htm
OSPF Shortest Path First Throttling
The OSPF Shortest Path First Throttling feature makes it possible to configure Shortest Path First (SPF) scheduling in millisecond intervals and to potentially delay SPF calculations during network instability. SPF is scheduled to calculate the Shortest Path Tree (SPT) when there is a change in topology. One SPF run may include multiple topology change events.
The interval at which the SPF calculations occur is chosen dynamically and is based on the frequency of topology changes in the network. The chosen interval is within the boundary of the user-specified value ranges. If network topology is unstable, SPF throttling calculates SPF scheduling intervals to be longer until the topology becomes stable. Refer to the following document for additional information:
OSPF Support for Fast Hello Packets
The OSPF Support for Fast Hello Packets feature provides a way to configure the sending of hello packets in intervals less than 1 second. Such a configuration would result in faster convergence in an Open Shortest Path First (OSPF) network. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s23/fasthelo.htm
Per-User QoS via AAA Policy Name
The Per-User QoS via AAA Policy Name feature provides the ability to download a policy name that describes quality of service (QoS) parameters for a user session from a RADIUS server and apply them for the particular session. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_puq.htm
Per VRF AAA
The Per VRF AAA feature allows authentication, authorization, and accounting (AAA) on the basis of Virtual Private Network (VPN) routing and forwarding (VRF) instances. For Cisco IOS Release 12.2(15)T or later releases, you can use a customer template which may be stored either locally or remotely, and AAA services can be performed on the information that is stored in the customer template. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftvrfaaa.htm
PPPoE Connection Throttling
This feature will throttle the PPP over Ethernet (PPPoE) connection requests to prevent any denial of service attacks. It will implement per-mac/per-vc initiated session rate throttling in the PPPoE server to limit the session initiate count during a specific period of time.
PPPoE Profiles
The PPPoE Profiles feature introduces PPP over Ethernet (PPPoE) profiles, which contain configuration information for a group of PPPoE sessions. Multiple PPPoE profiles can be defined on a device, allowing different virtual templates and other PPPoE configuration parameters to be assigned to different Ethernet interfaces, VLANs, and ATM permanent virtual circuits (PVCs). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftpprfls.htm
PRI QSIG Protocol
QSIG is a standardized PBX signaling protocol used primarily in Europe over E1 and BRI trunks and occasionally in North America over T1 trunks. The PRI QSIG Protocol feature provides QSIG signalling over PRI trunks
RADIUS Support of 56-Bit Acct Session-Id
The Radius Support of 56-Bit Acct Session-Id feature introduces a new 32-bit authentication, authorization, and accounting (AAA) variable, acct-session-id-count. The first 8 bits of the acct-session-id-count variable are reserved for the unique-ident, a unique number assigned to the accounting session that is preserved between reloads. The acct-session-id-count variable is used in addition to the existing 32-bit acct-session-id variable, RADIUS Attribute 44. This provides 56 bits to represent the actual accounting session ID. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftradaid.htm
RADIUS Timeout Set During Pre-Authentication
The RADIUS Timeout Set During Pre-Authentication feature provides RADIUS timeout values during the pre-authentication phase of a session, and the values are not overwritten in later phases of the same session. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftattr27.htm
RSVP Message Authentication
RSVP Support for RTP Header Compression, Phase 1
SIP Call Transfer and Call Forwarding Supplementary Services
The SIP Call Transfer and Call Forwarding Supplementary Services feature introduces the ability of Session Initiation Protocol (SIP) gateways to initiate blind or attended call transfers. Release Link Trunking (RLT) functionality was also added with this feature. With RLT, SIP blind call transfers can now be triggered by channel-associated signaling (CAS) trunk signaling. Finally, the SIP Call Transfer and Call Forwarding Supplementary Services feature implements SIP support of call forwarding requests from a Cisco IOS gateway.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftsipcal.htm
SIP—Configurable PSTN Cause Code Mapping
For calls to be established between a session initiation protocol (SIP) network and a PSTN network, the two networks must be able to interoperate. One aspect of their interoperation is the mapping of PSTN cause codes, which indicate reasons for Public Switched Telephone Network (PSTN) call failure or completion, for SIP status codes or events. The opposite is also true: SIP status codes or events are mapped to PSTN cause codes. Event mapping tables found in this document show the standard or default mappings between SIP and PSTN.
However, you may want to customize the SIP user agent software to override the default mappings between the SIP and PSTN networks. The Configurable PSTN Cause Code to SIP Response Mapping feature allows you to configure specific map settings between the PSTN and SIP networks. Thus, any SIP status code can be mapped to any PSTN cause code, or vice versa. When set, these settings can be stored in the NVRAM and are restored automatically on bootup.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmap.htm
Note
SIP Diversion Header Implementation for Redirecting Number
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SIP—DNS SRV RFC2782 Compliance
Session Initiation Protocol (SIP) on Cisco Voice over IP (VoIP) gateways uses Domain Name System Server (DNS SRV) query to determine the IP address of the user endpoint. The query string has a prefix in the form of "protocol.transport." and is attached to the fully qualified domain name (FQDN) of the next hop SIP server. This prefix style, from RFC 2052, has always been available; however, with this release, a second style is also available. The second style complies with RFC 2782 and prepends the protocol label with an underscore "_"; as in "_protocol._transport." The addition of the underscore reduces the risk of the same name being used for unrelated purposes. The form compliant with RFC 2782 is the default style. Use the srv version command to configure the DNS SRV feature.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/vvfresrv.htm
Note
SIP Gateway Support for Third Party Call Control
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SIP Gateway Support of RSVP and TEL URL
The SIP Gateway Support of RSVP and TEL URL feature also supports Telephone Uniform Resource Locators or TEL URLs. Currently Session Initiation Protocol (SIP) gateways support URLs in the SIP format. SIP URLs are used in SIP messages to indicate the originator, recipient, and destination of the SIP request. However, SIP gateways may also encounter URLs in other formats, such as TEL URLs. TEL URLs describe voice call connections. They also enable the gateway to accept TEL calls sent through the Internet and to generate TEL URLs in the request line of outgoing INVITE requests.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/vvfresrv.htm
Note
SIP Intra-gateway Hairpinning
SIP hairpinning is a call routing capability in which an incoming call on a specific gateway is signaled through the IP network and back out the same gateway. This call can be a public switched telephone network (PSTN) call routed into the IP network and back out to the PSTN over the same gateway.
Similarly, SIP hairpinning can be a call signaled from a line (for example, a telephone line) to the IP network and back out to a line on the same access gateway. With SIP hairpinning, unique gateways for ingress and egress are no longer necessary.
Note
SIP INVITE Request with Malformed Via Header
SIP INVITE requests that a user or service participate in a session. Each INVITE contains a Via header that indicates the transport path taken by the request so far and where to send a response. In the past, when an INVITE contained a malformed Via header, the gateway would print a debug message and discard the INVITE without incrementing a counter. However, the printed debug message was often inadequate, and it was difficult to detect that messages were being discarded.
The SIP INVITE Request with Malformed Via Header feature provides a response to the malformed request. A counter, Client Error: Bad Request, increments when a response is sent for a malformed Via field. Bad Request is a class 400 response and includes the explanation Malformed Via Field. The response is sent to the source IP address (the IP address where the SIP request originated) at User Datagram Protocol (UDP) port 5060.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmalvia.htm
Note
SIP: ISDN Suspend/Resume Support
The SIP: ISDN Suspend/Resume Support feature adds Session Initiation Protocol (SIP) call-hold support to SIP gateways when an ISDN Suspend event is triggered. Because Suspend and Resume support already exists for H.323, the SIP implementation of Suspend and Resume provides feature parity. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftsusres.htm
SIP—Session Initiation Protocol for VoIP Enhancements
Voice over IP (VoIP) currently implements the International Telecommunication Union (ITU)'s H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. The Session Initiation Protocol (SIP) is a new protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.
The Cisco SIP functionality, introduced in Cisco IOS Release 12.1(1)T and enhanced in Cisco IOS Release 12.1(3)T, enables Cisco access platforms to signal the setup of voice and multimedia calls over IP networks. This release ports the feature into the Cisco 1751 and Cisco 1760 platforms. The SIP feature also provides nonproprietary advantages in the areas of
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SIP Support for Media Forking
The SIP Support for Media Forking feature provides the ability for Session Initiation Protocol (SIP) networks to create midcall multiple streams (or branches) of audio. The multiple streams of audio are associated with a single call, but can be sent to several different destinations. The SIP Support for Media Forking feature allows service providers to use technologies such as speech recognition, voice authentication, and text-to-speech conversion to provide sophisticated services to their end-user customers. An example is a web-browsing application that uses voice recognition and text-to-speech (TTS) technology to make reservations, verify shipments, or order products. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftspfork.htm
SIP T.38 Fax Relay
The SIP T.38 Fax Relay feature adds standards-based fax support to session initiation protocol (SIP) and conforms to ITU-T T.38 Procedures for real-time Group 3 facsimile communication over IP networks. The ITU-T standard specifies real-time transmission of faxes between two regular fax terminals over an IP network. Much like a voice call, SIP T.38 Fax Relay requires call establishment, data transmission, and release signaling.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftsipfax.htm
Note
SIP User Agent MIB
The Session Initiation Protocol (SIP) User Agent Client (UAC) and User Agent Server (UAS) are manageable by an SNMP-based network management platform, such as the Cisco Voice Manager. This release ports the feature to the Cisco 1750 and Cisco 1761 platforms. The SIP MIB has been defined, will be submitted to the IETF, and will be implemented on those platforms.
To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB web site on Cisco.com at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml
Source Interface Selection for Outgoing Traffic with Certificate Authority
The Source Interface Selection for Outgoing Traffic with Certificate Authority feature allows you to specify the address of an interface to be used as the source address for all outgoing TCP connections when a designated trustpoint has been configured. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_asish.htm
Support for Bridged RFC 1483 Encapsulated Traffic over ATM SVCs
The Support for Bridged RFC 1483 Encapsulated Traffic over ATM SVCs feature allows you to send bridged RFC 1483 encapsulated packets over ATM switched virtual circuits (SVCs). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftbridge.htm
Support for IUA with SCTP for Cisco Access Servers
The Support for IUA with SCTP for Cisco Access Servers feature supports the IDSN User Adaptation (IUA) Layer with Stream Control Transmission Protocol (SCTP) for the Cisco AS5x00 network access servers (NASs) and the Cisco 2420, Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series. This feature is to be used as an alternative to the existing IP-based User Datagram Protocol-to-Reliable Link Manager (UDP-to-RLM) transport between the Cisco PGW2200 and Cisco gateways. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftgkrup.htm
T1 Channel Associated Signaling (CAS)
Channel Associated Signaling (CAS) is the transmission of signaling information within the voice channel. Support for CAS is now available on T1 interfaces.
T.37 for Cisco 7200
This feature adds T.37 standards-based store-and-forward fax protocol support for H.323 gateways and gatekeepers to the Cisco 7200 series. T.37 is an ITU-T recommended standard for store-and-forward fax that enables Cisco gateways and gatekeepers to interwork with other Cisco gateways and third-party H.323 devices that support the T.37 protocol.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm
T.37 store-and-forward fax was originally supported in Cisco IOS Release 12.1(5)T on the Cisco AS5300 platform. In Cisco IOS Release 12.2(8)T, support was added on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745. In Cisco IOS Release 12.2(13)T, support was added on the Cisco AS5350 and the Cisco AS5400. Cisco IOS Release 12.2(15)T adds support on the Cisco 7200 series.
Tokenless Call Authorization
The Tokenless Call Authorization feature provides a statically configured access list of authorized H.323 endpoints for the Cisco IOS gatekeeper. The gatekeeper accepts calls from endpoints on the list. This security feature is an alternative to Interzone ClearTokens (IZCTs) and Cisco Access Tokens (CATs), and can be used with Cisco CallManager. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_tklss.htm
Tunneled GR-303 for the Cisco Cable Modem
The Tunneled GR-303 Support feature enables the Cisco uBR925 cable access router to send and receive call control messages using GR-303 signaling, in addition to the Media Gateway Control Protocol (MGCP) signaling that was previously supported. This allows the Cisco uBR925 router to support advanced call features such as caller ID and call waiting, using both GR-303 and MGCP signaling. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/dtgrmgcp.htm
UDP Forwarding Support of IP Redundancy Virtual Router Group (VRG)
User Datagram Protocol (UDP) forwarding is used in Cisco IOS software to forward broadcast and multicast packets received for a specific IP address. Virtual Router Group (VRG) support is currently implemented with the Hot Standby Routing Protocol (HSRP), and it allows a set of routers to be grouped as a logical router that answers to a well known IP address. The UDP Forwarding Support of IP Redundancy Virtual Router Group (VRG) feature enables UDP forwarding to be VRG aware, resulting in forwarding only to the active router in the VRG. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftudpvrg.htm
V.92 and V.44 Support for Digital Modems
The V.92 and V.44 Support for Digital Modems feature supports the V.92 Modem on Hold and V.92 Quick Connect portions of the new V.92 modem standard, and the new V.44 LZJH compression standard based on Lempel-Ziv, on the Cisco 3600 and Cisco 3700 series router platforms. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122yt/122yt11/ftv92_44.htm
VRF-Aware IPSec
The VRF-Aware IPSec feature introduces IP Security (IPSec) tunnel mapping to Multiprotocol Label Switching (MPLS) Virtual Private Networks (VPNs). Using the VRF-Aware IPSec feature, you can map IPSec tunnels to virtual routing and forwarding (VRF) instances using single public-facing addresses.
A VRF instance is a per-VPN routing information repository that defines the VPN membership of a customer site attached to the provider edge (PE) router. A VRF comprises an IP routing table, a derived Cisco Express Forwarding (CEF) table, a set of interfaces that use the forwarding table, and a set of rules and routing protocol parameters that control the information that is included in the routing table. A separate set of routing and CEF tables is maintained for each VPN customer.
The MPLS distribution protocol is a high-performance packet-forwarding technology that integrates the performance and traffic management capabilities of data link layer switching with the scalability, flexibility, and performance of network-layer routing. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ft_vrfip.htm
XML Interface to Syslog Messages
The Cisco IOS system logging (Syslog) process allows the system to report and save important error messages, either locally or to a remote logging server. These Syslog messages include system error messages and debugging output sent during network operation to assist users and Cisco TAC engineers with identifying the type and severity of a problem. Syslog messages can be sent to the console, a monitor (TTY), a buffer, or a remote host.
The XML Interface to Syslog Messages features provides Command Line Interface (CLI) commands for enabling syslog messages to be sent in an XML format. XML (Extensible Markup Language), a derivative of SGML, provides a representation scheme to structuralize consistently formatted data such as that found in Syslog messages. This feature defines a closed set of meaningful XML tags for Syslog messages. Logs in a standardized XML format can be more readily used in external customized monitoring tools. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t15/ftxmlsys.htm
New Hardware Features Supported in Cisco IOS Release 12.2(13)T
The following new hardware features are supported in Cisco IOS Release 12.2(13)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
Catalyst 4224 Access Gateway Switch
The Cisco Catalyst 4224 Access Gateway Switch (Catalyst 4224) is an integrated switch/router that provides Voice over IP (VoIP) gateway and IP telephony services to a small branch office. The Cisco Catalyst 4224 provides an integrated switch and WAN/voice gateway for enterprise satellite offices with up to 24 users. It is intended to work in conjunction with a Cisco Call Manager cluster from the central site with fail over capabilities to allow local calls and basic PBX features.
For information about Cisco Catalyst 4224 configuration, see the following:
http://www.cisco.com/univercd/cc/td/doc/product/lan/cat4224/index.htm
Cisco 3631 Router Enhanced Functionality
In Cisco IOS Release 12.2(13)T, the Cisco 3631 will support additional functionality. Beginning in this release, this router will support the following interfaces:
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For more information about network module configuration, see the following:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis2600/hw_inst/nm_inst/nm-doc/index.htm
For more information about WAN interface card (WIC) configuration, see the following:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis2600/hw_inst/wic_inst/wic_doc/index.htm
Cisco 3725 Router, Cisco 3745 Router, Cisco 2691 Router Enhanced Functionality
In Cisco IOS Release 12.2(13)T, the Cisco 3725, Cisco 3745, and Cisco 2691 routers will support additional functionality. Beginning in this release, these routers will support the following interfaces:
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For more information about network module configuration, see the following:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis2600/hw_inst/nm_inst/nm-doc/index.htm
For more information about WAN interface card (WIC) configuration, see the following:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis2600/hw_inst/wic_inst/wic_doc/index.htm
Cisco 7401 ASR-BB and Cisco 7401 ASR-CP
The Cisco 7401 ASR-BB and Cisco 7401 ASR-CP are now supported on Cisco IOS Release 12.2T.
Content Engine Network Module for Caching and Content Delivery
The Content Engine (CE) Network Module for Caching and Content Delivery offers the ability to integrate the features of Cisco Application and Content Networking System (ACNS) software into branch office platforms. The CE network module combines the Content Caching, Content Filtering and Content Delivery features of ACNS with robust branch office routing and is supported on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers.
The CE network module can operate as a stand-alone cache or in an integrated enterprise content delivery network (E-CDN) environment. As one element of an E-CDN, the CE network module can be deployed with a combination of other content engines, content routers, content services switches, and content distribution managers to create a complete content delivery network system.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_1cenm.htm
PA-MC-8TE1+
The Cisco PA-MC-8TE1+ is a single-wide port adapter designed to provide a full eight-port PRI multichannel solution for the Cisco 7200 and Cisco 7400. The interfaces can be channelized, fractional or ISDN-PRI, or unframed (E1) with up to 256 independent HDLC channels definable for T1 and E1 applications.
SRP MIB for DPT-OC12 WAN Card
This feature provides the SRP MIB for PA-SRP-OC12xx and SRPIP-OC12xx cards for the Cisco 7200 and Cisco 7500 series routers.
Unchannelized support for PA-MC-2T3+ port adapter
The PA-MC-2T3+ is a single-width port adapter that provides two T3 interface connections. Each T3 interface can now be independently configured to be either channelized or unchannelized. A channelized T3 provides 28 T1 lines multiplexed into the T3. Each T1 line can be configured into one or more serial interface data channels.
Using the no channelized command, you can configure the T3 as a single, unchannelized serial interface data channel. You can configure this data channel to use all of the T3 bandwidth or a portion of it.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121e/121e5/5e_ct3.htm
Update to the Enhancements for the Cisco Voice Gateway 200
The Enhancements for the Cisco Voice Gateway 200 (Cisco VG200) feature provides the Cisco VG200 platform (also called CAG-VG200) with increased voice gateway feature parity to the Cisco 2600, Cisco 3600, and Cisco 3700 platforms. This update provides additional feature functionality on the Cisco VG200 platform.
The Cisco VG200 platforms provide the following default memory options: CAG-VG200—16 MB of Flash, 64 MB of DRAM
VPN Accelerator Module (VAM)
The VPN Acceleration Module (VAM) is a single-width acceleration module. It provides high-performance, hardware-assisted tunneling and encryption services suitable for virtual private network (VPN) remote access, site-to-site intranet, and extranet applications. It also provides platform scalability and security while working with all services necessary for successful VPN deployments — security, quality of service (QoS), firewall and intrusion detection, service-level validation, and management. The VAM off-loads IPSec processing from the main processor, thus freeing resources on the processor engines for other tasks.
The VAM provides hardware-accelerated support for multiple encryption functions:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122ye/1229ye/12ye_vam.htm
WIC-1-B-U-V2
Beginning in this release, the model number for the existing WIC-1-B-U interface card for the Cisco 1700 series, Cisco 2600 series, and Cisco 3600 series is changing to WIC-1-B-U-V2.
In addition, this interface card will now be supported on the Cisco 1760, Cisco 2691, Cisco 3725 and Cisco 3745 beginning with this release.
New Software Features in Cisco IOS Release 12.2(13)T
The following new features are supported in Cisco IOS Release 12.2(13)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
Advanced Encryption Standard (AES)
The Advanced Encryption Standard (AES) feature adds support for the new encryption standard AES, with CBC (Cipher Block Chaining) Mode, to IP Security (IPSec).
The National Institute of Standards and Technology (NIST) has created AES, which is a new Federal Information Processing Standard (FIPS) publication that describes an encryption method. AES is a privacy transforms for IPSec and Internet Key Exchange (IKE) and has been developed to replace the Data Encryption Standard (DES). AES is designed to be more secure than DES: AES offers a larger key size, while ensuring that the only known approach for an intruder to decrypt a message is to try every possible key. AES has a variable key length—the algorithm can specify a 128-bit key (the default), a 192-bit key, or a 256-bit.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_aes.htm
Analog DID (Direct Inward Dial)
Analog Direct Inward Dial (DID) is now supported on Cisco 1700 series routers.
Apollo Domain
The Apollo Domain networking protocol will no longer be offered after Cisco IOS Release 12.2. Apollo Domain commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
AppleTalk EIGRP
The AppleTalk Enhanced Interior Gateway Routing Protocol (Enhanced IGRP) will no longer be offered after Cisco IOS Release 12.2(13)T. AppleTalk EIGRP commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
ATM Multilink PPP Support on Multiple VCs
The ATM Multilink PPP Support on Multiple VCs feature supports the transport of real-time (voice) and other (data) traffic on Frame Relay and ATM virtual circuits (VCs).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftatmmlt.htm
ATM Policing by Service Category for SVC/SoftPVC
When configured, an ATM switch at the network side of a user-to-network (UNI) interface polices the flow of cells in the forward (into the network) direction of a virtual connection. These traffic policing mechanisms are known as usage parameter control (UPC). With UPC, the switch determines whether received cells comply with the negotiated traffic management values and takes one of the following actions on violating cells:
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The ATM Policing by Service Category for SVC/SoftPVC feature enables you to specify which traffic to police, based on service category, on switched virtual circuits (SVCs) or terminating VCs on the destination end of a soft VC.
For more information on UPC, refer to the "Traffic and Resource Management" chapter in the Guide to ATM Technology.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_4/svc_upc.htm
ATM Subinterface MIB/Traps
This feature adds support for the monitoring of ATM and Frame Relay (FR) subinterface status using SNMP. New CLI commands allow the enabling or disabling of ATM and Frame Relay notifications (traps and informs), and provide an option for limiting the rate of notifications sent ("trap throttling").
Automatic Protection Switching (APS)
This feature allows switchover of packet-over-SONET (POS) circuits in the event of circuit failure and is often required when connecting SONET equipment to telco equipment.
Banyan VINES
The Banyan Virtual Network System (VINES) protocol will no longer be offered after Cisco IOS Release 12.2(13)T. Banyan VINES commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
BGP 4 MIB Support for per-Peer Received Routes
BGP 4 MIB Support for per-Peer Received Routes introduces a new table in the CISCO-BGP4-MIB that provides the capability to query (by using Simple Network Management Protocol [SNMP] commands) for routes that are learned from individual Border Gateway Protocol (BGP) peers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftbgpmib.htm
BGP Policy Accounting
Border Gateway Protocol (BGP) policy accounting measures and classifies IP traffic that is sent to, or received from, different peers. Policy accounting is enabled on an input interface, and counters based on parameters such as community list, autonomous system number, or autonomous system path are assigned to identify the IP traffic.
Using the BGP table-map command, prefixes added to the routing table are classified by BGP attribute, autonomous system number, or autonomous system path. Packet and byte counters are incremented per input interface. A Cisco IOS policy-based classifier maps the traffic into one of eight possible buckets, representing different traffic classes.
Using BGP policy accounting, you can account for traffic according to the route it traverses. Service providers (SPs) can identify and account for all traffic by customer and bill accordingly. Customers are billed appropriately for traffic that is routed from a domestic, international, or satellite source.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_bgppa.htm
Bisync-to-IP Conversion for Automated Teller Machines
The Bisync-to-IP Conversion for Automated Teller Machines feature enables customers to attach a binary synchronous communication (bisync) automated teller machine to a serial interface on a Cisco router running bisync-to-IP (BIP) protocol translation, and then to route the data over a TCP/IP network directly to an IP-based application host.
As of Cisco IOS Release 12.2(13)T you can use the bstun peer-map-poll command in global configuration mode to map the ATM state to polling. The default is to not map the peer state to polling. If you configure this command, BIP activates polling when the BIP tunnel becomes active and stops polling when the tunnel connection is terminated. When the peer state-to-polling is not mapped, BIP waits for the host to issue an "active" status message across the BIP tunnel before polling the ATM device and polling is stopped when an "inactive" status message is received across the tunnel or the tunnel connection is terminated.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftbipatm.htm
Call Admission Control for H.323 VoIP Gateways
Before the call admission control feature, gateways did not have a mechanism to gracefully prevent calls from entering when certain resources were not available to process the call. This causes the new call to fail with unreported behavior, and could potentially cause the calls that are in progress to have quality related problems.
This feature set provides the ability to support resource-based call admission control processes. These resources include system resources such as CPU, memory, and call volume, and interface resources such as call volume.
If system resources are not available to admit the call, two kinds of actions are provided: system denial (which busyouts all of T1 or E1) or per call denial (which disconnects, hairpins, or plays a message or tone). If the interface-based resource is not available to admit the call, the call is dropped from the session protocol (such as H.323).
This feature was previously released in Cisco IOS Release 12.2(4)T on the Cisco 2600 and Cisco 3600 routers, and Cisco MC3810 multiservice concentrators. This release is porting the feature into the IAD2420 platform.
Call Release Source Reporting in Gateway-Generated Call Accounting Records
The Call Release Source Reporting in Gateway-Generated Call Accounting Records feature enables you to track the source of call release in a Voice over IP (VoIP) network. This call release information defines whether a call was released by the calling or called party or by an internal or external source.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_crsr.htm
CEF and Distributed CEF Switching for IPv6
Cisco Express Forwarding for IPv6 (CEFv6) is advanced, Layer 3 IP switching technology for the forwarding of IPv6 packets. Distributed CEF for IPv6 (dCEFv6) performs the same functions as CEFv6 but for distributed architecture platforms such as the Cisco 12000 series Internet routers and the Cisco 7500 series routers. dCEFv6 and CEFv6 function the same and offer the same benefits as dCEFv4 and CEFv4—network entries that are added, removed, or modified in the IPv6 Routing Information Base (RIB), as dictated by the routing protocols in use, are reflected in the Forwarding Information Bases (FIBs), and the IPv6 adjacency tables maintain Layer 2 next-hop addresses for all entries in each FIB.
CEFv6 was introduced in Cisco IOS Release 12.2(13)T for nondistributed architecture platforms, such as the Cisco 7200 series routers. dCEFv6 was introduced in Cisco IOS Release 12.0(21)ST for the Cisco 12000 series Internet routers, and was then integrated into Cisco IOS Release 12.2(13)T and later releases for other distributed architecture platforms, such as the Cisco 7500 series routers.
In Cisco IOS Release 12.0(21)ST, dCEFv6 included support for IPv6 addresses and prefixes. In Cisco IOS Release 12.2(13)T or later releases, dCEFv6 and CEFv6 were enhanced to include support for separate FIBs for IPv6 global, site-local, and link-local addresses.
Cisco Conferencing and Transcoding for Voice Gateway Routers
The feature enables voice conferencing to take place among conferees at small, remote branch offices or distributed sites using local resources, without calls having to traverse the company WAN to the central site that supports such services.
The feature also provides transcoding at the remote site. Different IP telephony devices support different codecs and, for communications to be enabled between them, transcoding is required. The feature provides transcoding at the remote site, without having to access transcoding services at the central site.
To provide these services, the feature takes advantage of unused DSP resources on a network module in an already existing small or midsize Cisco router at the remote site. The collection of DSP resources so made available is called a DSP farm. The DSP farm is managed by Cisco CallManager, the software-based call-processing component of the Cisco IP telephony solution, at a central office or branch office.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftdsp.htm
Cisco IOS Software Feature Removal
This feature permanently removes selected legacy features (or components) from the Cisco IOS code. These features will not be available in future releases of Cisco IOS software.
The features that have been removed in the 12.2(13)T release are as follows:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
Cisco IOS Telephony Service (ITS) Version 2.02
The new feature for Cisco IOS Telephony Service (ITS) Version 2.02 is an increase in directory numbers from 192 to 288 for the following platforms:
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The Cisco IOS Telephony Service V2.02 Feature Guide is located at the following location:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_feature_guides_list.html
Cisco Mobile Networks—Asymmetric Link
An asymmetric link environment such as satellite communications, with a separate uplink and downlink, provides challenges for the mobile router and foreign agent.Because each unidirectional link provides only one way traffic, the inherent mapping in the foreign agent of the return path to the mobile router for incoming messages does not apply. The Cisco Mobile Networks—Asymmetric Link feature solves this problem by extending the use of mobile networks to networks where the mobile router has unidirectional links to the foreign agent. The foreign agent is able to transmit packets back to the mobile router over a different link than the one on which it receives packets from the mobile router.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/asymmetr.htm
Cisco Mobile Networks—Dynamic Network Support
The Cisco Mobile Networks feature enables a mobile router and its subnets to be mobile and maintain all IP connectivity, transparent to the IP hosts connecting through this mobile router. Previously, this feature was a static network implementation that supported stub routers only.
Cisco IOS Release 12.2(13)T introduces dynamic network support, which means that the mobile router dynamically registers its mobile networks to the home agent, which reduces the amount of configuration required at the home agent. For example, if a home agent supports 2000 mobile routers, the home agent does not need 2000 configurations but only a range of home IP addresses to use for the mobile routers.This registration results in minimal configuration on the home agent making administration and set up easier.
Cisco Survivable Remote Site Telephony Service V2.02
The new feature for Cisco Survivable Remote (SRS) Telephony V2.02 is Unity Voice Mail integration, which introduces six new commands:
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For further information, see Cisco IOS Telephony Service V2.02 at the following location:
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/srs/index.htm
Class-Based RTP and TCP Header Compression
Real-time Transport Protocol (RTP) or Transmission Control Protocol (TCP) IP header compression is typically configured at the interface level. However, this feature now allows you to configure RTP or TCP IP header compression on a per-class basis, when a class in configured within a policy map. Policy maps are created using the Modular Quality of Service Command-Line Interface (MQC).
Thus, this feature extends the functionality of the MQC and allows you to configure and fine-tune IP header compression at a more granular level.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/fthdrcmp.htm
Clearable SIP-UA Statistics
This feature provides the ability to clear all Session Initiation Protocol (SIP) statistics counters that are displayed by the show sip-ua statistics command, which includes response, traffic and retry statistics. Prior to the implementation of the new feature, SIP counters could be cleared only by reloading or resetting the router. The new feature enhances both trouble-shooting and statistical analysis efforts by clearing SIP counters without reloading or resetting the router.
The new feature includes the following functionality:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftshadow.htm
Committed Access Rate (CAR)
Committed Access Rate (CAR) can rate limit traffic based on certain matching criteria, such as incoming interface, IP Precedence, or IP access list.
Connection-Oriented Media (Comedia) Enhancements for SIP
This feature provides the following functionality to symmetric Network Address Translation (NAT) traversal:
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The new feature implements one of many possible SIP solutions to address problems with different NAT types and traversals.With the Connection-Oriented Media (Comedia) Enhancements for SIP feature, the gateway can open an RTP session with the remote end and then update or modify the existing RTP session's remote address and port (raddr:rport) with the source address and port of the actual media packet received after passing through NAT.
Dial-Peer Support for Data Calls
The Dial-Peer Support for Data Calls feature enables the configuration and order assignment of dial peers so that a gateway can identify incoming calls as voice or data. The feature provides a unified call processing model that is scalable for voice and data calls through dial-peer provisioning. The feature also enables the capability of assigning separate number ranges for voice or data calls so that the calls will have the same preference level of matching.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftconcrt.htm
Distributed IPv6 for Cisco IOS software
This feature provides distributed CEF switching support for IPv6 on the Cisco 7500 platforms.
DLR Enhancements: PGM RFC-3208 Compliance
In compliance with RFC 3208, the DLR Enhancements feature adds off-tree designated local repairer (DLR) support and redirecting poll response (POLR) capability for upstream DLRs to the Cisco implementation of Pragmatic General Multicast (PGM).
Dual Serial Line Management to Interface Lucent 5ESS
This feature is a part of the Cisco IOS Telco Feature Set, a bundle of applications specific to the data communications network (DCN) environment. Specifically, this feature supports X.25-to-TCP protocol translation, and provides dual serial interfaces to preserve the redundancy and monitoring capability available from SCC0 and SCC1 links on a Lucent 5ESS switch in the DCN network.
Dynamic Multipoint VPN (DMVPN)
The Dynamic Multipoint VPN (DMVPN) feature combines GRE tunnels, IPSec encryption, and NHRP routing to provide users an ease of configuration via crypto profiles, which override the requirement for defining static crypto maps, and dynamic discovery of tunnel endpoints.
This feature relies on the following two Cisco technologies—NHRP and mGRE Tunnel Interface.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftgreips.htm
Dynamic Subscriber Bandwidth Selection
The Dynamic Subscriber Bandwidth Selection (DBS) feature enables wholesale service providers to sell different classes of service to retail service providers by controlling bandwidth at the ATM Virtual Circuit (VC) level. ATM Quality of Service (QoS) parameters from the subscriber domain are applied to the ATM PVC on which a PPPoE or PPPoA session is established.
Using DBS you can set the ATM permanent virtual circuit (PVC) traffic shaping parameters to be dynamically changed based on the RADIUS profile of a PPP over Ethernet (PPPoE) or PPP over ATM (PPPoA) user logging in on the PVC. If the user is the first user on that PVC, then the RADIUS profile values override the default values of the PVC. If users already exist on the PVC, then the new value overrides the existing configuration only if it is higher than the existing value. If multiple PPPoE sessions are allowed on a subscriber VC, then the highest peak cell rate (PCR) and sustainable cell rate (SCR) of all the sessions is selected as the PCR and SCR of the VC.
You can apply DBS QoS parameters per user as well as per domain. If you apply DBS QoS parameters under a domain profile, all users in that profile are assigned the same DBS QoS parameters. These parameters are assigned to the RADIUS profile for that domain. You can also apply distinctive DBS QoS parameters via the RADIUS user profile.
Traffic shaping parameters can be locally configured by IOS CLI in VC-mode, VC-class, range mode, or PVC-in-range mode. These parameters have a lower priority and are overridden by the shaping parameters specified in the domain service profile. Traffic shaping parameters that are CLI configured at the VC class interface or subinterface level are treated as the default QoS parameters for the PVCs to which they apply. These parameters are overridden by the domain service profile QoS parameters of the domain the user is logged in to. If no VC class is configured, the default is the unspecified bit rate (UBR).
When a network access server (NAS) sends a domain authorization request and receives an affirmative response from the RADIUS server, this response may include a "QoS-management" string via vendor-specific attribute (VSA) 26 for QoS management in the NAS. The QoS management values are configured as part of the domain service profile attributes on the RADIUS server. These values contain PCR and SCR values for a particular user or domain. If the QoS specified for a domain or user cannot be applied on the PVC that the session belongs to, the session is not established.
Changing PVC traffic parameters because of new simultaneous PPPoE sessions on the PVC does not cause existing PPPoE sessions that are already established to disconnect. Changing domain service profile QoS parameters on the RADIUS server does not cause traffic parameters to automatically change for PVCs that have existing sessions.
When you enter the dbs enable or no dbs enable commands to configure or unconfigure DBS, existing sessions are not disconnected. If you have a session that has been configured for DBS and you configure the no dbs enable command on a VC, additional sessions that are configured will display DBS configured QoS values until the first new session is up. After the first session is brought up, the VC has default and locally configured values. If you configure the dbs enable command after multiple sessions are already up on the VC, all sessions on that VC have DBS QoS parameters.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftdbs.htm
Enhanced Features for Local and Advanced Voice Busyout
This feature introduces 2 new commands, busyout monitor gatekeeper and busyout action graceful. The busyout monitor gatekeeper command busies out the gatekeeper if the gateway loses connection to the primary gatekeeper and removes the busyout state when the gateway restores connection to the primary or backup gatekeeper. The busyout action graceful command controls the busyout behavior that is triggered by the busyout monitor command. This command busies out the voice port immediately if the busyout behavior is triggered but if there is an active call on this voice port it will wait until the call is over.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_lavbo.htm
Enhanced ITU-T G.168 Echo Cancellation
This feature provides an alternative to the default, Cisco proprietary 32-millisecond G.165 echo canceller (EC). The new extended echo canceller provides improved performance for trunking gateway applications and provides a configurable tail length that supports up to 64 milliseconds (ms) of echo cancellation. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftecho.htm
Enhanced Packet Marking
The Enhanced Packet Marking feature allows you to map and convert the marking of a packet from one value to another (for example, the Precedence value can be mapped to the equivalent Class of Service (CoS) value) by using a kind of conversion chart called a table map.
The table map establishes an equivalency from one value to another. For example, the table map can map the CoS value of a packet to the Precedence or differentiated services code point (DSCP) value of the packet. For networks using MPLS, the MPLS EXP value can be mapped to the QoS group value, which can then be mapped to the Precedence or DSCP value of the packet. This value mapping can be propagated for use on the network, as needed. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftenpkmk.htm
Enhancements for the Cisco Voice Gateway 200
The Enhancements for the Cisco Voice Gateway 200 (VG200) feature provides the Cisco VG200 platform with increased voice gateway feature parity to the Cisco 2600, Cisco 3600, and Cisco 3700 platforms. This update provides additional feature functionality on the Cisco VG200 platform. Refer to the following document for additional information:
Exterior Gateway Protocol (EGP)
The Exterior Gateway Protocol (EGP) will no longer be offered after Cisco IOS Release 12.2(13)T. EGP commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
File System Check and Repair for PCMCIA ATA Disks
This feature introduces a File-System-Check (fsck) utility in Cisco IOS software for FAT file systems on PCMCIA disks. The utility performs functions such as checking the boot sector and partition table, checking file and directory structure, reclaiming unused disk space, and updating the FAT file structure. Prior to the introduction of this utility, corrupt files could not be removed from ATA disks using the Cisco IOS CLI. This utility is run using the fsck privileged EXEC mode command.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_fsck.htm
Frame Relay PVC Bundles with IP and MPLS QoS Support
Frame Relay PVC bundles allow you to associate a group of Frame Relay permanent virtual circuits (PVCs) with a single next-hop address. When Frame Relay PVC bundles are used with IP, packets are mapped to specific PVCs in the bundle on the basis of the precedence value or differentiated services code point (DSCP) settings in the type of service (ToS) field of the IP header. Each packet is treated differently according to the QoS configured for each PVC.
Frame Relay PVC bundles with MPLS QoS support extends Frame Relay PVC bundle functionality to support the mapping of Multiprotocol Label Switching (MPLS) packets to specific PVCs in the bundle. MPLS packets are mapped to PVCs according to the settings of the experimental (EXP) bits in the MPLS packet header.Waiting for information.
Frame Relay Queueing and Fragmentation at the Interface
The Frame Relay Queueing and Fragmentation at the Interface feature introduces support for low-latency queueing (LLQ) and FRF.12 end-to-end fragmentation on a Frame Relay interface. This new feature simplifies the configuration of low-latency, low-jitter quality of service (QoS) by enabling the queueing policy and fragmentation configured on the main interface to apply to all permanent virtual circuits (PVCs) and subinterfaces under that interface. Before the introduction of this feature, queueing and fragmentation had to be configured on each individual PVC. Subrate shaping can also be configured on the interface.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/frfrintq.htm
H.323 Call Redirection Enhancements
The user-to-user information element (UUIE) of the Facility message is used primarily for call redirection. The UUIE contains a field, facilityReason, that indicates the nature of the redirection. The H.323 Call Redirection Enhancements feature adds support for two of the reasons: routeCallToGatekeeper and callForwarded. It also provides a non-standard method for using the Facility message to effect call transfer.
This feature was previously released in Cisco IOS Release 12.2(2)T on Cisco 1700, Cisco 2600 series, Cisco 3600 series, Cisco MC3810, Cisco AS5300, Cisco uBR924 platforms. This release is porting the feature into the IAD2420 platform.
H.323 Dual Tone Multifrequency Relay Using Named Telephone Events
The NTE method of DTMF relay was originally available on Cisco gateways only for Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP) gateways. The H.323 DTMF Relay Using Named Telephone Events (NTE) feature adds support for this method for H.323 gateways.
Cisco H.323 gateways advertise capabilities using the H.245 capabilities messages. By default, they advertise that they can receive all DTMF relay modes. If the capabilities of the remote gateway do not match, the Cisco H.323 gateway transmits DTMF tones as in-band voice. Configuring DTMF relay on the Cisco H.323 gateway sets preferences for how the gateway handles DTMF transmission. If multiple methods are configured, the priority is as follows:
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In addition to support for NTE, the H.323 DTMF Relay Using NTE feature provides support for asymmetrical payload types. Payload types can differ between local and remote endpoints. Therefore, the Cisco gateway can transmit one payload type value and receive a different payload type value.
This feature was previously released in Cisco IOS Release 12.2(11)T on Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 platforms. This release is porting the feature into the IAD2420 platform.
H.323 (Gateway) Support
Support for H.323 Version 2 Gateway functionality is added to the Cisco IAD2420 series of integrated access devices (IADs). This support provides the Cisco IAD2420 IAD with more market opportunities than when the IAD only supported MGCP and SGCP call control protocols.
The Cisco IAD2420 IAD with 16 FXS analog ports delivers local voice and data service using VoIP in an Ethernet To The Building (ETTx) application. It aggregates the voice traffic from multiple tenants and transports it to an Ethernet switch, such as the Cisco 2950, over the Ethernet link. The built-in WAN interface (either a T1, ADSL or SHDSL module) is not used when using the IAD2420-16FXS.
H.323 Redundant Zone Support
The Redundant H.323 Zone Support feature allows users to configure multiple gatekeepers to service the same zone or technology prefix. This feature can be used with the Gateway Support for Alternate Gatekeepers feature, which allows a user to configure a gateway to point to two gatekeepers (one as the primary and the other as the alternate). Together, these features allow a user to configure a Cisco gateway to send location requests (LRQs) to two or more Cisco gatekeepers---one as a primary and the others as back up gatekeepers.
This feature was previously released in Cisco IOS Release 12.1(1)T on Cisco 2600 series, Cisco 3600 series, Cisco MC3810, Cisco AS5200, Cisco AS5300, and Cisco AS5800 platforms. This release is porting the feature into the Cisco IAD2420 platform.
H.323 Scalability and Interoperability Enhancements
The Cisco H.323 Scalability and Interoperability Enhancements feature upgrades the Cisco H.323 Gatekeeper and Cisco H.323 Gateway to comply with H.323 Version 3. The enhancements in this release include support for mandatory H.323 Version 3 elements in the gateway, support for H.225 call signalling over UDP, and address resolution using border elements.
For gatekeeper support, this feature was previously released in Cisco IOS Release 12.2(4)T on Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, Cisco MC3810, Cisco AS5850, and Cisco 7200 series platforms. For gateway support, this feature was previously released in Cisco IOS Release 12.2(4)T on Cisco 1700, Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5850, Cisco uBR900 series, and Cisco uBR924 platforms. This release is porting the feature into the IAD2420 platform.
H.323 Support for Virtual Interfaces
The H.323 Support for Virtual Interfaces feature allows users to configure the IP address of the gateway, so that the IP address include in the H.323 packet is deterministic and consistently indicates the same address for the source.
In previous releases of the Cisco IOS software, the source address included in the H.323 packet could vary depending on the protocol (RAS, H.225, H.245, or RTP). This makes it difficult to configure firewall applications to work with H.323 messages.
The H.323 Support for Virtual Interfaces feature addresses that difficulty by allowing the user to explicitly configure an IP address to be used for all protocols
This feature was previously released in Cisco IOS Release 12.1(2)T on Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco 7200 series, and Cisco uBR924 platforms. This release is porting the feature into the IAD2420 platform.
HP Probe
The HP Probe feature will no longer be offered after Cisco IOS Release 12.2(13)T. HP Probe commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
Interim Local Management Interface (ILMI)
The Interim Local Management Interface (ILMI) is a protocol defined by the ATM Forum for setting and capturing physical layer, ATM layer, virtual path, and virtual circuit parameters on ATM interfaces. ILMI uses simple network management protocol (SNMP) messages without User Datagram Protocol (UDP) and IP, and organizes managed objects into the following four management information bases (MIBs).
Interim-Interswitch Signaling Protocol (IISP)
The Interim-Interswitch Signalling Protocol (IISP) defines a static routing protocol (using manually configured prefix tables) for communication between ATM switches. IISP provides support for switched virtual circuits (SVCs) on ATM switches that do not support the Private Network-to-Network Interface (PNNI) protocol.
Interim Update at Call Connect
With this feature, Cisco IOS software generates and sends an additional updated interim accounting record to the accounting server when a call leg is connected. All attributes (for example, h323-connect-time and backward-call-indicators) available at the time of call connection are sent through this interim updated accounting record. Refer to the following document for additional information:
Interior Gateway Routing Protocol (IGRP)
The Interior Gateway Routing Protocol (IGRP) will no longer be offered after Cisco IOS Release 12.2(13)T. IGRP commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
IP Event Dampening
Interface state changes occur when interfaces are administratively brought up or down or if an interface changes state. When an interface changes state or flaps, routing protocols are notified of the status of the routes that are affected by the change in state. Every interface state change requires all affected devices in the network to recalculate best paths, install or remove routes from the routing tables, and then advertise valid routes to peer routers. An unstable interface that flaps excessively can cause other devices in the network to consume substantial amounts of system processing resources and cause routing protocols to lose synchronization with the state of the flapping interface.
The IP Event Dampening feature introduces a configurable exponential decay mechanism to suppress the effects of excessive interface flapping events on routing protocols and routing tables in the network. This feature allows the network operator to configure a router to automatically identify and selectively dampen a local interface that is flapping. Dampening an interface removes the interface from the network until the interface stops flapping and becomes stable. Configuring the IP Event Dampening feature improves convergence times and stability throughout the network by isolating failures so that disturbances are not propagated, which reduces the utilization of system processing resources by other devices in the network and improves overall network stability.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftipevdp.htm
IPSec NAT Transparency
Before the introduction of the IPSec NAT Transparency feature, a standard IPSec VPN tunnel would not work if there were one or more NAT or PAT points in the delivery path of the IPSec packet. This feature introduces support for IPSec traffic to travel through NAT or PAT points in the network by encapsulating IPSec packets in a User Datagram Protocol (UDP) wrapper, thereby, allowing remote access users to build IPSec tunnels to home gateways.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftipsnat.htm
IPSec Passive Mode
The IPSec Passive mode feature allows users to configure an intermediate mode—IPSec passive mode—that enables routers within an existing network to accept encrypted and unencrypted data. The routers will also attempt to negotiate an encrypted session when sending data, but they will send the data in unencrypted form as necessary.
IPSec passive mode is valuable for users who wish to migrate existing networks to IPSec because they no longer have wait for all routers to deploy IPSec; that is, all routers will continue to interact with routers that will encrypt data (that have been upgraded with IPSec) and routers that have yet to be upgraded.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftpasips.htm
Note
IPv6 ADSL and Dial Deployment Support
The IPv6 ADSL and Dial Deployment Support feature adds support for IPv6 prefix pools, and per-user IPv6 Radius attributes. It further enables deployment of IPv6 in DSL and dial access environments. This feature provides the extensions that make large scale IPv6 access possible for IPv6 environments, including IPv6 Radius attributes, stateless address configuration on PPP links, per-user static routes, and access lists (ACLs). Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ipv6/index.htm
IPv6 Extended Access Control Lists
In Cisco IOS Release 12.2(2)T or later releases, 12.0(21)ST, and 12.0(22)S, standard IPv6 access control list (ACL) functionality is used for basic traffic filtering functions—traffic filtering is based on source and destination addresses, inbound and outbound to a specific interface, and with an implicit deny statement at the end of each access list (functionality similar to standard ACLs in IPv4). IPv6 ACLs are defined and their deny and permit conditions are set by using the ipv6 access-list command with the deny and permit keywords in global configuration mode.
In Cisco IOS Release 12.2(13)T or later releases, and 12.0(23)S, the standard IPv6 ACL functionality is extended to support—in addition to traffic filtering based on source and destination addresses—filtering of traffic based on IPv6 option headers and optional, upper-layer protocol type information for finer granularity of control (functionality similar to extended ACLs in IPv4). IPv6 ACLs are defined by using the ipv6 access-list command in global configuration mode and their permit and deny conditions are set by using the deny and permit commands in IPv6 access list configuration mode. (Configuring the ipv6 access-list command places the router in IPv6 access list configuration mode, from which permit and deny conditions can be set for the defined IPv6 ACL.)
IPv6 Quality of Service
This feature provides for the application of all the Differentiated Services (DiffServ) QoS features to IPv6 packets. Specific QoS features include packet classification, traffic shaping, traffic policing, packet marking, and Drop based on Weighted Random Early Detect (WRED) on all applicable interfaces.
IPv6 RIP Enhancements
The IPv6 RIP Enhancements feature adds support for a separate IPv6 RIP routing table, the ability to delete routes from the IPv6 RIP routing table, and the ability to set route tags. The holddown timer default is now set to zero, and a maximum number of parallel routes can be configured.
IS-IS HMAC-MD5 Authentication
The IS-IS HMAC-MD5 Authentication feature adds an HMAC-MD5 digest to each Intermediate System-to-Intermediate System (IS-IS) protocol data unit (PDU). HMAC is a mechanism for message authentication codes (MAC) using cryptographic hash functions. The digest allows authentication at the IS-IS routing protocol level, which prevents unauthorized routing messages from being injected into the network routing domain.
IS-IS has five packet types: link-state packet (LSP), LAN Hello, Serial Hello, complete sequence number PDU (CSNP), and partial sequence number PDU (PSNP). The IS-IS HMAC-MD5 authentication or the cleartext password authentication can be applied to all five types of PDU. The authentication can be enabled on different IS-IS levels independently. The interface-related PDUs (LAN Hello, Serial Hello, CSNP and PSNP) can be enabled with authentication on different interfaces, with different levels and different passwords.
The HMAC-MD5 mode cannot be mixed with the clear text mode on the same authentication scope (LSP or interface). However, administrators can use one mode for LSP and another mode for some interfaces, for example. If mixed modes are intended, different keys should be used for different modes in order not to compromise the encrypted password in the PDUs.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftismd5.htm
L2TP Extended Failover
The L2TP Extended Failover feature extends Layer 2 Tunneling Protocol (L2TP) failover to occur if during tunnel establishment, a router receives a Stop-Control-Connection-Notification (StopCCN) message from its peer or during session establishment, a router receives a Call-Disconnect-Notify (CDN) message from its peer. In either case, the router selects an alternate peer to contact. This is in addition to the existing failover caused by excessive retransmission of Start-Control-Connection-Reply (SCCRQ) messages that indicate there is no response from the peer.
L2TP Extended Failover results in better load distribution and prevents congestion at a tunnel terminator by allowing the busy tunnel terminator to inform the tunnel initiator that it should try another tunnel terminator.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftl2tpef.htm
L2TP Redirect
The L2TP Redirect feature allows an L2TP network server (LNS) participating in Stack Group Bidding Protocol (SGBP) to send a redirect message to the L2TP access concentrator (LAC) if another LNS wins the bid. The LAC will then reinitiate the call to the newly redirected LNS. The feature provides two purposes:
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Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftl2tpmr.htm
Low Latency Queueing (LLQ) for IPSec
Low Latency Queueing (LLQ) for IPSec encryption engines helps reduce packet latency by introducing the concept of queueing before crypto engines. Prior to this, the crypto processing engine gave data traffic and voice traffic equal status. Administrators now designate voice traffic as priority. Data packets arriving at a router interface are directed into a data packet inbound queue for crypto engine processing. This queue is called the best effort queue. Voice packets arriving on a router interface are directed into a priority packet inbound queue for crypto engine processing. This queue is called the priority queue. The crypto engine undertakes packet processing in a ratio favorable for voice packets. Voice packets are guaranteed a minimum processing bandwidth on the crypto engine.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/llqfm.htm
LZ Software with Hardware Encryption
Before the LZ Software with Hardware Encryption feature was introduced, compression was not supported with the VPN encryption hardware advanced integration module (AIM) and network module (NM); that is, a user had to remove the VPN module from the router and run software encryption with software compression. This feature enables all VPN modules to support LZ compression in software when the VPN module is in Cisco 2600 and Cisco 3600 series routers, thereby, allowing users to configure and compress 2 128Kb/sec streams.
Manual Certificate Enrollment (TFTP and Cut-and-Paste)
The Manual Certificate Enrollment (TFTP and Cut-and-Paste) feature allows users to generate a certificate request and accept Certificate Authority (CA) certificates as well as the router's certificates; these tasks are accomplished via a TFTP server or manual cut-and-paste operations. Users may wish to utilize TFTP or manual cut-and-paste enrollment in the following situations:
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Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftmancrt.htm
Media Forking
Media Forking allows the gateway to create multiple streams (or forks) of media associated with a single call and send those streams to multiple destinations, which may include voice portals with speech recognition. Only the original media stream is bidirectional. Additional branches are unidirectional (transmit only), so additional participants are able to hear only the originating caller and not each other. Each media stream is independently configured and can be a variation of voice only, named telephone event (NTE) only, or voice plus NTE media stream.The content of the media stream is specified in the signaling when the media stream is established.
Although there can be more than one media destination, there is only one signaling destination, which might be the voice portal. The call leg that was originally signaled (for instance, from the originating gateway to the voice portal) is maintained for the life of the session. The media destinations are independent of the signaling destination, so media forks can be added and removed dynamically. The local telephony call leg must be maintained, and up to four media forks, including the destination of the original call, are supported. Fax calls are not supported on any media streams (including the original) when multiple forks are requested. No media forks can be created for a fax call session.
MGCP 1.0 and TGCP 1.0 Profiles
This feature implements the following MGCP protocols on the supported Cisco media gateways:
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MGCP1.0 is a protocol for the control of Voice over IP (VoIP) calls by external call-control elements known as media gateway controllers (MGCs) or call agents (CAs). It is described in the informational RFC 2705, published by the Internet Society.
PacketCable is an industry-wide initiative for developing interoperability standards for multimedia services over cable facilities using packet technology. PacketCable developed the NCS and TGCP protocols, which contain extensions and modifications to MGCP while preserving basic MGCP architecture and constructs. NCS is designed for use with analog, single-line user equipment on residential gateways, while TGCP is intended for use in VoIP-to-PSTN trunking gateways in a cable environment. To meet European cable requirements and equipment characteristics, the EuroPacketCable working group has adapted PacketCable standards under the name IP Cablecom.
MGCP Model
MGCP bases its call control and intelligence in centralized call agents, also called media gateway controllers. The call agents issue commands to simple, low-cost endpoints, which are housed in media gateways (MGs), and the call agents also receive event reports from the gateways. MGCP messages between call agents and media gateways are sent with Internet Protocol over User Datagram Protocol (IP/UDP).
The MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles feature provides protocols for RGWs and TGWs, which sit at the border of the packet network to provide an interface between traditional, circuit-based voice services and the packet network. Residential gateways offer a small number of analog line interfaces, while trunking gateways generally manage a large number of digital trunk circuits.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_24mg1.htm
MGCP Gateway Support for the Bind Command
Previous Media Gateway Control Protocols (MGCP) implementation did not allow the assignment of particular IP addresses for sourcing MGCP commands and media packets, which could cause firewall and security problems. With this feature, you can configure interfaces on which control and media packets can be exchanged. This new functionality allows you to separate signaling from voice by binding control (MGCP signaling) and media (Real-Time Transport Protocol, or RTP voice, fax, and modem) to specific gateway interfaces. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftxbind.htm
Mobile IP—Challenge/Response Extensions
The Mobile IP—Challenge/Response Extensions feature enables a foreign agent to authenticate a mobile node by sending mobile foreign challenge extensions (MFCE) and mobile node-AAA authentication extensions (MNAE) to the home agent in registration requests. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_chext.htm
Mobile IP—Fastswitching Support on Foreign Agent
The Mobile IP—Fastswitching Support on Foreign Agent feature enables packets to be fast switched from the foreign agent both in the direction of the mobile node and through the reverse tunnel. In the direction of the mobile node, packets will be properly fast-switched for global IP addresses. However, this feature does not support fast-switching to mobile nodes using private home addresses.
Fast-switching packets through the reverse tunnel is achieved by intercepting packets before cache lookup and dynamically switching them through the correct tunnel interface.
Mobile IP—Generic NAI Support and Home Address Allocation
The Mobile IP—Generic NAI Support and Home Address Allocation feature allows a mobile node to be identified by using a network access identifier (NAI) instead of an IP address (home address). The NAI is a character string similar to an email address in that it is formatted as either user or user@realm but it need not be a valid e-mail address.
The original purpose of the NAI was to support roaming between dialup ISPs. With the NAI, each ISP need not have all the accounts for all of its roaming partners in a single RADIUS database. RADIUS servers can proxy requests to remote servers for each realm.
These services are also valuable for mobile nodes using Mobile IP when the nodes are attempting to connect to foreign domains with AAA servers. The mobile node can identify itself by including the NAI along with the Mobile IP registration request.
Additionally, this feature allows you to configure the home agent to allocate addresses to mobile nodes either statically (including multiple static addresses per NAI flow) or dynamically. Home address allocation can be from address pools configured locally, through either DHCP server access, or from the AAA server.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftnaiadd.htm
Mobile IP Home Agent Policy Routing
The Mobile IP Home Agent Policy Routing feature supports route maps on Mobile IP tunnels created at the home agent. This feature allows an ISP to provide service to multiple customers. While reverse tunneling packets, the home agent looks up where the packet should go. For example, if an address corresponds to a configured network access identifier (NAI) realm name (such as cisco.com), the packet goes out interface 1, which has a connection to the Cisco network. If an address corresponds to another NAI realm name (such as nortel.com), the packet goes out interface 2, which has a connection to the Nortel network. This feature was designed to route traffics through VPNs back to an enterprise network.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/fthapoli.htm
Mobile IP —IPsec for Home Agent to Foreign Agent Tunnel
The Mobile IP—IPsec for Home Agent to Foreign Agent Tunnel enables the use of IPSec on the home agent to foreign agent tunnel.
Crypto map configuration must be applied to both the tunnel and physical interfaces. For details refer to the "Configuring Cisco Encryption Technology" chapter in the Release 12.2 Cisco IOS Security Configuration Guide.
Mobile IP—MIB Support for NAI and HA Redundancy
The CISCO-MOBILE-IP-MIB is enhanced to add support for following features:
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The following tables are defined in the MIB to support NAI based mobile nodes (MN):
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These tables are the same as the corresponding tables in the RFC2006-MIB (MIP MIB) in terms of the information they provide, but indices are changed so that entries for mobile nodes which are not identified by the IP address will also be included in the table.
The `cmiHaRegMobilityBindingTable' is augmented from `haMobilityBindingTable' of the RFC2006-MIB (MIP MIB) to provide the NAI information.
2.
Scalar objects have been added to MIB to monitor the message exchanges between peer home agents. These objects are under the `cmiHaRedun' subtree of the MIB.
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There are scalar objects under `cmiHaReg' subtree which gives statistics about the registration processing rate at home agent. Distinction is made between registration requests authenticated locally and those authenticated at the AAA server. There are scalar objects under the 'cmiMaReg' subtree which give statistics about the rate at which registration requests are received at the mobility agent (HA or FA).
Mobile IP—NAT Detect
The basic purpose of Network Address Translation (NAT) is to take traffic from the internal network and present it to the Internet as if it were coming from a single device having only one IP address. Traditional Mobile IP tunneling is incompatible with NAT. The Mobile IP—NAT Detect feature allows the home agent to tunnel traffic to Mobile IP clients with private IP addresses behind a NAT-enabled device. The home agent is capable of detecting a registration request traversing a NAT-enabled device and applying the appropriate tunnel to reach the Mobile IP client.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftnatrav.htm
Mobile IP—Private Addressing Support
The Mobile IP—Private Addressing Support feature allows the use of private IP addresses for mobile nodes. Enhancements have been made to the foreign agent to allow it to distinguish between mobile nodes using the same private home address, but with different home agents.
When a mobile node successfully registers with a foreign agent, a tunnel is set up between the foreign agent and the home agent. When a packet is received by the foreign agent for the mobile node, the foreign agent will identify which mobile node to route the packet to based on the address of the mobile node, as well as the home agent from which the packet came.
Mobile IP—Support for FA Reverse Tunneling
The Mobile IP—Support for Foreign Agent Reverse Tunneling feature prevents packets sent by a mobile node from being discarded by routers configured with ingress filtering by creating a reverse tunnel between the foreign agent and the home agent.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_farev.htm
Modular QoS CLI (MQC)-Based Frame-Relay Traffic Shaping
The Modular Quality of Service (QoS) Command Line Interface (CLI)-Based Frame-Relay Traffic Shaping feature provides users the ability to configure Frame Relay traffic shaping (FRTS) using Modular Quality of Service (QoS) Command Line Interface (CLI) commands. Modular QoS CLI is known as MQC.
Modular QoS CLI (MQC) Three-Level Hierarchical Policer
Earlier Cisco IOS traffic policing features allowed you to configure traffic policing at two levels of policy map hierarchies; the top level and a secondary level.
The Modular QoS CLI (MQC) Three-Level Hierarchical Policer extends the traffic policing functionality by allowing you to configure traffic policing at three levels of policy map hierarchies; a top level, a secondary level, and a third level. Traffic policing may be configured at any or all of these levels, depending on the needs of your network. The feature is configured using the Modular Quality of Service (QoS) Command-Line Interface (CLI) (MQC).
Configuring traffic policing in a three-level hierarchical structure provides a greater degree of granularity for traffic policing.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft3level.htm
Modular QoS CLI (MQC) Unconditional Packet Discard
This feature allows customers to classify traffic matching certain criteria and then configure the system to unconditionally discard any packets matching that criteria. This feature is configured using the Modular Quality of Service Command-Line Interface (MQC) feature.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftcbdrp.htm
MPLS DiffServ Tunneling Modes
MPLS DiffServ Tunneling Modes allows service providers to manage the QoS that a router will provide to an MPLS packet in an MPLS network. MPLS DiffServ Tunneling Modes conforms to the IETF draft standard for Uniform, Short Pipe, and Pipe modes, and to Cisco-defined extensions for scalable CLI management of those modes at customer edge, provider edge, and core routers.
Refer to the following document for more information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftdtmode.htm
MPLS Label Distribution Protocol (LDP) MIB
Multiprotocol label switching (MPLS) is a packet forwarding technology that uses a short, fixed-length value called a label in packets to determine the next hop for packet transport through an MPLS network by means of label switching routers (LSRs).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ldpmib13.htm
MPLS Virtual Private Networks
The Virtual Private Network (VPN) feature for Multiprotocol Label Switching (MPLS) allows a Cisco IOS network to deploy scalable IPv4 Layer 3 VPN backbone services.
This feature was originally introduced in 12.0(5)T. This release introduces the command.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftvpn13.htm
MPLS VPN—Carrier Supporting Carrier—IPv4 BGP Label Distribution
This feature enables you to configure your carrier supporting carrier network to enable Border Gateway Protocol (BGP) to transport routes and Multiprotocol Label Switching (MPLS) labels between the backbone carrier provider edge (PE) routers and the customer carrier customer edge (CE) routers. Previously you had to use Label Distribution Protocol (LDP) to carry the labels and an internal gateway protocol (IGP) to carry the routes between PE and CE routers to achieve the same goal.
This feature was originally introduced in Cisco IOS Release 12.0(21)ST. This release integrates the feature into Cisco IOS Release 12.2(13)T.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftcscl13.htm
MPLS VPN—Inter-AS—IPv4 BGP Label Distribution
This feature enables you to set up a VPN service provider network so that the autonomous system boundary routers (ASBRs) exchange IPv4 routes with MPLS labels of the provider edge (PE) routers. Route reflectors (RRs) exchange VPNv4 routes, using multihop, multiprotocol, External Border Gateway Protocol (EBGP). This configuration saves the ASBRs from having to store all the VPNv4 routes. Using the route reflectors to store the VPNv4 routes and forward them to the PE routers results in improved scalability.
This feature was originally introduced in Cisco IOS Release 12.0(21)ST. This release integrates the feature into Cisco IOS Release 12.2(13)T.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftiasl13.htm
MPLS VPN-MIB Notifications
The MPLS VPN technology allows service providers to offer intranet and extranet VPN services that directly connect their customers' remote offices to a public network with the same security and service levels that a private network offers. The Provider-Provisioned VPN (PPVPN)-MPLS-VPN MIB notifications provide SNMP notification for critical MPLS VPN events.
The MPLS VPN-MIB Notifications feature provides the following benefits:
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This feature was originally introduced in Cisco IOS Release 12.0(21)ST. This release integrates the feature into Cisco IOS Release 12.2(13)T.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftvpnm13.htm
MS CHAP Version 2
The MS CHAP Version 2 feature in Cisco IOS Release 12.2(13)T introduces the ability of Cisco routers to utilize Microsoft Challenge Handshake Authentication Protocol Version 2 (MS CHAP V2) authentication for PPP connections between a computer using a Microsoft Windows operating system and a network access server (NAS). MS CHAP V2 authentication is an updated version of MS CHAP that is similar to, but incompatible with MS CHAP Version 1 (V1). MS CHAP V2 introduces mutual authentication between peers and a change password feature.
Refer to the following document for additional information:
https://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guides_list.html
Multicast-VPN—IP Multicast Support for MPLS VPNs
The Multicast-VPN—IP Multicast Support for MPLS VPNs feature allows a service provider to configure and support multicast traffic in a Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN) environment. Because MPLS VPNs support only unicast traffic connectivity, deploying the Multicast-VPN feature in conjunction with MPLS VPN allows service providers to offer both unicast and multicast connectivity to MPLS VPN customers.
This feature supports routing and forwarding of multicast packets for each individual VPN routing and forwarding (VRF) instance, and it also provides a mechanism to transport VPN multicast packets across the service provider backbone.
The Multicast-VPN feature in Cisco IOS software provides the ability to support the multicast feature over a Layer 3 VPN. As enterprises extend the reach of their multicast applications, service providers can accommodate these enterprises over their MPLS core network. IP multicast is used to stream video, voice, and data to an MPLS VPN network core.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftmltvpn.htm
Multiclass Multilink PPP
Previous implementations of Cisco IOS Multilink PPP (MLP) include support for Link Fragmentation Interleaving (LFI). This feature allows the delivery of delay-sensitive packets, such as the packets of a Voice call, to be expedited by omitting the PPP Multilink Protocol header and sending them as raw PPP packets in between the fragments of larger data packets. This feature works well on bundles consisting of a single link. However, when the bundle contains multiple links there is no way to keep the interleaved packets in sequence with respect to each other.
The Multiclass Multilink PPP (MCMP) feature in Cisco IOS Release 12.2(13)T addresses the limitations of MLP LFI on bundles containing multiple links by introducing multiple data classes. Normal data traffic and delay-sensitive data traffic are divided into Class 0 and Class 1, respectively. Class 0 data traffic is subject to fragmentation just as regular Multilink packets are. Class 1 data traffic can be interleaved but never fragmented. The next transmit sequence number, expected sequence number, unassigned fragment list, working packet, lost fragment timer, fastswitching mode, and all statistics are managed per-class, rather than for the bundle as a whole.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftmmlppp.htm
NAT Default Inside Server
The NAT Default Inside Server feature provides for the need to forward packets from the outside to a specified inside local address. Traffic is redirected that does not match any Network Address Translation (NAT) entries and the packets are not dropped.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftnatis.htm
NAT Integration with MPLS VPNs
Network Address Translation (NAT) and MPLS VPNs can now be configured on a single device to work together. NAT can differentiate which MPLS VPN it receives IP traffic from even if the MPLS VPNs are all using the same IP addressing scheme. This enhancement enables MPLS VPN customers the ability to provide common shared services across multiple MPLS VPN customers while ensuring that each MPLS VPN is completely separate from the other.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftnatvpn.htm
NAT MIB (Read-Only)
This feature introduces support for the Network Address Translation (NAT) MIB. NAT provides tables for translating internal network addresses external network addresses. The NAT MIB provides objects for the monitoring and management of NAT bindings and session using SNMP. In this release, access to the MIB is limited to the read-only level. No new or modified Cisco IOS commands are associated with this MIB.
For details on the management options provided by the MIB, see the CISCO-IETF-NAT-MIB.my file available in the "SNMP v2 MIBs" section of the Cisco.com MIB page at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml. Additional information on the MIB is available in the form of an internet draft (draft-ietf-nat-natmib), available through www.ietf.org.
NAT Protocol Translation
Network Address Translation - Protocol Translation (NAT PT) is an IPv6 translation mechanism allowing IPv6-only devices to communicate with IPv4-only devices, and vice versa. NAT PT was designed using RFC 2766 as a migration tool to help customers transition their IPv4 networks to IPv6 networks. Using existing IPv4 NAT capability and adding a protocol translator allows NAT PT to provide direct communication between hosts speaking a different network protocol.
NAT Stateful Failover of Network Address Translation
There is an increasing need to provide highly resilient IP networks where application connectivity continues unaffected by potential failure to links and routers at the Network Address Translation (NAT) border. The Stateful NAT feature allows two or more network address translators to function as a translation group. A backup router running NAT provides translation services in the event of failure of the active translator.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsnat.htm
NAT Support of H.323 RAS
Cisco IOS NAT supports all H.225 and H.245 message types, including those sent in the RAS protocol. RAS provides a number of messages that are used by software clients and Voice over IP (VoIP) devices to register their location, request assistance in call setup, and control bandwidth. The RAS messages are directed toward an H.323 gatekeeper.
Some RAS messages include IP addressing information in the payload, typically meant to register a user with the gatekeeper or learn about another user already registered. If these messages are not known to NAT, they cannot be translated to an IP address that will be visible to the public.
Previously, NAT did not support H.323 v2 RAS messages. With this enhancement, embedded IP addresses can be inspected for potential address translation.
This feature was previously released in Cisco IOS Release 12.2(4)T on the Catalyst 2900, Catalyst 2900XL, Catalyst 4000 series, Catalyst 5000 family switches with an installed Route Switch Module, Catalyst 6000, Catalyst 8500 series, Cisco 800 series, Cisco 1000 series, Cisco1400 series, Cisco 1600 series, Cisco 1700, Cisco 2600 series, Cisco 3600 series, Cisco 4000 series, Cisco 6400 series, Cisco 7000 series, Cisco 8500 series, Cisco 12000 series, Cisco MC3810, Cisco uBR900 series, Cisco uBR7200, and LightStream 1010 series platforms. This release is porting the feature into the IAD2420 platform.
NAT-Support of H.323 v2 Call Signaling
Cisco IOS NAT supports all H.225 and H.245 message types, including FastConnect and Alerting, as part of the H.323 v2 specification.
Previously, NAT only supported H.323 version 1 and that was specific only to the Microsoft NetMeeting application. With this enhancement, any product that makes use of these message types will be able to pass through a Cisco IOS NAT configuration without any static configuration.
This feature was previously released in Cisco IOS Release 12.1(5)T on the Cisco Catalyst 2900, Cisco Catalyst 2900XL, Cisco Catalyst 4000 series, Cisco Catalyst 5000 family switches with an installed Route Switch Module, Cisco Catalyst 6000 series, Cisco Catalyst 8500 series, Cisco LightStream 1010 series, Cisco 800 series, Cisco 1000 series, Cisco 1400 series, Cisco 1600 series, Cisco 1700 series, Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, Cisco 4000 series, Cisco AS5300, Cisco AS5400, Cisco AS5800, Cisco 6400 series, Cisco 7000 series, Cisco 8500 series, Cisco 12000 series, Cisco MC3810, Cisco uBR900, and Cisco uBR7200 platforms. This release is porting the feature into the Cisco IAD2420 platform.
NetWare Link Services Protocol (NLSP)
The NetWare Link Services Protocol (NLSP) will no longer be offered after Cisco IOS Release 12.2(13)T. NLSP commands will not appear in future releases of the Cisco IOS software documentation set.
Next Hop Resolution Protocol (NHRP) for IPX
The Next Hop Resolution Protocol (NHRP) for IPX will no longer be offered after Cisco IOS Release 12.2(13)T. NHRP for IPX commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
OSPF Support for Multi-VRF on CE Routers
The OSPF Support for Multi-VRF on CE Routers feature provides the capability of suppressing provider edge (PE) checks which are needed to prevent loops when the PE is performing a mutual redistribution of packets between the OSPF and BGP protocols. When VRF is used on a router that is not a PE (that is, one that is not running BGP), the checks can be turned off to allow for correct population of the VRF routing table with routes to IP prefixes.
OSPF multi-VRF allows you to split the router into multiple virtual routers, where each router contains its own set of interfaces, routing table, and forwarding table.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120st/120st21/ospfvrfl.htm
Packet Classification Based on Layer 3 Packet Length
This feature allows customers to match and classify traffic on the basis of the layer 3 length in the IP header of a packet. The layer 3 length is the IP datagram plus the IP header.
Traffic that matches a particular layer 3 length can be organized into specific classes that can, in turn, receive specific user-defined quality of service (QoS) treatment (for example, a certain amount of bandwidth or an IP Precedence value) when that class is included in a policy map.
This feature provides the added capability of matching and classifying traffic on the basis of the layer 3 length in the IP packet header. This new match criterion is in addition to the other match criteria, such as the IP precedence, differentiated services code point (DSCP) value, class of service (CoS), currently available.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftmchpkt.htm
Packet Classification Using the Frame Relay DLCI Number
The Packet Classification Using the Frame Relay DLCI Number feature allows customers to match and classify traffic based on the Frame Relay data-link connection identifier (DLCI) number associated with a packet. This new match criterion is in addition to the other match criteria, such as the IP Precedence, Differentiated Service Code Point (DSCP) value, Class of Service (CoS), currently available.
The Packet Classification Using the Frame Relay DLCI Number feature extends the functionality of the Modular Quality of Service (QoS) Command-Line Interface (CLI) (MQC).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftpcdlci.htm
Per VRF AAA
Using the Per VRF AAA feature, Internet Service Providers (ISPs) can partition authentication, authorization, and accounting (AAA) services based on Virtual Route Forwarding (VRF). This feature permits the Virtual Home Gateway (VHG) to communicate directly with the customer's RADIUS server, which is associated with the customer's Virtual Private Network (VPN), without having to go through a RADIUS proxy. Thus, ISPs can scale their VPN offerings more efficiently because they no longer need to proxy AAA to provide their customers with the flexibility they demand.
This feature was originally introduced in Cisco IOS Release 12.2(1)DX. This release is porting the feature into the Cisco 7100 series, Cisco 7500 series, and Cisco 7700 series platforms.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftvrfaaa.htm
Percentage-Based Policing and Shaping
This feature provides the ability to configure traffic policing and traffic shaping based on a percentage of bandwidth available on the interface. Configuring traffic policing and traffic shaping in this manner enables customers to use the same policy map for multiple interfaces with differing amounts of bandwidth.
PPPoE Client DDR Idle-Timer
This feature supports the dial-on-demand routing (DDR) interesting traffic control list functionality of the dialer interface with a PPP over Ethernet (PPPoE) client, but also keeps original functionality (PPPoE connection up and always on after configuration) for those PPPoE clients that require it.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftppecls.htm
Privilege Command Enhancement
This feature simplifies the configuration of privilege levels for specific commands through the enhancement of the privilege level global configuration command. A privilege level can now be specified for all keyword options of a command with a single command-line interface (CLI) command. Previously, separate "privilege level" commands were required for each keyword combination of a command. This enhancement can significantly reduce the number of commands needed to configure user privilege levels and correspondingly reduce the size of configuration files.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftprienh.htm
RADIUS Attribute 52 and Attribute 53 Gigaword Support
The RADIUS Attribute 52 and Attribute 53 Gigaword Support feature introduces support for Attribute 52 (Acct-Input-Gigawords) and Attribute 53 (Acct-Output-Gigawords) in accordance with RFC 2869. Attribute 52 keeps track of the number of times the Acct-Input-Octets counter has rolled over the 32-bit integer throughout the course of the provided service; attribute 53 keeps track of the number of times the Acct-Output-Octets counter has rolled over the 32-bit integer throughout the delivery of service. Both attributes can be present only in Accounting-Request records where the Acct-Status-Type is set to "Stop" or "Interim-Update." These attributes can be used to keep accurate track of and bill for usage.
This feature was originally introduced in Cisco IOS Release 12.2(4)B. No additional platform support has been added.
RADIUS Attribute 77 for DSL
The RADIUS Attribute 77 for DSL feature introduces support for Attribute 77 (Connect-Info) to carry the textual name of the virtual circuit class associated with the given permanent virtual circuit (PVC). (Although attribute 77 does not carry the unspecified bit rate (UBR), the UBR can be inferred from the classname used if one UBR is set up on each class.) Attribute 77 is sent from the network access server (NAS) to the RADIUS server via Accounting-Request and Accounting-Response packets.
This feature was originally introduced in Cisco IOS Release 12.2(4)B. No additional platform support has been added.
RADIUS Centralized Filter Management
Before the RADIUS Centralized Filter Management feature, wholesale providers (who provide premium charges for customer services such as access control lists [ACLs]) were unable to prevent customers from applying exhaustive ACLs, which could impact router performance and other customers. This feature introduces a centralized administration point—a filter server—for ACL management. The filter server acts as a centralized RADIUS repository for ACL configuration.
Whether or not the RADIUS server that is used as the filter server is the same server that is used for access authentication, the network access server (NAS) will initiate a second access-request to the filter server. If configured, the NAS will use the filter-id name as the authentication username and the filter server password for the second access-request. The RADIUS server will attempt to authenticate the filter-id name, returning any required filtering configuration in the access-accept.
Because downloading ACLs is time consuming, a local cache is maintained on the NAS. If an ACL name exists on the local cache, that configuration will be used without consulting the filter server.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_ftrmn.htm
RADIUS EAP Support
The EAP RADIUS Support feature allows users to apply to the client authentication methods that may not be supported by the network access server; this is done via the Extensible Authentication Protocol (EAP). Before this feature was introduced, support for various authentication methods for PPP connections required custom vendor-specific work and changes to the client and NAS.
EAP is an authentication protocol for PPP that supports multiple authentication mechanisms that are negotiated during the authentication phase (instead of the link control protocol [LCP] phase). EAP allows a third-party authentication server to interact with a PPP implementation through a generic interface.
This feature was originally introduced in Cisco IOS Release 12.2(2)XB5. This release is porting the feature into the Catalyst 4000, Cisco AS5350, Cisco AS5800, Cisco AS5850, Cisco 05, Cisco 806, Cisco 820, Cisco 1400 series, Cisco 1600 series, Cisco 1600R, Cisco 2500 series, Cisco 2600 series, Cisco 3620, Cisco 7100 series, Cisco 7200 series, Cisco 7500 series, Cisco MC3810, Cisco SOHO 70 series, Cisco SOHO78, Cisco uBR7200, Cisco uBR920
RADIUS Logical Line ID
The RADIUS Logical Line ID feature enables users to track their customers on the basis of the physical lines in which the customers' calls originate. Thus, users can better maintain the profile database of their customers as the customers move from one physical line to another.
Logical Line Identification (LLID) is an alphanumeric string (which must be a minimum of one character and a maximum of 253 characters) that is a logical identification of a subscriber line. LLID is maintained in a RADIUS server customer profile database. This customer profile database is connected to a L2TP access concentrator (LAC) and is separate from the RADIUS server that the LAC and L2TP Network Server (LNS) use for the authentication and authorization of incoming users. When the customer profile database receives a preauthorization request from the LAC, the server sends the LLID to the LAC as the Calling-Station-ID attribute (attribute 31).
The LAC sends a preauthorization request to the customer profile database when the LAC is configured for preauthorization. Configure the LAC for preauthorization using the subscriber access pppoe pre-authorize command.
This feature was originally introduced in Cisco IOS Release 12.2(8)B. No additional platform support has been added.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftlineid.htm
RSVP Local Policy Support
The RSVP Local Policy Support feature allows network administrators to create default and access control list (ACL)-based policies. These policies, in turn, control how RSVP filters its signalling messages to allow or deny quality of service (QoS) to networking applications based on the IP addresses of the requesting hosts.
This feature is being introduced in Cisco IOS Release 12.2(13)T.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftrsvplp.htm
RSVP Refresh Reduction and Reliable Messaging
The RSVP Refresh Reduction and Reliable Messaging feature includes refresh reduction, which improves the scalability, latency, and reliability of RSVP signalling by introducing the following extensions:
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This feature was originally introduced in Cisco IOS Release 12.2(11)S. This release integrates the feature into Cisco IOS Release 12.2(13)T.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftrsvpre.htm
Session Limit Per VRF
The Session Limit Per VPN Routing and Forwarding Instance (VRF) feature enables session limits to be applied on all VPDN groups associated with a common VPDN virtual template. Before the implementation of Session Limit Per VRF, a single default template carrying the configuration values of a subset of VPDN group commands were associated with all VPDN groups configured on the router. Session Limit Per VRF enables you to create, define and name multiple VPDN templates. You can then associate a specific template with a VPDN group. A session limit can be configured at the VPDN template level to specify a combined session limit for all VPDN groups associated with the configured VPDN template.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_4/12b_vrf.htm
Show Command Output Redirection
This feature adds the capability to redirect output from Cisco IOS CLI show commands to a file. For each show command issued, a new file can be created, or the output can be appended to an existing file. Command output can optionally be displayed on-screen while being redirected to a file by using the tee keyword. Redirection is available using a pipe (|) character after any show command, combined with the redirect, append, or tee keywords.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftshowre.htm
Simple Multicast Routing Protocol (SMRP) for AppleTalk
The Simple Multicast Routing Protocol (SMRP) for AppleTalk will no longer be offered after Cisco IOS Release 12.2(13)T. NLSP commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
SIP and H.323 Fax Enhancements
The SIP and H.323 Fax Enhancements feature adds an assortment of fax transfer enhancements to the Cisco IOS gateway implementations of H.323 and Session Initiation Protocol (SIP) call control protocols. The enhanced areas include the use of:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsihfax.htm
SIP—Call Transfer Enhancements Using the Refer Method
The SIP—Call Transfer Enhancements Using the Refer Method feature provides blind and attended call transfer capabilities to supplement the Bye and Also methods already implemented on Cisco IOS Session Initiation Protocol (SIP) gateways. The SIP—Call Transfer Enhancements Using the Refer Method feature is compatible with the original forms of call transfer and with third-party call-control protocols. The SIP—Call Transfer Enhancements Using the Refer Method feature enables application service providers (ASPs) to provide attended transfer and blind transfer in accordance with emerging SIP standards.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsipref.htm
SIP Enhanced 180 Provisional Response Handling
This feature provides the ability to enable or disable early media cut-through on Cisco IOS gateways for SIP 180 response messages. The new feature allows you to specify whether 180 messages with Session Description Protocol (SDP) are handled in the same way as 183 responses with SDP. The 180 Ringing message is a provisional or informational response used to indicate that the INVITE message has been received by the user agent and that alerting is taking place. Both 180 and 183 messages may contain SDP which allow an early media session to be established prior to the call being answered.
Prior to the implementation of the new feature, Cisco gateways handled a 180 Ringing response with SDP in the same manner as a 183 Session Progress response; that is, the SDP was assumed to be an indication that the far end was going to send early media. Cisco gateways handled a 180 response without SDP by providing local ringback, rather than early media cut-through. The new feature provides the capability to ignore the presence or absence of SDP in 180 messages, and as a result, treat all 180 messages in a uniform manner.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft180sdp.htm
SIP Extensions for Caller Identity and Privacy
This feature provides support for privacy indication, as well as network verification and screening of a call participant's name and number. Cisco implements the new feature on Cisco SIP IOS trunking gateways by supporting a new header, Remote-Party-ID. In previous SIP implementations, the From header was used to indicate calling party identity, and once defined in the initial INVITE request, could not be modified for the duration of that session. Implementing the Remote-Party-ID header, which can be modified, added or removed as a call session is being established, overcomes previous limitations and enables call participant verification and screening
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsipext.htm
SIP Gateway Compliance to RFC2543-bis-04
RFC2543-bis-04 contains several changes to Session Initiation Protocol (SIP) gateway code. The SIP Gateway Compliance to RFC2543-bis-04 feature updates Cisco SIP Voice over IP (VoIP) gateways with the latest RFC changes. All changes are compatible with older RFC versions. Some of the changes include:
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SIP Redirect Processing Enhancements
The SIP Redirect Processing Enhancements feature allows flexibility in the handling of incoming redirect or 3xx class of responses so they can be enabled or disabled through the command-line interface (CLI). The default mode is enabled, in which Session Initiation Protocols (SIP) gateways handle incoming 3xx messages as per RFC 2543. RFC 2543 states that redirect response messages are used by SIP user agents (UA) to initiate a new Invite when a UA learns that a user has moved from a previously known location. If redirect handling is disabled through the CLI, the UA treats incoming 3xx responses as 4xx error class responses. The call is not redirected, and is instead released with the appropriate PSTN cause code.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsipmaz.htm
SNMP Notification Logging
Systems that support Simple Network Management Protocol (SNMP) often need a mechanism for recording notification information as a hedge against lost notifications, whether those are traps or informs that exceed retransmission limits. The Notification Log MIB provides a common infrastructure for other MIBs in the form of a local logging function. The SNMP Notification Logging feature adds Cisco IOS command-line interface (CLI) commands to change the size of the notification log, to set the global ageout value for the log, and to display logging summaries at the command line.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s22/ftmiblog.htm
SSG Autologoff
The SSG Autologoff feature enables the Cisco Service Selection Gateway (SSG) to verify connectivity with each host at configured intervals. If SSG detects that the host is not reachable from SSG, then it automatically initiates the logoff for that host.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ssg/index.htm
SSG Port-Bundle Host Key
The SSG Port-Bundle Host Key feature enhances communication and functionality between the Service Selection Gateway (SSG) and the Cisco Subscriber Edge Services Manager (SESM) by introducing a mechanism that uses the host source IP address and source port to identify and monitor subscribers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ssg/index.htm
SSG TCP Redirect for Services
The SSG TCP Redirect for Services feature redirects certain packets, which would otherwise be dropped, to captive portals that can handle the packets in a suitable manner. For example, packets sent upstream by unauthorized users are forwarded to a captive portal that can redirect the users to a logon page. Similarly, if users try to access a service to which they have not logged on, the packets are redirected to a captive portal that can provide a service logon screen.
The captive portal can be any server that is programmed to respond to the redirected packets. If the Cisco Subscriber Edge Services Manager (SESM) is used as a captive portal, unauthenticated subscribers can be sent automatically to the SESM logon page when they start a browser session. In SESM Release 3.1(3), captive portal applications can also redirect to service logon pages, advertising pages, and message pages. The SESM captive portal application can also capture a URL in a subscriber's request and redirect the browser to the originally requested URL after successful authentication. Redirected packets are always sent to a captive portal group that consists of one or more servers. SSG selects one server from the group in a round-robin fashion to receive the redirected packets.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ssg/index.htm
Subscriber Service Switch (SSS)
The Subscriber Service Switch (SSS) was developed in response to a need by Internet service providers for increased scalability and extensibility for remote access service selection and Layer 2 subscriber policy management. This Layer 2 subscriber policy is needed to manage tunneling of PPP, Ethernet, Frame Relay, and other link-level protocols in a policy-based bridging fashion
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_sss.htm
Support for IPsec ESP Through NAT
The ability to support multiple concurrent IPsec ESP tunnels or connections through a router configured with Network Address Translation (NAT) can now be utilized when the NAT router is configured in overload or Port Address Translation (PAT) mode.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftnatesp.htm
T.38 Fax Relay for VoIP H.323
T.38 Fax Relay for VoIP H.323 provides standards-based fax relay protocol support for H.323 gateways and gatekeepers. T.38 is an ITU-T recommended standard for fax relay. Since T.38 is a standards-based implementation for fax relay, Cisco gateways and gatekeepers are able to interwork with third-party H.323 devices that support T.38 protocol.
This feature was previously released in Cisco IOS Release 12.1(3)T on Cisco 2600 series, Cisco 3640, and Cisco MC3810 platforms. This release is porting the feature into the IAD2420 platform.
Terminal Line Security for PAD Connections
X.25 closed user group (CUG) service is a network service that allows subscribers to be segregated into private subnetworks with limited outgoing and incoming access. A data terminal equipment (DTE) device becomes a member of a CUG by subscription; the DTE must obtain membership from its network service for the set of CUGs to which it needs access.
The Terminal Line Security for PAD Connections feature allows CUG services to be configured on terminal lines, enabling terminal lines to participate in X.25 CUG security for packet assembler/disassembler (PAD) connections. CUG services can be applied to console lines, auxiliary lines, and tty and vty devices. Configuring CUG services on terminal lines allows you to specify CUG protection for lines that are part of the point of presence (POP). Before the introduction of this feature, CUG services could be configured only on X.25 synchronous data communications equipment (DCE) interfaces.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftcugpad.htm
Update to the Interworking of Cisco MGCP Voice Gateways and Cisco CallManager Version 3.2 Feature
This document describes updates to the Interworking of Cisco MGCP Voice Gateways and Cisco Call Manager Version 3.2 feature. This update introduces the mgcp validate domain-name command, which enables you to check if the domain name or host name and the IP address received as part of the endpoint names sent from the Call Agent (CA) or Cisco CallManager (CCM) match with the ones that have been configured on the gateway (GW). This check is valid for the MGCP messages received from the CA or CCM only.
Use the new mgcp validate domain-name command first before configuring MGCP in a Voice over IP (VoIP) network.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftvalid.fm
Update to the playout-delay Command
In environments with long network delays, T.38 fax relay can be unsuccessful. The fax keyword was added to the playout-delay command to allow users to decrease the playout delay value to compensate for long network delays when necessary.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ft_pdfax.htm
Virtual Router Redundancy Protocol (VRRP)
There are several ways a LAN client can determine which router should be the first hop to a particular remote destination. The client can use a dynamic process or static configuration. Examples of dynamic router discovery are as follows:
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The drawback to dynamic discovery protocols is that they incur some configuration and processing overhead on the LAN client. Also, in the event of a router failure, the process of switching to another router can be slow.
An alternative to dynamic discovery protocols is to statically configure a default router on the client. This approach simplifies client configuration and processing, but creates a single point of failure. If the default gateway fails, the LAN client is limited to communicating only on the local IP network segment and is cut off from the rest of the network.
The Virtual Router Redundancy Protocol (VRRP) feature can solve the static configuration problem. VRRP enables a group of routers to form a single virtual router. The LAN clients can then be configured with the virtual router as their default gateway. The virtual router, representing a group of routers, is also known as a VRRP group.
VRRP is supported on Ethernet, Fast Ethernet, and Gigabit Ethernet interfaces, and on MPLS VPNs and VLANs.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120st/120st18/st_vrrpx.htm
VLAN Range
Using the VLAN Range feature, you can group VLAN subinterfaces together so that any command entered in a group applies to every subinterface within the group. This capability simplifies configurations and reduces command parsing.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_4/12b_rang.htm
Voice and Quality of Service Features for ADSL and G.SHDSL on Cisco 1700, Cisco 2600, and Cisco 3600 Series Routers
Cisco 1700 series, Cisco 2600 series, and Cisco 3600 series routers with ADSL or G.SHDSL WAN interface cards support the integration of voice and data over the same ADSL or G.SHDSL circuit using Voice over IP (VoIP). Cisco 2600 series and Cisco 3600 series routers with ADSL or G.SHDSL WAN interface cards also support the integration of voice and data over the same ADSL or G.SHDSL circuit using Voice over ATM (VoATM).
This feature was originally introduced in Cisco IOS Release 12.2(4)XL. This release is porting the feature into the Cisco 1700 series platforms.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xl/122xl4/ft_qgdsl.htm
Voice Call Tuning
This feature provides tools for quickly taking spot performance measurements of voice call performance while the call is up. You also have the ability to change the echo canceller and jitter buffer parameters of a call while the call is in progress. Audible effects can be immediately noticed, aiding in problem determination and resolution. The feature provides real-time call monitor and manipulation on the interface between Cisco IOS software and the digital signalling processors (DSPs) by addressing the following two items:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftvdsptn.htm
VPDN Multihop by DNIS
The Cisco VPDN Multihop by DNIS feature allows dialed number identification service (DNIS)-based multihop capability in a virtual private dial-up network (VPDN), which enables customers that dial in to a network using a standard telephone line to take advantage of the aggregation capability offered by multihop switching.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_8/ftvmhopd.htm
VRRP Support
There are several ways a LAN client can determine which router should be the first hop to a particular remote destination. The client can use a dynamic process or static configuration. Examples of dynamic router discovery are as follows:
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The drawback to dynamic discovery protocols is that they incur some configuration and processing overhead on the LAN client. Also, in the event of a router failure, the process of switching to another router can be slow.
An alternative to dynamic discovery protocols is to statically configure a default router on the client. This approach simplifies client configuration and processing, but creates a single point of failure. If the default gateway fails, the LAN client is limited to communicating only on the local IP network segment and is cut off from the rest of the network.
The Virtual Router Redundancy Protocol (VRRP) feature can solve the static configuration problem. VRRP enables a group of routers to form a single virtual router. The LAN clients can then be configured with the virtual router as their default gateway. The virtual router, representing a group of routers, is also known as a VRRP group.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120st/120st18/st_vrrpx.htm
X.25 Suppression of Security Signaling Facilities
This feature allows the X.25 Call Redirection/Call Deflection Notification (CRCDN) and Called Line Address Modified Notification (CLAMN) security signaling facilities to be disabled (suppressed) in packets that transit data communication equipment that uses a mix of International Telecommunication Union Telecommunication Standardization Sector T (ITU-T) 1980 and 1984 X.25 protocols.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsupsgx.htm
Xerox Network Systems (XNS)
The Xerox Network Systems (XNS) feature will no longer be offered after Cisco IOS Release 12.2(13)T. XNS commands will not appear in future releases of the Cisco IOS software documentation set.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftjencrg.htm
New Hardware Features Supported in Cisco IOS Release 12.2(11)T1
The following new hardware features are supported in Cisco IOS Release 12.2(11)T1. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
Cisco 3640A Router
The Cisco 3640A is identical to the Cisco 3640 router in terms of physical characteristics, interface support, performance and memory. The Cisco 3640A router will support the same Cisco IOS feature sets as the Cisco 3640 router, but requires a different minimum version of Cisco IOS software.
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(11)T
The following hardware platforms and modules are now supported in Cisco IOS Release 12.2(11)T. These platforms and modules were first introduced in earlier Cisco IOS software releases.
16-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series
The 16-port Ethernet switch network module was originally introduced in Cisco IOS Release 12.2(8)T. Cisco IOS Release 12.2(11)T adds stacking and flow control features to the previously released feature.
See the "16-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series" section or refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft1636nm.htm.
Cisco 1760 Router
The Cisco 1760 router is a voice-and-data-capable router that provides Voice-over-IP (VoIP) functionality and can carry voice traffic (for example, telephone calls and faxes) over an IP network. Using one or two WAN connections, the router links small-to-medium-size remote Ethernet and Fast Ethernet LANs to central offices.
The Cisco 1760 router is available in two models. The Cisco 1760 runs data and data-plus-voice images, providing digital and analog voice support. The Cisco 1760-V includes all the features needed for immediate integration of data and voice services with support for multiple voice channels.
Refer to the documents at the following location for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/5300/index.htm.
Cisco AS5350 Universal Gateway
The Cisco AS5350 Universal Gateway is the only one-rack-unit, two, four, or eight PRI gateway that provides universal services—data, voice, and fax services on any service, any port. The Cisco AS5350 offers high performance and high reliability in a compact, modular design. This cost-effective platform is ideally suited for Internet service providers (ISPs) and enterprises that require innovative universal services.
Refer to the documents at the following location for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5350/index.htm.
Cisco AS5850 Universal Gateway
The Cisco AS5850 Universal Gateway provides the highest concentration of port and Integrated Services Digital Network (ISDN) terminations available in a single remote access server product. The Cisco AS5850 is specifically designed to meet the demands of large service providers such as Post, Telephone, and Telegraphs (PTTs), regional bell operating companies (RBOCs), inter-exchange carriers (IXCs), and large Internet service providers (ISPs).
Refer to the documents at the following location for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5850/index.htm.
Cisco Signaling Link Terminal (SLT) Dual Ethernet
The Cisco Signaling Link Terminal (SLT) Dual Ethernet feature adds Cisco Cisco Signaling Link Terminal dual Ethernet support to the virtual switch controller (VSC). This enhanced Cisco SLT support provides two IP networks and two additional Session Manager sessions (for a total of four Session Manager sessions) for improved backhaul communication. These additions increase the resilience of Cisco SLT and VSC communications by supporting two Reliable User Datagram Protocol (RUDP) sessions from each Ethernet interface to each VSC. These VSC enhancements help to determine when to switch Ethernets and when to switch VSC activity.
The Cisco SLT, which is based on the Cisco 2611 router, is shipped with two Ethernet interfaces. Until this feature was released, the Cisco SLT and VSC solution supported only one of the two Ethernet interfaces. Both Session Manager sessions had to travel over this single Ethernet interface. The Cisco Signaling Link Terminal Dual Ethernet feature supports the second Ethernet, which improves the resilience of the backhaul IP communications.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftsltdua.htm.
New Software Features in Cisco IOS Release 12.2(11)T
The following new features are supported in Cisco IOS Release 12.2(11)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
AAA-PPP-VPDN Non-Blocking
Previously, Cisco IOS created a statically configurable number of processes to authenticate calls. Each of these processes would handle a single call, but in some situations the limited number of processes could not keep up with the incoming call rate. This resulted in some calls timing out. The AAA-PPP-VPDN Non-Blocking feature changes the software architecture such that the number of processes will not limit the rate of call handling.
Note
Accounting of VPDN Disconnect Cause
In the past, whenever a Layer 2 Tunneling Protocol (L2TP) or Layer 2 Forwarding (L2F) session fails or disconnects, the network access server (NAS) and Home GateWay (HGW) report a very generic disconnect-cause code, such as "LOST CARRIER". These generic codes do not provide enough detailed information for accounting and debugging purposes, creating a need for disconnect-cause codes that provide more detailed information. The Accounting of VPDN Disconnect Cause feature adds eight new disconnect-cause codes. These eight disconnect-cause codes describe the status of Virtual Private Dialup Network (VPDN) failures and disconnects more specifically than existing generic disconnect-cause codes. These new disconnect-cause codes can be found in the Cisco IOS Security Configuration Guide, Release 12.2 located at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fsecur_c/fappendx/scgrdat3.htm.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftacldir.htm.
Note
ACL Authentication of Incoming rsh and rcp Requests
To enable the Cisco IOS software to receive incoming remote shell (rsh) protocol and remote copy (rcp) protocol requests, customers must configure an authentication database to control access to the router. This configuration is accomplished by using the ip rcmd remote-host command.
Currently, when using this command, customers must specify the local user, the remote host, and the remote user in the database authentication configuration. For users who can execute commands to the router from multiple hosts, multiple database authentication configuration entries must be used, one for each host.
This feature allows customers to specify an access list for a given user. The access list identifies the hosts to which the user has access. A new argument, access-list, has been added that can be used with this command to specify the access list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftauth.htm.
Note
ACL Default Direction
The ACL Default Direction feature allows you to change the filter direction (where filter direction is not specified) to inbound packets only; that is, you can configure your server to filter packets that are coming toward the network.
This feature introduces the radius-server attribute 11 direction default command, which allows you to change the default direction of filters for your access control lists (ACL) via RADIUS. (RADIUS attribute 11 (Filter-Id) indicates the name of the filter list for the user.) Enabling this command allows you to change the filter direction to inbound—which stops traffic from entering a router, thereby reducing resource consumption—rather than the outbound default direction, which waits until the traffic is about to leave the network before filtering. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftacldir.htm.
Note
Advanced Voice Busyout
The local voice busyout feature provides a way to busy out a voice port or DS-0 group (time slot) if a state change is detected in a monitored network interface (or interfaces). When a monitored interface changes to a specified state—to out-of-service or in-service—the voice port presents a seized/busyout condition to the attached PBX or other customer premises equipment (CPE). The PBX or other CPE can then attempt to select an alternate route.
Advanced Voice Busyout adds the following functionality to the local voice busyout feature:
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Using the Advanced Voice Busyout feature you can perform the following tasks:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft_cacbo.htm.
Note
Analog Centralized Automatic Message Accounting E911 Trunk
Cisco IOS Release 12.2(11)T is the first Cisco IOS release that introduces the Analog Centralized Automatic Message Accounting (CAMA) E911 feature that adds E911 connectivity features to the Cisco 2600 series and Cisco 3600 series routers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/acam_911.htm.
Asynchronous Line Monitoring
Before Cisco IOS Release 12.2(4)T, the Cisco IOS software did not provide a method for displaying asynchronous character mode traffic flowing out of an asynchronous line. Therefore, when a user tried to troubleshoot difficult asynchronous problems, the user had to use RS-232 datascopes to examine the data stream. This method is detailed and cumbersome. The Asynchronous Line Monitoring feature that is available in Cisco IOS Release 12.2(4)T allows the monitoring of inbound and outbound character mode asynchronous traffic on another terminal line. To monitor inbound or outbound asynchronous character mode traffic on the port to be monitored, enter the monitor traffic line command in privileged EXEC mode.
This feature increases the efficiency of the user who performs troubleshooting on asynchronous character mode traffic problems.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftasync.htm.
Note
ATM Service Level Monitoring (SLM)
The Cisco Service Assurance Agent (SA Agent) is an embedded performance monitoring utility in Cisco IOS software. The ATM Service Level Monitoring (SLM) feature expands the capabilities of the SA Agent to provide detailed monitoring statistics for your ATM network. Monitoring service levels for ATM connections allows service providers to ensure that their networks are meeting or exceeding the performance outlined in service level agreements (SLAs).
The ATM Service Level Monitoring feature can also be used with Cisco Networking Services (CNS). A device running CNS, such as the IE2100, can be used to retrieve the ATM performance statistics generated by the SA Agent. Additionally, these results can be passed to other devices running third-party monitoring software.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftatmslm.htm
Barge-In and Busy Line Verify Operator Services
The Barge-In and Busy Line Verify Operator Services feature enhances Simple Gateway Control Protocol (SGCP)/Media Gateway Control Protocol (MGCP) gateway conferencing capabilities to support the Busy Line Verification/Operator Interrupt (BLV/OI) feature. The Busy Line Verification feature permits an operator to establish a connection to a customer's line to verify a busy condition for a calling party. The Operator Interrupt feature allows the operator to speak to the customer and to connect the calling party and customer, if appropriate. These enhancements support other call flows such as call pickup with barge-in that require the ability to conference a second call into an existing two-party call without intervention by parties in the existing call. No explicit configuration is required to enable this feature.
The MGCP Basic CLASS and Operator Services feature introduced conferencing to support three-way calling on SGCP and MGCP gateways. It is described in MGCP Basic CLASS and Operator Services at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmgcpgr.htm.
Basic Service Relationships (H.225 Annex G)
Cisco's H.225 Annex G implements the minimal set of Annex G features needed to allow Cisco border elements to interoperate with any ClearingHouse border element. This feature enhances Cisco's H.225.0 Annex G support to include basic Service Relationships and Usage Reporting. The feature provides enhanced interoperability with a ClearingHouse border element and third party border element as well as address resolution for interdomain call routing.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_srang.htm.
BGP Conditional Route Injection
Cisco IOS software provides several methods in which you can originate a prefix into the Border Gateway Protocol (BGP). The existing methods include using the network or aggregate-address commands and redistribution. These methods assume the existence of more specific routing information (matching the route to be originated) in either the routing table or the BGP table.
The BGP Conditional Route Injection feature enables you to originate a prefix into BGP without the corresponding match. The routes are injected into the BGP table only if certain conditions are met. The most common condition is the existence of a less-specific prefix.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11bpri.htm.
Note
BGP Hide Local-Autonomous System
The BGP Hide Local-Autonomous System feature introduces the no-prepend keyword to the neighbor local-as command. The use of the no-prepend keyword allows a network operator to configure a Border Gateway Protocol (BGP) speaker to not prepend the local autonomous system number to any routes that are received from external peers. This feature can be used to help transparently change the autonomous system number of a BGP network and ensure that routes can be propagated throughout the autonomous system, while the autonomous system number transition is incomplete. Because the local autonomous is not prepended to these routes, external routes will not be rejected by internal peers during the transition from one autonomous system number to another.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11bhla.htm.
Note
BGP Link Bandwidth
The Border Gateway Protocol (BGP) Link Bandwidth feature is used to advertise the bandwidth of an autonomous system exit link as an extended community. The BGP Link Bandwidth feature is supported by the internal BGP (iBGP) and external BGP (eBGP) multipath features. The link bandwidth extended community indicates the preference of an autonomous system exit link in terms of bandwidth. The link bandwidth extended community attribute may be propagated to all iBGP peers and used with the BGP multipath features to configure unequal cost load balancing. When a router receives a route from a directly connected external neighbor and advertises this route to iBGP neighbors, the router may advertise the bandwidth of that link. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11b_lb.htm.
Note
BGP Multipath Load Sharing for Both eBGP and iBGP in an MPLS-VPN
The BGP Multipath Load Sharing for eBGP and iBGP feature allows you to configure multipath load balancing with both external BGP (eBGP) and internal BGP (iBGP) paths in Border Gateway Protocol (BGP) networks that are configured to use Multiprotocol Label Switching (MPLS) Virtual Private Networks (VPNs). This feature provides improved load balancing deployment and service offering capabilities and is useful for multi-homed autonomous systems and Provider Edge (PE) routers that import both eBGP and iBGP paths from multihomed and stub networks. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11bmpl.htm.
Note
BGP Prefix-Based Outbound Route Filtering
The BGP Prefix-Based Outbound Route Filtering feature uses Border Gateway Protocol (BGP) outbound route filter (ORF) send and receive capabilities to minimize the number of BGP updates that are sent between peer routers. The configuration of this feature can help reduce the amount of resources required for generating and processing routing updates by filtering out unwanted routing updates at the source. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11borf.htm.
Note
Call Admission Control Based on CPU Utilization
The Preauthentication with ISDN PRI feature permits the Cisco AS5300 and AS5800 universal access servers to deny incoming calls exceeding a preconfigured threshold, permitting the selection of a system CPU load level value. This feature helps ensure the quality of service (QoS) of existing calls and reliability of system processes by preventing system overload that is caused by excessive incoming calls. The feature rejects new digital calls (PRI, channel-associated signaling [CAS], and ISDN), with minor disruption to system users.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5800/sw_conf/ios_122/dt61294.htm.
Call Admission Control for H.323 VoIP Gateways
Before the call admission control feature was available, gateways did not have a mechanism to prevent calls from entering when certain resources were not available to process the call. This situation caused new calls to fail with unreported behavior and potentially caused the calls in progress to have quality-related problems.
This feature set provides the ability to support resource-based call admission control processes. These resources include system resources such as CPU, memory, and call volume and interface resources such as call volume.
If system resources are not available to admit the call, the following two kinds of actions are provided: system denial (which busy outs all of T1 or E1) or per-call denial (which disconnects, hairpins, or plays a message or tone). If the interface-based resource is not available to admit the call, the call is dropped from the session protocol (such as H.323).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftcac58.htm.
Note
Call Status Tracking Optimization
In an H.323 Voice-over-IP (VoIP) network, gatekeepers use information request (IRQ) messages to obtain information about a certain call or all calls from an endpoint (for example, an originating gateway). The gatekeeper can send an IRQ to request information from the endpoint, which responds with an information request response (IRR). The gatekeeper can also use the IRR Frequency field in the initial admission confirm (ACF) message to instruct the endpoint to periodically report with IRR messages during call admission.
Currently, the Cisco gatekeeper maintains the call states of all calls it has admitted to track bandwidth usage. In addition, the gatekeeper must be able to reconstruct call structures for a newly transferred gateway from an alternate gatekeeper, if a gatekeeper switchover has occurred. In a gatekeeper switchover, the new gatekeeper sends an IRQ message with the call reference value (CRV) set to zero to the newly registered gateway to obtain information about existing calls before the switchover.
If a gateway supports a large volume of calls, the number of IRR messages as responses to an IRQ with the CRV set to zero could be CPU intensive and cause congestion. Additionally, if a gatekeeper serves many endpoints or high-capacity gateways, the IRQ requests and the resulting IRR messages received can flood the network, causing high CPU utilization and network congestion.
The Call Status Tracking Optimization feature provides the following methods to address this potential problem:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_csto2.htm.
Call Tracker show Commands Extensions
Before Cisco IOS Release 12.2(11)T, the show calltracker active EXEC command and the show calltracker history EXEC command provided a simple way to examine the Call Tracker active table and Call Tracker history table in chronological order. The extensions to these commands available in Cisco IOS Release 12.2(11)T allow the command output to be reverse collated (output from most recent to least recent) or to be filtered by call category or service type. Historical data for disconnected call sessions can be filtered by subsystem type.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftcall.htm.
CEF on Multipoint GRE Tunnels
The CEF on Multipoint GRE Tunnels feature enables Cisco Express Forwarding (CEF) switching of IP traffic to and from multipoint generic routing encapsulation (GRE) tunnels. Tunnel traffic can be forwarded to a prefix through a tunnel destination when both the prefix and the tunnel destination are specified by the application.
Note
Certificate Autoenrollment
The Certificate Autoenrollment feature allows you to configure your router to automatically request a certificate from the certification authority (CA) that is using the parameters in the configuration. Thus, operator convention is no longer required at the time the enrollment request is sent to the CA server.
Automatic enrollment will be performed on startup for any trustpoint CA that is configured and does not have a valid certificate. When the certificate—which is issued by a trustpoint CA that has been configured for autoenrollment—expires, a new certificate is requested. Although this feature does not provide seamless certificate renewal, it does provide unattended recovery from expiration.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftautoen.htm.
Note
Certificate Enrollment Enhancements
The Certificate Enrollment Enhancements feature introduces five new subcommands to the crypto ca trustpoint command—ip-address (ca-trustpoint), password (ca-trustpoint), serial-number, subject-name, and usage. These commands provide new options for certificate requests and allow users to specify fields in the configuration instead of having to go through prompts. (However, the prompting behavior remains the default if this feature is not enabled.) Thus, users can preload all necessary information into the configuration, allowing each router to obtain its certificate automatically when it is booted.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftenrol2.htm.
Note
Circuit Interface Identification Persistence for SNMP
The Circuit Interface MIB (CISCO-CIRCUIT-INTERFACE-MIB) provides a MIB object (cciDescr) which can be used to identify individual circuit-based interfaces for SNMP monitoring. The Circuit Interface Identification Persistence for SNMP feature maintains this user-defined name of the circuit across reboots, allowing the consistent identification of circuit interfaces. Circuit Interface Identification Persistence is enabled using the snmp mib persist circuit global configuration command.
Note
Cisco Gateway Management Agent
The Cisco Gateway Management Agent (CGMA) feature provides an eXtensible Markup Language (XML) interface to support real-time management of a Cisco IOS gateway (GW). Currently, GWs provide statistics using Simple Network Management Protocol (SNMP) and do not support real-time polling. The CGMA feature allows GWs to communicate with third-party management applications using XML over TCP/IP.
Note
Cisco H.323 Multizone Enhancements
The Cisco H.323 Multizone Enhancements feature enables the Cisco gateway to provide information to the gatekeeper with the use of additional fields in the RAS (registration, admission, and status) messages.
Previously, the source gateway attempted to set up a call to a destination IP address as provided by the gatekeeper in an Admission Confirm (ACF) message. If the gatekeeper was unable to resolve the destination E.164 phone number to an IP address, the incoming call was terminated.
This version of the H.323 software adds support to allow a gatekeeper to provide additional destination information and modify the destinationInfo field in the ACF. The gateway will include the canMapAlias associated destination information in setting up the call to the destination gateway.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5800/sw_conf/ios_122/pul0244x.htm.
Note
Cisco IOS Telephony Service Version 2.0
The Cisco IOS Telephony Service Version 2.0 feature was previously released in Cisco IOS Release 12.2(8)T. In Cisco IOS Release 12.2(11)T, there are minor enhancements to this feature, which is now referred to as Cisco IOS Telephony Service Version 2.01. Refer to the following document for information about the enhancements added to this release:
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/ipkey2.htm.
Cisco VCWare Version Checker
The Cisco VCWare Version Checker feature adds Cisco VCWare version checker warning output at bootup and when you use the show vfc version vcware and show vfc version dspware commands.
This new version checker feature detects possible mismatches between Cisco IOS software and Cisco VCWare and DSPWare. If a software mismatch is found, a compatibility mismatch warning is output at bootup and when the show vfc version commands are used. If no mismatch is found, there is no advisory output. Because the new information is advisory only, there is no action taken whether the software is compatible or incompatible.
This feature applies only to the Cisco AS5300. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftvdspck.htm
CISCO-BULK-FILE-MIB Enhancements
The Cisco Bulk File Creation MIB (CISCO-BULK-FILE-MIB.my) is a MIB module for creating and deleting bulk files of SNMP data for file transfer. The CISCO-BULK-FILE-MIB Enhancements feature enhances the Cisco Bulk File Creation MIB to support selective-row-transfer and notification-on-file-creation. Prior to this enhancement, when the MIB was used to dump large tables (for example, the ccHistoryTable), much of the data transfer consisted of duplicated data. This feature allows the SNMP manager to specify a starting row in the SNMP Get request.
This feature also introduces a notification that can be sent when file creation is complete or when there is an error during file creation. Specifically, this feature modifies the CISCO-BULK-FILE-MIB by introducing four new MIB objects (cbfDefineFileNotifyOnCompletion, cbfDefineObjectTableInstance, cbfDefineObjectNumEntries, cbfDefineObjectLastPolledInst) and a new notification object (cbfDefineFileCompletion). For details, refer to the CISCO-BULK-FILE-MIB.my file, available through Cisco.com MIB FTP site.
Note
CISCO-SIP-UA-MIB Enhancements Providing Functional Parity to SIP related CLI
The CISCO-SIP-UA-MIB Enhancements Providing Functional Parity to session initiation protocol (SIP) related CLI feature has Simple Network Management Protocol (SNMP)/command-line interface (CLI) MIB enhancements to maintain parity with SIP features released to date.
No documentation work is required. The MIB is "self-documenting."
CNS Agents SSL Security
CNS Agents SSL Security is a Cisco IOS software feature that allows for the configuration of a secure connection between the CNS Agent, running on the Cisco IOS software-based device, and a CNS Server. Secure Socket Layer (SSL) encryption for CNS connections is enabled on the Cisco IOS device (CNS Agent) side using the encrypt keyword with the cns config initial or cns config partial global configuration mode commands.
Note
CNS Configuration Agent
CNS is a foundation technology for linking users to network services. CNS SDK accomplishes this by making applications network-aware and increasing the intelligence of the network elements. CNS SDK provides building blocks to a range of customers in market segments such as Enterprise, service provider, independent software vendors, and system integrators.
The CNS Configuration Agent supports routing devices by providing:
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Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcns_ca.htm.
Note
CNS Event Agent
CNS is a foundation technology for linking users to network services. CNS SDK accomplishes this by making applications network-aware and increasing the intelligence of the network elements. CNS SDK provides building blocks to a range of customers in market segments such as Enterprise, service provider, independent software vendors, and system integrators.
The CNS Event Agent is part of the Cisco IOS infrastructure that allows Cisco IOS applications, for example CNS Configuration Agent, to publish and subscribe to events on a CNS Event Bus. CNS Event Agent works in conjunction with CNS Configuration Agent.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcns_ea.htm.
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Configuring a Gatekeeper to Provide Nonavailability Information for Terminating Endpoints
An H.323 Location Request (LRQ) message is sent by a gatekeeper to another gatekeeper to request a terminating endpoint. The second gatekeeper determines the appropriate endpoint on the basis of the information contained in the LRQ message. However, sometimes all the terminating endpoints are busy servicing other calls and none are available. If you configure the lrq reject-resource-low command, the second gatekeeper will reject the LRQ request if no terminating endpoints are available. If the command is not configured, the second gatekeeper will allocate and return a terminating endpoint address to the sending gatekeeper even if all the terminating endpoints are busy.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_lrqrj.htm.
Connect-Info RADIUS Attribute 77
The Connect-Info RADIUS Attribute 77 feature enables the network access server (NAS) to report Connect-Info (attribute 77) in accounting "start" and "stop" records that are sent to the RADIUS client. The "start" and "stop" records allow you to compare transmit and receive speeds and have a realistic view of a user session. Comparing transmit and receive speeds is important because many modem speeds are often different at the end of the modem connection (after negotiation).
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftattr77.htm.
Customer Profile Idle Timer Enhancements for Interesting Traffic
The Customer Profile Idle Timer Enhancements for Interesting Traffic feature supports a PPP idle timer based on interesting traffic for dialer interfaces.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftprfidl.htm.
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Default VPDN Group Template
The Default VPDN Group Template feature introduces the ability to configure global default values for virtual private dialup network (VPDN) parameters in a VPDN template. These global default values are applied to all VPDN groups, unless specific values are configured for individual VPDN groups. Previously, the Cisco IOS software required that VPDN parameters be configured for each individual VPDN group if the system default values were not desired.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdevpdn.htm.
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DF Bit Override Functionality with IPSec Tunnels
The DF Bit Override Functionality with IPSec Tunnels feature allows customers to configure the setting of the DF bit when encapsulating tunnel mode IPSec traffic on a global or per-interface level. Thus, if the DF bit is set to clear, routers can fragment packets regardless of the original DF bit setting. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftdfipsc.htm.
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DHCP Client—Dynamic Subnet Allocation API
The DHCP Client-Dynamic Subnet Allocation API feature is an application program interface (API) that is called by the DHCP Server-On-Demand Address Pool Manager feature for obtaining a subnet or releasing a subnet to the source server via DHCP. This feature allows automated configuration of layer 3 devices for simplified deployment.
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DHCP Client on WAN Interfaces
The DHCP Client on WAN Interfaces feature extends the Dynamic Host Configuration Protocol (DHCP) to allow PPP over ATM (PPPoA) and certain ATM interfaces to acquire an IP address through DHCP. By using DHCP rather than the IP Control Protocol (IPCP), a DHCP client can acquire other useful information such as DNS server addresses, the DNS default domain name, and default route.
Previously, the ip address dhcp interface configuration command could only be used on Ethernet interfaces. This feature allows the ip address dhcp command to be used on WAN interfaces.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftwandhp.htm.
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DHCP Relay—MPLS VPN Support
The DHCP relay agent information option (option 82) enables a Dynamic Host Configuration Protocol (DHCP) relay agent to include information about itself when forwarding client-originated DHCP packets to a DHCP server. The DHCP server can use this information to implement IP address or other parameter-assignment policies. The DHCP relay agent information option is organized as a single DHCP option that contains one or more suboptions that convey information known by the relay agent.
In some environments, a relay agent resides in a network element that also has access to one or more Multiprotocol Label Switching (MPLS) Virtual Private Networks (VPNs). A DHCP server that wants to offer service to DHCP clients on those different VPNs needs to know the VPN in which each client resides. The network element that contains the relay agent typically knows about the VPN association of the DHCP client and includes this information in the relay agent information option.
The DHCP Relay-MPLS VPN Support feature allows the relay agent to forward this necessary VPN-related information to the DHCP server using the following three suboptions of the DHCP relay agent information option:
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The DHCP Relay-MPLS VPN Support feature enables a network administrator to conserve address space by allowing overlapping addresses. The relay agent can now support multiple clients on different VPNs, and many of these clients from different VPNs can share the same IP address.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdhmpls.htm.
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DHCP Relay Agent Support for Unnumbered Interfaces
Relay agents are used to forward requests and replies between clients and servers when they are not on the same physical subnet. Relay agent forwarding is distinct from the normal forwarding of an IP router, where IP datagrams are switched between networks somewhat transparently. Relay Agents receive Dynamic Host Configuration Protocol (DHCP) messages and then generate a new DHCP message to send out on another interface.
The Cisco IOS DHCP relay agent supports IP unnumbered interfaces. The DHCP relay agent automatically adds a static host route specifying the unnumbered interface as the outbound interface.
DHCP Server—On-Demand Address Pool Manager
The DHCP Server-On-Demand Address Pool Manager is a feature in which pools of IP addresses can be dynamically increased or reduced in size depending on the address utilization level. This feature supports address assignment using the Dynamic Host Configuration Protocol (DHCP) for customers using private addresses. Each on-demand address pool (ODAP) is configured and associated with a particular Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN).
When configured, the ODAP is populated with one or more subnets leased from a source server and is ready to serve address requests from DHCP clients or from PPP sessions. The source server can be a remote DHCP server or a RADIUS server (via AAA). Currently, only the Cisco Access Registrar RADIUS server supports ODAPs. Subnets can be added to the pool when a certain utilization level (high utilization mark) is achieved. When the utilization level falls below a certain level (low utilization mark), a subnet can be returned to the server from which it was originally leased.
This feature allows customers to optimize their use of IP addresses, thus conserving address space.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftondhcp.htm.
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DHCP Server—Option to Ignore All BOOTP Requests
The DHCP Server—Option to Ignore All BOOTP Requests feature introduces the following new global configuration command: ip dhcp bootp ignore. This command allows the Cisco IOS DHCP server to ignore received BOOTP requests.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdbootp.htm.
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Dialer CEF
The Dialer CEF feature introduces Cisco Express Forwarding (CEF) support for dialer interfaces. The Dialer CEF feature allows packets to be CEF switched across dialer interfaces rather than being low-end switched (LES) or fast switched. Compared to fast switching, CEF switching support improves switching performance by decreasing CPU utilization and lowering the packet loss rate. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftdlrcef.htm.
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Dialer Persistent
The Dialer Persistent feature allows the connection settings in a dial-on-demand routing (DDR) dialer profile to be configured as persistent, that is, the connection is not torn down until the shutdown EXEC command is entered on the dialer interface. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftdperst.htm.
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Dialer Watch Connect Delay
The Dialer Watch Connect Delay feature introduces the ability to configure a delay in bringing up a secondary link when a primary link that is monitored by Dialer Watch goes down and is removed from the routing table. Previously, the router would instantly dial a secondary route without allowing time for the primary route to come back up. When the Dialer Watch Connect Delay feature is configured, the router will check for availability of the primary link at the end of the specified delay time before dialing the secondary link.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdialwl.htm.
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Distributed Management Event and Expression MIB Persistence
The MIB Persistence feature allows the SNMP data of an MIB to be persistent across reloads; that is, MIB information retains the same set object values each time the user reboots. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftmibpr1.htm.
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Distributed Management Event MIB Conformance to RFC 2981
Prior to Cisco IOS Release 12.2(4)T3, Event MIB support in Cisco IOS software was based on the IETF internet draft version. In Cisco IOS Release 12.2(4)T3, the Cisco implementation of the EVENT-MIB was updated to comply with the finalized version of the Event MIB, as defined in RFC 2981. For details, see RFC 2981, available through the IETF web site at http://www.ietf.org.
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DLSw+ Enhanced Load Balancing
In a network with multiple capable paths, the Data Link Switch Plus (DLSw+) Load Balancing Enhancements feature improves traffic load balancing between peers by distributing new circuits based on existing loads and the desired ratio.
For each capable peer (peers that have the lowest or equal cost specified), the DLSw+ Load Balancing feature calculates the difference between the desired and the actual ratio of circuits being used on a peer. It detects the path that is underloaded in comparison to the other capable peers and assigns new circuits to that path until the desired ratio is achieved.
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DLSw+ Peer Group Clusters
The DLSw+ Peer Group Clusters feature reduces the explorer packet replication that typically occurs in a large Data Link Switch Plus (DLSw+) peer group design, where there are multiple routers connected to the same LAN.
The DLSw+ Peer Group Clusters feature associates DLSw+ peers (that are connected to the same LAN) into logical groups. Once the multiple peers are defined in the same peer group cluster, the DLSw+ Border Peer recognizes that it does not have to forward explorers to more than one member within the same peer group cluster.
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DTMF Events Through SIP Signaling
The DTMF Events Through SIP Signaling feature adds support for sending dual tone multifrequency (DTMF) notifications using NOTIFY messages from a session initiation protocol (SIP) gateway. The use of DTMF signaling for this feature enables support for advanced telephony services. Currently there are a number of application servers and service creation platforms that do not support media connections. To provide value added services to the network, these servers and platforms need to be aware of signaling events from a specific participant in the call. After the server or platform is aware of the DTMF events that are being signaled, it can use third-party call control, or other signaling mechanisms, to provide enhanced services. Examples of the types of services and platforms that are supported by this feature are various voice web browser services, Centrex switches/business service platforms, calling card services, and unified message servers. All of these applications require a method for the user to communicate with the application outside of the media connection. The Preauthentication with ISDN PRI feature provides this signaling capability.
The output generated by the show sip-ua statistics command displays the enhanced SIP Response and Total Traffic Statistics available with the new feature.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftnotify.htm.
DTMF Relay for SIP calls Using Named Telephone Events
The DTMF Relay using Named Telephone Events feature adds support for relaying dual tone multifrequency (DTMF) tones and hookflash events in session initiation protocol (SIP) on Cisco Voice over IP (VoIP) gateways (note that this feature is implemented for SIP only). Using Named Telephone Events (NTE) to relay DTMF tones provides a standardized means of transporting DTMF tones in RTP packets according to section 3 of RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, developed by the Internet Engineering Task Force (IETF) Audio/Video Transport (AVT) working group. RFC 2833 defines formats of NTE RTP packets used to transport DTMF digits, hookflash, and other telephony events between two peer endpoints.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ft_dtmf.htm.
Easy VPN Server
The Easy VPN Server feature introduces server support for the Cisco VPN Client Release 3.x software clients and Cisco VPN hardware clients. It allows a remote end user to communicate using IP Security (IPSec) with any Cisco IOS Virtual Private Network (VPN) gateway. Centrally managed IPSec policies are "pushed" to the client by the server, minimizing configuration by the end user.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftunity.htm.
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Enable Multilink PPP via RADIUS for Preauthentication User
The Enable Multilink PPP via RADIUS for Preauthentication User feature allows you to selectively enable and disable Multilink PPP (MLP) negotiation for different users via RADIUS vendor-specific attribute (VSA) preauth:ppp-multilink=1.
You can enable MLP by configuring the ppp multilink command on an interface, but then this command enables MLP negotiation for all connections and users on that interface; that is, you cannot selectively enable or disable MLP negotiation for specific connections and users on an interface.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftppprad.htm
Encrypted Vendor-Specific Attributes
The Encrypted Vendor-Specific Attributes feature introduces support for the following three types of string vendor-specific attributes (VSAs):
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftencvsa.htm.
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Enhanced Codec Support for SIP Using Dynamic Payloads
The Enhanced Codec Support for SIP Using Dynamic Payloads feature enhances codec selection and payload negotiation between originating and terminating session initiation protocol (SIP) gateways. The Enhanced Codec Support for SIP Using Dynamic Payloads feature provides the following SIP enhancements:
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The feature adds support, which varies on different platforms, for eight additional codecs:
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The feature adds support for dynamic payloads by expanding SIP ability to advertise and negotiate available codecs. The feature also expands the Session Description Protocol (SDP) message body of SIP INVITE requests, which describe codec capabilities of the SIP gateway.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftcodec.htm.
Enhanced Debug Capabilities for Cisco Voice Gateways
The Enhanced Debug Capabilities for Cisco Voice Gateways feature provides a uniform call identifier to track calls end-to-end, filter calls based on certain criteria, and provide more concise debug commands. Previously if all debugs were turned on, the debug output would wrap around, so viewing a smaller amount of debug output to effectively identify the problem areas was critical.
Another requirement was single-call tracing that enables a single call, based on certain criteria, to be traced end-to-end in the gateway. A generic format to identify the trace call was required also and was needed across Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.323, voice telephony service providers (VTSPs), session applications, interactive voice response (IVR), Call Control Applications Programming Interface (CCAPI) and digital signal processors (DSPs).
The voice call debug command was implemented to give the user the choice of displaying the full GUID or reduces the length of GUID by displaying the headers only. The full-guid keyword displays a full call trace and the default is displaying the header only.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_dbgsy.htm
Enhanced Password Security
The Enhanced Password Security feature allows you to configure Message Digest 5 (MD5) encryption for username passwords. Before the introduction of this feature, there were two types of passwords associated with usernames: Type 0, which is a clear text password visible to any user who has access to privileged mode on the router, and type 7, which is a password with a weak, exclusive, or type encryption. Type 7 passwords can be retrieved from the encrypted text by using publicly available tools.
Use the username secret command to configure a username and an associated MD5-encrypted secret.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_md5.htm.
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Enhanced Test Command
The Enhanced Test Command feature introduces two new commands—aaa user profile and aaa attribute—that allow you to create a named user profile with calling line identification (CLID) or dialed number identification service (DNIS) attribute values, which can be associated with a test aaa group command.
Use the aaa attribute command to add CLID or DNIS attribute values to a user profile, which is created by using the aaa user profile command. The CLID or DNIS attribute values can be associated with the record that is going out with the user profile (via the test aaa group command), thereby providing the RADIUS server with access to CLID or DNIS attribute information for all incoming calls. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftaaacmd.htm.
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Enhanced VoiceXML Diagnostics
With the Enhanced VoiceXML Diagnostics feature, debugging output can be filtered for all VoiceXML applications except the application named in the debug condition application voice command. When this command is configured, the gateway displays debugging messages only for the specified VoiceXML application when using the debug vxml and debug http client commands.
Refer to the following documents for additional information:
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http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ivrapp/index.htm.
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http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/rel_docs/vxmlprg/index.htm.
Enhancements for the Cisco VG200 Voice Gateway
The Enhancements for the Cisco VG200 Voice Gateway feature provides the Cisco VG200 platform with increased voice gateway feature parity to the Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series. This feature is also supported on the Cisco VG200XM platform upgrade. The Cisco VG200XM is new for Cisco IOS Release 12.2(11)T and is a more powerful version of the Cisco VG200, offering higher processing power and improved performance.
The Cisco VG200 platforms provide the following default memory options:
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The Enhancements for the Cisco VG200 Voice Gateway feature includes the following features:
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Enhancing Raw Buffer Management: Audit and Prepopulation for Channel-Associated Signaling
This feature implements an audit process to reclaim leaking raw buffers on a channel-associated signaling (CAS) interface.
Buffers pass voice data between subsystems in a voice-call-control infrastructure. However, pool management and the improper usage of buffers result in either process memory exhaustion or system crashes. Both are relatively difficult to troubleshoot. Auditing raw buffers allows you to detect and reclaim leaking buffers on the CAS interface and thus to improve buffer availability and be memory efficient.
Auditing raw buffers provides the following monitor scheme: When a raw message is allocated, the start time stamp is recorded. An audit process periodically (every 2 minutes) reclaims active raw messages that are more than 10 minutes old and returns the buffers to the appropriate pool. The 10-minute window allows enough time for all the call-related events to pass through.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftrawr11.htm
Event Tracer
The Event Tracer feature provides a binary trace facility for troubleshooting Cisco IOS software. This feature gives Cisco service representatives additional insight into the operation of the Cisco IOS software and can be useful in helping to diagnose problems in the unlikely event of an operating system malfunction or, in the case of redundant systems, route processor switchover.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s18/evnttrcr.htm.
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Fax and Modem Pass-Through over VoIP
Fax and modem pass-through are supported on the Cisco 2600 series, Cisco 3600 series, and the Cisco 3700 series modular access routers beginning in Cisco IOS Release 12.2(11)T.
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On detection of the fax or modem tone on an established VoIP call, the gateways switch into modem fax or pass-through mode: the voice codec and configuration is suspended and the pass-through parameters are loaded for the duration of the fax or modem session. This changes the bandwidth needed for the call to the equivalent of G.711.
With pass-through, the fax or modem traffic is carried between the two gateways in RTP packets, using an uncompressed format resembling the G.711 codec. Packet redundancy may be used to mitigate the effects of packet loss in the IP network. Even so, fax and modem pass-through remain susceptible to packet loss, jitter and latency in the IP network. The two endpoints must be clocked synchronously for this type of transport to work predictably.
The Fax and Modem Pass-Through feature is also known as Voice Band Data (VBD) by the International Telecommunication Union (ITU). VBD refers to the transport of fax or modem signals over a voice channel through a packet network with an encoding appropriate for fax or modem signals. The minimum set of coders for VBD mode is G.711 ulaw and alaw with VAD disabled. For modem transport, Echo cancellation is also be disabled.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm.
Fax Detection (Single-number Voice and Fax)
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On Cisco AS5300, Cisco AS5350, and Cisco AS5400 gateways that are equipped with voice feature cards (VFCs), the fax detection feature lets service providers deploy unified communication applications in which each subscriber has a single E.164 number for both voice mail and fax mail. When configured for fax detection, the gateway automatically listens to incoming calls to discriminate between voice and fax. The gateway then routes the calls to the appropriate application or server.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm.
Fax Detection for VoiceXML
With the Fax Detection for VoiceXML feature, when a VoiceXML fax detection application is configured on the gateway, callers can dial a single number for both voice and fax calls. The gateway automatically detects that a call is a fax transmission by listening for comfort noise generation (CNG), the distinctive fax "calling" tone. When configured for fax detection, the Cisco VoiceXML gateway continuously listens to incoming calls to determine which calls are voice or fax. The gateway then routes the calls to the appropriate application or media server.
Refer to the following documents for additional information:
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http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ivrapp/index.htm.
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http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/rel_docs/vxmlprg/index.htm.
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Fax Relay Packet Loss Concealment
The Fax Relay Packet Loss Concealment feature improves the current real-time fax over IP (commonly known as fax relay) implementation in Cisco gateways, allowing fax transmissions to work reliably over higher packet loss conditions.
In addition, this feature includes enhanced real-time fax debug capabilities and statistics. These debugs and statistics will give better visibility into the real-time fax operation in the gateway, allowing for improved field diagnostics and troubleshooting.
These improvements include configuration of fax relay Error Correction Mode (ECM) on the Voice over IP (VoIP) dial peer. ECM provides for error-free page transmission. This mode is available on fax machines that include memory for storage of the page data (usually high-end fax machines).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ft_0393.htm.
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G.Clear, GSMFR, and G.726 Codecs and Modem and Fax Pass-Through for Cisco Universal Gateways
The following features are now available on Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850 universal gateways.
The G.Clear, GSMFR, and G.726 Codecs and Modem and Fax Pass-Through for Cisco Universal Gateways feature provides support for G.Clear, GSMFR, and G.726 codecs, as well as support for modem and fax Pass-through for Cisco universal gateways.
G.Clear guarantees bit integrity when transferring a DS-0 through a gateway server, supports the transporting of nonvoice circuit data sessions through a Voice over IP (VoIP) network, and enables the VoIP networks to transport ISDN and switched 56 circuit-switched data calls. With the availability of G.Clear, ISDN data calls that do not require bonding can be supported.
The GSMFR codec was introduced in 1987. The GSMFR speech coder has a frame size of 20 ms and operates at a bit rate of 13 kbps. GSMFR is an Regular Pulse Excited - Linear Predictive (RPE-LTP) coder.
The G.726 Adaptive Differential PCM (ADPCM) voice codec operates at bit rates of 16, 24, and 32 kbps. ADPCM provides the following:
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In addition, modem and fax pass-through services are supported. When service providers and aggregators are implementing VoIP, they sometimes cannot separate fax or data traffic from voice traffic. These carriers that aggregate voice traffic over VoIP infrastructures require service offerings to carry fax and data as easily as voice.
On detection of the modem answer tone, the gateways switch into modem pass-through mode. With modem pass-through, the modem traffic is carried between the two gateways in RTP packets, using an uncompressed or lightly compressed voice codec—G.711 ulaw, G.711 alaw, or Voice Band Data (VBD). Packet redundancy may be used to mitigate the effects of packet loss in the IP network. Even so, modem pass-through remains susceptible to packet loss and jitter and latency in the IP network.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_ghost.htm.
Gatekeeper Endpoint Control Enhancements
The Gatekeeper Endpoint Control Enhancements feature provides enhancements to the Cisco IOS gatekeeper, including commands to allow both forced unregistration of an endpoint and rejection of new registrations or calls when a Gatekeeper Transaction Message Protocol (GKTMP) server is down or unreachable. This feature also provides both forced unregistration of an endpoint using a GKTMP command from an application server and a command to enable faster reconnection to a GKTMP server when its TCP connection fails.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftgkece2.htm
Gatekeeper-to-Gatekeeper Authentication
The Gatekeeper-to-Gatekeeper Authentication feature provides additional security for H.323 networks by introducing the ability to validate intra-domain and interdomain gatekeeper-to-gatekeeper Location Request (LRQ) messages on a per-hop basis. When used in conjunction with per-call security using the interzone ClearToken (IZCT), network resources and security holes are protected from hackers. The IZCT was introduced in the Inter-Domain Gatekeeper Security Enhancement feature released in Cisco IOS Release 12.2(2)XA and Cisco IOS Release 12.2(4)T.
The Gatekeeper-to-Gatekeeper Authentication feature provides a Cisco Access Token (CAT) to carry authentication within zones. The CAT is used by adjacent gatekeepers to authenticate each other and is configured on a per-zone basis. In addition, service providers can specify inbound passwords to authenticate LRQ messages coming from foreign domains and outbound passwords to be included in LRQ messages to foreign domains.
This release documents two new commands: security password-group and security zone.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_idlrq.htm.
Generic Routing Encapsulation (GRE) Tunnel Keepalive
The GRE Tunnel Keepalive feature provides the capability of configuring keepalive packets to be sent over IP-encapsulated generic routing encapsulation (GRE) tunnels. You can specify the rate at which keepalives will be sent and the number of times that a device will continue to send keepalive packets without a response before the interface becomes inactive.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/grekpliv.htm.
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Global Modem Counters
The Global Modem Counters feature adds two new global call counters for ISDN calls to the Cisco IOS software. In Cisco IOS Release 12.2(11)T, this feature is supported only on the Cisco AS5800 universal access server. The CISCO-POP-MGMT-MIB has been updated with two new objects, cpmCallVolSuccISDNDigital and cpmCallVolAnalogCallClearedNormally. The cpmCallVolSuccISDNDigital object allows the Cisco IOS software to track the number of successful incoming and outgoing ISDN digital data calls that have occurred since the system was started. The cpmCallVolAnalogCallClearedNormally object allows the Cisco IOS to track the number of successful incoming and outgoing analog data calls.
No new commands have been introduced with this feature. To use this feature, enable System Network Management Protocol (SNMP) and the corresponding OIDs for these new objects. To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
Globalized Cadence and Tone for Cisco IOS Gateways
Because previously Cisco CallManager and Cisco IOS gateways were configured independently and may lead to configuration mismatches, Cisco CallManager is now preconfigured to provide cadences and tones for the user's locale. It is no longer necessary for you to configure the cptone command on the gateway. This feature shows the user how to verify which locale is preconfigured on Cisco CallManager.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_glbl.htm.
GTD for GKTMP using SS7 Interconnect version 2.0
The GTD for GKTMP using SS7 Interconnect version 2.0 feature consists of the following two features:
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The GTD for GKTMP Using SS7 Interconnect version 2.0 feature provides additional functionality to Cisco gateways and gatekeepers in a Cisco SS7 Interconnect for Voice Gateways Solution. The generic transparency descriptor or generic telephony descriptor (GTD) format is defined in the a Cisco proprietary draft. GTD format defines parameters and messages of existing SS7 ISUP protocols in text format and allows SS7 messages to be carried as a payload in the H.225 registration, admission, and status (RAS) messages between the GW and GK. GTD messages can also be transported between GWs and GKs in H.323 messages. With the GTD feature, the GK extracts the GTD message and the external route server derives routing and accounting information based upon the GTD information provided from the Cisco Gatekeeper Transaction Message Protocol (GKTMP).
Currently routing on Cisco GWs is based on generic parameters such as originating number, destination number, and port source. Adding support for SS7 ISUP messages allows the VoIP network to use additional routing enhancements found in traditional TDM switches.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftgtdpy2.htm
H.323 Call Redirection Enhancements
The user-to-user information element (UUIE) of the Facility message is used primarily for call redirection. The UUIE contains a field, facilityReason, that indicates the nature of the redirection. The H.323 Call Redirection Enhancements feature adds support for two of the reasons: routeCallToGatekeeper and callForwarded. It also provides a nonstandard method for using the Facility message to effect call transfer.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcallrd.htm.
Note
H.323 Dual Tone Multifrequency (DTMF) Relay Using Named Telephone Events
Until now, the Named Telephone Event (NTE) method of dual tone multifrequency (DTMF) relay was available on Cisco gateways only for Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP) gateways. The H.323 Dual Tone Multifrequency Relay Using Named Telephone Events feature adds support for this method for H.323 gateways.
Cisco H.323 gateways advertise capabilities using the H.245 capabilities messages. By default, they advertise that they can receive all DTMF relay modes. If the capabilities of the remote gateway do not match, the Cisco H.323 gateway transmits DTMF tones as in-band voice. Configuring DTMF relay on the Cisco H.323 gateway sets preferences for how the gateway handles DTMF transmission. If multiple methods are configured, the priority is as follows:
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In addition to support for NTE, the H.323 Dual Tone Multifrequency Relay Using Named Telephone Events feature provides support for asymmetrical payload types. Payload types can differ between local and remote endpoints. Therefore, the Cisco gateway can transmit one payload type value and receive a different payload type value.
There are no new or modified commands.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/fth3dtmf.htm
H.323 Redundant Zone Support
The Redundant H.323 Zone Support feature allows users to configure multiple gatekeepers to service the same zone or technology prefix. This feature can be used with the Gateway Support for Alternate Gatekeepers feature, which allows a user to configure a gateway to point to two gatekeepers (one as the primary and the other as the alternate). Together, these features allow a user to configure a Cisco gateway to send location requests (LRQs) to two or more Cisco gatekeepers—one as a primary and the others as back up gatekeepers. All gatekeepers are active. The gateway can choose to register with any one (but not all) at a given time.
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iBGP Multipath Load Sharing
When a Border Gateway Protocol (BGP) speaker router with no local policy configured receives multiple network layer reachability information (NLRI) from the internal BGP for the same destination, the router will choose one internal BGP path as the best path. The best path is then installed in the IP routing table of the router.
The Internal BGP Multipath Load Sharing feature enables the BGP speaker router to select multiple internal BGP paths as the best paths to a destination. The best paths or multipaths are then installed in the IP routing table of the router.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11bmls.htm
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IGMP MIB Support Enhancements for SNMP
The Internet Group Management Protocol (IGMP) is used by IP hosts to report their multicast group memberships to neighboring multicast routers. The IGMP MIB describes objects that enable users to remotely monitor and configure IGMP using Simple Network Management Protocol (SNMP). It also allows users to remotely subscribe and unsubscribe from multicast groups. The IGMP MIB Support Enhancements for SNMP feature adds full support of RFC 2933 (Internet Group Management Protocol MIB) in Cisco IOS software. There are no new or modified Cisco IOS commands associated with this feature.
For complete details on the IGMP MIB, see the IGMP-STD-MIB.my file available from the Cisco MIB website on Cisco.com at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
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IKE—Initiate Aggressive Mode
The IKE—Initiate Aggressive Mode feature allows you to specify RADIUS Tunnel attributes (Tunnel-Client-Endpoint [66] and Tunnel-Password [69]) for an IPSec peer and to initiate an IKE aggressive mode negotiation with the tunnel attributes. This feature is best implemented in a crypto hub-and-spoke scenario, in which the spokes initiate IKE aggressive mode negotiation with the hub by using the preshared keys that are specified as tunnel attributes and stored on the AAA server. This scenario is scalable because the preshared keys are kept at a central repository (the AAA server) and more than one hub router and one application can use the information.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ikeag.htm.
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Integrated Signaling Link Terminal
The Integrated Signaling Link Terminal feature pulls existing Cisco distributed Message Transfer Part (MTP) SS7 signaling architecture functionality—previously available only on Cisco 26xx-based signaling link terminals (SLTs)—directly onto a single Cisco AS5350 or Cisco AS5400. Like the Cisco 26xx-based SLT, the Integrated SLT on a Cisco AS5350 or Cisco AS5400 backhauls upper-layer Signaling System 7 (SS7) protocols across an IP network using Cisco Reliable User Datagram Protocol (RUDP), terminating the MTP1 and MTP2 layers of the SS7 protocol stack at the Media Gateway Controller (MGC).
Using the 2-, 4-, or 8-PRI dial feature card (DFC) or the CT3 (28-PRI) DFC card, this feature is designed for small points of presence (POPs) that require only one or two network access servers (NASs) or Voice-over-IP (VoIP) gateways as part of a dial or VoIP solution. This feature eliminates the use of the Cisco 26xx-based SLT in the product configuration.
When the Integrated Signaling Link Feature feature is implemented, a Cisco AS5350 or Cisco AS5400 functions as an SS7 signaling data link terminal and as a NAS, voice gateway, or both when universal ports are used.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftintslt.htm.
Integrated IS-IS Point-to-Point Adjacency Over Broadcast Media
When a network consists of only two networking devices that are connected to broadcast media and using the integrated IS-IS protocol, it is better for the system not to have to handle the link as a broadcast link but rather as a point-to-point link. The Integrated IS-IS Point-to-Point Adjacency Over Broadcast Media feature introduces a new command to make IS-IS behave as a point-to-point link between the networking devices. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftissp2p.htm.
Note
Interactive Voice Response Version 2.0 on VoIP Gateways
Interactive Voice Response (IVR) consists of simple voice prompting and digit collection to gather caller information for authenticating the user and identifying the destination. IVR applications can be assigned to specific ports or invoked on the basis of dialed number identification service (DNIS). An IP Public Switched Telephone Network (PSTN) gateway can have several IVR applications to accommodate many different gateway services, and you can customize the IVR applications to present different interfaces to the various callers.
IVR systems provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words, or more commonly, dual tone multifrequency (DTMF) signaling. IVR uses Tool Command Language (TCL) scripts to gather information and to process accounting and billing.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ivr72.htm.
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Inter-Domain Gatekeeper Security Enhancement
The Inter-Domain Gatekeeper Security Enhancement feature provides a means of authenticating and authorizing H.323 calls between the administrative domains of Internet Telephone Service Providers (ITSPs).
An interzone ClearToken (IZCT) is generated in the originating gatekeeper (OGK) when a location request (LRQ) is initiated or an admission confirmation (ACF) is about to be sent for an intrazone call within an ITSP's administrative domain. As the IZCT traverses the routing path, each gatekeeper (GK) stamps the IZCT's destination GK ID with its own ID. This identifies when the IZCT is being passed over to another ITSP's domain. The IZCT is then sent back to the OGW in the location confirmation (LCF) message. The OGW passes the IZCT to the terminating gateway (TGW) in the SETUP message. The TGW forwards the IZCT in the admission request (ARQ) answerCall field to the terminating gatekeeper (TGK), which then validates it.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft_ctoke.htm.
Interface Alias Long Name Support
The Interface Alias (ifAlias) is a user-specified description of an interface used for SNMP network management. The ifAlias is an object in the Interfaces Group MIB (IF-MIB), which can be set by a network manager to "name" an interface. The ifAlias value for an interface or subinterface can be set using the description command in interface configuration mode, or by using a Set operation from a Network Management System.
Before Cisco IOS Release 12.2(2)T, ifAlias descriptions for subinterfaces were limited to 64 characters. A new Cisco IOS software command, snmp ifmib ifalias long, configures the system to handle ifAlias descriptions of up to 256 characters. IfAlias descriptions appear in the output of the show interfaces CLI command. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftshowif.htm.
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Interface Index Display
The Interface Index (IfIndex) is a user-specified identification number for an interface used in SNMP network management. The IfIndex is an object in the Interfaces Group MIB (IF-MIB), which can be set by a network manager to consistently identify an interface. A new Cisco IOS software command, show snmp mib ifmib ifindex, allows the user to display the IfIndex identification numbers assigned to interfaces and subinterfaces using the CLI. The IFIndex provides a way to display these values without the need for a Network Management Station. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftshowif.htm.
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Internal Cause Code Consistency Between SIP and H.323
The Internal Cause Code Consistency Between SIP and H.323 feature establishes a standard set of categories for internal causes of voice call failures. Before this feature, the cause code passed when an internal failure occurred was not standardized or based on any defined rules. The nonstandardization led to confusing or incorrect cause code information, and possibly contributed to billing errors.
The H.323 and SIP standard cause codes that are now generated accurately reflect the nature of each internal failure. This makes H.323 and SIP consistent with cause codes generated for common problems. Also, for each internal failure, an ITU-T Q.850 release cause code is also assigned.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftbibble.htm.
Interworking of Cisco MGCP Voice Gateways and Cisco CallManager Version 3.2
The Interworking of Cisco MGCP Voice Gateways and Cisco CallManager Version 3.2 feature allows Cisco voice gateways to act as redundant fail-over MGCP gateways. The new functionality includes the following configurable options:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_ccm1.htm.
Interworking Signaling Enhancements for H.323 and SIP VoIP
The Interworking Signaling Enhancements for H.323 and SIP VoIP feature enables VoIP networks to properly signal the setup and tear-down of calls when interworking with PSTN networks. These enhancements ensure that in-band tones and announcements are generated when needed so that the voice path is cut-through at the appropriate point of call setup and that early alerting (ringing) does not occur. In addition, support for network-side ISDN and the reducing of speech clipping is addressed.
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ip dhcp-client default-router distance value Command
Previous to Cisco IOS Release 12.2, Dynamic Host Configuration Protocol (DHCP) originated default routes that always had an admin. distance of 254. This distance allowed a metric of 255 as a backup route, but some routing protocols would interpret 255 as "route unavailable." You can now configure the default admin. distance with the new ip dhcp-client default-router distance value command.
Refer to the following document for additional information:
http://www.cisco.com/en/US/docs/ios/12_3/ipaddr/command/reference/ip1_i1g.html#wp1099050
IP Multicast MIB Enhancements
This feature enhances the IP multicast routing protocol in Cisco IOS software by adding MIB variables to query the number of (S, G) and (*, G) entries. It also adds support for high-speed interface counters.
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IPSec VPN High Availability Enhancements
The IPSec VPN High Availability feature consists of two new features—Reverse Route Injection and Hot Standby Router Protocol and IPSec—that work together to provide users with a simplified network design for VPNs and reduced configuration complexity on remote peers with respect to defining gateway lists.
Reverse Route Injection
Reverse Route Injection (RRI) is a feature designed to simplify network design for Virtual Private Network (VPNs) in which there is a requirement for redundancy or load balancing. RRI works with both dynamic and static crypto maps.
In the dynamic case, as remote peers establish IPSec security associations (SAs) with an RRI-enabled router, a static route is created for each subnet or host protected by that remote peer. For static crypto maps, a static route is created for each destination of an extended access-list rule.
When routes are created, they are injected into any dynamic routing protocol and distributed to surrounding devices. This traffic flows, requiring IPSec to be directed to the appropriate RRI router for transport across the correct SAs to avoid IPSec policy mismatches and possible packet loss.
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Hot Standby Router Protocol and IPSec
Hot Standby Router Protocol (HSRP) is designed to provide high network availability by routing IP traffic from hosts on Ethernet networks without relying on the availability of any single router. HSRP is particularly useful for hosts that do not support a router discovery protocol, such as ICMP Router Discovery Protocol (IRDP), and do not have the functionality to switch to a new router when their selected router reloads or loses power. Without this functionality, a router that loses its default gateway because of a router failure is unable to communicate with the network.
HSRP is configurable on LAN interfaces using standby command line interface (CLI) commands. It is now possible to use the standby IP address from an interface as the local IPSec identity, or local tunnel endpoint.
By using the standby IP address as the tunnel endpoint, failover can be applied to VPN routers by using HSRP. Remote VPN gateways connect to the local VPN router via the standby address that belongs to the active device in the HSRP group. In the event of failover, the standby device takes over ownership of the standby IP address and begins to service remote VPN gateways.
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Further Documentation
Refer to the following document for further information about the IPSec VPN High Availability Enhancements feature:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122y/122ye/1229ye/12yipsec.htm
IPv6 for Cisco IOS Software
IPv6, formerly called IPng (next generation), is a replacement for the current version of IP (version 4). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/index.htm.
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ISDN and V.120 Support for NextPort DSPs
The ISDN and V.120 Support For NextPort DSPs feature provides full coverage for digital calls and performance enhancement for V.120 calls. The feature permits terminating synchronous ISDN and V.120 sessions without customer intervention. This feature allows the Cisco AS5350 and Cisco AS5400 to terminate more than 256 ISDN sessions per channelized T3 (CT3) controller by adding ISDN capacity. This feature is mandatory for wholesale dial installations in which ISDN is being used. This feature permits V.120 calls to operate on the NextPort digital signal processor (DSP) instead of on the CT3 controller to reduce activity on the CPU and to increase the V.120 call capability. Support for these enhancements is automatic, and no configuration steps are required.
ISDN-NFAS with D Channel Backup
ISDN Non-Facility Associated Signaling (NFAS) allows a single D channel to control multiple PRI interfaces. A backup D channel can be configured for use when the primary NFAS D channel fails. Any hard failure causes a switchover to the backup D channel and currently connected calls remain connected. The ISDN-NFAS with D Channel Backup feature also supports the DMS100 and NI2 switch types.
Once the channelized T1controllers are configured for ISDN PRI, only the NFAS primary D channel must be configured; its configuration is distributed to all the members of the associated NFAS group.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5800/sw_conf/ios_122/ft_nfas.htm.
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IVR: Configuring Dynamic Prompts
The functionality of dynamic prompts, an existing Cisco IOS feature, has been expanded in Cisco IOS Release 12.2(11)T to play out International Organization for Standardization (ISO) formatted time and date, and visible noncontrol ASCII characters. Dynamic prompts allow a TCL application to play the date and time information on a Cisco voice gateway. The information is first retrieved by using the clock command in the Toolkit Command Language (TCL) library and then played through dynamic prompts using the multilanguage script.
The media play command in the TCL library plays the specified dynamic prompt on the specified call leg. The English version of the multilanguage TCL script must be enabled before you use the media play command; it allows a dynamic prompt to play string and visible noncontrol ASCII characters.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/actap/index.htm.
IVR: Customizing Accounting Templates
You can create an accounting template to customize your accounting records based on your billing needs. An accounting template is a text-based interface that allows you to customize and define the content of that template and helps reduce billing traffic from the gateway to the accounting servers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/actap/index.htm.
IVR: Directing AAA Requests
Cisco IOS Release 12.2(11)T introduces the capability of splitting authentication, authorization, and accounting (AAA) requests to RADIUS servers based on account number, called party number, and incoming trunk groups.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/actap/index.htm.
IVR: Enhanced Multilanguage Support
This feature releases the infrastructure to support Tool Command Language (TCL)-based script interpreters, which allow you to easily add new languages to your router or access server. You can add a new language by creating a TCL script that interprets prompts into a sequence of audio files or silences. The underlying Cisco IOS dynamic prompting code interfaces with the TCL script to translate the message into a sequence of URLs that point to audio files. Then, the Cisco IOS software plays the sequence of audio files as a dynamic prompt. New TCL-script language interpreters operate simultaneously with the current built-in languages: Spanish, Chinese/Mandarin, and English. Adversely, new TCL-script language interpreters can replace one or more of the built-in languages by overwriting the built-in language functionality.
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmultil.htm.
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L2TP Large-Scale Dial-Out
The L2TP Large-Scale Dial-Out feature enables the router to dial multiple Layer 2 Tunnel Protocol (L2TP) access concentrators (LACs) from a single L2TP network server (LNS). The LACs are signaled through the LNS and use L2TP to establish the dial sessions. User-defined profiles can be configured on an authentication, authorization, and accounting (AAA) server and retrieved by the LNS when dial-out occurs.The L2TP Large-Scale Dial-Out feature also supports multiple LACs bound into one stack group, call traffic load balancing, and outbound call congestion management.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftl2lsdo.htm.
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L2TP Security
The L2TP Security feature provides enhanced security for tunneled PPP frames between the Layer 2 Transport Protocol (L2TP) access concentrator (LAC) and the L2TP network server (LNS). Previous releases of the Cisco IOS software provided only a one-time, optional mutual authentication during tunnel setup with no authentication of subsequent data packets or control messages. In situations in which the L2TP is used to tunnel PPP sessions over an untrusted infrastructure such as the Internet, the security attributes of L2TP and PPP are inadequate. PPP provides no protection of the L2TP tunnel, and current PPP encryption protocols provide inadequate key management and no authentication or integrity mechanisms. The L2TP Security feature allows the robust security features of IP Security (IPSec) to protect the L2TP tunnel and the PPP sessions within the tunnel. In addition, the L2TP Security feature provides built-in keepalives and standardized interfaces for user authentication and accounting to authentication, authorization, and accounting (AAA) servers.
The deployment of Microsoft Windows 2000 demands the integration of IPSec with L2TP because this is the default virtual private dialup network (VPDN) networking scenario. This integration of protocols is also used for LAN-to-LAN VPDN connections in Microsoft Windows 2000. The L2TP Security feature provides integration of IPSec with L2TP in a solution that is scalable to large networks with minimal configuration.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftl2tsec.htm.
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Location Confirmation (LCF) Enhancements for Alternate Endpoints
This feature was originally introduced in Cisco IOS Release 12.2(4)T; see the "Location Confirmation Enhancements for Alternate Endpoints" section. Cisco IOS Release 12.2(11)T documents the new endpoint alt-ep collect command. In addition, effective with Cisco IOS Release 12.2(11)T, duplicate alternate endpoints that are received in a Location Confirmation (LCF) message are removed from the consolidated list of endpoints.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_lcfep.htm.
Low Latency Queueing with Priority Percentage Support
This feature allows you to configure bandwidth as a percentage within low latency queueing (LLQ). Specifically, you can designate a percentage of the bandwidth to be allocated to an entity (such as a physical interface, a shaped ATM permanent virtual circuit (PVC), or a shaped Frame Relay PVC) to which a policy map is attached. Traffic associated with the policy map will then be given priority treatment. This feature also allows you to specify the percentage of bandwidth to be allocated to nonpriority traffic classes.
This feature modifies two existing commands—bandwidth and priority. This feature adds a new keyword to the bandwidth command—remaining percent. The feature also changes the functionality of the existing percent keyword. These changes result in the following commands for bandwidth: bandwidth percent and bandwidth remaining percent. The bandwidth percent command configures bandwidth as an absolute percentage of the total bandwidth on the interface. The bandwidth remaining percent command allows you to allocate bandwidth as a relative percentage of the total bandwidth available on the interface. This command allows you to specify the relative percentage of the bandwidth to be allocated to the classes of traffic.
This feature also adds the percent keyword to the priority command. The priority percent command indicates that the bandwidth will be allocated as a percentage of the total bandwidth of the interface. You can then specify the percentage (that is, a number from 1 to 100) to be allocated by using the percentage argument with the priority percent command.
Unlike the bandwidth command, the priority command provides a strict priority to the traffic class, which ensures low latency to high priority traffic classes. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftllqpct.htm.
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MD5 File Validation
The MD5 File Validation feature allows you to check the integrity of a Cisco IOS software image by comparing its MD5 checksum value against a known MD5 checksum value for the image. MD5 values are now made available on Cisco.com for all Cisco IOS software images for comparison against local system image values.
To perform the MD5 integrity check, execute the verify command using the new "/md5" keyword. For example, executing the verify flash:c7200-is-mz.122-2.T.bin /md5 command will calculate and display the MD5 value for the software image. Compare this value with the value available on Cisco.com for this image.
Alternatively, you can get the MD5 value from Cisco.com first, then specify this value in the command syntax. For example, executing the verify flash:c7200-is-mz.122-2.T.bin /MD5 8b5f3062c4caeccae72571440e962233 command will display a message verifying that the MD5 values match or that there is a mismatch.
A mismatch in MD5 values means that either the image is corrupt or the wrong MD5 value was entered.
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Media Gateway Control Protocol-Based Fax (T.38) and Dual Tone Multifrequency (IETF RFC 2833) Relay
The MGCP-Based Fax (T.38) and DTMF (IETF RFC 2833) Relay feature adds support for fax relay and DTMF relay with MGCP. This feature provides two modes of implementation for each component: gateway (GW)-controlled mode and call agent (CA)-controlled mode. In GW-controlled mode, GWs negotiate DTMF and fax relay transmission by exchanging capability information in Session Description Protocol (SDP) messages. That transmission is transparent to the CA. GW-controlled mode allows use of the MGCP-Based Fax (T.38) and DTMF (IETF RFC 2833) Relay feature without upgrading the CA software to support the feature. In CA-controlled mode, CAs use MGCP messaging to instruct GWs to process fax and DTMF traffic. For MGCP T.38 Fax Relay, the CAs can also instruct GWs to revert to GW-controlled mode if the CA is unable to handle the fax control messaging traffic; for example, in overloaded or congested networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmgcpfx.htm.
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MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
The MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles feature implements the following Media Gateway Control Protocol (MGCP) protocols on the supported Cisco media gateways:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_24mg1.htm
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MGCP Basic CLASS and Operator Services
The MGCP BCOS are a set of calling features, sometimes called "custom calling" features, that use MGCP to transmit voice, video, and data over the IP network. These features are usually found in circuit-based networks. MGCP BCOS brings them to the Cisco IOS gateways on packet-based networks.
The MGCP BCOS software is built on the MGCP CAS PBX and AAL2 software package, and supports MGCP 0.1 and the earlier protocol versions Simple Gateway Control Protocol (SGCP) 1.1 and 1.5.
The following MGCP BCOS features are available on Residential Gateways (RGWs) and Business Gateways (BGWs):
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The following two features can be run as RGW or trunking gateway (TGW) features:
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This feature is supported in SGCP mode on Cisco 3660 and Cisco AS5300 platforms and in MGCP mode on all five supported platforms.
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This feature is supported on the Cisco 3660 and Cisco AS5300 TGW platforms and on the Cisco MC3810 series, and Cisco 2600 RGW platforms. This feature cannot be supported on the G.728 and G.723 codecs.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmgcpgr.htm.
Note
MGCP CAS PBX and AAL2 PVC
The MGCP CAS PBX and AAL2 PVC software package is a solutions-oriented program that focuses on several customer gateway scenarios. These scenarios require features that address residential, business, and trunking gateway needs on a variety of hardware platforms:
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To answer these needs, the MGCP CAS PBX and AAL2 PVC feature combines and expands existing feature sets on the merged Simple Gateway Control Protocol (SGCP)/MGCP software platform as follows:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmgcptk.htm.
Note
MGCP Generic Configuration Support for Call Manager (IP-PBX)
The MGCP Generic Configuration Support for Call Manager (IP-PBX) feature provides generic configuration support for Cisco IOS Media Gateway Control Protocol (MGCP) gateways with Call Manager. The gateways receive voice configuration from Call Manager by way of an eXtensible Markup Language (XML) file that is downloaded from a TFTP server.
MGCP Line Package Enhancements for Loop Current Feed Open (LCFO)
The MGCP Line Package Enhancements for Loop Current Feed Open (LFCO) feature enhances Media Gateway Control Protocol (MGCP) residential gateway capabilities to support the generation of the LFCO signal at the request of the call agent. LFCO is a new signal in the line package. This enhancement supports call flows that involve answering machines or other automated devices that act as the terminating party and will facilitate the notification of the originating party's on-hook to such devices. There is no explicit configuration required to enable this feature.
MGCP PRI Backhaul and T1-CAS Support for Call Manager (IP-PBX)
ISDN PRI backhaul provides a method for transporting complete IP telephony signaling information from an ISDN PRI interface of an MGCP voice gateway to Cisco CallManager through a highly reliable TCP connection.
This feature works by terminating all the ISDN PRI Layer 2 (Q.921) signaling functions in the Cisco IOS software on the MGCP voice gateway while, at the same time, packaging all the ISDN PRI Layer 3 (Q.931) signaling information into packets for transmission to the Cisco CallManager through an IP tunnel over a highly reliable TCP connection. This methodology ensures the integrity of the Q.931 signaling information being passed through the network for managing IP telephony devices.
A rich set of user-side and network-side ISDN PRI calling functions is supported by the ISDN PRI backhaul feature. A single TCP connection is used by the gateway to backhaul all the ISDN D channels to Cisco CallManager. The "SAP/Channel ID" parameter in the header of each message identifies individual D channels. In addition to carrying the backhaul traffic, the inherent TCP keepalive mechanism is also used to determine MGCP voice gateway connectivity to an available call agent.
The MGCP voice gateway also establishes a TCP link to the backup (secondary) Cisco CallManager server. In the event of Cisco CallManager switchover, the ISDN PRI backhaul functions are assumed by the secondary Cisco CallManager server. During this switchover, all active ISDN PRI calls are preserved, and the affected MGCP gateway is registered with the new Cisco CallManager server through a Restart-in-Progress (RSIP) message to ensure continued gateway operation.
T1 CAS is supported in non-backhaul fashion and supported CAS signaling types on the Cisco CallManager are E&M, wink-start, and E&M delay-dial. E1 CAS is not supported.
MGCP Voice on Cisco AS5850 Universal Gateway
Although the documents listed below were not written specifically for the Cisco AS5850, they still apply to the Cisco AS5850. MGCP Voice on Cisco AS5850 Universal Gateway include the following features:
FGD-OS 911 Calls
The 911 feature can be run as residential gateway (RGW) or trunking gateway (TGW) feature
Interactive Voice Response Version 2.0 on Cisco VoIP Gateway
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dt_skyn.htm.
Note
Media Gateway Control Protocol Residential Gateway Support
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/mgcp1213.htm
MGCP based Fax (T.38) and DTMF (IETF Ver.) Relay
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmgcpfx.htm
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MGCP VoIP Call Admission Control
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_04mac.htm
Network Access Server Package for Media Gateway Control Protocol
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ft_mgnas.htm
PRI/Q.931 Signaling Backhaul for Call Agent Applications
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/5300/sw_conf/ios_121/0144cors.htm
Route-Switch-Controller Handover Redundancy on the Cisco AS5850
See the "Route Switch Controller (RSC) Handover Redundancy" section or refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/handred.htm
Note
MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_24mg1.htm
MGCP CAS PBX and AAL2 PVC
Refer to the following document for information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xm/121xm_5/ftmgcpba.htm.
Further Documentation
Refer to the following document for further information about the MGCP Voice on Cisco AS5850 Universal Gateway feature:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/pull_daz.htm
MGCP VoIP Call Admission Control
MGCP VoIP Call Admission Control (CAC) determines if calls can be accepted on the IP network on the basis of available network resources. Before this release, Media Gateway Control Protocol (MGCP) Voice over IP (VoIP) calls were established regardless of the available resources on the gateway or network. The gateway had no mechanism for gracefully refusing calls if resources were not available to process the call. New calls would fail with unexpected behavior and in-progress calls would experience quality-related problems.
The MGCP VoIP Call Admission Control feature provides three CAC mechanisms to address the need for improved quality and predictable gateway behavior. The first mechanism is local/system CAC, which provides the ability to gracefully refuse calls on the basis of the availability of local gateway call processing resources such as CPU utilization and memory. The second CAC mechanism provides synchronization with Resource Reservation Protocol (RSVP) and reports the reservation request to the call agent. The third mechanism provides network congestion detection to gracefully refuse calls on the basis of a measured level of congestion.
Modem Relay Support on VoIP Platforms
When service providers and aggregators are implementing Voice over IP (VoIP), they sometimes cannot separate fax or data traffic from voice traffic. These carriers that aggregate voice traffic over VoIP infrastructures require service offerings to carry fax and data as easily as voice.
Modem relay demodulates a modem signal at one voice gateway by decomposing the modem signal to digital form and then passing this signal as packet data to another voice gateway, where the signal is remodulated and sent to a receiving modem. The relay process distinguishes that the call is in fact a modem call. On detection of the modem answer tone, the gateways switch into modem pass-through mode. If the CM (call menu) signal is detected, the two gateways switch into modem relay mode.
There are two ways to transport modem traffic over VoIP networks:
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In this implementation, the call starts out as a voice call and then switches into modem pass-through mode and then into modem relay mode.
This feature significantly reduces the effects that dropped packets, latency, and jitter have on the modem session. Compared to modem pass-through, it also reduces the amount of bandwidth used. Primary applications for this feature are transport of modem dial-up traffic over IP networks.
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftmodrly.htm
Modem Script and System Script Support in Large-Scale Dial-Out
Modem connection and system login chat scripts are often used when asynchronous dial-on-demand routing (DDR) is configured. Currently, however, the large-scale dial-out network architecture does not allow chat scripts for a particular session to be passed through the network. Cisco IOS Release 12.2(2)T allows modem and system chat scripts to pass through large-scale dial-out networks by allocating two new authentication, authorization, and accounting (AAA) attributes for outbound service.
The AAA attributes define specific AAA elements in a user profile. Large-scale dial-out supports Cisco attribute-value (AV) pairs and TACACS+ attributes. The Modem Script and System Script Support in Large-Scale Dial-Out feature provides two new outbound service attributes for passing chat scripts: modem-script and system-script. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftlschat.htm.
Note
Monitoring Voice and Fax Services on the Cisco AS5350 and Cisco AS5400 Universal Gateways
The Universal Port Dial Feature Card (DFC) is a hardware card that processes voice and data services port technology for the Cisco AS5350 and Cisco AS5400.
The ports on the Universal Port DFC support multiple types of service including modem, digital, voice, and fax. Ports can be aggregated at the slot level of the Universal Port module, the Service Processing Element (SPE) level within the Universal Port module, and the individual port level.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xm/121xm_5/ftupspe.htm.
MPLS Label Distribution Protocol (LDP)
Cisco MPLS label distribution protocol (LDP) allows the construction of highly scalable and flexible IP Virtual Private Networks (VPNs) that support multiple levels of services.
LDP provides a standard methodology for hop-by-hop distribution of labels in an Multiprotocol Label Switching (MPLS) network by assigning labels to routes that have been chosen by the underlying Interior Gateway Protocol (IGP) routing protocols. The resulting label switch paths (LSPs) forward label traffic across an MPLS backbone to particular destinations. These capabilities enable service providers to implement Cisco MPLS-based IP VPNs and IP+ATM services across multivendor MPLS networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ldp7t.htm.
Note
MPLS VPN ID
Using Multiprotocol Label Switching (MPLS) VPN ID you can identify virtual private networks (VPNs) by a VPN identification number, as described in RFC 2685. This implementation of the MPLS VPN ID feature is used for identifying a VPN. The MPLS VPN ID feature is not used to control the distribution of routing information or to associate IP addresses with MPLS VPN ID numbers in routing updates.
Multiple VPNs can be configured in a router. You can use a VPN name (a unique ASCII string) to reference a specific VPN configured in the router. Alternately, you can use a VPN ID to identify a particular VPN in the router. The VPN ID follows a standard specification (RFC 2685). To ensure that the VPN has a consistent VPN ID, assign the same VPN ID to all the routers in the service provider network that services that VPN.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftvpnid.htm.
Note
Multicast Music on Hold Support for Call Manager (IP-PBX)
The Multicast Music on Hold Support for Call Manager (IP-PBX) feature provides the functionality to stream music from a Multicast Music on Hold (MOH) server to the voice interfaces of on-net and off-net callers that have been placed on hold.
This integrated multicast capability of Cisco CallManager 3.1 is implemented through the H.323 signaling plane in Cisco CallManager.
In an MOH environment, whenever caller A places caller B on hold, Cisco CallManager requests the MOH server to stream RTP packets to the "on-hold" interface through the preconfigured multicast address. In this way, RTP packets can be relayed to appropriately configured voice interfaces in a VoIP network that have been placed on hold.
Multiple MOH servers can be present in the same network, but each server must have a different Class D IP address, and the address must be preconfigured in Cisco CallManager and the Cisco IOS MGCP voice gateways.
The MOH feature enables you to subscribe to a music streaming service when using a Cisco IOS MGCP voice gateway. By means of a preconfigured multicast address on a gateway, the gateway can "listen for" Real-Time Transport Protocol (RTP) packets that are broadcast from a default router in the network and can relay the packets to designated voice interfaces in the network.
RTP is the Internet-standard protocol for transporting real-time data across a network, including audio and video information. Thus, RTP is well suited for media on demand and interactive services, such as IP telephony.
The default router in the network for handling multicast traffic must have the following enabled:
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When you configure a multicast address on a gateway, the gateway sends an Internet Gateway Management Protocol (IGMP) "join" message to the default router, indicating to the default router that the gateway is to receive RTP multicast packets.
Multiple RSA Keypair Support
The Multiple RSA Keypair Support feature allows the Cisco IOS software to maintain a distinct key pair for each certification authority (CA) with which it is dealing. Thus, the Cisco IOS software can match policy requirements for each CA without compromising the requirements specified by the other CAs, such as key length, key lifetime, and general-purpose versus special-usage keys.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftmltkey.htm.
Note
NAT Support for SIP
The Session Initiation Protocol (SIP) is an application layer signaling protocol used for creating and controlling multimedia sessions with two or more participants. SIP is transported over TCP or UDP. The messages used in the protocol may have IP addresses embedded in the packet payload. If a message passes through a router configured with Network Address Translation (NAT), the embedded information must be translated and encoded back to the packet. An Application Layer Gateway (ALG) is used with NAT to enable SIP.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftnatsip.htm.
Note
NetFlow Multiple Export Destinations
The NetFlow Multiple Export Destinations feature enables configuration of multiple destinations of the NetFlow data. With this feature enabled, two identical streams of NetFlow data are sent to the destination host. Currently, the maximum number of export destinations allowed is two.
The NetFlow Multiple Export Destinations feature improves the chances of receiving complete NetFlow data by providing redundant streams of data. Because the same export data is sent to more than one NetFlow collector, fewer packets will be lost. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dtnfdest.htm.
Note
NetFlow ToS-Based Router Aggregation
The NetFlow ToS-Based Router Aggregation feature provides the ability to enable limited router-based type of service (ToS) aggregation of NetFlow Export data, which results in summarized NetFlow Export data to be exported to a collection device. The result is lower bandwidth requirements for NetFlow Export data and reduced platform requirements for NetFlow data collection devices. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s15/dtnfltos.htm.
Note
Network Access Server (NAS) Package for MGCP
The Network Access Server (NAS) Package for MGCP feature adds support for the Media Gateway Control Protocol (MGCP) NAS package on the Cisco AS5350, Cisco AS5400, and Cisco AS5850. With this implementation, data calls can be terminated on a trunking media gateway that is serving as a NAS. Trunks on the NAS are controlled and managed by a call agent that supports MGCP for both voice and data calls. The call agent must support the MGCP NAS package.
These capabilities are enabled by the universal port functionality of the Cisco AS5350, Cisco AS5400, and Cisco AS5850, which allows these platforms to operate simultaneously as network access servers and voice gateways to deliver universal services on any port at any time. These universal services include dial access, real-time voice and fax, wireless data access, and unified communications.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ft_mgnas.htm.
Network Side ISDN PRI Signaling, Trunking, and Switching
The Network Side ISDN PRI Signaling, Trunking, and Switching feature enables Cisco IOS software to replicate the public switched network interface to a PBX that is compatible with the National ISDN (NI) switch types and European Telecommunications Standards Institute (ETSI) Net5 switch types.
Routers and PBXs are both traditionally CPE devices with respect to the public switched network interfaces. However, for Voice over IP (VoIP) applications, it is desirable to interface access servers to PBXs with the access server representing the public switched network.
Enterprise organizations use the current VoIP features with Cisco products as a method to reduce costs for long distance phone calls within and outside their organizations. However, there are times that a call cannot go over VoIP and the call needs to be placed using the Public Switched Telephone Network (PSTN). The customer then must have two devices connected to a PBX to allow some calls to be placed using VoIP and some calls to be placed over the PSTN. In contrast, this feature allows Cisco access servers to connect directly to user-side CPE devices such as PBXs and allows voice calls and data calls to be placed without requiring two different devices to be connected to the PBXs.
The ISDN Network Side ISDN PRI Signaling, Trunking, and Switching feature allows Cisco ISDN-enabled access servers to switch calls across interfaces as legacy phone switches do today and to mimic the behavior of the legacy phone switches.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtpri_ni.htm.
Nonblocking Gatekeeper AAA Interface
The Nonblocking Gatekeeper AAA Interface feature enables Cisco gatekeepers to perform authentication, authorization, and accounting (AAA) through the gatekeeper interface at much higher call rates.
There are no new or modified commands.
Optimized PPP Negotiation
The Optimized PPP Negotiation feature optimizes the time needed for PPP negotiation when a connection is made. PPP negotiation can include several cycles before the negotiation options are acknowledged. These negotiation cycles can cause a significant user-perceived delay, especially in networks with slow links such as a wireless data connection. Additionally, the PPP negotiation time can add significantly to the total time the user stays connected in these types of connections. Changes to the PPP link control protocol (LCP) and PPP Internet Protocol Control Protocol (IPCP) negotiation strategies as part of Cisco IOS Release 12.2(4)T and later releases make a reduction in the negotiation time possible.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftcphneg.htm
Note
OSP Debug Enhancement
The OSP Debug Enhancement feature documents the new debug voip settlement ssl command. Use this command if you find a connection or I/O error with the Secure Socket Layer (SSL) connection after using the debug voip settlement error command. Turning on the debug voip settlement ssl command allows the Open Settlement Protocol (OSP) to display detailed information for the SSL connection.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftdbgosp.htm
OSPF ABR Type 3 LSA Filtering
The OSPF ABR Type 3 link-state advertisement (LSA) Filtering feature extends the ability of an ABR that is running the OSPF protocol to filter type 3 LSAs between different OSPF areas. This feature allows only specified prefixes to be sent from one area to another area and restricts all other prefixes. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11at3f.htm.
Note
OSPF Sham-Link Support for MPLS VPN
A sham link is a logical path within an Open Shortest Path First (OSPF) area; it represents an unnumbered point-to-point connection between two provider edge (PE) devices. All routers within the area see the link and use it during the shortest path first (SPF) computation.
On PE routers the VPN Route Forwarding (VRF) routing table is populated by OSPF routes over the sham link. The sham link gives users the capability of specifying which path will be used for traffic.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ospfshmk.htm
Note
OSPF Stub Router Advertisement
The OSPF Stub Router Advertisement feature allows you to bring a new router into a network without immediately routing traffic through the new router and allows you to gracefully shut down or reload a router without dropping packets that are destined for other networks. This feature introduces three configuration options that allow you to configure a router that is running the Open Shortest Path First (OSPF) protocol to advertise a maximum or infinite metric to all neighbors. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11osra.htm.
Note
OSPF Update Packet-Pacing Configurable Timers
The OSPF Update Packet-Pacing Configurable Timers feature allows you to configure the rate at which Open Shortest Path First (OSPF) link-state advertisement (LSA) flood pacing, group pacing, and retransmission pacing updates occur. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11opct.htm.
Note
Particle Drivers
The Particle Drivers feature is a collection of performance and reliability improvements for the Cisco AS5350, Cisco AS5400, and Cisco AS5400HPX universal gateways. It includes particles-based packet drivers for improved performance. These particle drivers optimize Cisco IOS fast switching code and significantly improve the way Cisco IOS uses processor cache memory. Data packets for some protocols, such as MLPPP, IP Multicast, and cRTP, are fast switched with particle drivers. Cisco IOS CEF switching paths are highly optimized with particle drivers.
PIAFS Wireless Data Protocol Version 2.1 for Cisco MICA Modems
The PIAFS Wireless Data Protocol Version 2.1 for Cisco MICA Modems feature adds support for the Personal Handyphone Internet Access Forum Standard (PIAFS) using Cisco MICA technologies modems for the Cisco AS5300 and Cisco AS5800. PIAFS provides data connectivity between a client computer and a remote access server (RAS) using the Personal-Handyphone-System (PHS) digital cellular telephone system. PIAFS 2.1 allows the modem to shift speed during a connection between 32,000 and 64,000 bps when initiated by a remote terminal adapter (TA).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftblknt.htm.
PIM Multicast Scalability
The PIM Multicast Scalability feature enhances the Protocol Independent Multicast (PIM) protocol in Cisco IOS software by adding a new level of scalability. With this feature, edge devices can have a large number of multicast groups and users without increasing the CPU utilization of the router.
Note
PIM MIB Extension for IP Multicast
Protocol Independent Multicast (PIM) is an IP Multicast routing protocol used for routing multicast data packets to multicast groups. RFC 2934 defines the Protocol Independent Multicast for IPv4 MIB, which describes managed objects that enable users to remotely monitor and configure PIM using Simple Network Management Protocol (SNMP).
The PIM MIB Extension for IP Multicast feature introduces support in Cisco IOS software for the CISCO-PIM-MIB, which is an extension of RFC 2934 and an enhancement to the existing Cisco implementation of the PIM MIB.
This feature introduces the following new classes of PIM notifications:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftpimmib.htm.
Note
Preauthentication with ISDN PRI and Channel-Associated Signalling Enhancements
Preauthentication allows a Cisco network access server (NAS) to decide—on the basis of the Dialed Number Identification Service (DNIS) number—whether to answer an incoming call. When an incoming call arrives from the public network switch but before it is answered, the NAS sends the DNIS number to a RADIUS server for authorization.
The Preauthentication with ISDN PRI and Channel-Associated Signaling Enhancements feature provides additional support for preauthentication, which was introduced in a previous Cisco IOS release. For more information about preauthentication, refer to the Cisco IOS Release 12.1(3)T feature module titled Preauthentication with ISDN PRI and Channel-Associated Signaling.
This feature supports the use of attribute 44 by the RADIUS server application, which allows user authentication on the basis of the Calling Line Identification (CLID) number in the same transaction. For more information about attribute 44 and how it works with preauthentication, refer to the Cisco IOS Release 12.0(7)T feature module titled RADIUS Attribute 44 (Accounting Session ID) in Access Requests.
This feature also supports the use of new RADIUS attributes. These RADIUS attributes are configured in the RADIUS preauthentication profiles to specify preauthentication behavior. They may also be used, for instance, to specify whether subsequent authentication should occur and, if so, what authentication method should be used. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtdt1.htm.
Note
PRI Backhaul Using the Stream Control Transmission Protocol and the ISDN Q.921 User Adaptation Layer
The PRI Backhaul Using the Stream Control Transmission Protocol and the ISDN Q.921 User Adaptation Layer feature fulfills the need for a standards-based PRI signaling backhaul that works with third-party call agents to enable solutions like Integrated Access, IP PBX, and Telecommuter.
This feature provides the following:
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This feature provides a configuration interface for Cisco IOS software implementation and implements the protocol message flows for SCTP and IUA.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_0546.htm.
Note
PRI/Q.931 Signaling Backhaul for Call Agent Applications
The PRI/Q.931 Signaling Backhaul for Call Agent Applications feature implements PRI/Q.931 signaling backhaul support for call agent applications on the Cisco 2600 and Cisco 3600 series routers and Cisco MC3810 series access concentrators. PRI/Q.931 signaling backhaul is the transport of PRI signaling (Q.931 and above layers) between a media gateway (such as a Cisco access server, router, or concentrator) and a media gateway controller (Cisco VSC3000).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5400/sw_conf/ios_122/122_2x/122xb/pul0144.htm.
Note
PSTN Fallback
The goal of PSTN fallback is to monitor congestion in the IP network and either redirect calls to the PSTN or reject calls based on the network congestion. Calls can be rerouted to an alternate IP destination or to the PSTN if the IP network is found unsuitable for voice traffic at that time. The user defines the congestion thresholds based on the configured network. This functionality enables the service provider to give a reasonable guarantee about the quality of the conversation to their VoIP users at the time of call admission.
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PSTN fallback includes the following features:
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For more information, including configuration tasks and examples, and command references for PSTN fallback, please refer to PSTN Fallback. Refer to the following document for additional information about the Call Admission Control for H.323 VoIP Gateways feature:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122
xa_2/ft_pfavb.htm.
Note
QSIG for TCL IVR 2.0
Q.SIG support is required for European countries to interconnect enterprise customers to a wholesale voice solution. The Q.SIG for TCL IVR 2.0 feature provides transparent Q.SIG interworking when using a TCL IVR version 2.0 voice application on a Cisco IOS voice gateway. This functionality can be enabled using a new CLI on the POTS or VoIP dial peer. Before this feature, Q.SIG messages were interpreted by the TCL IVR 2.0 application, rather than passed transparently to the remote endpoint.
R2 and ISUP Transparency and R2-to-ISUP Interworking Enhancements
The R2 and ISUP Transparency and R2-to-ISUP Interworking Enhancements feature provides enhancements to ISDN User Part (ISUP) transparency, R2-to-ISUP interworking, and R2 transparency using Generic Transparency Descriptor (GTD) objects in Cisco IOS Release 12.2(11)T. This release also provides support for Calling Line ID Presentation (CLIP) and Calling Line ID Restriction (CLIR) and is part of the Cisco SS7 Interconnect for Voice Gateways Solution.
This feature adds the following functionality:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_isup1.htm.
RADIUS Attribute 66 (Tunnel-Client-Endpoint) Enhancements
Virtual private networks (VPNs) use Layer 2 Forwarding (L2F) or Layer 2 Tunnel Protocol (L2TP) tunnels to tunnel the link layer of high-level protocols (for example, PPP) or asynchronous High-Level Data Link Control (HDLC)). Internet service providers (ISPs) configure their network access servers (NASs) to receive calls from users and forward the calls to the customer tunnel server. Usually, the ISP maintains only information about the tunnel server—the tunnel endpoint. The customer maintains the IP addresses, routing, and other user database functions of the tunnel server users.
The RADIUS Attribute 66 (Tunnel-Client-Endpoint) Enhancements feature adds the ability to specify the host name of the NAS—rather than the IP address of the NAS—in RADIUS attribute 66 (Tunnel-Client-Endpoint). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtdt4.htm.
Note
RADIUS Attribute 82: Tunnel Assignment ID
The RADIUS Attribute 82: Tunnel Assignment ID feature allows the Layer 2 Transport Protocol access concentrator (LAC) to group users from different per-user or domain RADIUS profiles into the same active tunnel. Previously, Cisco IOS software assigned a separate virtual private dialup network (VPDN) tunnel for each per-user or domain RADIUS profile, even if tunnels with identical endpoints already existed. The RADIUS Attribute 82: Tunnel Assignment ID feature defines a new AV pair, Tunnel-Assignment-ID, which allows the LAC to group users from different RADIUS profiles into the same tunnel if the chosen endpoint, tunnel type, and Tunnel-Assignment-ID are identical. This feature introduces new software functionality. No new commands are introduced with this feature.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftrada82.htm.
Note
RADIUS Attribute Value Screening
The RADIUS Attribute Value Screening feature allows users to configure a list of "accept" or "reject" RADIUS attributes on the network access server (NAS) for purposes such as authorization or accounting.
If a NAS accepts and processes all RADIUS attributes received in an Access-Accept packet, unwanted attributes may be processed, creating a problem for wholesale providers who do not control their customers' authentication, authorization, and accounting (AAA) servers. For example, there may be attributes that specify services to which the customer has not subscribed, or there may be attributes that may degrade service for other wholesale dial users. The ability to configure the NAS to restrict the use of specific attributes has therefore become a requirement for many users.
The RADIUS Attribute Value Screening feature should be implemented in one of the following ways:
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RADIUS Number Translation VSAs for VoIP
The RADIUS Number Translation VSAs for VoIP feature enables a Cisco AS5x00 voice gateway to export pre- and post-translated called and calling numbers to a RADIUS server in the form of generic vendor-specific attributes (VSA). Cisco gateways can be configured to present gateway received, gatekeeper translated, and final translated numbers to the RADIUS server.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/vsaig3.htm.
RADIUS Packet of Disconnect
The RADIUS Packet of Disconnect feature consists of a method for terminating a call that has already been connected. This "Packet of Disconnect" (POD) is a RADIUS access_request packet and is intended to be used when the authenticating agent server wants to disconnect the user after the session has been accepted by the RADIUS access_accept packet. This may be needed in at least two situations:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_pod1.htm.
RADIUS Packet Suppression for VoIP GW Rotary Dial-Peer Attempts
The RADIUS Packet Suppression for VoIP GW Rotary Dial-Peer Attempts feature enables the suppression of excess RADIUS start and stop requests that are sent when the originating or terminating gateway does rotary dial-peer retries for outbound call legs. When the rotary retry suppression feature is enabled, only one set of start and stop accounting packets is generated once a connection is successful or once the connection fails in the last rotary dial-peer attempt.
The rotary retry suppression feature gives you more control over authentication, authorization, and accounting (AAA) functions by enabling or disabling accounting on outgoing call legs. Standard RADIUS accounting enabled on the voice gateway sends a start and stop accounting request to RADIUS on every attempt using a rotary dial peer for making a connection. Every attempt can generate a pair of accounting requests even when the connection is not successful. The rotary retry suppression feature eliminates unnecessary traffic flow to the RADIUS server or other Voice over IP (VoIP) billing servers. When the rotary retry feature is activated, no matter how many dial peers are used for the outgoing call leg, only one pair of accounting start and stop records is sent to the billing server.
There is one modified command: suppress rotary—the keyword, rotary, was added to enable rotary retry suppression.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftsuppre.htm.
RADIUS Preauthentication for H.323 and SIP Voice Calls
The RADIUS Preauthentication for H.323 and SIP Voice Calls feature provides the means for service providers to accept or reject H.323 or SIP voice calls that come in to their networks before the calls are answered. This feature allows a wholesale service provider to screen an originating (PSTN-to-IP network) or terminating (IP-to-PSTN) voice call by using information about the call to determine which customer the call belongs to and whether the call should be admitted to the network. The type of information that can be used for screening includes the called number, the called number prefix, the originating H.323 zone and the originating voice gateway address. The service provider can use this feature in conjunction with a RADIUS-based port-policy management (PPM) server such as Cisco Resource Policy Management Server (RPMS) to make admission control decisions on the basis of information such as the total number of calls in the network, the total number of calls allowed for this customer and the current number of calls from this customer.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_trg.htm.
RADIUS Progress Codes
The RADIUS Progress Codes feature adds additional progress codes—10, 31, 32, 60, 65, 67—to RADIUS attribute 196 (Ascend-Connect-Progress), which indicates the connection state before the call is disconnected via progress codes.
Attribute 196 is sent in network, exec, and resource accounting start and stop records. This attribute can facilitate call failure debugging because each progress code identifies accounting information relevant start or stop record is requested, authentication, authorization, and accounting (AAA) will add attribute 196 into the record as part of the standard attribute list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftatr196.htm.
RADIUS Route Download
The RADIUS Route Download feature allows users to configure their network access server (NAS) to send static route download requests to authentication, authorization, and accounting (AAA) servers specified by a named method list. Before this feature, all RADIUS authorization requests for static route download could be sent only to AAA servers specified by the default method list.
This feature extends the functionality of the aaa route download command to allow users to specify the name of the method list that will be used to direct static route download requests to the AAA servers. The aaa route download command must be used to add separate method lists; however, users will continue to enable the aaa authorization configuration default command to download static route configuration information from the AAA server specified by the default method list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftradrou.htm.
Note
RADIUS Tunnel Preference for Load Balancing and Fail-over
Tunnel servers may be load balanced or failed-over from a single tunnel initiator, as selected by the RADIUS Tunnel Preference for Load Balancing and Fail-Over attribute. There is no configuration associated with this feature. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftradtun.htm.
Note
Reverse Path Forwarding - Source Exists Only
The Reverse Path Forwarding - Source Exists Only feature allows you to verify if the source IP address is valid in the Forwarding Information Base (FIB) for unicast Reverse Path Forwarding (uRPF) traffic. Packets that have not be allocated on the Internet, being used for spoofed source addresses, will be dropped. Packets with an entry in the FIB will be passed. This uRPF option can be used on internet service provider (ISP) peering routing devices with other ISPs.
Rotating Through Dial Strings
The Rotating Through Dial Strings feature allows you to specify the dialing order when multiple dial strings are configured. Options for dialing order include:
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This feature takes advantage of information available from a previous call attempt, such as whether the call was unsuccessful or the line was busy, and thereby increases the rate of successful calls.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftrotdls.htm.
Note
Route Switch Controller (RSC) Handover Redundancy
Route-Switch-Controller Handover Redundancy on the Cisco AS5850, with its provision of handover-split mode, provides the first phase of high availability to the Cisco AS5850 platform.
If your gateway contains two route-switch-controller (RSC) cards, you can configure your Cisco AS5850 into either of two split modes: classic split or handover split.
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/handred.htm.
Router-Shelf Redundancy for the AS5800 Series
This feature provides AS5800 router-shelf redundancy by using a second router shelf that automatically takes over the other shelf's resources (dial-shelf cards) if it appears that the other router has died. The failover is disruptive in that there is no attempt to maintain calls that were established on the failing router; the dial-shelf cards controlled by the failing router are restarted under the control of the backup router and hence become available again.
Two router shelves are connected to the same dial shelf (as in split mode) but with only one router active at a time. Both router shelves are configured for normal mode as opposed to split mode. Each router shelf contains the same configuration, being whatever configuration is appropriate for the full set of cards in the dial shelf. The active router controls all the cards in the dial shelf, while the other router functions purely as a backup. If the active router fails, all dial-shelf cards restart under the control of the backup router, which then functions as the active router.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xv/121xv_5/ftred3.htm.
SGCP RSIP and AUEP Enhancement
The SGCP RSIP and AUEP Enhancement feature provides additional messaging capabilities that allow an endpoint on an SGCP 1.5 gateway to synchronize with a call agent after the endpoint returns to service from the disconnected procedure. The additional messaging capabilities provide the following:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_rsip.htm.
Shell-Based Authentication of VPDN Users
The Shell-Based Authentication of VPDN Users feature provides terminal services for VPDN users to support rollout of wholesale dial networks. Terminal services (shell login or exec login) on the network access server (NAS) provide the following capabilities:
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With the terminal services, user authentication methods other than PAP and CHAP can be applied to PPP users. With the Shell-Based Authentication of VPDN Users feature, PPP authentication data is preconfigured or entered before PPP starts. Authentication is completed without any further input from the user. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122defer/ftexvpnt.htm.
Note
SIP—Call Transfer Using Refer Method
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The Refer method provides call transfer capabilities to supplement the Bye and Also methods already implemented on Cisco IOS Session Initiation Protocol (SIP) gateways.
Call transfer allows a wide variety of decentralized multiparty call operations. These decentralized call operations form the basis for third-party call control and thus are important features for Voice over IP (VoIP) and SIP. Call transfer is also critical for conference calling, where calls can transition smoothly between multiple point-to-point links and IP level multicasting.
The following are components of call transfer:
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Refer to the following document for additional information:
Note
SIP Carrier Identification Code
The SIP Carrier Identification Code feature enables the transmission of the Carrier Identification Code (CIC) parameter from the Session Initiation Protocol (SIP) network to the ISDN. The CIC parameter is a three- or four- digit code that is used in routing tables to identify the network serving the remote user when a call is routed over many different networks. The CIC parameter is carried in SIP INVITE requests and 302 REDIRECTs and maps to the ISDN Transit Network Selection Information Element (TNS IE) in the outgoing ISDN SETUP message. The TNS IE identifies the requested transportation networks and allows different providers equal access support based on customer choice.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftsipcic.htm.
SIP—Configurable PSTN Cause Code Mapping
For calls to be established between a session initiation protocol (SIP) network and a PSTN network, the two networks must be able to interoperate. One aspect of their interoperation is the mapping of PSTN cause codes, which indicate reasons for Public Switched Telephone Network (PSTN) call failure or completion, for SIP status codes or events. The opposite is also true: SIP status codes or events are mapped to PSTN cause codes. Event mapping tables found in this document show the standard or default mappings between SIP and PSTN.
However, you may want to customize the SIP user agent software to override the default mappings between the SIP and PSTN networks. The Configurable PSTN Cause Code to SIP Response Mapping feature allows you to configure specific map settings between the PSTN and SIP networks. Thus, any SIP status code can be mapped to any PSTN cause code, or vice versa. When set, these settings can be stored in the NVRAM and are restored automatically on bootup.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmap.htm.
Note
SIP—DNS SRV RFC2782 Compliance
Session Initiation Protocol (SIP) on Cisco Voice over IP (VoIP) gateways uses Domain Name System Server (DNS SRV) query to determine the IP address of the user endpoint. The query string has a prefix in the form of "protocol.transport." and is attached to the fully qualified domain name (FQDN) of the next hop SIP server. This prefix style, from RFC 2052, has always been available; however, with this release, a second style is also available. The second style complies with RFC 2782 and prepends the protocol label with an underscore "_"; as in "_protocol._transport." The addition of the underscore reduces the risk of the same name being used for unrelated purposes. The form compliant with RFC 2782 is the default style. Use the srv version command to configure the DNS SRV feature.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/vvfresrv.htm.
Note
SIP Diversion Header Implementation for Redirecting Number
SIP is a new protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to the ITU-T H.323 specification. SIP is defined by RFC 2543 and is used for multimedia call session setup and control over IP networks. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/sipcf2.htm.
Note
SIP—Enhanced Billing Support for Gateways
The Enhanced Billing Support for SIP Gateways feature provides changes to authentication, authorization, and accounting (AAA) records and the RADIUS implementations on Cisco session initiation protocol (SIP) gateways. These changes were introduced to provide customers and partners the ability to effectively bill for traffic transported over SIP networks.
Refer to the following document for additional information:
Note
SIP Gateway Support for the Bind Command
In previous releases of Cisco IOS software, the source address of a packet going out of the gateway was never deterministic. That is, the session protocols and Voice over IP (VoIP) layers always depended on the IP layer to give the best local address. The best local address was then used as the source address (the address showing where the SIP request came from) for signaling and media packets. Using this nondeterministic address occasionally caused confusion for firewall applications, because a firewall could not be configured with an exact address and would take action on several different source address packets.
However, the bind interface command allows you to configure the source IP address of signaling and media packets to a specific interface's IP address. Thus, the address that goes out on the packet is bound to the IP address of the interface specified with the bind command. Packets that are not destined to the bound address are discarded.
When you do not want to specify a bind address, or if the interface is down, the IP layer still provides the best local address.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftbind.htm.
Note
SIP Gateway Support for Third Party Call Control
SIP is a new protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to the ITU-T H.323 specification. SIP is defined by RFC 2543 and is used for multimedia call session setup and control over IP networks. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/sipcf2.htm.
Note
SIP Gateway Support of RSVP and TEL URL
The SIP Gateway Support of RSVP and TEL URL feature also supports Telephone Uniform Resource Locators or TEL URL. Currently session initiation protocol (SIP) gateways support URLs in the SIP format. SIP URLs are used in SIP messages to indicate the originator, recipient, and destination of the SIP request. However, SIP gateways may also encounter URLs in other formats, such as TEL URLs. TEL URLs describe voice call connections. They also enable the gateway to accept TEL calls sent through the Internet and to generate TEL URLs in the request line of outgoing INVITE requests.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/vvfresrv.htm.
Note
SIP INFO Method for DTMF Tone Generation
The SIP INFO Method for DTMF Tone Generation feature adds support for dual tone multifrequency (DTMF) tone generation to allow out-of-band signaling. The SIP INFO method is used to generate DTMF tones on the telephony call leg. The SIP INFO method or request message is used by a user agent (UA) to send call signaling information to another UA with which it has an established media session. The SIP INFO message is sent along the signaling path of the call. Upon receipt of a SIP INFO message with DTMF relay content, the gateway generates the specified DTMF tone on the telephony end of the call.
The SIP INFO Method for DTMF Tone Generation feature is always enabled and is invoked when a SIP INFO message is received with DTMF relay content. This feature is related to the DTMF Events Through SIP Signaling feature, which provides the ability for an application to be notified about DTMF events using SIP NOTIFY messages. Together, the two features provide a mechanism to both send and receive DTMF digits along the signaling path.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftinfo.htm.
SIP Intra-gateway Hairpinning
SIP hairpinning is a call routing capability in which an incoming call on a specific gateway is signaled through the IP network and back out the same gateway. This call can be a public switched telephone network (PSTN) call routed into the IP network and back out to the PSTN over the same gateway.
Similarly, SIP hairpinning can be a call signaled from a line (for example, a telephone line) to the IP network and back out to a line on the same access gateway. With SIP hairpinning, unique gateways for ingress and egress are no longer necessary.
Note
SIP INVITE Request with Malformed Via Header
SIP INVITE requests that a user or service participate in a session. Each INVITE contains a Via header that indicates the transport path taken by the request so far and where to send a response. In the past, when an INVITE contained a malformed Via header, the gateway would print a debug message and discard the INVITE without incrementing a counter. However, the printed debug message was often inadequate, and it was difficult to detect that messages were being discarded.
The SIP INVITE Request with Malformed Via Header feature provides a response to the malformed request. A counter, Client Error: Bad Request, increments when a response is sent for a malformed Via field. Bad Request is a class 400 response and includes the explanation Malformed Via Field. The response is sent to the source IP address (the IP address where the SIP request originated) at User Datagram Protocol (UDP) port 5060.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmalvia.htm.
Note
SIP Multiple 18x Responses
The SIP Multiple 18x Responses feature enhances forking support on the user agent client (UAC) by supporting sequential forking. With sequential forking the UAC receives multiple provisional responses (18x) but treats each response as a separate call leg. This allows the proxy to initiate a new INVITE if the called party does not pick up.
SIP—Session Initiation Protocol for VoIP
Voice over IP (VoIP) currently implements the ITU H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. Session Initiation Protocol (SIP) is a protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. The Cisco SIP functionality equips Cisco routers to signal the setup of voice and multimedia calls over IP networks. SIP provides an alternative to H.323 within the VoIP internetworking software.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_sip72.htm.
Note
SIP—Session Initiation Protocol for VoIP Enhancements
Voice over IP (VoIP) currently implements the International Telecommunication Union (ITU)'s H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. The Session Initiation Protocol (SIP) is a new protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.
The Cisco SIP functionality, introduced in Cisco IOS Release 12.1(1)T and enhanced in Cisco IOS Release 12.1(3)T, enables Cisco access platforms to signal the setup of voice and multimedia calls over IP networks. The SIP feature also provides nonproprietary advantages in the areas of
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftsipgv.htm.
SIP Session Timer Support
The SIP Session Timer Support feature adds the capability of a periodic refresh of session initiation protocol (SIP) sessions by sending repeated INVITE requests. The repeated INVITE requests, or re-INVITEs, are sent during an active call leg to allow user agents (UAs) or proxies to determine the status of a SIP session. Without this keepalive mechanism, proxies that remember incoming and outgoing requests (stateful proxies) may continue to retain call state needlessly. If a UA fails to send a BYE message at the end of a session or if the BYE message gets lost because of network problems, a stateful proxy does not know that the session has ended. The re-INVITES ensure that active sessions stay active and that completed sessions are terminated.
The SIP Session Timer Support feature also adds two new general headers that are used to negotiate the value of the refresh interval.
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftsiptim.htm.
SIP T.37 Store and Forward Fax
SIP T.37 is an ITU specification that enables store-and-forward fax applications, as well as toggling from voice to fax, for example, providing an Interactive Voice Response (IVR) front end to a store-and-forward fax application.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm
Note
SIP T.38 Fax Relay
The SIP T.38 Fax Relay feature adds standards-based fax support to session initiation protocol (SIP) and conforms to ITU-T T.38 Procedures for real-time Group 3 facsimile communication over IP networks. The ITU-T standard specifies real-time transmission of faxes between two regular fax terminals over an IP network. Much like a voice call, SIP T.38 Fax Relay requires call establishment, data transmission, and release signaling.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftsipfax.htm.
Note
SIP User Agent MIB
The Session Initiation Protocol (SIP) User Agent Client (UAC) and User Agent Server (UAS) are manageable by an SNMP-based network management platform, such as the Cisco Voice Manager. The SIP UAC/UAS exists on the AS5300 and AS5400 platforms. The SIP MIB has been defined, will be submitted to the IETF, and will be implemented on those platforms.
To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB web site on Cisco.com at the following location:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
SNMP Support over VPN
The SNMP Support over VPN feature allows the sending and receiving of SNMP notifications using VPN Routing Forwarding table (VRF).
SNMP is an application-layer protocol that provides a message format for communication between SNMP managers and agents.
A VPN is a network that provides high connectivity transfers on a shared system with the same usage guidelines as a private network. A VPN can be built on the Internet or on the service provider IP, Frame Relay, or ATM system.
A VRF stores per-VPN routing data. It defines the VPN membership of a customer site attached to the network access server (NAS). A VRF consists of an IP routing table, a derived Cisco Express Forwarding (CEF) table, guidelines, and routing protocol parameters that control the information that is included in the routing table.
The SNMP Support over VPN feature provides configuration commands that allow users to associate SNMP agents and managers with specific VRFs. The specified VRF is used for the sending of SNMP notifications (traps and informs) and responses between agents and managers. If a VRF is not specified, the default routing table for the VPN is used.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftnm_vpn.htm
Note
SNMPv3 Community MIB Support
The SNMPv3 Community MIB Support feature implements support for the SNMP Community MIB (SNMP-COMMUNITY-MIB) module, defined in RFC 2576, in Cisco IOS software.
The SNMPv1/v2c Message Processing Model and Security Model require mappings between parameters used in SNMPv1 and SNMPv2c messages and the version independent parameters used in the Simple Network Management Protocol (SNMP) architecture. The SNMP Community MIB contains objects for mapping between these community strings and version-independent SNMP message parameters.
The mapped parameters consist of the SNMPv1/v2c community name and the SNMP securityName and contextEngineID/contextName pair. This MIB provides mappings in both directions, that is, a community name may be mapped to a securityName, contextEngineID, and contextName, or the combination of securityName, contextEngineID, and contextName may be mapped to a community name. This MIB also augments the snmpTargetAddrTable with a transport address mask value and a maximum message size value.
For implementation details, refer to the SNMP-COMMUNITY-MIB.my file, available through Cisco.com .
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Speech Recognition and Synthesis for Voice Applications
The Speech Recognition and Synthesis for Voice Applications feature adds support for automatic speech recognition (ASR) and text-to-speech (TTS) capabilities for VoiceXML and TCL applications. This feature provides interfaces to ASR and TTS media servers using Media Resource Control Protocol (MRCP), an application-level protocol developed by Cisco and its ASR and TTS media server partners. Client devices that process audio or video streams use MRCP to control media resources on the external ASR and TTS servers.
Refer to the following documents for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ivrapp/index.htm.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/rel_docs/vxmlprg/index.htm.
Static Cache Entry for IPv6 Neighbor Discovery
The Static Cache Entry for IPv6 Neighbor Discovery feature enables the configuring of static entries in the IPv6 neighbor discovery cache, which provides functionality in IPv6 that is equivalent to static Address Resolution Protocol (ARP) entries in IPv4. Static entries in the IPv6 neighbor discovery cache are not modified by the neighbor discovery process. Cisco IOS software uses static ARP entries in IPv4 to translate 32-bit IP addresses into 48-bit hardware addresses. In IPv6, Cisco IOS software uses static entries in the IPv6 neighbor discovery cache to translate 128-bit IPv6 addresses into 48-bit hardware addresses.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/ftipv6s.htm.
Note
Survivable Remote Site Telephony Version 2.0
The Survivable Remote Site Telephony Version 2.0 feature was previously released in Cisco IOS Release 12.2(8)T. In Cisco IOS Release 12.2(11)T, there are minor enhancements to this feature, which is now referred to as Survivable Remote Site Telephony Version 2.01. Refer to the following document for information about the enhancements added to this release:
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/srs/fallbak2.htm.
T.37/T.38 Fax Gateway
This feature adds Store-and-Forward Fax to the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850. Store-and-Forward Fax, previously documented in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2, enables routers to send and receive faxes across packet-based networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm.
Note
T.38 Fax Relay for VoIP H.323
The T.38 Fax Relay for VoIP H.323 feature provides standards-based Fax Relay protocol support on Cisco 2600 series, Cisco 3600 series, Cisco 7200 series and Cisco MC3810 series multiservice gateways. The Cisco proprietary Fax Relay solution is sometimes not an ideal solution for Enterprise and Service Provider customers who have implemented a mixed vendor network. Because the T.38 Fax Relay protocol is standards based, Cisco gateways and gatekeepers will now be able to interoperate with third-party T.38-enabled gateways and gatekeepers in a mixed vendor network where real time Fax Relay capabilities are required.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm.
Note
TCL IVR 2.0 Call Initiation and Callback
The TCL IVR 2.0 Call Initiation and Callback feature allows Tool Command Language (TCL) Interactive Voice Response (IVR) applications to make outbound calls without specifying an incoming call leg in the setup command.
The TCL IVR 2.0 Call Initiation and Callback feature modifies the following TCL IVR Version 2.0 verbs:
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In addition, the following new information tags were added to support the above changes:
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Finally, the following additions were made to the callInfo array:
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Refer to the following TCL IVR API Command Reference for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/chapter3.htm.
TCL IVR Disconnect Cause-Code Manipulation
The leg disconnect command disconnects one or more call legs that are not part of any connection. The cause_code argument, which has been added in Cisco IOS Release 12.2(1)T, is an integer ISDN cause code for the disconnect. It is of the form di-xxx or just xxx, where xxx is the ISDN cause code. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2.htm.
Note
TCL-Enabled Signaling Parameter Mapping
The TCL-Enabled Signaling Parameter Mapping feature provides control over call signaling information elements from a Tool Command Language (TCL) Interactive Voice Response (IVR) script to make the Cisco Media-Gateway (that is, the Cisco AS5300 and Cisco AS5800 platforms) interoperable with British Telecom and France Telecom networks. New parameters were introduced under the set callinfo command. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/chapter3.htm.
TCP Window Scaling
TCP Window Scaling adds support for the Window Scaling extension option in RFC 1323. To improve TCP performance in network paths with a large bandwidth-delay product, Long Fat Networks (LFNs), a larger window size is recommended. This TCP Window Scaling enhancement provides that support.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/tcpwslfn.htm.
Note
Timer and Retry Enhancements for L2TP and L2F
The Timer and Retry Enhancements for L2TP and L2F feature allows the user to configure certain adjustable timers for the L2TP and L2F protocols. For L2F, the settings for control packet retries and control packet timeouts are now both configurable. Initial tunnel packet retries and initial tunnel packet timeouts are now configurable for both the L2F and L2TP protocols.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftretreh.htm.
Note
Trustpoint CLI
The Trustpoint CLI feature introduces the crypto ca trustpoint command, which combines and replaces the functionality of the existing crypto ca identity and crypto ca trusted-root commands.
Although both of the existing commands allow you to declare the certification authority (CA) that your router should use, only the crypto ca identity command supports enrollment (the requesting of a router certificate from a CA). With the crypto ca trustpoint command, you can declare the CA and specify any characteristics for the CA that the existing commands supported.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/fttrust.htm.
Note
Tunnel Type of Service (ToS)
The Tunnel Type of Service (ToS) feature allows you to configure the ToS and Time-to-Live (TTL) byte values in the encapsulating IP header of tunnel packets for an IP tunnel interface on a router. The Tunnel ToS feature is supported on Cisco Express Forwarding (CEF), fast switching, and process switching forwarding modes.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s17/12s_tos.htm.
Note
Universal Port Resource Pooling for Voice and Data Services
With Cisco Resource Pool Manager (RPM), telephone companies and Internet service providers (ISPs) can share dial resources for wholesale and retail dial network services in a single network access server (NAS) or across multiple NAS stacks. Call management and call discrimination can be configured to occur before the call is answered, and customers are differentiated by using configurable customer profiles that are based on the dial number identification service (DNIS) and call type determined at the time of an incoming call. As a result, Cisco RPM enables service providers to count, control, and manage resources and provide accounting for shared resources when implementing different service-level agreements.
The Universal Port Resource Pooling for Voice and Data Services feature enables service providers to mix voice and data services using resource pool management. With the implementation of the new voice command in resource-pool profile service configuration mode, a resource group with voice service is designated under a particular customer profile, and voice resource pool service is enabled after resource pool management is configured.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ftuprp.htm.
V.44 LZJH Compression for Cisco AS5300 and Cisco AS5800 Universal Access Servers
Note
The V.44 LZJH Compression for Cisco AS5300 and Cisco AS5800 Universal Access Servers feature introduces the V.44 Lempel-Ziv-Jeff-Heath (LZJH) compression algorithm International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard on Cisco MICA portware platforms.
V.44 LZJH is a new compression standard based on Lempel-Ziv that uses a new string-matching algorithm that increases upload and download speeds to make Internet access and web browsing faster. The V.44 call success rate (CSR) is similar to V.42bis with significant compression improvement for most file types, including HTML files. V.44 applies more millions of instructions per second (MIPS) than V.42bis toward the same application data stream and yields better compression rates in almost any data stream in which V.42bis shows positive results.
V.44 supports automatic switching between compressed and transparent modes on Cisco MICA portware platforms. Automatic switching allows overall performance gains without loss in throughput for file streams that are not compressible.
V.44 is globally controlled through dialed number identification service (DNIS), calling line ID (CLID), and resource pool manager server (RPMS) virtual groups, and performance improvement is determined by the LZJH algorithms. The Cisco MICA portware is responsible for the ITU implementation of V.44 and the collection of statistics related to the new feature.
To support V.44 LZJH compression, the control switch module (CSM) has been modified. MIBs that show the status of V.42bis have been extended to show V.44 configuration status. New disconnect reasons help manage V.44 session status and debugging.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/122xb2_2/ftv44mca.htm.
V.44 LZJH Compression for Cisco AS5350, Cisco AS5400, and Cisco AS5850 Universal Gateways and Cisco AS5800 Universal Access Servers
Note
The V.44 LZJH Compression for Cisco AS5350, Cisco AS5400, and Cisco AS5850 Universal Gateways and Cisco AS5800 Universal Access Servers feature introduces the V.44 Lempel-Ziv-Jeff-Heath (LZJH) compression algorithm International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard on Cisco MICA portware platforms.
V.44 LZJH is a new compression standard based on Lempel-Ziv that uses a new string-matching algorithm that increases upload and download speeds to make Internet access and web browsing faster. The V.44 call success rate (CSR) is similar to V.42bis with significant compression improvement for most file types, including HTML files. V.44 applies more millions of instructions per second (MIPS) than V.42bis toward the same application data stream and yields better compression rates in almost any data stream in which V.42bis shows positive results.
V.44 supports automatic switching between compressed and transparent modes on Cisco MICA portware platforms. Automatic switching allows overall performance gains without loss in throughput for file streams that are not compressible.
V.44 is globally controlled through dialed number identification service (DNIS), calling line ID (CLID), and resource pool manager server (RPMS) virtual groups, and performance improvement is determined by the LZJH algorithms. The Cisco MICA portware is responsible for the ITU implementation of V.44 and the collection of statistics related to the new feature.
To support V.44 LZJH compression, the control switch module (CSM) has been modified. MIBs that show the status of V.42bis have been extended to show V.44 configuration status. New disconnect reasons help manage V.44 session status and debugging.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/122xb2_2/ft_v44.htm.
V.92 Modem on Hold for Cisco AS5300 and Cisco AS5800 Universal Access Servers
Note
The V.92 Modem on Hold for Cisco AS5300 and Cisco AS5800 Universal Access Servers introduces the V.92 International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard Modem on Hold (MOH) feature with Cisco MICA portware.
To remain current with industry needs, the ITU-T V.90 modem standard recommendations have been enhanced. The new standard, V.92, meets the need for a digital modem and analog modem pair on the Public Switched Telephone Network (PSTN). V.92 improves the upstream data signaling rate and adds new features that enhance modem usability.
V.92 is implemented at the modem level as new modem protocols and standards. The new V.92 features co-reside with existing portware features and have no impact on the hardware configuration of either the HMM or DMM (including memory requirements). Cisco IOS software is responsible for controlling the features and displaying the new statistics. V.92 and V.44 support is bound with the rest of the Cisco IOS device driver components.
V.92 Modem on Hold allows a dial-in customer to suspend a modem session to answer an incoming voice call or to place an outgoing call while engaged in a modem session. When the dial-in customer uses Modem on Hold to suspend an active modem session to engage in an incoming voice call, the Internet service provider (ISP) modem listens to the original modem connection and waits for the dial-in customer's modem to resume the connection. When the voice call ends, the modem signals the telephone system to end the second call and return to the original modem connection, then the modem signals the ISP modem that it is ready to resume the modem call. Both modems renegotiate the connection, and the original exchange of data continues.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/122xb2_2/ft92mmoh.htm.
V.92 Modem on Hold for Cisco AS5350, Cisco AS5400, and Cisco AS5850 Universal Gateways and Cisco AS5800 Universal Access Servers
Note
The V.92 Modem on Hold for Cisco AS5350, Cisco AS5400, and Cisco AS5850 Universal Gateways and Cisco AS5800 Universal Access Servers feature introduces the V.92 International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard Modem on Hold (MOH) feature with Cisco MICA portware.
To remain current with industry needs, the ITU-T V.90 modem standard recommendations have been enhanced. The new standard, V.92, meets the need for a digital modem and analog modem pair on the Public Switched Telephone Network (PSTN). V.92 improves the upstream data signaling rate and adds new features that enhance modem usability.
V.92 is implemented at the modem level as new modem protocols and standards. The new V.92 features co-reside with existing portware features and have no impact on the hardware configuration of either the HMM or DMM (including memory requirements). Cisco IOS software is responsible for controlling the features and displaying the new statistics. V.92 and V.44 support is bound with the rest of the Cisco IOS device driver components.
V.92 Modem on Hold allows a dial-in customer to suspend a modem session to answer an incoming voice call or to place an outgoing call while engaged in a modem session. When the dial-in customer uses Modem on Hold to suspend an active modem session to engage in an incoming voice call, the Internet service provider (ISP) modem listens to the original modem connection and waits for the dial-in customer's modem to resume the connection. When the voice call ends, the modem signals the telephone system to end the second call and return to the original modem connection, then the modem signals the ISP modem that it is ready to resume the modem call. Both modems renegotiate the connection, and the original exchange of data continues.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/122xb2_2/ftv92moh.htm.
V.92 Quick Connect for Cisco AS5300 and Cisco AS5800 Universal Access Servers
Note
The V.92 Quick Connect for Cisco AS5300 and Cisco AS5800 Universal Access Servers feature introduces the V.92 International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard Quick Connect (QC) feature with Cisco MICA portware platforms.
V.92 Quick Connect speeds up the client-to-server startup negotiation, reducing the overall connect time up to 30 percent. The client modem retains line condition information and characteristics of the connection of the Internet service provider (ISP), which reduces connect time by avoiding some of the initial signal handshaking.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/122xb2_2/ft92mqc.htm.
V.92 Quick Connect for Cisco AS5350, Cisco AS5400, and Cisco AS5850 Universal Gateways and Cisco AS5800 Universal Access Servers
Note
The V.92 Quick Connect for Cisco AS5350, Cisco AS5400, and Cisco AS5850 Universal Gateways and Cisco AS5800 Universal Access Servers feature introduces the V.92 International Telecommunication Union Telecommunication Standardization Sector (ITU-T) standard Quick Connect (QC) feature with Cisco MICA portware platforms.
V.92 Quick Connect speeds up the client-to-server startup negotiation, reducing the overall connect time up to 30 percent. The client modem retains line condition information and characteristics of the connection to the Internet service provider (ISP), which reduces connect time by avoiding some of the initial signal handshaking.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/122xb2_2/ftv92qc.htm.
VoAAL2 Profile 9 Support for Broadband Loop Emulation Services Specification Interoperability
The VoAAL2 Profile 9 Support for Broadband Loop Emulation Services Specification Interoperability feature allows the Cisco IAD2420 series integrated access device (IAD) to provide Voice over ATM Adaptation Layer 2 (VoAAL2) Profile 9 using G.711 u-law or G.711 a-law with a 44-byte voice payload. Profile 9 is part of the Broadband Loop Emulation Services (BLES) specification put forth by the ATM Forum. This feature enables Cisco IAD2420 series IADs to offer standards-based interoperability with V5.2 and GR.303 VoAAL2 gateways from various third party vendors that are BLES compliant. The Profile 9 feature allows the Cisco IAD2420 series IADs to deliver business-class voice services from Class 5 switches over T1 ATM and xDSL WAN links. Profile 9 establishes the foundation for accepting packet voice architectures for the carriers and allows the transition to the call agent architectures.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_iadp9.htm.
Voice Application Access To SS7 Signaling
The Voice Application Access To SS7 Signaling feature provides a means of transporting ISUP signaling messages from SS7 networks to VoIP networks. ISUP messages and parameters are converted to Generic Transparency Descriptor (GTD) format and transported by the underlying call signalling messages to each node transited by the call.
Refer to the following document for information about the information tags that are associated with this feature:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/chapter4.htm
Voice DSP Control Message Logger
The Voice DSP Control Message Logger feature provides improved debugging capabilities through Cisco IOS software by allowing you to log control messages that pass through the Cisco IOS software and TI-based voice DSP firmware on the Host Port Interface (HPI). The logged messages can later be examined when voice problems are diagnosed.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftvdsplg.htm.
Voice over IP Q.SIG Network Transparency
Integration of Q.SIG with the Cisco AS5400 universal access server enables Cisco voice switching services to connect private branch exchanges (PBXs), key systems (KTs), and central office switches (COs) that communicate by using the Q.SIG protocol.
The Q.SIG protocol is a variant of ISDN D-channel voice signaling. It is based on the ISDN Q.921 and Q.931 standards and is becoming a worldwide standard for PBX interconnection. By using Q.SIG signaling, the Cisco AS5300 can route incoming voice calls from a private integrated services network exchange (PINX) across a wide-area network (WAN) to a peer Cisco AS5400, which can then transport the signaling and voice packets to a second PINX.
Q.SIG on the AS5400 allows the user to place Q.SIG calls into and receive Q.SIG calls from Cisco Voice-over-IP (VoIP) networks. The Cisco packet network appears to PBXs as a large, distributed transit PBX that can establish calls to any destination served by a Cisco voice node. The switched voice connections are established and torn down in response to Q.SIG control messages that come over an ISDN PRI D channel. The Q.SIG message is passed transparently across the IP network and the message appears to the attached PINXs as a transit network. The PINXs are responsible for processing and provisioning the attached services.
Note
VoiceXML For Cisco IOS
Applications written in Voice eXtensible Markup Language (VoiceXML) provide access through a voice browser to content and services over the telephone, just as Hypertext Markup Language (HTML) provides access through a web browser running on a PC. The universal accessibility of the telephone and its ease of use makes VoiceXML applications a powerful alternative to HTML for accessing the information and services of the World Wide Web.
The Cisco IOS VoiceXML feature provides a platform for interpreting VoiceXML documents. When a telephone call is made to the Cisco VoiceXML-enabled gateway, VoiceXML documents are downloaded from web servers, providing content and services to the caller, typically in the form of pre-recorded audio in an IVR application. Customers can access online business applications over the telephone, providing for example, stock quotes, sports scores, or bank balances.
VoiceXML brings the advantages of web-based development and content delivery to voice applications. It is similar to HTML in its simplicity and in its presentation of information. The Cisco IOS VoiceXML feature is based on the W3C VoiceXML 2.0 Working Draft and is designed to provide web developers great flexibility and ease in implementing VoiceXML applications.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ivrapp/index.htm.
VoiceXML SS7 ISUP Session Variables
The ISUP signaling message set used in SS7 networks contains information that is used for call establishment, routing, and billing functions. To help transport these messages from SS7 networks (using ISUP based messages) to VoIP networks (using H.323 and SIP based messages), ISUP messages and parameters are represented in generic transparency descriptor (GTD) format and transported by the underlying call signaling messages to each node transited by the call. These GTD parameters and fields are extracted and mapped to TCL and VoiceXML variables for access by Tool Command Language (TCL) and VoiceXML scripts.
VoiceXML Media Volume and Rate Controls
With the VoiceXML Media Volume and Rate Controls feature, the volume of audio prompts played out by VoiceXML applications can now be adjusted during playback. Audio prompts that are played out from memory or chunked transfer mode using G.711 or GSM-FR codecs can also be speeded up or slowed down. A VoiceXML variable contains the rate and duration of the last prompt that was played. The rate and volume of prompts is controlled by using Cisco-specific attributes in the VoiceXML document.
VoiceXML Transfer Enhancements
THe VoiceXML Transfer Enhancements feature enhances the transfer functionality in the Cisco VoiceXML implementation by introducing specific Cisco parameters as attributes for the transfer element.
VoiceXML Voice Store and Forward
The VoiceXML Voice Store and Forward feature expands Cisco IOS VoiceXML to include streaming-based recording and playout. It enables the input and processing of form field entries using recorded audio clips, rather than numeric input only. Audio clips can be captured and then submitted to an external web server using HTTP or Real Time Streaming Protocol (RTSP), or to a messaging server using Simple Mail Transfer Protocol (SMTP) for additional processing.
VoIP Call Admission Control using RSVP
The VoIP Call Admission Control Using RSVP feature synchronizes Resource Reservation Protocol (RSVP) procedures with H.323 Version 2 (Fast Connect) setup procedures to guarantee that the required Quality of Service (QoS) for VoIP calls is maintained across the IP network. In older Cisco IOS releases, VoIP gateways used H.323 Version 1 (Slow Connect) procedures when initiating calls requiring bandwidth reservation. This feature, which is enabled by default, allows gateways to use H.323 Version 2 (Fast Connect) for all calls, including those requiring RSVP.
Note
VoIP Interoperability with Cisco Express Forwarding and Policy Based Routing
The VoIP Interoperability with Cisco Express Forwarding and Policy Based Routing feature consists of the following two features:
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The VoIP Interoperability with Cisco Express Forwarding and Policy Based Routing feature enables CEF for switching voice signaling and voice payloads from voice interfaces to other LAN/WAN interfaces for applications, such as Tollbypass. This feature also enables Policy Based Routing of VoIP traffic that originates or terminates on the specified voice gateways and introduces voice packet Differentiated Services Code Point (DSCP) marking for Media Gateway Control Protocol (MGCP) voice gateways.
This feature modifies the Voice over IP (VoIP) and interactive voice response (IVR) programming so that they can interoperate with features that are supported only in the CEF path (not in the fast switching path that VoX uses). Voice and IVR currently only work in the fast path on the routers where they are originated and terminated (Voice and IVR on "transit" routers are just data packets and of course can be CEF switched). Cisco Express Forwarding
Cisco Express Forwarding (CEF) is advanced Layer 3 IP switching technology. CEF optimizes network performance and scalability for networks with large and dynamic traffic patterns, such as the Internet, on networks characterized by intensive Web-based applications, or interactive sessions.
Although you can use CEF in any part of a network, it is designed for high-performance, highly resilient Layer 3 IP backbone switching.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_cef26.htm.
VoIP Gatekeeper Trunk and Carrier Based Routing Enhancements
The VoIP Gatekeeper Trunk and Carrier Based Routing Enhancements feature implements the capability to report the Public Switched Telephone Network (PSTN)-side interfaces for incoming and outgoing calls to the H.323 gatekeeper and to the peer H.323 gateway and endpoint. The feature permits identification, by means of labeling individual PSTN trunks or trunk groups, the circuit that is sending a call. The software routes the call to a specific outbound circuit using some criteria, such as inbound circuit, time period, or cost, and then forwards the call to a circuit that is connected to the specified outbound carrier.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftgkrenb.htm.
VoIP Gateway Trunk and Carrier Based Routing Enhancements
Voice wholesalers use multiple ingress and egress carriers to route traffic. A call that comes in to a gateway on a particular ingress carrier must be routed to an appropriate egress carrier. As networks grow and become more complicated, the dial plans needed to route the carrier traffic efficiently become more complex and the need for carrier sensitive routing (CSR) increases.
The VoIP Gateway Trunk and Carrier Based Routing Enhancements feature implements Carrier Sensitive Routing (CSR) for Cisco voice gateways. The VoIP Gateway Trunk and Carrier Based Routing Enhancements feature adds the following routing features:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftgwrepg.htm
VoIP Outgoing Trunk Group Identification and Carrier ID for Gateways
The VoIP Outgoing Trunk Group Identification and Carrier ID for Gateways feature provides an enhancement to Registration, Admission, and Status (RAS) Admission Confirmation and Location Confirm messages. RAS messages include a circuitInfo field that provides trunk group label or carrier ID information for remote endpoints (gateways) in H.323 networks. The Voice over IP (VoIP) Outgoing Trunk Group Identification and Carrier ID for Gateways feature also adds trunk group label and carrier ID support for the alternate endpoint field in the Gatekeeper Transaction Message Protocol (GKTMP) Response Admission Request (ARQ), Admission Confirmation (ACF), Location Request (LRQ), and Location Confirm (LCF) messages.
The carrier-id keyword and carrier-name arguments were introduced for the endpoint alt-ep h323id command in Cisco IOS Release 12.2(11)T.
VPDN Group Session Limiting
Before the introduction of the VPDN Group Session Limiting feature, you could only globally limit the number of virtual private dialup network (VPDN) sessions on a router with limits applied equally to all VPDN groups. Using the VPDN Group Session Limiting feature, you can limit the number of VPDN sessions allowed per VPDN group. This feature is implemented with the introduction of the session-limit number command in VPDN group configuration mode. VPDN group session limiting is applied after the global VPDN session limiting (which is configured via the vpdn session-limit session command in configuration mode) is enforced.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftvpdngs.htm
Note
VPN Routing Forwarding (VRF) Framed Route (Pool) Assignment via PPP
The VPN Routing Forwarding (VRF) Framed Route (Pool) Assignment via PPP feature introduces support to make the following RADIUS attributes VRF aware: attribute 22 (Framed-Route), a combination of attribute 8 (Framed-IP-Address) and attribute 9 (Framed-IP-Netmask), and the Cisco VSA route command. Thus, static IP routes can be applied to a particular VRF routing table rather than the global routing table.
Note
WRED Enhancement—Explicit Congestion Notification (ECN)
Currently, the congestion control and avoidance algorithms for TCP are based on the idea that packet loss is an appropriate indication of congestion on networks that transmit data using the best-effort service model. When a network uses the best-effort service model, the network delivers data if it can, without any assurance of reliability, delay bounds, or throughput. However, these algorithms and the best-effort service model are not suited to applications that are sensitive to delay or packet loss (for instance, interactive traffic including Telnet, web browsing, and transfer of audio and video data). Weighted random early detection (WRED), and by extension, Explicit Congestion Notification (ECN), helps to solve this problem.
To indicate congestion, WRED drops packets on the basis of the average queue length exceeding a specific threshold value. ECN is an extension to WRED in that ECN marks packets instead of dropping them when the average queue length exceeds a specific threshold value. When configured with the WRED Enhancement—Support for Explicit Congestion Notification feature, routers and end hosts would use this marking as a signal that the network is congested and slow down sending packets.
This feature provides an improved method for congestion avoidance by allowing the network to mark packets for transmission later, rather than dropping them from the queue. Marking the packets for transmission later accommodates applications that are sensitive to delay or packet loss and provides improved throughput and application performance.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftwrdecn.htm.
Note
X.25 Over TCP Profiles
The Cisco X.25 over TCP (XOT) service was originally developed as an X.25 class of service that was only designed to switch X.25 traffic across an IP network. This service allowed network administrators to connect X.25 devices across the rich connectivity and media features available to IP traffic. XOT uses a set of default parameters to make this type of network easy to design.
When the XOT' capabilities were enhanced to support packet assembler/disassembler (PAD) traffic on an XOT session, network designers saw a need to be able to configure parameters for increased flexibility. For instance, because XOT does not have any physical interfaces that an administrator can configure, PAD over XOT sessions cannot be configured with interface map or facility commands to establish a PAD connection using nondefault values.
The introduction of X.25 profiles for XOT allows the network designer added flexibility to control the X.25 class services of XOT for PAD and XOT switching usage.
Another important aspect of this feature is that it allows you to associate access lists with XOT connections, enabling you to apply security on the basis of IP addresses and to have a unique X.25 configuration for specified IP addresses.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_xotp.htm.
Note
X.25 Record Boundary Preservation for Data Communications Networks
The X.25 Record Boundary Preservation for Data Communications Networks feature enables hosts using TCP/IP-based protocols to exchange data with devices that use the X.25 protocol, retaining the logical record boundaries indicated by use of the X.25 "more data" bit (M-bit).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdcnrbp.htm.
Note
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(8)T1
The following hardware platforms and modules are now supported in Cisco IOS Release 12.2(8)T1. These platforms and modules were first introduced in earlier Cisco IOS software releases.
36-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series
The 36-Port Ethernet switch network module is a modular, high-density voice network module that provides Layer 2 switching across Ethernet ports. The 36-port Ethernet switch network module has thirty-six 10/100BASE-TX ports, and an optional power module can also be added to provide inline power for Cisco IP telephones.
The 36-port Ethernet switch network module supports the same features as the 16-port Ethernet switch network module introduced in Cisco IOS Release 12.2(8)T.
Cisco 1721 Router
The Cisco 1721 data-only modular access router is an enhanced Cisco = 1720 router that provides higher performance, additional functionality, and increased memory capacity. The router supports WAN access, Virtual Private Network (VPN), and firewall technology for secure Internet, intranet, and extranet access. Cisco 1721 routers also support standards-based Institute of IEEE 802.1Q VLAN routing, which enables enterprises to set up and route between multiple VLANs for additional security in an internal corporate network.
Cisco 2600XM Series Routers
The Cisco 2600XM series provides new product enhancements to the current Cisco 2600 series. The Cisco 2600XM series is available in three performance levels and six base configurations:
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Each model also has two WAN interface card (WIC) slots, one Network Module slot, and an Advanced Integration Module.
Cisco 2691 Series Router
Cisco IOS Release 12.2(8)T1 supports a new platform, the Cisco 2691 series router.
The Cisco 2691 router is part of the next generation Modular Multiservice platform for deployment of advanced IP Telephony Solutions and Integrated Services. This platform is the fourth in a series of Cisco 2600 products that offer additional performance levels.
The Cisco 2691 provides two 10/100BASE-T Fast Ethernet (FE) ports with one Network Module (NM) slot, three WAN Interface Cards (WICs) slots, and two Advanced Interface Module (AIM) slots. Many of the current NMs, WICs and AIMs used today on the Cisco 2600 and Cisco 3600 series routers are supported on the Cisco 2691 series router.
New Software Features in Cisco IOS Release 12.2(8)T1
The following new features are supported in Cisco IOS Release 12.2(8)T1. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
MPLS Label Switch Controller and Enhancements
The Multiprotocol Label Switching (MPLS) Label Switch Controller (LSC), combined with a slave ATM switch, supports scalable integration of IP services over an ATM network. The MPLS LSC enables the slave ATM switch to:
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This feature was originally introduced in Cisco IOS Release 11.1CT as the Tag Switch Controller. Cisco IOS Release 12.2(8)T1 adds support for the Cisco 8400 IGX Switch with a Universal Router Module as an MPLS ATM-LSR. In addition, support is added for the Virtual Circuit (VC) Merge and MPLS Diff-Serv-aware features.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftlsc.htm.
Virtual Circuit (VC) Merge
The Virtual Circuit (VC) Merge feature allows multiple incoming VCs to be merged into a single outgoing VC. The feature is only available on frame-based connections carrying ATM Adaptation Layer 5 (AAL5) frames consisting of multiple cells. VC Merge helps scale Multiprotocol Label Switching (MPLS) networks, because it allocates only one VC to each destination on a link.
VC merge maps several incoming labels to one single outgoing label. Cells from different virtual channel identifiers (VCIs) traveling to the same destination are transmitted to the same outgoing VC using multipoint-to-point connections.
VC merge allows the switch to transmit cells coming from different VCIs over the same outgoing VCI to the same destination. In other words, VC merge queues AAL5 frames in input buffers until the switch receives the last frame. Then the switch transmits the cells from that AAL5 frame before it sends any cells from other frames. VC merge requires the switch to provide buffering, but no more buffering than is required in IP networks. VC merge slightly delays the transfer of frames; however, VC merge is for IP traffic and not for traffic that requires speed. IP traffic tolerates delays better than other traffic on the ATM network.
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(8)T
The following hardware platforms and modules are now supported in Cisco IOS Release 12.2(8)T. These platforms and modules were first introduced in earlier Cisco IOS software releases.
1- and 2-Port V.90 Modem WICs for Cisco 2600 and 3600 Series
Three applications are available for the V.90 modem WAN interface card (WIC) on the Cisco 2600 and Cisco 3600 series multiservice platforms.
Remote Router Management and Out-of-Band Access
In this mode, the modem WIC is used as a dial-in modem for remote terminal access to the router's command-line interface (CLI) for configuration, troubleshooting, and monitoring. The modem WIC acts similar to a modem that is connected to the auxiliary (AUX) port of a router, but the integrated nature of the modem WIC greatly decreases customer configuration time and deployment and sustaining costs. Typically, the 1-port modem WIC is used for this application. Connection speeds of up to 33.6 kbps are possible.
Asynchronous Dial-on-Demand Routing and Dial Backup
In this mode, the V.90 modem WIC transports network traffic. When ISDN service is not available and the traffic load does not justify a leased line or Frame Relay connection, asynchronous dial-on-demand routing (DDR) is often the only choice for making a WAN connection. Even at sites that do have a leased line or Frame Relay connection, asynchronous DDR can increase bandwidth during sustained traffic load. In addition, when the primary leased line or Frame Relay link is down during an outage, asynchronous dial backup provides a secondary way to make the WAN connection. Both the 1-port and 2-port versions of the V.90 modem WIC can be used for this application.
Two ports on one modem WIC (or even three or more ports spanning multiple modem WIC cards) can be combined using Multilink PPP (MLP) to increase connection speeds in a scalar manner. Each connection is capable of V.90 speeds (up to 56 kbps) when connecting to a digital V.90 server modem.
Low-Density Analog RAS Access
In this application, the V.90 modem WIC enables the platform to provide the services of a typical small remote access server (RAS). One service allows remote users to dial in and gain access to resources on the LAN (or even across the WAN). The analog modems in the modem WIC allow dial-in connection speeds of up to 33.6 kbps, but MLP can bind multiple links together and increase the throughput.
Another service allows PCs (running Cisco DialOut Utility) on the LAN to use the modems for dial-out. Users can connect to other modems (bulletin boards, AOL, ISPs, and so on) or fax machines. The modem WIC allows dial-out connection speeds of up to 56 kbps when dialing a digital V.90 server modem or up to 33.6 kbps when dialing another analog modem. Fax calls connect at up to 14.4 kbps.
Typical RAS deployments with the V.90 modem WIC use the 2-port modem version. With enough slots, the V.90 modem WIC can be used to scale to up to 24 modems in a Cisco 3660 multiservice platform.
There is no limit for lines in the MLP bundle with WICs and population of WICs on any Cisco 2600 series or Cisco 3600 series multiservices platforms.
Additional Information
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft12pwi8.htm.
8FXO DID for IAD24xx Platform
Note
Direct Inward Dialing (DID) is a service offered by telephone companies that enables callers to dial directly to an extension on a private branch exchange (PBX) without the assistance of an operator or automated call attendant. This service makes use of DID trunks, which forward only the last three to five digits of a phone number to the PBX. If, for example, a company has a PBX with extensions 555-1000 to 555-1999, and a caller dials 555-1234, the local central office (CO) would forward 234 to the PBX. The PBX would then ring extension 234. This entire process is transparent to the caller.
The Foreign Exchange Office (FXO) ports on the analog FXO voice module supports the Direct Inward Dialing (DID) for Cisco IAD2420 platform. An eight-port FXO voice interface module for the Cisco IAD2420 platform provides higher FXO port density than was previously available in the Cisco IAD2420 platform. These analog voice ports can be used to support analog voice connections from the Cisco IAD2420 chassis to the PBX on the CO side of the interface.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdidiad.htm.
16-Port Ethernet Switch Module for Cisco 2600 Series and Cisco 3600 Series
The16-Port Ethernet switch network module is a modular, high-density voice network module that provides Layer 2 switching across Ethernet ports. The 16-port Ethernet switch network module has sixteen 10/100BASE-TX ports, and an optional power module can also be added to provide inline power for Cisco IP telephones.
Features included on this network module include the following:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xt/122xt_2/ft1636nm.htm.
7100 PA Support
The 7100 PA is a port of support for the following features and port adapters on the Cisco 7100: Turbo ACLs, cRTP Acceleration, PA-GE, PA-MC-2T1, PA-MC-2E1/120, PA-POS, PA-MC-4T1, PA-MC-8T1, PA-MC-8E1, PA 2FE, PA-T3+, and PA-2T3+.
AIM-ATM, AIM-VOICE-30, and AIM-ATM-VOICE-30 on the Cisco 2600 Series and Cisco 3660
Three types of Advanced Integration Modules (AIMs) provide components that provide segmentation and reassembly (SAR) of packets for ATM transport over a WAN and voice digital signal processing (DSP) services. The Cisco 2600 series has one internal slot for an AIM, and the Cisco 3660 has two. The three types of AIMs are as follows:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_04gin.htm.
Analog Station Interface (ASI) Cards
Analog station interface (ASI) cards enable you to connect to analog telephones, fax machines, and teleconferencing stations. The following two ASI cards are available:
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ATM OC-12 Port Adapter
Platforms: Cisco 7500/RSP series with Versatile Interface Processor (VIP)
The ATM OC-12 Port Adapter is a dual-width ATM port adapter that provides a single-port, 622.08 Mbps connection from Cisco 7500 series routers to any ATM switch. The PA-A3 OC-12 includes two hardware versions (PA-A3-OC12MM and PA-A3-OC12SMI) that support the following standards-based physical interfaces:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/cable/cab_rout/cfig_nts/6228oc12/6228ovrn.htm.
Cisco 806 Broadband Gateway Router
The Cisco 806 Broadband Gateway Router adds business-class functionality to affordable broadband access for small offices and corporate telecommuters. Through the power of Cisco IOS technology, the Cisco 806 provides business-class security, remote management, and quality of service capabilities. These value-added features, with the proven reliability of Cisco IOS technology, provide the mission-critical networking required by today's agile businesses.
Cisco 1721 Router
The Cisco 1721 data-only modular access router is an enhanced Cisco 1720 router that provides higher performance, additional functionality, and increased memory capacity. The router supports WAN access, VPN, and firewall technology for secure Internet, intranet, and extranet access. Cisco 1721 routers also support standards-based IEEE 802.1Q VLAN routing, which enables enterprises to set up and route between multiple VLANs for additional security in an internal corporate network.
Cisco 3631 Series Router
Cisco IOS Release 12.2(8)T supports a new platform, the Cisco 3631 series router.
The Cisco 3631 is a new midrange router for Data Communication Network (DCN) applications that provides two network modules and two WICs, one Fast Ethernet port, one console port, and an auxiliary port. The Cisco 3631 is two rack units high in an 11-inch NEBS/ETSI-compliant chassis that functions at 70,000 pps.
Cisco 3725 Application Service Router
Cisco IOS Release 12.2(8)T supports a new platform, the Cisco 3725 router.
The Cisco 3725 Series Application Service Router is part of a new family of modular routers that enable flexible and scalable deployment of new e-business applications in an integrated branch office access platform.
The Cisco 3700 series are new access platforms optimized for the modular integration and consolidation of branch applications and services. The Cisco 3725 is a two-rack unit (RU) router equipped with two on-board Fast Ethernet (FE) interfaces, three WAN Interface Card (WIC) slots and two Advanced Integration Module (AIM) slots, and two network module (NM) slots. The Cisco 3725 also includes optional -48vDC integrated inline power to support IP Telephony when used with an EtherSwitch network module.
Cisco 3745 Application Service Router
Cisco IOS Release 12.2(8)T supports a new platform, the Cisco 3745 router.
The Cisco 3745 Series Application Service Router is part of a new family of modular routers that enable flexible and scalable deployment of new e-business applications in an integrated branch office access platform.
The Cisco 3745 is a three-rack unit (RU) router equipped with two on-board Fast Ethernet (FE) interfaces, three WAN Interface Card (WIC) slots and two Advanced Integration Module (AIM) slots, and two network module (NM) slots. The Cisco 3745 also includes optional 48vDC integrated inline power, internal redundant AC or DC Power options, and Online Insertion and Removal (OIR) capabilities for like network modules.
The Cisco 3700 series is ideal for sites and solutions requiring the highest levels of integration at the edge, such as:
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Cisco High-Density Analog Voice and Fax Network Module
The Cisco High-Density Analog Voice and Fax Network Module provides dual tone multifrequency (DTMF) detection, voice compression and decompression, call progress tone generation, voice activity detection (VAD), echo cancellation, and adaptive jitter buffering for up to 16 ports.
The base card supports four foreign exchange station (FXS) ports. The addition of an eight-port FXS expansion module can increase the capacity to twelve FXS ports. The addition of two four-port FXO expansion modules can increase the capacity to eight FXO ports and four FXS ports. The addition of one each of the FXS and FXO expansion modules can increase the capacity to twelve FXS ports and four FXO ports. The FXO expansion module supports a power failure port, which connects directly to the central office (CO) in case of failure.
The digital signal processors (DSPs) on the network module support up to eight ports of high-complexity codecs or up to sixteen ports of medium-complexity and low-complexity codecs. The number of DSPs must be increased if more than eight ports of high-complexity codecs are needed. In this case, a DSP expansion module must be installed.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xt/
122xt_2/ft_hdanm.htm.Cisco IOS Voice Features on IGX 8400 Series Universal Router Module
The Universal Router Module (URM) is a Cisco IOS-based IP router blade that enables users to provision Voice over IP (VoIP) and Voice over ATM (VoATM) on a Cisco IGX 8400 series platform. The voice and routing capabilities of the URM have been derived from the Cisco 3660, while the ATM capabilities have been derived from the ATM OC/3 network module for the Cisco 2600 series and Cisco 3600 series routers. The embedded UXM-E processor supports one OC3 ATM port, and the embedded router supports one OC3 ATM port similar to the 1-port OC-3/STM-1 ATM Circuit Emulation Service network module for the Cisco 3600 series routers. These ATM ports are connected to each other internally.
In addition to VoIP and VoATM, IP routing and Cisco IOS command-line interface (CLI) commands, which enable configuration of the voice ports and dial peers, are now available on the Cisco IGX 8400 series platforms.
The URM interoperates with all Cisco IOS-based voice products and supports 30 voice channels with high-complexity codec types and 60 voice channels with medium-complexity codec types. Note that only digital voice ports are supported on the URM; analog ports are not supported.
The URM also provides support for MPLS, IP Security (IPSec), remote embedded router configuration of the URM (a RAS feature), and support for Enterprise Plus features. Using the BC-URI-2FE back card, you can use the URM for data-only access. Also, support for VPN-AIM/HP enables the URM to provide hardware-accelerated encryption for scalable IPSec-VPN networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/
122xb_2/ft_igxxb.htm.Digital J1 Voice Interface Card
The J1 interface card provides the proper interface for directly connecting Cisco multiservice access routers to Japanese Private Branch Exchanges (PBXs) that use a J1 interface (2.048 Mbps TDM interface). This interface card supports 30 voice channels per port.
It provides the software and hardware features required to connect to over 80 percent of the Japanese PBXs that use digital interfaces. This new J1 voice interface card (VIC) provides a TTC JJ-20.11 compliant interface between high-density voice network modules (NM-HDV) and a Japanese PBX.
The digital J1 card provides a single-port line interface in a VIC form factor. It is specifically designed to conform to the TTC JJ-20.10-12 standards that define the interface between a PBX and time-division multiplexer (TDM).
For additional information about the Digital J1 Voice Interface Card, refer to the following document:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftj1voip.htm.
G.SHDSL Symmetric DSL Support
Note
G.SHDSL is an ATM-based, multirate, high-speed (up to 2.3 MB), symmetrical digital subscriber line technology for data transfer between a single customer premises equipment (CPE) subscriber and a central office.
G.SHDSL is supported on the G.SHDSL WAN interface card (WIC-1SHDSL), a 1-port WAN interface card (WIC) for Cisco 2600 series and Cisco 3600 series routers.
The G.SHDSL WIC is compatible with the Cisco 6015, Cisco 6130, Cisco 6160, and Cisco 6260 Digital Subscriber Line Access Multiplexers (DSLAMs). The DSLAM must be equipped with G.SHDSL line cards that are compatible with the DSL service to be configured.
The G.SHDSL WIC supports ATM Adaptation Layer 2 (AAL2), ATM Adaptation Layer 5 (AAL5), and various classes of service for ATM.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_gdsl8.htm.
Multichannel STM-1 Port Adapter
The Multichannel STM-1 Port Adapter is a high-speed, single-port multichannel STM-1 port adapter. You can configure the PA-MC-STM-1 as a multichannel E1/E0 STM-1 port.
The PA-MC-STM-1 can be configured into 63 individual E1 links. Each E1 link can carry a single channel at full or fractional rates or be broken down into multiple DS0 or nx64 Kbps rates. The PA-MC-STM-1 supports up to three TUG-3/AU-3 transport slots numbered 1 through 3. You can configure each TUG-3/AU-3 to carry 21 SDH TU-12s. Each SDH TU-12 is capable of carrying a channelized E1 frame, which can be unchannelized to nx64-Kbps time slots.
For additional information about the Multichannel STM-1 Port Adapter, refer to the following document:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_stm5.htm.
NM-AIC-64, Contact Closure Network Module
The NM-AIC-64, Contact Closure Network Module (also known as the AIC) is an optional card that expands network management capabilities for customer-defined alarms. The AIC has its own CPU that communicates with the router and external media through serial communication channels. The AIC reduces service provider and enterprise operating costs by providing a flexible, low-cost network solution for migrating existing data communications networks (DCNs) to IP-based DCNs. The AIC provides its users with a single box solution because it can be configured in the same router along with other operations, alarm administration, maintenance management, and provisioning (OAM&P) interfaces.
This feature was first introduced on the Cisco 2600 series and Cisco 3600 series platforms in Cisco IOS Release 12.2(2)XG. For Cisco IOS Release 12.2(8)T, platform support for the Cisco 3631 has been added.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_aicnm.htm.
URM LAN
On the Cisco IGX 8400 series, the Universal Router Module (URM) has been enhanced by new LAN features such as Security and VPN. Installed URMs can be enabled with the new features by upgrading to IGX switch software 9.3.30 and Cisco IOS Release 12.2(2)XX as well as by adding an AIM-VPN daughter module to the URM. Also, a new, price-reduced back card for the URM with 2 FE ports (BC-URI-2FE) for LAN services will be supported. URM together with the voice-enabled back cards (BC-URI-2FE2V-E1/T1) will support the new LAN.
New Software Features in Cisco IOS Release 12.2(8)T
The following new features are supported in Cisco IOS Release 12.2(8)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
ACL Authentication of Incoming rsh and rcp Requests
To enable the Cisco IOS software to receive incoming remote shell (rsh) protocol and remote copy (rcp) protocol requests, customers must configure an authentication database to control access to the router. This configuration is accomplished by using the ip rcmd remote-host command.
Currently, when using this command, customers must specify the local user, the remote host, and the remote user in the database authentication configuration. For users who can execute commands to the router from multiple hosts, multiple database authentication configuration entries must be used, one for each host.
This feature allows customers to specify an access list for a given user. The access list identifies the hosts to which the user has access. A new argument, access-list, has been added that can be used with this command to specify the access list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftauth.htm.
Asynchronous Serial Traffic Over User Datagram Protocol (UDP)
The Asynchronous Serial Traffic Over User Datagram Protocol (UDP) feature provides the ability to encapsulate asynchronous data into UDP packets, and then unreliably transmit this data without needing to establish a connection with a receiving device.
You load the data you want to transmit through an asynchronous port, and then transmit it, optionally, as a multicast or a broadcast. The receiving device(s) can then receive the data whenever it wants. If the receiver ends reception, the transmission is unaffected.
This process is referred to as UDP Telnet (UDPTN), although it does not---and cannot---use the Telnet protocol. UDPTN is similar to Telnet in that both are used to transmit data, but UDPTN is unique in that it does not require that a connection be established with a receiving device.
ATM PVC Bundle Enhancement—MPLS EXP-Based PVC Selection
The ATM PVC Bundle Enhancement — MPLS EXP-Based PVC Selection feature is an extension to the IP to ATM Class of Service feature suite. The IP to ATM Class of Service feature suite, using virtual circuit (VC) support and bundle management, maps quality of service (QoS) characteristics between IP and ATM. It provides customers who have multiple VCs (with varying qualities of service to the same destination) the ability to build a QoS differentiated network.
The IP to ATM Class of Service feature suite allowed customers to use IP precedence level as the selection criteria for packet forwarding. This new feature now gives customers the option of using the Multiprotocol Label Switching (MPLS) experimental (EXP) level as an additional selection criteria for packet forwarding.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftmpls.htm.
ATM Software Segmentation and Reassembly (SAR)
The ATM Software Segmentation and Reassembly (SAR) feature allows the Cisco 2600 series to carry voice and data traffic over ATM networks using ATM Adaptation Layer 2 (AAL2) and AAL5 and allows the Cisco 3660 router to support AAL2 voice traffic.
For the Cisco 2600 series, this feature works in conjunction with the T1/E1 multiflex voice/WAN interface card (VWIC), which is plugged into a WIC slot to provide one ATM WAN interface at a T1/E1 rate supporting up to 24/30 voice channel.
T1/E1 ATM support is a time-to-market feature that helps service providers take advantage of the inherent quality of service (QoS) features of ATM multiservice applications. FR-ATM (FRF.5 and FRF.8) internetworking is supported on the Cisco 2600 series.
On the Cisco 3660, a T1 IMA network module is used as the Inverse Multiplexing ATM (IMA) interface providing a maximum of one ATM IMA interface that supports up to 48/60 voice channels. Up to eight T1/E1s and multiple IMA groups are permitted, but only the first IMA group supports voice over AAL2 for up to 48/60 voice channels.
NM-IMA already supports AAL5 on both the Cisco 2600 series and Cisco 3600 series (not just the Cisco 3660).
The Cisco 2600 Series T1/E1 ATM portion of this feature provides a shared implementation of the ATM features currently available on the Cisco MC3810 with the Cisco 2600 series.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ft_t1atm.htm.
ATM SVC Troubleshooting Enhancements
The ATM SVC Troubleshooting Enhancements feature introduces the following two new debug commands: debug atm native and debug atm nmba. These commands can be used to troubleshoot ATM switched virtual circuits (SVCs). The debug atm nbma and debug atm native commands are used to debug problems with Resource Reservation Protocol (RSVP) SVC creation and teardown. The debug atm native command can also be used to debug problems with SVCs created using static maps.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftsvctrb.htm.
BGP Hide Local-Autonomous System
The BGP Hide Local-Autonomous System feature introduces the no-prepend keyword to the neighbor local-as command. The use of the no-prepend keyword allows a network operator to configure a Border Gateway Protocol (BGP) speaker to not prepend the local autonomous system number to any routes that are received from external peers. This feature can be used to help transparently change the autonomous system number of a BGP network and ensure that routes can be propagated throughout the autonomous system, while the autonomous system number transition is incomplete. Because the local autonomous is not prepended to these routes, external routes will not be rejected by internal peers during the transition from one autonomous system number to another.
Refer to the following document for additional information:
https://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guides_list.html.
BGP Named Community Lists
The BGP Named Community Lists feature introduces a new type of community list called the named community list. The BGP Named Community Lists feature allows the network operator to assign meaningful names to community lists and increases the number of community lists that can be configured. A named community list can be configured with regular expressions and with numbered community lists. All rules of numbered communities apply to named community lists except that there is no limitation on the number of community attributes that can be configured for a named community list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftbgpncl.htm.
BIP—BSC to IP Conversion for Automated Teller Machines
The Bisync-to-IP (BIP) Conversion for Automated Teller Machines feature enables customers to attach a binary synchronous (bisync) communication automated teller machine to a serial interface on a Cisco router running bisync-to-IP (BIP) protocol translation and then to route the data over a TCP/IP network directly to an IP-based application host.
Call Admission Control for H.323 VoIP Gateways
Call Admission Control for H.323 VoIP Gateways feature set provides the ability to support resource-based call admission control processes. These resources include system resources such as CPU, memory, and call volume and interface resources such as call volume.
If system resources are not available to admit the call, the following two kinds of actions are provided: system denial (which busyouts all of T1 or E1) or per-call denial (which disconnects, hairpins, or plays a message or tone). If the interface-based resource is not available to admit the call, the call is dropped from the session protocol (such as H.323).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_cac7x.htm.
CDP and ODR Support for ATM PVCs
This feature introduces support for the Cisco Discovery Protocol (CDP) over ATM point-to-point permanent virtual circuits (PVCs). Prior to this release, CDP discovery messages were not supported over ATM interfaces.
CDP is a Cisco proprietary device discovery protocol. Each Cisco device periodically sends messages to a multicast address. These messages advertise information about that device, such as the system ID (name), capabilities, Cisco IOS software version, and the network address of the connected interface. This information will be picked up by any neighboring Cisco devices on the same medium, which are listening for CDP advertisements. The information learned about neighboring devices is available through the Cisco IOS CLI show cdp commands and through SNMP monitoring using the CDP MIB.
This feature also adds support for On-Demand Routing (ODR) over ATM PVCs. ODR uses CDP to propagate IP address information in hub-and-spoke topologies. When ODR is enabled, spoke routers automatically advertise their subnets using CDP.
CDP is disabled by default for ATM PVC interfaces. To enable CDP, use the cdp run global configuration mode command and the cdp enable interface configuration mode command on both ends of the PVC. To enable ODR, use the router odr global configuration mode command on the hub router and turn off any dynamic routing protocols in the spoke routers.
For details on configuring CDP, refer to the following documentation:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/ffun_c/fcfprt3/fcf015.htm.
For details on configuring ODR, refer to the following documentation:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fipr_c/ipcprt2/1cfodr.htm.
Cisco Discovery Protocol (CDP)— IPv6 Address Family Support for Neighbor Information
The CDP IPv6 Address Family Support for Neighbor Information feature adds the ability to transfer IPv6 addressing information between two Cisco devices using Cisco Discovery Protocol (CDP). CDP in IPv6 functions the same as and offers the same benefits as CDP in IPv4. IPv6 enhancements to CDP allow CDP to exchange IPv6 and neighbor addressing information. IPv6 CDP provides IPv6 information to network management products and provides troubleshooting tools.
CEF-Switched Multipoint GRE Tunnels
The CEF-Switched Multipoint GRE Tunnels feature enables CEF switching of IP traffic to and from multipoint GRE tunnels. Tunnel traffic can be forwarded to a prefix through a tunnel destination when both the prefix and the tunnel destination are specified by the application.
Certificate Autoenrollment
The Certificate Autoenrollment feature allows you to configure your router to automatically request a certificate from the certification authority (CA) that is using the parameters in the configuration. Thus, operator convention is no longer required at the time the enrollment request is sent to the CA server.
Automatic enrollment will be performed on startup for any trustpoint CA that is configured and does not have a valid certificate. When the certificate—which is issued by a trustpoint CA that has been configured for autoenrollment—expires, a new certificate is requested. Although this feature does not provide seamless certificate renewal, it does provide unattended recovery from expiration.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftautoen.htm.
Certificate Enrollment Enhancements
The Certificate Enrollment Enhancements feature introduces five new subcommands to the crypto ca trustpoint command—ip-address (ca-trustpoint), password (ca-trustpoint), serial-number, subject-name, and usage. These commands provide new options for certificate requests and allow users to specify fields in the configuration instead of having to go through prompts. (However, the prompting behavior remains the default if this feature is not enabled.) Thus, users can preload all necessary information into the configuration, allowing each router to obtain its certificate automatically when it is booted.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftenrol2.htm.
CISCO-BULK-FILE-MIB Enhancements
The Cisco Bulk File Creation MIB (CISCO-BULK-FILE-MIB.my) is a MIB module for creating and deleting bulk files of SNMP data for file transfer. The CISCO-BULK-FILE-MIB Enhancements feature enhances the Cisco Bulk File Creation MIB to support selective-row-transfer and notification-on-file-creation. Prior to this enhancement, when the MIB was used to dump large tables (for example, the ccHistoryTable), much of the data transfer consisted of duplicated data. This feature allows the SNMP manager to specify a starting row in the SNMP Get request.
This feature also introduces a notification that can be sent when file creation is complete or when there is an error during file creation. Specifically, this feature modifies the CISCO-BULK-FILE-MIB by introducing four new MIB objects (cbfDefineFileNotifyOnCompletion, cbfDefineObjectTableInstance, cbfDefineObjectNumEntries, cbfDefineObjectLastPolledInst) and a new notification object (cbfDefineFileCompletion). For details, refer to the CISCO-BULK-FILE-MIB.my file, available through Cisco.com MIB FTP site at the following URL:
http://tools.cisco.com/Support/SNMP/do/BrowseMIB.do?local=en&mibName=CISCO-BULK-FILE-MIB
Cisco Gateway Management Agent (CGMA) Phase 2
The Cisco Gateway Management Agent (CGMA) Phase 2 feature provides additional enhancements for the Cisco Gateway Management Agent (CGMA) feature. The CGMA provides an eXtensible Markup Language (XML) interface to support real-time management of a Cisco IOS gateway. Currently, gateways provide statistics using Simple Network Management Protocol (SNMP) and do not support real-time polling. The CGMA feature allows gateways to communicate with third-party management applications using XML over TCP/IP.
Refer to the following document for additional information:
https://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guides_list.html.
Cisco Hoot and Holler over IP
The Cisco Hoot and Holler over IP feature can now be ported to the Cisco 1750, 1751, and 1760 routers in Cisco IOS Release 12.2(8)T. The feature is already available on the Cisco 2600 and 3600 series routers.
Cisco VoIP technology, which was initially focused on traditional PBX toll-bypass applications, can be used to combine hoot and holler networks with data networks. While some customers may have integrated hoot and data to some level in the late 1980s with time-division multiplexing (TDM), this form of integration does not allow for dynamic sharing of bandwidth that is characteristic of VoIP. This dynamic sharing of bandwidth is even more compelling with hoot and holler than with a toll-bypass application because some hoot circuits may be active for an hour or two for morning reports but dead for the rest of the day. The idle bandwidth can be used by the data applications during these long periods of inactivity.
Beginning with Cisco IOS Release 12.1(2)XH, Cisco hoot and holler over IP can be implemented using Cisco VoIP technology. This solution leverages Cisco IOS expertise in VoIP, quality of service (QoS), and IP multiplexing and is available on Cisco 1750, 1751, and 1760 routers and on Cisco 2600 and 3600 series multiservice routers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_hhip.htm.
Cisco IOS Firewall Performance Improvements
The Cisco IOS Firewall Performance Improvements feature introduces the following three performance metrics for Context-Based Access Control (CBAC):
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftfirewl.htm.
Cisco IOS Telephony Service Version 2.0
The Cisco IOS Telephony Service, under the IP Telephony services umbrella, provides basic Cisco IP phone call-handling capabilities in a LAN environment on the Cisco routers. This feature enables the Cisco multiservice routers to act as the Cisco IOS Telephony Service for the Cisco IP Phone 7960, Cisco IP Phone 7940, Cisco IP Phone 7910, and Cisco IP Conference Station 7935. This feature also helps download phone software images and configures and manages the Cisco IP phones in your LAN. The Cisco IOS Telephony Service provides you with a telephony system perfect for a small office with a small number of extensions.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/ipkey2.htm.
Cisco Service Assurance Agent Support for the Cisco 820 Series and SOHO 70 Series
Cisco IOS Release 12.2(8)T adds support for the Cisco Service Assurance Agent feature to Cisco 820 series and Cisco SOHO 70 series routers. For information on configuration for the Cisco Service Assurance Agent, refer to the following location:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/ffun_c/fcfprt3/fcf017.htm.
The Cisco 820 series and SOHO 70 series do not currently support the Cisco Service Assurance Agent Application Performance Monitor (APM) feature.
Class-Based Weighted Fair Queueing (CBWFQ)
Class-based weighted fair queueing (CBWFQ) extends the standard WFQ functionality to provide support for user-defined traffic classes. For CBWFQ, you define traffic classes based on match criteria including protocols, access control lists (ACLs), and input interfaces. Packets satisfying the match criteria for a class constitute the traffic for that class. A queue is reserved for each class, and traffic belonging to a class is directed to the queue for that class.
CNS Agents SSL Security
CNS Agents SSL Security is a Cisco IOS software feature that allows for the configuration of a secure connection between the CNS Agent, running on the Cisco IOS software-based device, and a CNS Server. Secure Socket Layer (SSL) encryption for CNS connections is enabled on the Cisco IOS device (CNS Agent) side using the encrypt keyword with the cns config initial or cns config partial global configuration mode commands.
CNS Flow-Through Provisioning
The CNS Flow-Through Provisioning feature provides the infrastructure for automated configuration of network devices on a mass scale. Based on the already released 12.2T CNS event and configuration agents, this extra functionality facilitates the industry's first true "zero-touch" network deployment solution, eliminating the need for the traditional technician truck-roll associated with initial device turn-up. This IOS infrastructure interoperates with CNS IE2100 Intelligent Network Engine, creating the foundation for a closed loop binding of the service provider's operational systems, business systems, and Cisco's order process into a single e-business solution. The result is the first automated work flow ranging from initial subscriber order-entry, through Cisco manufacturing and shipping, to final device provisioning and subscriber billing. This process focuses on a root problem of today's service provider business model—use of human labor in the mass production process of subscriber service activation.
Configurable PSTN Cause Code to SIP Response Mapping
For calls to be established between a Session Initiation Protocol (SIP) network and a Public Switched Telephone Network (PSTN), the two networks must be able to interoperate. One aspect of their interoperation is the mapping of PSTN cause codes, which indicate reasons for PSTN call failure or completion, to SIP status codes or events. The opposite is also true: SIP status codes or events are mapped to PSTN cause codes. Event mapping tables in the document referenced below show the standard or default mappings between SIP and PSTN.
However, you may want to customize the SIP user agent software to override the default mappings between the SIP and PSTN networks. The Configurable PSTN Cause Code to SIP Response Mapping feature allows you to configure specific map settings between the PSTN and SIP networks. Thus, any SIP status code can be mapped to any PSTN cause code, and vice versa. When set, these settings can be stored in NVRAM and are restored automatically on bootup.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmap.htm
Default VPDN Group Template
The Default VPDN Group Template feature introduces the ability to configure global default values for virtual private dialup network (VPDN) parameters in a VPDN template. These global default values are applied to all VPDN groups, unless specific values are configured for individual VPDN groups. Previously, the Cisco IOS software required that VPDN parameters be configured for each individual VPDN group if the system default values were not desired.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdevpdn.htm.
DHCP Client—Dynamic Subnet Allocation API
The DHCP Client-Dynamic Subnet Allocation API feature is an application program interface (API) that is called by the DHCP Server-On-Demand Address Pool Manager feature for obtaining a subnet or releasing a subnet to the source server via DHCP. This feature allows automated configuration of layer 3 devices for simplified deployment.
DHCP Client on WAN Interfaces
The DHCP Client on WAN Interfaces feature extends the Dynamic Host Configuration Protocol (DHCP) to allow PPP over ATM (PPPoA) and certain ATM interfaces to acquire an IP address through DHCP. By using DHCP rather than the IP Control Protocol (IPCP), a DHCP client can acquire other useful information such as DNS server addresses, the DNS default domain name, and default route.
Previously, the ip address dhcp interface configuration command could only be used on Ethernet interfaces. This feature allows the ip address dhcp command to be used on WAN interfaces.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftwandhp.htm.
DHCP Relay—MPLS VPN Support
The DHCP relay agent information option (option 82) enables a Dynamic Host Configuration Protocol (DHCP) relay agent to include information about itself when forwarding client-originated DHCP packets to a DHCP server. The DHCP server can use this information to implement IP address or other parameter-assignment policies. The DHCP relay agent information option is organized as a single DHCP option that contains one or more suboptions that convey information known by the relay agent.
In some environments, a relay agent resides in a network element that also has access to one or more Multiprotocol Label Switching (MPLS) Virtual Private Networks (VPNs). A DHCP server that wants to offer service to DHCP clients on those different VPNs needs to know the VPN in which each client resides. The network element that contains the relay agent typically knows about the VPN association of the DHCP client and includes this information in the relay agent information option.
The DHCP Relay-MPLS VPN Support feature allows the relay agent to forward this necessary VPN-related information to the DHCP server using the following three suboptions of the DHCP relay agent information option:
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The DHCP Relay-MPLS VPN Support feature enables a network administrator to conserve address space by allowing overlapping addresses. The relay agent can now support multiple clients on different VPNs, and many of these clients from different VPNs can share the same IP address.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdhmpls.htm.
DHCP Server—On-Demand Address Pool Manager
The DHCP Server-On-Demand Address Pool Manager is a feature in which pools of IP addresses can be dynamically increased or reduced in size depending on the address utilization level. This feature supports address assignment using the Dynamic Host Configuration Protocol (DHCP) for customers using private addresses. Each on-demand address pool (ODAP) is configured and associated with a particular Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN).
When configured, the ODAP is populated with one or more subnets leased from a source server and is ready to serve address requests from DHCP clients or from PPP sessions. The source server can be a remote DHCP server or a RADIUS server (via AAA). Currently, only the Cisco Access Registrar RADIUS server supports ODAPs. Subnets can be added to the pool when a certain utilization level (high utilization mark) is achieved. When the utilization level falls below a certain level (low utilization mark), a subnet can be returned to the server from which it was originally leased.
This feature allows customers to optimize their use of IP addresses, thus conserving address space.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftondhcp.htm.
DHCP Server—Option to Ignore All BOOTP Requests
The DHCP Server—Option to Ignore All BOOTP Requests feature introduces the following new global configuration command: ip dhcp bootp ignore. This command allows the Cisco IOS DHCP server to ignore received BOOTP requests.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdbootp.htm.
DHCP Server Options Import and Autoconfiguration
The Cisco IOS DHCP server was enhanced to allow configuration information to be updated automatically. Network administrators can configure one or more centralized DHCP servers to update specific DHCP options within the DHCP pools. The remote servers can request or "import" these option parameters from the centralized servers.
This feature was originally introduced in Cisco IOS Release 12.1(2)T. This release is porting the feature into the Cisco 800 series platform.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t2/dt_dhcpi.htm.
Dialer Map VRF-Aware for an MPLS VPN
The Cisco IOS Release 12.2(8)T dialer software is "VRF-aware for an MPLS VPN," which means that it can distinguish between two destinations with the same IP address using information stored in a virtual routing and forwarding instance (VRF). The VRF is identified based on the incoming interface of the packet and is used with a defined destination IP address to determine the telephone number to be dialed.
The Dialer Map VRF-Aware for an MPLS VPN feature allows the dialer software to dial out in a Multiprotocol Label Switching (MPLS)-based Virtual Private Network (VPN). The MPLS VPN model simplifies network routing by allowing several sites to transparently interconnect through the service provider network. One service provider network can support several different IP VPNs, each of which appears to its users as a separate, private network. Within a VPN, each site can send IP packets to any other site in the same VPN because each VPN is associated with one or more VRFs. The VRF is a key element in the VPN technology because it maintains the routing information that defines a customer VPN site.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftmapvrf.htm.
Dialer Watch Connect Delay
The Dialer Watch Connect Delay feature introduces the ability to configure a delay in bringing up a secondary link when a primary link that is monitored by Dialer Watch goes down and is removed from the routing table. Previously, the router would instantly dial a secondary route without allowing time for the primary route to come back up. When the Dialer Watch Connect Delay feature is configured, the router will check for availability of the primary link at the end of the specified delay time before dialing the secondary link.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdialwl.htm.
Diff-Serv-aware MPLS Traffic Engineering
MPLS traffic engineering allows constraint-based routing of IP traffic. One of the constraints satisfied by constant bit rate (CBR) is the availability of required bandwidth over a selected path. Diff-Serv-aware Traffic Engineering extends MPLS traffic engineering to enable you to perform constraint-based routing of "guaranteed" traffic, which satisfies a more restrictive bandwidth constraint than that satisfied by CBR for regular traffic. This ability to satisfy a more restrictive bandwidth constraint translates into an ability to achieve higher quality of service performance (in terms of delay, jitter, or loss) for the guaranteed traffic. Results include virtual leased lines and voice-trunking services.
This release adds support for label-controlled ATM (LC-ATM) interfaces. Previous releases supported Packet-over-SONNET (POS) and ATM permanent virtual circuit (PVC) interfaces.
Disabling V.110 Padding
In networks with devices such as terminal adapters (TAs) and global system for mobile communication (GSM) handsets that do not fully conform to the V.110 modem standard, you will need to disable V.110 padding. To disable the padded V.110 modem speed report required by the V.110 modem standard, use the no isdn v110 padding command in interface configuration mode.
DistributedDirector Boomerang Support
Boomerang is a Director Response Protocol (DRP) metric for DistributedDirector. The boomerang server provides a way to select a content server with the fastest response time from a group of redundant content servers. Instead of relying on static maps, boomerang dynamically recognizes problems such as congestion and link failures and avoids them. The content server with the fastest response time, as determined by the priority of the configured metrics, is determined to be the best site.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftddboom.htm
DistributedDirector Cache Auto Refresh
The DistributedDirector Cache Auto Refresh feature works in the background to continuously update all entries in the DistributedDirector cache. When this background refresh feature is initiated, DistributedDirector periodically updates all expired cache entries. The DistributedDirector cache saves the latest answers to all past Domain Name System (DNS) queries that were received since cache auto refresh was initiated, and any repeat request is served directly from the cache when caching is enabled.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftrefrsh.htm.
DistributedDirector Configurable Cache
DistributedDirector maintains an internal cache of entries, which is dynamically configurable. This internal configurable cache consists of sorting events that occur on a per-client basis. Users can configure both the variable size of this internal cache and the amount of time the DistributedDirector system will retain per-client sorting information.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftddcach.htm.
DistributedDirector MIB Support
The Cisco DistributedDirector MIB provides MIB support for DistributedDirector. This MIB contains DistributedDirector statistics, configurations, and status.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftddmib.htm.
Distributed LFI/dQoS over Leased Lines
Note
The Distributed LFI/dQoS over Leased Lines feature extends distributed link fragmentation (dLFI) and interleaving functionality on the VIP-enabled Cisco 7500 series routers to leased lines. Previously, Distributed Link Fragmentation and Interleaving was only available for Frame Relay and ATM.
Note
The dLFI feature supports the transport of real-time traffic, such as voice, and non-real-time traffic, such as data, on lower-speed Frame Relay and ATM virtual circuits (VCs) and on leased lines without causing excessive delay to the real-time traffic.
This feature is implemented using multilink PPP (MLP) over Frame Relay, ATM, and leased lines on VIP-enabled Cisco 7500 series routers. The feature enables delay-sensitive real-time packets and non-real-time packets to share the same link by fragmenting the large data packets into a sequence of smaller data packets (fragments). The fragments are then interleaved with the real-time packets. On the receiving side of the link, the fragments are reassembled and the packet reconstructed.
The dLFI feature is often useful in networks that send real-time traffic using Distributed Low Latency Queueing, such as voice, but have bandwidth problems that delay this real-time traffic due to the transport of large, less time-sensitive data packets. The dLFI feature can be used in these networks to disassemble the large data packets into multiple segments. The real-time traffic packets then can be sent between these segments of the data packets. In this scenario, the real-time traffic does not experience a lengthy delay waiting for the low-priority data packets to traverse the network. The data packets are reassembled at the receiving side of the link, so the data is delivered intact.
The ability to configure Quality of Service (QoS) using the Modular QoS CLI while also using distributed MLP (dMLP) is also introduced as part of the dLFI feature. The ability to configure QoS using the Modular QoS CLI while using dMLP was not supported prior to the introduction of the dLFI feature.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdlfi2.htm.
Distributed Multilink Point-to-Point Protocol
The Distributed Multilink Point-to-Point Protocol (dMLPPP) feature allows you to combine T1/E1 lines in a Versatile Interface Processor (VIP) on a Cisco 7500 series router into a bundle that has the combined bandwidth of multiple T1/E1 lines. This is done by using a VIP MLPPP link. You choose the number of bundles and the number of T1/E1 lines in each bundle. This allows you to increase the bandwidth of your network links beyond that of a single T1/E1 line without having to purchase a T3 line. Non-distributed MLPPP can only perform limited links, with CPU utilization quickly reaching 90% with only a few T1/E1 lines running MLPPP. With distributed MLP, you can increase the router's total capacity. DMLP supports bundling of fractional T1/E1 starting from DS0(64KBps) onwards.
Multiprotocol Label Switching (MPLS) and MPLS-VPN configurations are supported on DMLP bundle interfaces. As of Cisco IOS Release 12.2(8)T, Class-Based Weighted Fair Queueing (CBWFQ) and Low Latency Queueing (LLQ) are supported on DMLP.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120t/120t3/multippp.htm.
DNS Client AAAA Record Lookups over IPv6
The DNS Lookups over an IPv6 Transport feature adds support for IPv6 AAAA record types over an IPv6 transport in the Domain Name System (DNS) name-to-address and address-to-name lookup processes.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/index.htm.
DRP Agent—Boomerang Support
Boomerang is a Director Response Protocol (DRP) metric for DistributedDirector. When the boomerang metric is active, DistributedDirector instructs the DRP to send Domain Name Service (DNS) responses directly back to the querying client. The DNS response contains the addresses of the sites associated with the respective DRP agent. All involved DRPs send back their DNS responses at the same time. The packet of the DRP that is at shortest delay to the client will arrive first. The client may take the first answer and ignore subsequent ones, a standard behavior of all local DNS server implementations. The DRP agent allows configuration for full boomerang support. The boomerang client is the DRP agent.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdrpcli.htm.
Dual Tone Multifrequency (DTMF) Relay for SIP Calls Using Named Telephone Events
The Dual Tone Multifrequency (DTMF) Relay for SIP Calls Using Named Telephone Events (NTE) feature provides reliable digit relay between Cisco VoIP gateways when a low bandwidth codec is used. Using NTE to relay DTMF tones provides a standardized means of transporting DTMF tones in Real-Time Transport Protocol (RTP) packets. This feature also adds SIP phone support.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ft_dtmf.htm.
Easy VPN Server
The Easy VPN Server feature introduces server support for the Cisco VPN Client Release 3.x software clients and Cisco VPN hardware clients. It allows a remote end user to communicate using IP Security (IPSec) with any Cisco IOS Virtual Private Network (VPN) gateway. Centrally managed IPSec policies are "pushed" to the client by the server, minimizing configuration by the end user.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftunity.htm.
Enabling Fax Rate on POTS to POTS Fax Calls
This command line interface (CLI) change was made to enable a fax relay between two plain old telephone service (POTS) dial peers to cover the case in which a fax call fails if it is made without DSP (digital signal processor) involvement.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftfxpots.htm.
Encrypted Vendor-Specific Attributes
The Encrypted Vendor-Specific Attributes feature introduces support for the following three types of string vendor-specific attributes (VSAs):
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftencvsa.htm.
Enhanced Billing Support for SIP Gateways
The Enhanced Billing Support for SIP Gateways feature describes the changes to authentication, authorization, and accounting (AAA) records and the RADIUS implementations on Cisco Session Initiation Protocol (SIP) gateways. These changes were introduced to provide customers and partners with the ability to effectively bill for traffic transported over SIP networks.
Username Attribute
The username attribute is included in all AAA records and is the primary means for the billing system to identify an end user. The password attribute is included in authentication and authorization messages of inbound Voice over IP (VoIP) call legs.
For most implementations, the SIP gateway populates the username attribute in the SIP INVITE request with the calling number from the FROM: header and the password attribute with null or with data from an Interactive Voice Response (IVR) script. If a Proxy-Authorization header exists, it is ignored. A new Cisco IOS command, aaa username, determines the information with which to populate the username attribute.
Within the Microsoft Passport authentication service that authenticates and identifies users, the passport user ID (PUID) is used. The PUID and a password are passed from a Microsoft network to the Internet telephony service provider (ITSP) network in the Proxy-Authorization header of a SIP INVITE request as a single, base-64 encoded string. For example,
Proxy-Authorization: basic MDAwMzAwMDA4MDM5MzJlNjouThe new Cisco IOS aaa username command enables parsing of the Proxy-Authorization header; decoding of the PUID and password; and populating the PUID into the username attribute and the decoded password into the password attribute. The decoded password is generally a "." because a Microsoft Network (MSN) authenticates users prior to this point. For example,
Username = "123456789012345"
Password = "Z\335\304\326KU\037\301\261\326GS\255\242\002\202"The password in the example above is an encrypted "." and is the same for all users.
SIP Call ID
From the Call ID header of the SIP INVITE request, the SIP Call ID is extracted and populated in a Cisco vendor-specific attribute (VSA) as a new attribute-value pair call-id=string. The attribute-value pair can be used to correlate RADIUS records from Cisco Session Initiation Protocol (SIP) gateways with RADIUS records from other SIP network elements, for example, proxies. For complete information on this attribute-value pair, refer to the RADIUS Vendor-Specific Attributes Voice Implementation Guide.
Session Protocol
Session Protocol is another new attribute-value pair that indicates if the call is using Session Initiation Protocol (SIP) or H.323 as the signaling protocol. For complete information on this attribute-value pair, refer to the RADIUS Vendor-Specific Attributes Voice Implementation Guide.
Silent Authentication Script
As part of the Enhanced Billing Support for SIP Gateways feature, a new Tool Command Language (TCL) Interactive Voice Response (IVR) API 2.0 Silent Authorization script has been developed. The Silent Authorization script allows users to be authorized without having to separately enter a username or password into the system. The script automatically extracts the passport user ID (PUID) and password from the SIP INVITE request and then authenticates that information through RADIUS authentication and authorization records. The script is referred to as silent because neither the caller nor the called party hears any prompts.
Further Documentation
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmsnbil.htm.
Enhanced Password Security
The Enhanced Password Security feature allows you to configure Message Digest 5 (MD5) encryption for username passwords. Before the introduction of this feature, there were two types of passwords associated with usernames: Type 0, which is a clear text password visible to any user who has access to privileged mode on the router, and type 7, which is a password with a weak, exclusive, or type encryption. Type 7 passwords can be retrieved from the encrypted text by using publicly available tools.
Use the username secret command to configure a username and an associated MD5-encrypted secret.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_md5.htm.
Error Log Count Enhancement
The Cisco IOS logging facility allows you to save error messages locally or to a remote host. When these error messages exceed the capacity of the local buffer dedicated to storing them, the oldest messages are removed. To provide you with more information about messages that have occurred and may have been removed from the local buffer, an error log counter tabulates the occurrences of each error message and time-stamps the most recent occurrence.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/fterrlog.htm.
Event Tracer
The Event Tracer feature provides a binary trace facility for troubleshooting Cisco IOS software. This feature gives Cisco service representatives additional insight into the operation of the Cisco IOS software and can be useful in helping to diagnose problems in the unlikely event of an operating system malfunction or, in the case of redundant systems, route processor switchover.
This feature was originally introduced in Cisco IOS Release 12.0(18)S.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s18/
evnttrcr.htm.Fax Detection for Cisco 2600 Series and Cisco 3600 Series Routers
Note
On Cisco 2600 series and Cisco 3600 series routers equipped with digital and analog voice network modules, the fax detection feature enables service providers to deploy unified communications, in which each subscriber has a single E.164 number for both voice and fax by providing the capability to detect automatically whether an incoming call is voice or fax. Supported network modules are NM-HDV with voice interface cards (VIC)/voice WAN interface cards (VWIC) for digital T1connections and Voice 2V with VIC FXS for analog connections. VWIC and VIC FXS are the voice interface cards within the network modules. When configured for fax detection, the gateway automatically listens to incoming calls to discriminate between voice and fax. The gateway then routes the calls to the appropriate application or server.
Note
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/pull2snf.htm.
Firewall Feature Set
The Cisco IOS Firewall feature set provides firewall-specific security features to the Cisco CVA122 Cable Voice Adapter. When this feature is enabled, the router acts as a buffer between the Internet and other public networks and the private network that is connected to the router. Security is provided by access lists, as well as by examining incoming traffic for suspicious activity.
The firewall-specific security features include the following:
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Gatekeeper Transaction Message Protocol Interface Resiliency Enhancement
Gatekeeper Transaction Message Protocol (GKTMP) is used between the Cisco IOS Gatekeeper and a server to provide enhanced call routing and address translation services. The GKTMP Interface Resiliency Enhancement feature adds additional parameters in the disengage request (REQUEST DRQ) message sent from the gatekeeper (GK) to the server. It also provides new request alive (REQUEST ALV) and response alive (RESPONSE ALV) messages between the gatekeeper and server, server failure detection, and a flow control command.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/
122xb_2/ftgkire.htm.Generic Routing Encapsulation (GRE) Tunnel Keepalive
The GRE Tunnel Keepalive feature provides the capability of configuring keepalive packets to be sent over IP-encapsulated generic routing encapsulation (GRE) tunnels. You can specify the rate at which keepalives will be sent and the number of times that a device will continue to send keepalive packets without a response before the interface becomes inactive.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/grekpliv.htm.
GKTMP Security Token Enhancement
The GKTMP Security Token Enhancement feature provides support for ClearTokens in messages between a NetSpeak route server and the Cisco IOS gatekeeper. The Request ARQ, Response ARQ, Response ACF, Request LRQ, Response LRQ, Request LCF, and Response LCF messages between the Cisco IOS gatekeeper and the route server now include ClearTokens. In addition, the Response ARQ messages include both gateways in a local domain or zone and remote zone gatekeepers and allow prioritization of the resulting sets of gateways. The Response LRQ messages support a combination of endpoint addresses and a list of remote zone gatekeepers to which to forward the LRQ message.
G.SHDSL Symmetric DSL Support
G.SHDSL is a new multirate symmetric high-speed digital subscriber line (DSL) technology for the local loop that connects customer premises equipment (CPE) to the central office (CO) in the access network. This access technology for business applications is important because of its symmetric and multirate functionality. G.SHDSL refers to the approved standard officially designated in International Telecommunication Union-Telecommunications Standards Section (ITU-T) G.991.2.
IGMP Version 3—Explicit Tracking of Hosts, Groups, and Channels
The Internet Group Management Protocol (IGMP) is used by IP hosts to report their multicast group memberships to neighboring multicast routers. IGMP is available in Versions 1, 2, and 3. IGMP Version 3 (IGMPv3) is supported in Cisco IOS Release 12.0(15)S, 12.1(5)T, 12.1(8)E, and later releases.
The IGMP Version 3—Explicit Tracking of Hosts, Groups, and Channels feature enables a multicast router to explicitly track the membership of all multicast hosts in a particular multiaccess network. This enhancement to the Cisco IOS implementation of IGMPv3 enables the router to keep track of each individual host that is joined to a particular group or channel. The main benefits are that this feature provides minimal leave latencies, faster channel changing, and improved diagnostics capabilities for IGMP.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_xtrk.htm.
IKE—Initiate Aggressive Mode
The IKE—Initiate Aggressive Mode feature allows you to specify RADIUS Tunnel attributes (Tunnel-Client-Endpoint [66] and Tunnel-Password [69]) for an IPSec peer and to initiate an IKE aggressive mode negotiation with the tunnel attributes. This feature is best implemented in a crypto hub-and-spoke scenario, in which the spokes initiate IKE aggressive mode negotiation with the hub by using the preshared keys that are specified as tunnel attributes and stored on the AAA server. This scenario is scalable because the preshared keys are kept at a central repository (the AAA server) and more than one hub router and one application can use the information.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ikeag.htm.
Integrated IS-IS Point-to-Point Adjacency Over Broadcast Media
When a network consists of only two networking devices that are connected to broadcast media and using the integrated IS-IS protocol, it is better for the system not to have to handle the link as a broadcast link but rather as a point-to-point link. The Integrated IS-IS Point-to-Point Adjacency Over Broadcast Media feature introduces a new command to make IS-IS behave as a point-to-point link between the networking devices.
Refer to the following document for additional information:
http://www.cisco.com/en/US/docs/ios/12_2t/12_2t8/feature/guide/ftissp2p.html
Integrated IS-IS Support for IPv6
IPv6 supports Interior Gateway Protocols (IGPs) and Exterior Gateway Protocols (EGPs). Routing Information Protocol (RIP) and Integrated Intermediate System-to-Intermediate System (IS-IS) protocols are the supported IGPs for IPv6. Multiprotocol Border Gateway Protocol (BGP) is the supported EGP for IPv6.
IS-IS in IPv6 functions the same as and offers many of the same benefits as IS-IS in IPv4. IPv6 enhancements to IS-IS allow IS-IS to advertise IPv6 prefixes in addition to IPv4 and Open System Interconnection (OSI) routes. Extensions to the IS-IS CLI allow configuration of IPv6-specific parameters. IS-IS in IPv6 extends the address families supported by IS-IS to include IPv6, in addition to OSI and IPv4.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/ftipv6s.htm.
Interactive Voice Response Version 2.0 on VoIP Gateways
Interactive Voice Response (IVR) consists of simple voice prompting and digit collection to gather caller information for authenticating the user and identifying the destination. IVR applications can be assigned to specific ports or invoked on the basis of dialed number identification service (DNIS). An IP Public Switched Telephone Network (PSTN) gateway can have several IVR applications to accommodate many different gateway services, and you can customize the IVR applications to present different interfaces to the various callers.
IVR systems provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words, or more commonly, dual tone multifrequency (DTMF) signaling. IVR uses Tool Command Language (TCL) scripts to gather information and to process accounting and billing.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ivr72.htm.
IP-FORWARDING-TABLE-MIB
This release introduces support for the new IP-FORWARD-MIB (IP Forwarding Table MIB). The current version of the IP Forwarding Table MIB is defined in RFC 2096. (RFC 2096 replaces RFC 1354.) The Cisco implementation of this MIB does not support the ipCiderRouteNextHopAS object. Additionally, all entries for the ipCidrRouteTos object (the IP Type-of-Service field) remain set to zero, which indicates a default TOS policy.
For details, refer to the IP-FORWARD-MIB.my file, available through the Cisco MIB FTP site at the following URL:
http://tools.cisco.com/Support/SNMP/do/SearchOID.do?local=en&step=1
IP Multicast MIB Enhancements
The IP Multicast MIB Enhancements feature enhances the IP multicast routing protocol in Cisco IOS software by adding MIB variables to query the number of (S, G) and (*, G) entries. It also adds support for high-speed interface counters.
IPSec VPN High Availability Enhancements
The IPSec VPN High Availability feature consists of two new features—Reverse Route Injection and Hot Standby Router Protocol and IPSec—that work together to provide users with a simplified network design for VPNs and reduced configuration complexity on remote peers with respect to defining gateway lists.
Reverse Route Injection
Reverse Route Injection (RRI) is a feature designed to simplify network design for Virtual Private Network (VPNs) in which there is a requirement for redundancy or load balancing. RRI works with both dynamic and static crypto maps.
In the dynamic case, as remote peers establish IPSec security associations (SAs) with an RRI-enabled router, a static route is created for each subnet or host protected by that remote peer. For static crypto maps, a static route is created for each destination of an extended access-list rule.
When routes are created, they are injected into any dynamic routing protocol and distributed to surrounding devices. This traffic flows, requiring IPSec to be directed to the appropriate RRI router for transport across the correct SAs to avoid IPSec policy mismatches and possible packet loss.
Hot Standby Router Protocol and IPSec
Hot Standby Router Protocol (HSRP) is designed to provide high network availability by routing IP traffic from hosts on Ethernet networks without relying on the availability of any single router. HSRP is particularly useful for hosts that do not support a router discovery protocol, such as ICMP Router Discovery Protocol (IRDP), and do not have the functionality to switch to a new router when their selected router reloads or loses power. Without this functionality, a router that loses its default gateway because of a router failure is unable to communicate with the network.
HSRP is configurable on LAN interfaces using standby command line interface (CLI) commands. It is now possible to use the standby IP address from an interface as the local IPSec identity, or local tunnel endpoint.
By using the standby IP address as the tunnel endpoint, failover can be applied to VPN routers by using HSRP. Remote VPN gateways connect to the local VPN router via the standby address that belongs to the active device in the HSRP group. In the event of failover, the standby device takes over ownership of the standby IP address and begins to service remote VPN gateways.
Further Documentation
Refer to the following document for further information about the IPSec VPN High Availability Enhancements feature:
http://www.cisco.com/en/US/docs/ios/12_1/12_1e9/feature/guide/ft_ipsha.html.
Large-Scale Dial-Out (LSDO) VRF Aware
The Large-Scale Dial-Out (LSDO) VRF Aware feature allows LSDO to support the Layer 2 Tunnel Protocol (L2TP) in an Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN). The basic operation of LSDO relies on per-user static routes stored in an authentication, authorization, and accounting (AAA) server and redistributed static routes and redistributed connected routes to put better routes that point to the same remote network or host on the alternate network access server (NAS). When using LSDO, overlapping IP addresses are often present in virtual routing and forwarding instances (VRFs), so that a unique key is needed to retrieve the correct route from the AAA server. With virtual private dial network (VPDN) as a dial-out resource, a virtual access interface is created for maintaining each PPP session. Software before Cisco IOS Release 12.2(8)T did not update the VRF information on the virtual access interface; rather, this information was cloned from the dialer interface. Now, the VRF table identifier is retrieved from the incoming packet and is mapped to the VRF name. This VRF name and the destination IP address are combined to make the unique key needed to retrieve the dial string and other user profile information from the AAA server.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftlsdvpn.htm.
Media Gateway Control Protocol Based Fax (T.38) and Dual Tone Multifrequency (IETF RFC 2833) Relay
The MGCP-Based Fax (T.38) and DTMF (IETF RFC 2833) Relay feature adds support for fax relay and DTMF relay with MGCP. This feature provides two modes of implementation for each component: gateway (GW) controlled mode and call agent (CA) controlled mode. In GW controlled mode, GWs negotiate DTMF and fax relay transmission by exchanging capability information in Session Definition Protocol (SDP) messages. That transmission is transparent to the CA. GW-controlled mode allows use of the MGCP-Based Fax (T.38) and DTMF (IETF RFC 2833) Relay feature without upgrading the CA software to support the feature. In CA-controlled mode, CAs use MGCP messaging to instruct GWs to process fax and DTMF traffic. For MGCP T.38 Fax Relay, the CAs can also instruct GWs to revert to GW-controlled mode if the CA is unable to handle the fax control messaging traffic; for example, in overloaded or congested networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmgcpfx.htm.
MGCP VoIP Call Admission Control
MGCP CAC determines if calls can be accepted on the IP network on the basis of available network resources. Before this release, MGCP Voice over IP (VoIP) calls were established regardless of the available resources on the gateway or network. The gateway had no mechanism for gracefully refusing calls if resources were not available to process the call. New calls would fail with unexpected behavior and in-progress calls would experience quality-related problems.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_04mac.htm.
MPLS Label Distribution Protocol (LDP)
Cisco MPLS label distribution protocol (LDP) allows the construction of highly scalable and flexible IP Virtual Private Networks (VPNs) that support multiple levels of services.
LDP provides a standard methodology for hop-by-hop distribution of labels in an Multiprotocol Label Switching (MPLS) network by assigning labels to routes that have been chosen by the underlying Interior Gateway Protocol (IGP) routing protocols. The resulting label switch paths (LSPs) forward label traffic across an MPLS backbone to particular destinations. These capabilities enable service providers to implement Cisco MPLS-based IP VPNs and IP+ATM services across multivendor MPLS networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ldp7t.htm.
MPLS Over ATM: Virtual Circuit (VC) Merge
VC merge maps several incoming labels to one single outgoing label. Cells from different virtual channel identifiers (VCIs) that travel to the same destination are transmitted to the same outgoing VC using multipoint-to-point connections.
VC merge allows the switch to transmit cells that come from different VCIs over the same outgoing VCI to the same destination. In other words, VC merge queues ATM Adaptation Layer 5 (AAL5) frames in input buffers until the switch receives the last frame. Then the switch transmits the cells from that AAL5 frame before it sends any cells from other frames. VC merge requires the switch to provide buffering, but no more buffering than is required in IP networks. VC merge slightly delays the transfer of frames; however, VC merge is for IP traffic and not for traffic that requires speed. IP traffic tolerates delays better than other traffic on the ATM network.
MPLS Traffic Engineering (TE) MIB
Simple Network Management Protocol (SNMP) agent code operating in conjunction with the MPLS Traffic Engineering MIB (MPLS TE MIB) enables a standardized, SNMP-based approach to be used in managing the Multiprotocol Label Switching (MPLS) Traffic Engineering (TE) features in Cisco IOS software.
The MPLS TE MIB is based on the Internet Engineering Task Force (IETF) draft MIB entitled draft-ietf-mpls-te-mib-05.txt, which includes objects describing features that support MPLS traffic engineering. This IETF draft MIB, which undergoes revisions from time to time, is being evolved toward becoming a standard. Accordingly, Cisco's implementation of the MPLS TE MIB is expected to track the evolution of the IETF draft MIB.
The SNMP objects defined in the MPLS TE MIB can be viewed by any standard SNMP utility. All MPLS TE MIB objects are based on the IETF draft MIB; thus, no specific Cisco SNMP application is required to support the functions and operations pertaining to the MPLS TE MIB.
The MPLS TE MIB provides the following benefits:
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Refer to the following document for additional information.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/temib28t.htm.
MPLS VPN Carrier Supporting Carrier
The carrier supporting carrier feature enables one MPLS VPN-based service provider to allow other service providers to use a segment of its backbone network. The service provider that provides the segment of the backbone network to the other provider is called the backbone carrier. The service provider that uses the segment of the backbone network is called the customer carrier.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftcsc8.htm.
Note
MPLS VPN ID
Using Multiprotocol Label Switching (MPLS) VPN ID you can identify virtual private networks (VPNs) by a VPN identification number, as described in RFC 2685. This implementation of the MPLS VPN ID feature is used for identifying a VPN. The MPLS VPN ID feature is not used to control the distribution of routing information or to associate IP addresses with MPLS VPN ID numbers in routing updates.
Multiple VPNs can be configured in a router. You can use a VPN name (a unique ASCII string) to reference a specific VPN configured in the router. Alternately, you can use a VPN ID to identify a particular VPN in the router. The VPN ID follows a standard specification (RFC 2685). To ensure that the VPN has a consistent VPN ID, assign the same VPN ID to all the routers in the service provider network that services that VPN.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftvpnid.htm.
Multilink Frame Relay
The Multilink Frame Relay feature introduces functionality based on the Frame Relay Forum Multilink Frame Relay UNI/NNI Implementation Agreement (FRF.16). This feature provides a cost-effective way to increase bandwidth for particular applications by enabling multiple serial links to be aggregated into a single bundle of bandwidth. Multilink Frame Relay is supported on User-to-Network Interfaces (UNIs) and Network-to-Network Interfaces (NNIs) in Frame Relay networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_mfr.htm.
Multiple RSA Key Pair Support
The Multiple RSA Key Pair Support feature allows the Cisco IOS software to maintain a distinct key pair for each certification authority (CA) with which it is dealing. Thus, the Cisco IOS software can match policy requirements for each CA without compromising the requirements specified by the other CAs, such as key length, key lifetime, and general-purpose versus special-usage keys.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftmltkey.htm.
Multiprotocol BGP (MP-BGP) Support for CLNS
The Multiprotocol BGP Support for CLNS feature allows Border Gateway Protocol (BGP) to be used as an interdomain routing protocol in networks that use Connectionless Network Service (CLNS) as the network-layer protocol. This feature was developed to solve a scaling issue with a data communications network (DCN) where large numbers of routers are managed remotely. The benefits of using Multiprotocol BGP (MP-BGP) to support CLNS are not confined to DCN networks and can be implemented to help scale any network using Open System Interconnection (OSI) routing protocols with CLNS.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/
fm_bgpmc.htm.NAT Support for SIP
The Session Initiation Protocol (SIP) is an application layer signaling protocol used for creating and controlling multimedia sessions with two or more participants. SIP is transported over TCP or UDP. The messages used in the protocol may have IP addresses embedded in the packet payload. If a message passes through a router configured with Network Address Translation (NAT), the embedded information must be translated and encoded back to the packet. An Application Layer Gateway (ALG) is used with NAT to enable SIP.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftnatsip.htm.
Network-Based Application Recognition RTP Payload Type Classification
The RTP Payload Type Classification enhancement has been added to the Network-Based Application Recognition (NBAR) feature. With the addition of NBAR RTP Payload Type Classification, RTP traffic can now be classified as a protocol within the Modular QoS CLI framework.
For additional information on the NBAR feature, including NBAR RTP Payload Type Classification, refer to the following document:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/dtnbarad.htm.
Nonstop Forwarding Enhanced FIB Refresh
Operational networks must minimize traffic disruption and offer the most uptime possible. The Nonstop Forwarding Enhanced FIB Refresh feature provides users the capability to continue forwarding IP traffic while Cisco Express Forwarding (CEF) database tables are being rebuilt. IP forwarding on the router is therefore uninterrupted.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftfibeps.htm.
OSPF Sham-Link Support for MPLS VPN
A sham link is a logical path within an Open Shortest Path First (OSPF) area; it represents an unnumbered point-to-point connection between two provider edge (PE) devices. All routers within the area see the link and use it during the shortest path first (SPF) computation.
On PE routers the VPN Route Forwarding (VRF) routing table is populated by OSPF routes over the sham link. The sham link gives users the capability of specifying which path will be used for traffic.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ospfshmk.htm
PIM Multicast Scalability
This feature enhances the Protocol Independent Multicast (PIM) protocol in Cisco IOS software by adding a new level of scalability. With this feature, edge devices can have a large number of multicast groups and users without increasing the CPU utilization of the router.
Plain NFAS Support on NM-HDV
The current Non-Facility Associated Signaling (NFAS) support on the High-Density Voice network modules (NM-HDV) is joined with the Redundant Link Manager/Signaling System 7 (RLM/SS7). When a user configures an ISDN PRI NFAS group via the Cisco command line interface (CLI), all channels within the PRI are treated as B channels. A D channel is not created and, thus no signaling will be passed to the ISDN stack.
This feature modifies the existing implementation of NFAS/RLM on NM-HDV to activate the generic NFAS feature on Cisco 2600 and 3600 routers and to allow the coexistence of plain NFAS and NFAS/RLM/SS7 on the Cisco 3660 router.
Policer Enhancement—Multiple Actions
This feature further extends the functionality of the Cisco IOS Traffic Policing feature (a single-rate policer) and the Two-Rate Policer feature. The Traffic Policing and Two-Rate Policer features are traffic policing mechanisms that allow you to control the maximum rate of traffic sent or received on an interface. Both of these traffic policing mechanisms mark packets as conforming to, exceeding, or violating a specified rate. After a packet is marked, you can specify an action to be taken on the packet on the basis of that marking.
With both the Traffic Policing feature and the Two-Rate Policer feature, you can specify only one conform action, one exceed action, and one violate action. Now with the new Policer Enhancement—Multiple Actions feature, you can specify multiple conform, exceed, and violate actions for the marked packets. You specify the multiple actions by using the action argument of the police command.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftpolenh.htm.
PPPoE MTU Adjustment
The syntax of the ip adjust-mss command has changed to the following:
ip tcp adjust-mss mss
where the value of the mss argument must be 1452 or less to fix the Point-to-Point Protocol over Ethernet (PPPoE) maximum transmission unit (MTU) problem.
PPPoE Session-Count MIB
The PPPoE Session-Count MIB provides the ability to use Simple Network Management Protocol (SNMP) to monitor in real time the number of PPP over Ethernet sessions configured on permanent virtual circuits (PVCs) and on a router.
This new MIB also introduces two SNMP traps that generate notification messages when a PPPoE session-count threshold is reached on any PVC or on the router.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftpscmib.htm.
Radius Packet of Disconnect
This feature consists of a method for terminating a call that has already been connected. The "Packet of Disconnect" (POD) is a RADIUS access_request packet and is intended to be used in situations in which the authenticating agent server wants to disconnect the user after the session has been accepted by the RADIUS access_accept packet. This may be needed in at least two situations:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_pod1.htm.
RADIUS Route Download
The RADIUS Route Download feature allows users to configure their network access server (NAS) to send static route download requests to authentication, authorization, and accounting (AAA) servers specified by a named method list. Before this feature, all RADIUS authorization requests for static route download could be sent only to AAA servers specified by the default method list.
This feature extends the functionality of the aaa route download command to allow users to specify the name of the method list that will be used to direct static route download requests to the AAA servers. The aaa route download command must be used to add separate method lists; however, users will continue to enable the aaa authorization configuration default command to download static route configuration information from the AAA server specified by the default method list.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftradrou.htm.
Rotating Through Dial Strings
The Rotating Through Dial Strings feature allows you to specify the dialing order when multiple dial strings are configured. Options for dialing order include:
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This feature takes advantage of information available from a previous call attempt, such as whether the call was unsuccessful or the line was busy, and thereby increases the rate of successful calls.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftrotdls.htm.
Secure Shell (SSH) Support over IPv6
Secure Shell (SSH) in IPv6 functions the same and offers the same benefits as SSH in IPv4—the SSH Server feature enables an SSH client to make a secure, encrypted connection to a Cisco router, and the SSH Client feature enables a Cisco router to make a secure, encrypted connection to another Cisco router or to any other device that is running an SSH server. IPv6 enhancements to SSH consist of support for IPv6 addresses that enable a Cisco router to accept and establish secure, encrypted connections with remote IPv6 nodes over an IPv6 transport.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/index.htm.
Secure Shell (SSH) Version 1 Server Support
Secure Shell (SSH) is an application and a protocol that provide a secure replacement to the Berkeley r-tools. The protocol secures the sessions using standard cryptographic mechanisms, and the application can be used much like the Berkeley rexec and rsh tools are used. There are currently two versions of SSH available: SSH Version 1 and SSH Version 2. Only SSH Version 1 is implemented in Cisco IOS software. For more information on this feature, refer to the "Configuring Secure Shell" chapter of the Cisco IOS Security Configuration Guide, Release 12.2.
This feature was originally introduced in Cisco IOS Release 12.1(1)T. This release adds support for the Cisco 826, Cisco 827, and Cisco 827-4V platforms.
Service Selection Gateway
Service Selection Gateway (SSG) is a switching solution for service providers who offer intranet, extranet, and Internet connections to subscribers using broadband access technology such as digital subscriber lines, cable modems, or wireless to allow simultaneous access to network services.
SSG works in conjunction with the Cisco Service Selection Dashboard (SSD) or its successor product, the Cisco Subscriber Edge Services Manager (SESM).
Together with the SSD or SESM, SSG provides subscriber authentication, service selection, and service connection capabilities to subscribers of Internet services.
Subscribers interact with an SSD or SESM web application using a standard Internet browser.
SSG communicates with the authentication, authorization, and accounting (AAA) management network where RADIUS, Dynamic Host Configuration Protocol (DHCP), and Simple Network Management Protocol (SNMP) servers reside and with the Internet service provider (ISP) network, which may connect to the Internet, corporate networks, and value-added services.
A licensed version of SSG works with SESM to present to subscribers a menu of network services that can be selected from a single graphical user interface (GUI). This functionality improves flexibility and convenience for subscribers and enables service providers to bill subscribers for connect time and services used, rather than charging a flat rate.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_4/122b4_sg/ft_ssg.htm.
Session Initiation Protocol (SIP) for VoIP
Voice over IP (VoIP) currently implements the ITU H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. Session Initiation Protocol (SIP) is a protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. The Cisco SIP functionality equips Cisco routers to signal the setup of voice and multimedia calls over IP networks. SIP provides an alternative to H.323 within the VoIP internetworking software.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_sip72.htm.
Simple Network-Enabled Auto-Provisioning for Cisco IAD2420 Series IADs
The Simple Network-Enabled Auto-Provisioning (SNAP) feature on the Cisco IAD2420 series IAD allows service providers to rapidly deploy and configure services to the Cisco IAD platforms at customer premises without requiring configuration of the IADs at the customer site and with little or no onsite technician intervention. SNAP is part of the Cisco Networking Services (CNS) technology, which allows network products to be installed and automates many of the configuration tasks. SNAP consists of two basic functions: learning and setting the IP address and downloading the configuration for the IAD.
Each Cisco IAD2420 series IAD using SNAP includes a CNS Configuration Agent and CNS Event Agent that communicates with the CNS Configuration Registrar to enable the configuration of the IAD.
Using SNAP in conjunction with a Cisco aggregation router, a CNS Configuration Registrar, and an optional Domain Name System (DNS) server, SNAP performs the IAD configuration on the CNS Configuration Registrar and downloads the configuration to the IAD at the customer premises.
SIP Gateway Support for the Bind Command
In previous releases of Cisco IOS software, the source address of a packet going out of the gateway was never deterministic. That is, the session protocols and Voice over IP (VoIP) layers always depended on the IP layer to give the best local address. The best local address was then used as the source address (the address showing where the SIP request came from) for signaling and media packets. Using this nondeterministic address occasionally caused confusion for firewall applications, because a firewall could not be configured with an exact address and would take action on several different source address packets.
However, the bind interface command allows you to configure the source IP address of signaling and media packets to a specific interface's IP address. Thus, the address that goes out on the packet is bound to the IP address of the interface specified with the bind command. Packets that are not destined to the bound address are discarded.
When you do not want to specify a bind address, or if the interface is down, the IP layer still provides the best local address.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftbind.htm.
SIP Gateway Support of RSVP and TEL URL
The SIP Gateway Support of RSVP and TEL URL feature also supports Telephone Uniform Resource Locators or TEL URL. Currently SIP gateways support URLs in the SIP format. SIP URLs are used in SIP messages to indicate the originator, recipient, and destination of the SIP request. However, SIP gateways may also encounter URLs in other formats, such as TEL URLs. TEL URLs describe voice call connections. They also enable the gateway to accept TEL calls sent through the Internet and to generate TEL URLs in the request line of outgoing INVITE requests.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/vvfresrv.htm.
SIP Intra-gateway Hairpinning
SIP hairpinning is a call routing capability in which an incoming call on a specific gateway is signaled through the IP network and back out the same gateway. This call can be a public switched telephone network (PSTN) call routed into the IP network and back out to the PSTN over the same gateway.
Similarly, SIP hairpinning can be a call signaled from a line (for example, a telephone line) to the IP network and back out to a line on the same access gateway. With SIP hairpinning, unique gateways for ingress and egress are no longer necessary.
SIP INVITE Request with Malformed Via Header
A SIP INVITE requests that a user or service participate in a session. Each INVITE contains a Via header that indicates the transport path taken by the request so far and where to send a response.
In the past, when an INVITE contained a malformed Via header, the gateway would print a debug message and discard the INVITE without incrementing a counter. However, the printed debug message was often inadequate, and it was difficult to detect that messages were being discarded.
The SIP INVITE Request with Malformed Via Header feature provides a response to the malformed request. A counter, Client Error: Bad Request, increments when a response is sent for a malformed Via field. Bad Request is a class 400 response and includes the explanation Malformed Via Field. The response is sent to the source IP address (the IP address where the SIP request originated) at User Datagram Protocol (UDP) port 5060.
This feature applies to messages arriving on UDP, because the Via header is not used to respond to messages arriving on TCP.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftmalvia.htm.
SIP T.37 Store and Forward Fax
SIP T.37 is an ITU specification that enables store-and-forward fax applications, as well as toggling from voice to fax, for example, providing an Interactive Voice Response (IVR) front end to a store-and-forward fax application.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/index.htm
SIP T.38 Fax Relay
The SIP T.38 Fax Relay feature adds standards-based fax support to Session Initiation Protocol (SIP) and conforms to ITU-T T.38, Procedures for Real-Time Group 3 Facsimile Communication over IP Networks. The ITU-T standard specifies real-time transmission of faxes between two regular fax terminals over an IP network.
The SIP T.38 Fax Relay feature also includes the following functionality:
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UDPTL, as defined in ITU-T T.38, is a transport layer that is used on top of UDP. UDPTL makes the delivery of packets more reliable by providing data redundancy.
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SIP T.38 Fax Relay supports QoS when establishing T.38 sessions. If the dial peer is already configured for QoS, the T.38 stream maintains the QoS support. QoS ensures certain bandwidth reservations for calls.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftsipfax.htm.
SIP—Call Transfer Using Refer Method
Note
The Refer method provides call transfer capabilities to supplement the Bye and Also methods already implemented on Cisco IOS Session Initiation Protocol (SIP) gateways.
Call transfer allows a wide variety of decentralized multiparty call operations. These decentralized call operations form the basis for third-party call control and thus are important features for Voice over IP (VoIP) and SIP. Call transfer is also critical for conference calling, where calls can transition smoothly between multiple point-to-point links and IP level multicasting.
The following are components of call transfer:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftrefer.htm.
SIP—DNS SRV RFC2782 Compliance
Session Initiation Protocol (SIP) on Cisco Voice over IP (VoIP) gateways uses Domain Name System Server (DNS SRV) query to determine the IP address of the user endpoint. The query string has a prefix in the form of "protocol.transport." and is attached to the fully qualified domain name (FQDN) of the next hop SIP server. This prefix style, from RFC 2052, has always been available; however, with this release, a second style is also available. The second style complies with RFC 2782 and prepends the protocol label with an underscore "_"; as in "_protocol._transport." The addition of the underscore reduces the risk of the same name being used for unrelated purposes. The form compliant with RFC 2782 is the default style. Use the srv version command to configure the DNS SRV feature.
SNMP IF-MIB Support for VLAN (ISL, 802.1Q) Subinterfaces
This feature updates the Cisco implementation of the Interfaces Group MIB (abbreviated "IF-MIB" and defined in RFC 2233) to completely support ifTable and ifXTable entries for Inter-Switch Link (ISL) or 802.1Q encapsulated subinterfaces.
The Interface Table (the ifTable object) contains information on an Simple Network Management Protocol (SNMP) management entity's interfaces. Each sublayer of a network interface is considered to be an interface. An ifTable is a list of interface entries in which each entry contains management information applicable to that interface. The ifXTable is an extension to the ifTable. It contains replacements for objects of the ifTable that were deprecated. The ifXTable also contains 64-bit versions of the counters defined in the ifTable. Cisco IOS software can support both interfaces and subinterfaces in the ifTable.
ISL is a Cisco protocol for interconnecting switches and maintaining VLAN information as traffic is exchanged between switches. It can also be used to configure routing between any number of VLANs in a network by creating subinterfaces for each VLAN.
802.1Q (also referred to as "DOT1Q") is an IEEE standard protocol for interconnecting bridges/switches and maintaining VLAN information as traffic is exchanged between the devices. 802.1Q can also be used to configure routing between any number of VLANs in a network by creating subinterfaces for each VLAN.
The following objects of the ifTable have been updated: ifIndex, ifDescr, ifType, ifMtu, ifSpeed, ifPhysAddress, ifInOctets, ifInUcastPkts, ifInNUcastPkts, ifOutOctets, ifOutUcastPkts, ifOutNUcastPkts.
The following objects of the ifXTable have been updated: ifName, ifInMulticastPkts, ifInBroadcastPkts, ifOutMulticastPkts, ifOutBroadcastPkts, ifHCInOctets, ifHCInUcastPkts, ifHCInMulticastPkts, ifHCInBroadcastPkts, ifHCOutOctets, ifHCOutUcastPkts, ifHCOutMulticastPkts, ifHCOutBroadcastPkts.
Static Cache Entry for IPv6 Neighbor Discovery
The Static Cache Entry for IPv6 Neighbor Discovery feature enables the configuring of static entries in the IPv6 neighbor discovery cache, which provides functionality in IPv6 that is equivalent to static Address Resolution Protocol (ARP) entries in IPv4. Static entries in the IPv6 neighbor discovery cache are not modified by the neighbor discovery process. Cisco IOS software uses static ARP entries in IPv4 to translate 32-bit IP addresses into 48-bit hardware addresses. In IPv6, Cisco IOS software uses static entries in the IPv6 neighbor discovery cache to translate 128-bit IPv6 addresses into 48-bit hardware addresses.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/ftipv6s.htm.
Stream Control Transmission Protocol (SCTP) Release 2
Stream Control Transmission Protocol (SCTP) is a reliable datagram-oriented IP transport protocol specified by RFC 2960. It provides the layer between an SCTP user application and an unreliable end-to-end datagram service such as IP. The basic service offered by SCTP is the reliable transfer of user datagrams between peer SCTP users. It performs this service within the context of an association between two SCTP hosts. SCTP is connection-oriented, but SCTP association is a broader concept than the TCP connection, for example.
SCTP is not explicitly configured on routers, but it underlies several Cisco applications. The commands described in the document referenced below are useful for troubleshooting when SCTP issues are suspected as the cause of problems.
The SCTP feature was originally introduced in Cisco IOS Release 12.2(4)T as Release 1. SCTP Release 2 includes updated output for the show ip sctp association parameters and show ip sctp association statistics commands.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_sctp2.htm.
Survivable Remote Site Telephony Version 1.0
The Survivable Remote Site (SRS) Telephony feature, under the IP Telephony services umbrella, provides the Cisco CallManager with fallback support for the Cisco IP phones attached to the Cisco router on your local Ethernet. The SRS Telephony feature enables the routers to provide call handling support for the Cisco IP phones when the Cisco IP phones lose connection to the remote primary, secondary, or tertiary Cisco CallManager or when the WAN connection is down.
Survivable Remote Site Telephony Version 2.0
The Survivable Remote Site (SRS) Telephony feature, under the IP Telephony services umbrella, provides the Cisco CallManager with fallback support for the Cisco IP phones attached to the Cisco router on your local Ethernet. The SRS Telephony feature enables the routers to provide call handling support for the Cisco IP phones when the Cisco IP phones lose connection to the remote primary, secondary, or tertiary Cisco CallManager or when the WAN connection is down.
Cisco CallManager 3.0 supports Cisco IP phones at remote sites attached to Cisco branch office multiservice routers across the WAN. Prior to the SRS Telephony feature, when the WAN connection between the remote branch office router and the Cisco CallManager failed or connectivity with the Cisco CallManager was lost for some reason, the Cisco IP phones at the branch office became unusable for the duration of the failure. The SRS Telephony feature overcomes this problem and enables the basic features of the Cisco IP phones by providing call-handling support on the branch office router for its attached Cisco IP phones. The system automatically detects the failure and uses the Simple Network Auto Provisioning (SNAP) technology to autoconfigure the branch office router to provide call processing for the local Cisco IP phones. When the WAN link or connection to the primary Cisco CallManager is restored, call-handling capabilities for the Cisco IP phones switch back to the primary Cisco CallManager. During a failure when SRS Telephony feature is enabled, the Cisco IP phone displays a message to inform you that the Cisco IP phones are in the Cisco CallManager fallback mode and are able to perform limited functions.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/srs/fallbak2.htm.
T.37 Store-and-Forward Fax for Cisco 1751 Modular Access Routers
Fax applications enable Cisco 1751 modular access routers to send and receive faxes across packet-based networks by using voice interface cards (VICs) that support Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), Ear and Mouth (E&M), and BRI NT/TE signaling protocols.
The Cisco 1751 modular access routers support carrier-class Voice over IP (VoIP) and fax over IP services. Because the Cisco 1751 modular access routers are H.323 compliant, they support a family of industry-standard voice codecs and provide echo cancellation and voice activity detection (VAD) and silence suppression. There is an interactive voice response (IVR) application that provides voice prompts and digit collection in order to authenticate the user and identify the call destination.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/plfxrl17.htm.
T.37 Store-and-Forward Fax for the Cisco 2600 Series and Cisco 3600 Series Routers
This feature adds fax detection and store and forward fax to the Cisco 2600 series and Cisco 3600 series routers. When equipped with digital and analog voice network modules, these routers support configuration of the T.37/T38 fax gateway. Supported network modules are NM-HDV with voice interface cards (VICs) for digital T1 connections and Voice 2V with VICs FXS for analog connections. VWIC and VIC FXS are the voice interface cards within the network modules.
Voice network modules installed in Cisco 2600 series or Cisco 3600 series routers convert telephone voice signals into data packets that can be transmitted over an IP network. VWICs/VICs work with existing telephone and fax equipment and are compatible with H.323 standards for audio and video conferencing.
Cisco 2600 series and Cisco 3600 series routers that support carrier-class Voice over IP (VoIP) and fax over IP services provide echo cancellation and voice activity detection (VAD)/silence suppression. An interactive voice response (IVR) application provides voice prompts and digit collection to authenticate the user and identify the call destination.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/plfxrl17.htm.
TCP Window Scaling
TCP Window Scaling adds support for the Window Scaling extension option in RFC 1323. To improve TCP performance in network paths with a large bandwidth-delay product, Long Fat Networks (LFNs), a larger window size is recommended. This TCP Window Scaling enhancement provides that support.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/tcpwslfn.htm.
Trustpoint CLI
The Trustpoint CLI feature introduces the crypto ca trustpoint command, which combines and replaces the functionality of the existing crypto ca identity and crypto ca trusted-root commands.
Although both of the existing commands allow you to declare the certification authority (CA) that your router should use, only the crypto ca identity command supports enrollment (the requesting of a router certificate from a CA). With the crypto ca trustpoint command, you can declare the CA and specify any characteristics for the CA that the existing commands supported.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/fttrust.htm.
Tunnel Type of Service (ToS)
The Tunnel Type of Service (ToS) feature allows you to configure the ToS and Time-to-Live (TTL) byte values in the encapsulating IP header of tunnel packets for an IP tunnel interface on a router. The Tunnel ToS feature is supported on Cisco Express Forwarding (CEF), fast switching, and process switching forwarding modes.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s17/12s_tos.htm.
Unspecified Bit Rate Plus (uBR+) and ATM Enhancements for Service Provider Integrated Access
The uBR+ and ATM Enhancements for Service Provider Integrated Access feature includes:
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When uBR CPE to ATM switch is configured, a file transfer from one virtual circuit (VC) utilizes the entire trunk bandwidth when no other VCs (data or voice) are active. When other VCs become active with fixed committed information rates (CIRs), because uBR+ is not configured, the new VCs are not guaranteed their intended CIR. UBR+ resolves this by reallocating the configured CIRs to guarantee that all VCs achieve the appropriate throughput. If there is any remaining bandwidth, bursting up to that availability is still permitted. Because uBR allows for a continuous burst, bandwidth could be conserved by assigning a uBR class of service (CoS) to the VC. However, uBR has a variable bit rate (VBR) that constrains the burst period to a maximum burst size (MBS), rather than allowing a continuous burst. The uBR+ and ATM Enhancements for Service Provider Integrated Access feature does not have an MBS constraint.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/
122xb_2/ft_ubr.htm.Update to the MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
The voice-port (MGCP profile) command has been replaced by the port (MGCP profile) command. This command associates a voice port with the MGCP profile that is being configured.
The port (MGCP profile) command is used with the MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles feature that was released in Cisco IOS Release 12.2(4)T. Only the name of the command has been updated. The syntax and functionality have not changed.
Platforms supported are:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_mgupd.htm
VLAN Range
Using the VLAN Range feature, you can group VLAN subinterfaces together so that any command entered in a group applies to every subinterface within the group. This feature simplifies configurations and reduces command parsing.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122b/122b_4/
12b_rang.htm.VoAAL2 Profile 9 Support for BLES Interoperability
This feature allows Cisco routers to provide VoAAL2 (Voice over ATM Adaption Layer 2) Profile 9 (G.711ulaw and G.711alaw with 44-byte voice payload) for interoperability with V5.2 and GR.303 Voice GW to Class 5 switches. This feature allows service providers to deliver voice services over xDSL and T1 ATM networks from Class 5 switches.
Voice Support for Japan on Cisco 800 Series Routers, Phase 2
The Enhanced Voice Services for Japan for Cisco 800 Series Routers, Phase 2 features consist of the following voice capabilities for the Cisco 800 series routers:
Intercom
This feature establishes a voice connection between the two plain old telephone service (POTS) ports within the router. No B channels are used for calling between ports, so the B channels are available for data calls. During an intercom call, call waiting is disabled. If an external call comes to either POTS port, no call waiting tone is generated. The calling party will hear a busy signal. The flash hook and dual tone multifrequency (DTMF) keys are also disabled.
An intercom call is established by pressing **0# on the handset of either POTS port. If either port is busy with an external voice call, the intercom call will not be established.
Redial
This feature allows the user on each POTS port to redial the last number dialed on that port. Redial is activated when the user presses **4# on the handset. The router will store a number of up to 65 digits for each port. Feature access codes starting with an asterisk (*), interactive voice response (IVR) digits, or the pound (#) key are not stored.
The redial feature is supported separately on each POTS port.
Local Call Transfer
An external call received on either POTS port can be transferred to the other port. The transfer is initiated by pressing the flash hook followed by **0# on the handset.
This feature does not support conference calls.
Volume Adjustments
This feature allows the adjustment of the receiver volume on each POTS port. Volume adjustment is configured using command-line interface (CLI) commands, separately for each port.
To configure the telephone receiver volume on each port, use the CLI (the volume command).
Distinctive Ringing Based on Caller ID
This feature allows the user to register with the router up to 20 different numbers for each POTS port and to assign distinctive ring cadences to each of these numbers. Three different cadences are available. One of the cadences is the normal ring cadence as defined by Nippon Telegraph and Telephone (NTT) and is the default cadence for unregistered numbers. Numbers are registered, and ring cadences are assigned using CLI commands.
This feature is similar to the Nariwake feature available by subscription from NTT. However, this feature does not require the user to subscribe to any special service from the service provider. If the user already subscribes to Nariwake, Nariwake takes precedence over this feature.
The ring cadences used for this feature are the same as those used by the Nariwake feature.
Distinctive ringing based on caller ID is configured using the CLI (the caller-number command).
Subaddresses for POTS Ports
This feature allows the router to assign ISDN subaddresses to the POTS ports. With the subaddressing properly configured on the router, an external call is able to reach the dialed destination directly.
The subaddress for each POTS port is configured separately using the CLI (the subaddress command).
Silent Fax Calls
This feature allows either POTS port to be configured as a Type 2 Smart Fax port. When configured in this way, the router will not generate a ring alert when a call comes into the port. Instead, a silent fax tone will be generated, to which the Type 2 Smart Fax machine will respond. The fax machine does not ring, but the fax call gets connected. If a telephone is connected instead of a fax machine, the telephone does not ring.
This feature is configured using the CLI (the silent-fax command).
PIAFS Support
This feature provides support for the Personal Handyphone System (PHS) Internet Access Forum Standard (PIAFS). PIAFS is a standard error correction protocol for cellular data communication that has been developed in Japan. It is designed to pass data over the PHS cellular system. It also provides transmission control procedures (comparable to OSI reference model Layer 2) for high-quality data transmission. Both PIAFS version 2.0 and version 2.1 are supported on Cisco 803, 804, and 813 routers.
The common applications that are supported using PIAFS in PHS data communications are as follows:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_ktna2.htm.
VPN Routing Forwarding (VRF) Framed Route (Pool) Assignment via PPP
The VPN Routing Forwarding (VRF) Framed Route (Pool) Assignment via PPP feature introduces support to make the following RADIUS attributes VRF aware: attribute 22 (Framed-Route), a combination of attribute 8 (Framed-IP-Address) and attribute 9 (Framed-IP-Netmask), and the Cisco VSA route command. Thus, static IP routes can be applied to a particular VRF routing table rather than the global routing table.
WRED Enhancement—Explicit Congestion Notification (ECN)
Currently, the congestion control and avoidance algorithms for TCP are based on the idea that packet loss is an appropriate indication of congestion on networks that transmit data using the best-effort service model. When a network uses the best-effort service model, the network delivers data if it can, without any assurance of reliability, delay bounds, or throughput. However, these algorithms and the best-effort service model are not suited to applications that are sensitive to delay or packet loss (for instance, interactive traffic including Telnet, web browsing, and transfer of audio and video data). Weighted random early detection (WRED), and by extension, Explicit Congestion Notification (ECN), helps to solve this problem.
To indicate congestion, WRED drops packets on the basis of the average queue length exceeding a specific threshold value. ECN is an extension to WRED in that ECN marks packets instead of dropping them when the average queue length exceeds a specific threshold value. When configured with the WRED Enhancement—Support for Explicit Congestion Notification feature, routers and end hosts would use this marking as a signal that the network is congested and slow down sending packets.
This feature provides an improved method for congestion avoidance by allowing the network to mark packets for transmission later, rather than dropping them from the queue. Marking the packets for transmission later accommodates applications that are sensitive to delay or packet loss and provides improved throughput and application performance.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftwrdecn.htm.
X.25 Over TCP Profiles
The Cisco X.25 over TCP (XOT) service was originally developed as an X.25 class of service that was only designed to switch X.25 traffic across an IP network. This service allowed network administrators to connect X.25 devices across the rich connectivity and media features available to IP traffic. XOT uses a set of default parameters to make this type of network easy to design.
When the XOT' capabilities were enhanced to support packet assembler/disassembler (PAD) traffic on an XOT session, network designers saw a need to be able to configure parameters for increased flexibility. For instance, because XOT does not have any physical interfaces that an administrator can configure, PAD over XOT sessions cannot be configured with interface map or facility commands to establish a PAD connection using nondefault values.
The introduction of X.25 profiles for XOT allows the network designer added flexibility to control the X.25 class services of XOT for PAD and XOT switching usage.
Another important aspect of this feature is that it allows you to associate access lists with XOT connections, enabling you to apply security on the basis of IP addresses and to have a unique X.25 configuration for specified IP addresses.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ft_xotp.htm.
X.25 Record Boundary Preservation for Data Communications Networks
The X.25 Record Boundary Preservation for Data Communications Networks feature enables hosts using TCP/IP-based protocols to exchange data with devices that use the X.25 protocol, retaining the logical record boundaries indicated by use of the X.25 "more data" bit (M-bit).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/ftdcnrbp.htm.
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(4)T
The following hardware platforms and modules are now supported in Cisco IOS Release 12.2(4)T. These platforms and modules were first introduced in earlier Cisco IOS software releases.
1-Port ADSL WAN Interface Card
The 1-Port ADSL WAN Interface Card (WIC) provides ADSL high-speed digital data transfer between a single customer premises equipment (CPE) subscriber and the central office.
The ADSL WIC is compatible with the Alcatel Digital Subscriber Loop Access Multiplexer (DSLAM) and the Cisco 6130, Cisco 6160, and Cisco 6260 DSLAMs with Flexi-line cards. It supports ATM Adaptation Layer 2 (AAL2) and AAL5 for the Cisco 2600 series and Cisco 3600 series routers and AAL5 only for the Cisco 1700 series routers, for both voice and data service.
Refer to the following documents for additional information:
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http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_adsl4.htm.
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http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xr/121xr_5/ftiaddsl.htm.
Note
1-Port T1/E1 Digital Voice Port Adapters for Cisco 7200 and Cisco 7500
The PA-VXB and the PA-VXC are multichannel packet voice port adapters that allow Cisco 7200 series routers, Cisco 7200 VXR routers, Cisco 7401ASR routers, and Cisco 7500 series routers to become dedicated packet voice hubs or packet voice gateways that connect to both private branch exchanges (PBXs) and the Public Switched Telephone Network (PSTN). With this technology, packet voice and packet fax calls can be placed over the WAN and sent through the gateway into the traditional circuit-switched voice infrastructure. The PA-VXB and PA-VXC are single-width port adapters with two universal ports that are configurable for either T1 or E1 connections. The PA-VXB contains 12 high-performance digital signal processors (DSPs) that support up to 48 medium-complexity or 24 high-complexity channels of compressed voice. The PA-VXC contains 30 high-performance DSPs that support up to 120 medium-complexity or 60 high-complexity channels of compressed voice.
In Voice over IP (VoIP), the DSP segments the voice signal into frames, which are then coupled in groups of two and stored in voice packets. These voice packets are transported using IP in compliance with ITU-T specification H.323. Because VoIP is a delay-sensitive application, you must have a well-engineered end-to-end network to use it successfully. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward quality of service (QoS). Traffic shaping considerations must be taken into account to ensure the reliability of the voice connection.
8-Port Mix-Enabled T1/E1/PRI PA
The PA-MC-8TE1+ port adapter is a single-wide port adapter that provides eight T1 or E1 interfaces for Cisco 7200 series routers. The PA-MC-8TE1+ interfaces can be channelized, fractional, or unframed (E1 only).
The PA-MC-8TE1+ provides the following features:
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Refer to the PA-MC-8TE1+ Port Adapter Installation and Configuration Note for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/core/7200vx/portadpt/multicha/8port_t1/index.htm.
Cisco 1751 Router
The voice-and-data capable Cisco 1751 router provides global Internet and company intranet access and includes the following:
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Cisco uBR925 Cable Access Router
The Cisco uBR925 cable access router acts as a cable modem to connect computers and other customer premises equipment (CPE) devices at a subscriber site to the service provider cable, hybrid fiber-coaxial (HFC), and IP backbone network. The Cisco uBR925 router is based on the Data-over-Cable Service Interface Specifications (DOCSIS) and interoperates with any bidirectional, DOCSIS-qualified cable modem termination system (CMTS).
The Cisco uBR925 cable access router supports both data and Voice over IP (VoIP) traffic via a shared two-way cable system and IP backbone network. PCs and other CPE devices can connect to the Cisco uBR925 router either through a four-port Ethernet hub or through the Universal Serial Bus (USB) port of the router. Single-line telephones, fax, or modems can be connected to two RJ-11 analog voice ports of the router. The Cisco uBR925 router supports DOCSIS-compliant bridging data operations, and it can also function as an advanced router, providing WAN data and VoIP connectivity in a variety of configurations.
Cisco CVA122 Cable Voice Adapter
The Cisco CVA122 Cable Voice Adapter acts as a cable modem to connect computers and other customer premises equipment (CPE) devices at a subscriber site to the service provider cable, hybrid fiber-coaxial (HFC), and IP backbone network. The Cisco CVA122 Cable Voice Adapter is based on the Data-over-Cable Service Interface Specifications (DOCSIS) and interoperates with any bidirectional, DOCSIS-qualified cable modem termination system (CMTS).
The Cisco CVA122 Cable Voice Adapter supports both data and Voice over IP (VoIP) traffic via a shared two-way cable system and IP backbone network. PCs and other CPE devices can connect to the cable voice adapter either through an Ethernet port or through a Universal Serial Bus (USB) port. Single-line telephones, fax, or modems can be connected to two RJ-11 analog voice ports of the cable voice adapter. The cable voice adapter supports DOCSIS-compliant bridging data operations, and it can also function as an advanced router, providing WAN data and VoIP connectivity in a variety of configurations.
Cisco CVA122E Cable Voice Adapter
The Cisco CVA122E Cable Voice Adapter acts as a cable modem to connect computers and other customer premises equipment (CPE) devices at a subscriber site to the service provider cable, hybrid fiber-coaxial (HFC), and IP backbone network. The Cisco CVA122E Cable Voice Adapter is based on the European Data-over-Cable Service Interface Specifications (EuroDOCSIS) and interoperates with any bidirectional, EuroDOCSIS-qualified cable modem termination system (CMTS).
The Cisco CVA122 Cable Voice Adapter supports both data and Voice over IP (VoIP) traffic via a shared two-way cable system and IP backbone network. PCs and other CPE devices can connect to the cable voice adapter either through an Ethernet port or through a Universal Serial Bus (USB) port. Single-line telephones, fax, or modems can be connected t o two RJ-11 analog voice ports of the cable voice adapter. The cable voice adapter supports EuroDOCSIS-compliant bridging data operations, and it can also function as an advanced router, providing WAN data and VoIP connectivity in a variety of configurations.
New Software Features in Cisco IOS Release 12.2(4)T
The following new features are supported in Cisco IOS Release 12.2(4)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
AAA-PPP-VPDN Non-Blocking
Previously, Cisco IOS created a statically configurable number of processes to authenticate calls. Each of these processes would handle a single call, but in some situations the limited number of processes could not keep up with the incoming call rate. This resulted in some calls timing out. The AAA-PPP-VPDN Non-Blocking feature changes the software architecture such that the number of processes will not limit the rate of call handling.
Ability to Disable Xauth for Static IPSec Peers
The Ability to Disable Xauth for Static IPSec Peers feature allows users to disable extended authentication (Xauth), which prevents the routers from being prompted for Xauth information—username and password.
Without the ability to disable Xauth, a user cannot select which peer on the same crypto map should use Xauth. That is, if a user has router-to-router IP Security (IPSec) on the same crypto map as a Virtual Private Network (VPN)-client-to-Cisco-IOS IPSec, both peers will be prompted for a username and password. Removing Xauth while configuring the preshared key for router-to-router IPSec, prevents duplicate Xauth information from being exchanged, thereby, reducing traffic on your network. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftnxauth.htm.
ACL Default Direction
The ACL Default Direction feature allows you to change the filter direction (where filter direction is not specified) to inbound packets only; that is, you can configure your server to filter packets that are coming toward the network.
This feature introduces the radius-server attribute 11 direction default command, which allows you to change the default direction of filters for your access control lists (ACL) via RADIUS. (RADIUS attribute 11 (Filter-Id) indicates the name of the filter list for the user.) Enabling this command allows you to change the filter direction to inbound—which stops traffic from entering a router, thereby reducing resource consumption—rather than the outbound default direction, which waits until the traffic is about to leave the network before filtering.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftacldir.htm.
Accounting of VPDN Disconnect Cause
In the past, whenever a Layer 2 Tunneling Protocol (L2TP) or Layer 2 Forwarding (L2F) session fails or disconnects, the network access server (NAS) and Home GateWay (HGW) report a very generic disconnect-cause code, such as "LOST CARRIER". These generic codes do not provide enough detailed information for accounting and debugging purposes, creating a need for disconnect-cause codes that provide more detailed information. The Accounting of VPDN Disconnect Cause feature adds eight new disconnect-cause codes. These eight disconnect-cause codes describe the status of Virtual Private Dialup Network (VPDN) failures and disconnects more specifically than existing generic disconnect-cause codes. These new disconnect-cause codes can be found in the Cisco IOS Security Configuration Guide, Release 12.2 located at the following URL:
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftacldir.htm.
Adaptive Frame Relay Traffic Shaping for Interface Congestion
The Adaptive Frame Relay Traffic Shaping for Interface Congestion feature enhances Frame Relay traffic shaping functionality by adjusting permanent virtual circuit (PVC) sending rates based on interface congestion. When this new feature is enabled, the traffic-shaping mechanism monitors interface congestion. When the congestion level exceeds a configured value called queue depth, the sending rate of all PVCs is reduced to the minimum committed information rate (minCIR). As soon as interface congestion drops below the queue depth, the traffic-shaping mechanism changes the sending rate of the PVCs back to the committed information rate (CIR). This process guarantees the minCIR for PVCs when there is interface congestion.
This new feature works in conjunction with backward explicit congestion notification (BECN) and Foresight functionality.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_afrts.htm.
Advanced Voice Busyout
The local voice busyout feature provides a way to busy out a voice port or DS-0 group (time slot) if a state change is detected in a monitored network interface (or interfaces). When a monitored interface changes to a specified state—to out-of-service or in-service—the voice port presents a seized/busyout condition to the attached PBX or other customer premises equipment (CPE). The PBX or other CPE can then attempt to select an alternate route.
Advanced Voice Busyout adds the following functionality to the local voice busyout feature:
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Using the Advanced Voice Busyout feature you can perform the following tasks:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft_cacbo.htm.
Note
Asynchronous Line Monitoring
Before Cisco IOS Release 12.2(4)T, the Cisco IOS software did not provide a method for displaying asynchronous character mode traffic flowing out of an asynchronous line. Therefore, when a user tried to troubleshoot difficult asynchronous problems, the user had to use RS-232 datascopes in order to examine the data stream. This method is very detailed and cumbersome. The Asynchronous Line Monitoring feature available in Cisco IOS Release 12.2(4)T allows the monitoring of inbound and outbound character mode asynchronous traffic on another terminal line. To monitor inbound or outbound asynchronous character mode traffic on the port to be monitored, enter the monitor traffic line command in privileged EXEC mode.
This feature increases the efficiency of the user who performs troubleshooting on asynchronous character mode traffic problems. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftasync.htm.
ATM SNMP Trap and OAM Enhancements
The ATM SNMP Trap and OAM Enhancements feature introduces the following enhancements to the Simple Network Management Protocol (SNMP) notifications for ATM permanent virtual circuits (PVCs) and to operation, administration, and maintenance (OAM) functionality:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftpvctrp.htm.
AutoInstall over Frame Relay-ATM Interworking Connections
The AutoInstall over Frame Relay-ATM Interworking Connections feature extends the functionality of the existing Cisco IOS AutoInstall feature. After you connect a new router to the network and turn on the new router, Cisco IOS AutoInstall automatically configures the router from a preexisting configuration file that is downloaded from the network. This process was designed to facilitate the centralized management of router installation.
The AutoInstall over Frame Relay-ATM Interworking Connections feature allows you to configure an ATM permanent virtual circuit (PVC) to accept the BOOTP and TFTP requests from a new router performing AutoInstall. This feature also allows the use of a central router with ATM or Frame Relay IETF encapsulation to run a BOOTP server and provide an initial IP address to the new router.
For details, refer to the "AutoInstall over Frame Relay-ATM Interworking Connections" feature module document. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftautatm.htm.
Automatic Bandwidth Adjustment for MPLS Traffic Engineering Tunnels
Traffic engineering automatic bandwidth adjustment provides the means to automatically adjust the bandwidth allocation for traffic engineering tunnels based on their measured traffic load.
Traffic engineering autobandwidth samples the average output rate for each tunnel marked for automatic bandwidth adjustment. For each marked tunnel, it periodically (for example, once per day) adjusts the tunnel's allocated bandwidth to be the largest sample for the tunnel since the last adjustment.
The frequency with which tunnel bandwidth is adjusted and the allowable range of adjustments is configurable on a per-tunnel basis. In addition, the sampling interval and the interval over which to average tunnel traffic to obtain the average output rate is user-configurable on a per-tunnel basis.
There are three new commands:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftbwadjm.htm.
BGP Conditional Route Injection
Cisco IOS software provides several methods in which you can originate a prefix into the Border Gateway Protocol (BGP). The existing methods include using the network or aggregate-address commands and redistribution. These methods assume the existence of more specific routing information (matching the route to be originated) in either the routing table or the BGP table.
The BGP Conditional Route Injection feature enables you to originate a prefix into BGP without the corresponding match. The routes are injected into the BGP table only if certain conditions are met. The most common condition is the existence of a less-specific prefix.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftbgpri.htm.
BGP Link Bandwidth
The Border Gateway Protocol (BGP) Link Bandwidth feature is used to advertise the bandwidth of an autonomous system exit link as an extended community. The BGP Link Bandwidth feature is supported by the internal BGP (iBGP) and external BGP (eBGP) multipath features. The link bandwidth extended community indicates the preference of an autonomous system exit link in terms of bandwidth. The link bandwidth extended community attribute may be propagated to all iBGP peers and used with the BGP multipath features to configure unequal cost load balancing. When a router receives a route from a directly connected external neighbor and advertises this route to iBGP neighbors, the router may advertise the bandwidth of that link.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftbgplb.htm.
BGP Multipath Load Sharing for Both eBGP and iBGP in an MPLS-VPN
The BGP Multipath Load Sharing for eBGP and iBGP feature allows you to configure multipath load balancing with both external BGP (eBGP) and internal BGP (iBGP) paths in Border Gateway Protocol (BGP) networks that are configured to use Multiprotocol Label Switching (MPLS) Virtual Private Networks (VPNs). This feature provides improved load balancing deployment and service offering capabilities and is useful for multi-homed autonomous systems and Provider Edge (PE) routers that import both eBGP and iBGP paths from multihomed and stub networks.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/fteibmpl.htm.
BGP Prefix-Based Outbound Route Filtering
The BGP Prefix-Based Outbound Route Filtering feature uses Border Gateway Protocol (BGP) outbound route filter (ORF) send and receive capabilities to minimize the number of BGP updates that are sent between peer routers. The configuration of this feature can help reduce the amount of resources required for generating and processing routing updates by filtering out unwanted routing updates at the source.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftbgporf.htm.
Call Admission Control for H.323 VoIP Gateways
Call Admission Control, Call Treatment, and Busyout Components
Before the call admission control feature, gateways did not have a mechanism to gracefully prevent calls from entering when certain resources were not available to process the call. This causes the new call to fail with unreported behavior, and could potentially cause the calls that are in progress to have quality related problems.
This feature set provides the ability to support resource-based call admission control processes. These resources include system resources such as CPU, memory, and call volume, and interface resources such as call volume.
If system resources are not available to admit the call, two kinds of actions are provided: system denial (which busyouts all of T1 or E1) or per call denial (which disconnects, hairpins, or plays a message or tone). If the interface-based resource is not available to admit the call, the call is dropped from the session protocol (such as H.323).
For further information on busyout, please refer to Advanced Voicebusyout, Cisco IOS Release 12.2(2)XA. For further information on the call denial aspects of this feature, please refer to Call Admission Control Based on CPU Utilization.
User Selected Threshold
This feature allows a user to configure call admission thresholds for local resources as well as memory and CPU resources. The list of local resources that are configured for call admission are described in the command description of "call threshold poll-interval."
With the call threshold command, a user is allowed to configure two thresholds, high and low, for each resource. Call treatment is triggered when the current value of a resource goes beyond the configured high. The call treatment remains in effect until current resource value falls below the configured low. Having high and low thresholds prevents call admission flapping and provides hysteresis in call admission decision making.
With the call spike command, a user is allowed to configure the limit for incoming calls during a specified time period. A call spike is the term for when a large number of incoming calls arrive from the PSTN in a very short period of time (for example: 100 incoming calls in 10 milliseconds).
Configurable Call Treatment
With the call treatment command, users are allowed to select how the call should be treated when local resources are not available to handle the call. For example, when the current resource value for any one of the configured triggers for call threshold has reached beyond the configured threshold, the call treatment choices are as follows:
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Resource Unavailable Signaling
This feature set supports the autobusyout feature where channels are busied out when local resources are not available to handle the call. Autobusyout is supported on both channel-associated signaling (CAS) and PRI channels.
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PSTN Fallback
The goal of PSTN fallback is to monitor congestion in the IP network and either redirect calls to the PSTN or reject calls based on the network congestion. Calls can be rerouted to an alternate IP destination or to the PSTN if the IP network is found unsuitable for voice traffic at that time. The user defines the congestion thresholds based on the configured network. This functionality enables the service provider to give a reasonable guarantee about the quality of the conversation to their VoIP users at the time of call admission.
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PSTN fallback includes the following features:
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For more information, including configuration tasks and examples, and command references for PSTN fallback, please refer to PSTN Fallback. Refer to the following document for additional information about the Call Admission Control for H.323 VoIP Gateways feature:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft_pfavb.htm.
Circuit Interface Identification Persistence for SNMP
The Circuit Interface MIB (CISCO-CIRCUIT-INTERFACE-MIB) provides a MIB object (cciDescr) which can be used to identify individual circuit-based interfaces for SNMP monitoring. The Circuit Interface Identification Persistence for SNMP feature maintains this user-defined name of the circuit across reboots, allowing the consistent identification of circuit interfaces. Circuit Interface Identification Persistence is enabled using the snmp mib persist circuit global configuration command.
Cisco H.323 Scalability and Interoperability Enhancements
The Cisco H.323 Scalability and Interoperability Enhancements feature upgrades the Cisco H.323 Gatekeeper and Cisco H.323 Gateway to comply with H.323 Version 3. The enhancements in this release include support for mandatory H.323 Version 3 elements in the gateway, support for H.225 call signalling over User Datagram Protocol (UDP), and address resolution using border elements.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/fth323v3.htm.
Cisco Mobile Networks
The Cisco Mobile Networks feature enables a Mobile Router and its subnets to be mobile and maintain all IP connectivity, transparent to the IP hosts connecting through this Mobile Router.
Mobile IP, as defined in standard RFC 2002, provides the architecture that enables the Mobile Router to connect back to its home network. Mobile IP allows devices to roam while appearing to be at their home network. Such a device is called a mobile node. A mobile node is a node, for example, a personal digital assistant, a laptop computer, or a data-ready cellular phone, that can change its point of attachment from one network or subnet to another. This mobile node can travel from link to link and maintain ongoing communications while using the same IP address.
The Mobile Router functions similarly to the mobile node with one key difference—the Mobile Router allows entire networks to roam. For example, a plane with a Mobile Router can fly around the world while passengers stay connected to the Internet. This communication is accomplished by Mobile IP aware routers tunneling packets, which are destined to hosts on the mobile networks, to the location where the Mobile Router is visiting. The Mobile Router then forwards the packets to the destination device.
These devices can be mobile nodes running mobile IP client software or nodes without the software. The Mobile Router eliminates the need for a mobile IP client. In fact, the nodes on the mobile network are not aware of any IP mobility at all. The Mobile Router "hides" the IP roaming from the local IP nodes so that the local nodes appear to be directly attached to the home network.
The Cisco Mobile Networks feature is a static network implementation that supports stub routers only. The Mobile Router avoids convergence problems by statically defining which networks it can address. The Mobile Router can do the following:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftmbrout.htm.
Cisco Modem User Interface
The Cisco Modem User Interface feature enables Cisco routers to behave like a modem and be configured using standard Hayes modem commands. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftcmodui.htm.
Configuring AAL2 and AAL5 for the High Performance ATM Advanced Integration Module on the Cisco 2600 Series
The High Performance ATM Advanced Integration Module (AIM) is an internally mounted card that offers a cost-effective solution for supporting low-speed ATM WAN connections on the Cisco 2600 family of products. When using the voice DSP capability of a digital T1/E1 packet voice trunk network module (NM-HDV) and a T1/E1 multiflex VWIC, it supports as many as 60 channels of compressed voice over a T1/E1 trunk using AAL2 or AAL5, without using a dedicated ATM network module. AAL2 and AAL5 are the most bandwidth-efficient standards-based trunking methods for transporting compressed voice, voice-band data, circuit-mode data, and frame-mode data over ATM infrastructures. This feature provides a cost-effective, low-density ATM T1 or E1 solution for the Cisco 2600 series.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_24aim.htm.
CNS Configuration Agent
CNS is a foundation technology for linking users to network services. CNS SDK accomplishes this by making applications network-aware and increasing the intelligence of the network elements. CNS SDK provides building blocks to a range of customers in market segments such as Enterprise, service provider, independent software vendors, and system integrators.
The CNS Configuration Agent supports routing devices by providing:
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Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcns_ca.htm.
CNS Event Agent
CNS is a foundation technology for linking users to network services. CNS SDK accomplishes this by making applications network-aware and increasing the intelligence of the network elements. CNS SDK provides building blocks to a range of customers in market segments such as Enterprise, service provider, independent software vendors, and system integrators.
The CNS Event Agent is part of the Cisco IOS infrastructure that allows Cisco IOS applications, for example CNS Configuration Agent, to publish and subscribe to events on a CNS Event Bus. CNS Event Agent works in conjunction with CNS Configuration Agent.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcns_ea.htm.
Crashinfo Support for Cisco 3600 Series
Crashinfo is a mechanism to reliably and quickly store useful information related to unexpected system shutdowns directly to a local flash card. This information can be retrieved after a system reload to aid in the analysis and resolution of a system error.
Cisco IOS Release 12.2(4)T introduces crashinfo support for the Cisco 3600 series. To enable this feature, use the exception crashinfo file device:filename in global configuration mode. Use the device and filename arguments to specify the flashcard and file to be used for storing the diagnostic information. To change the size of the crashinfo buffer, use the exception crashinfo buffersize command. The default buffer size is 32 Kilobytes.
DFP Support in DistributedDirector
DistributedDirector can obtain load information from Cisco LocalDirector, Catalyst 4840g, and other clients using Dynamic Feedback Protocol (DFP). This protocol allows the user to configure the DistributedDirector to communicate with various DFP agents. The DistributedDirector tells the DFP agents how often they should report load information; then the DFP agent can tell the DistributedDirector which LocalDirector cluster to remove from providing service.
Dialer CEF
The Dialer CEF feature introduces Cisco Express Forwarding (CEF) support for dialer interfaces. The Dialer CEF feature allows packets to be CEF switched across dialer interfaces rather than being low-end switched (LES) or fast switched. Compared to fast switching, CEF switching support improves switching performance by decreasing CPU utilization and lowering the packet loss rate. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftdlrcef.htm.
Dialer Persistent
The Dialer Persistent feature allows the connection settings in a dial-on-demand routing (DDR) dialer profile to be configured as persistent, that is, the connection is not torn down until the shutdown EXEC command is entered on the dialer interface. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftdperst.htm.
Diff-Serv-aware Traffic Engineering
MPLS traffic engineering allows constraint-based routing of constant bit rate (CBR) IP traffic. One of the constraints satisfied by CBR is the availability of required bandwidth over a selected path. Diff-Serv-aware Traffic Engineering extends MPLS traffic engineering to enable you to perform constraint-based routing of "guaranteed" traffic, which satisfies a more restrictive bandwidth constraint than that satisfied by CBR for regular traffic. This ability to satisfy a more restrictive bandwidth constraint translates into an ability to achieve higher Quality of Service performance (in terms of delay, jitter, or loss) for the guaranteed traffic.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_ds_te.htm.
Distinguished Name Based Crypto Maps
The Distinguished Name Based Crypto Maps feature allows you to restrict access to selected encrypted interfaces to peers with specific certificates, especially certificates with particular Distinguished Names (DNs).
Initially, if the router accepted a certificate or a shared secret from the encrypting peer, Cisco IOS did not have a method of preventing the peer from communicating with any encrypted interface other than the restrictions on the IP address of the encrypting peer. This feature allows you to configure which crypto maps are usable to a peer based on the DN that a peer used to authenticate itself. Thus, enabling you to control which encrypted interfaces a peer with a specified DN can access.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftdnacl.htm.
Distributed Link Fragmentation and Interleaving
The Distributed Link Fragmentation and Interleaving feature extends Link Fragmentation and Interleaving functionality to VIP-enabled Cisco 7500 series routers.
The Distributed Link Fragmentation and Interleaving feature supports the transport of real-time traffic, such as voice, and non real-time traffic, such as data, on lower-speed Frame Relay and ATM virtual circuits (VCs) without causing excessive delay to the real-time traffic.
This feature implements link fragmentation and interleaving (LFI) using multilink PPP (MLP) over Frame Relay and ATM. The feature enables delay-sensitive real-time packets and packets that are not real-time data to share the same link by fragmenting the large data packets into a sequence of smaller data packets (fragments). The fragments are then interleaved with the real-time packets. On the receiving side of the link, the fragments are reassembled and the packet is reconstructed.
Distributed LFI is often useful in networks that send real-time traffic, such as voice, but have bandwidth problems that delay this real-time traffic due to the transport of large, less time-sensitive data packets. Distributed LFI can be used in these networks to disassemble the large data packet into multiple segments. The real-time traffic packets can then be sent between these segments of the data packet. In this scenario, the real-time traffic does not experience a lengthy delay waiting for the low-priority data packet to traverse the network. The data packet is reassembled at the receiving side of the link, so the data is delivered intact.
DistributedDirector Enhancements
This release of Cisco DistributedDirector contains two new commands. The new ip director default priorities command specifies the default priorities for each type of metric. If a metric does not have a default priority specified, DistributedDirector does not use that metric. The default priorities take effect if no priorities are specified in the DNS TXT record for that host.
The new ip director drp rttprobe tcp | icmp command enables DistributedDirector to instruct a DRP agent to send ICMP-echo packets to measure the RTT.
The new show ip director default priority command verifies the default priority configurations.
The ip director default-weights command has been modified. It is now ip director default weights.
The show ip director default-weights command has been modified. It is now show ip director default weights.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftdd1224.htm.
Distributed Management Event and Expression MIB Persistence
The MIB Persistence feature allows the SNMP data of an MIB to be persistent across reloads; that is, MIB information retains the same set object values each time the user reboots. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftmibpr1.htm.
DNS Server Support for NS Records
DistributedDirector has improved server load-balancing capacity with the DNS Server Support for NS Records feature. This feature adds support for name server (NS) records to the Cisco IOS Domain Name System (DNS) server. With this feature, the DistributedDirector can distribute the server-selection process to multiple DistributedDirectors, improving overall server capacity.
Enhanced Test Command
The Enhanced Test Command feature introduces two new commands—aaa user profile and aaa attribute—that allow you to create a named user profile with calling line identification (CLID) or dialed number identification service (DNIS) attribute values, which can be associated with a test aaa group command.
Use the aaa attribute command to add CLID or DNIS attribute values to a user profile, which is created by using the aaa user profile command. The CLID or DNIS attribute values can be associated with the record that is going out with the user profile (via the test aaa group command), thereby providing the RADIUS server with access to CLID or DNIS attribute information for all incoming calls.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftaaacmd.htm.
Enhancements to H.323 Call Statistics
Beginning with Cisco IOS Release 12.2(4)T, enhancements to H.323 call statistics allow you to clear the gateway counters, display H.323 messages that have been sent and received, obtain statistics on the reasons calls are disconnected, and display debug output for various components within the H.323 subsystem. To enable these enhancements, the following commands have been added or modified: clear h323 gateway, show h323 gateway, and debug cch323.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftcallst.htm.
Firewall Authentication Proxy
The Cisco IOS Firewall authentication proxy feature allows network administrators to apply specific security policies on a per-user basis. Previously, user identity and related authorized access was associated with the IP address of a user, or a single security policy had to be applied to an entire user group or sub network. Now, users can be identified and authorized on the basis of their per-user policy, and access privileges tailored on an individual basis are possible, as opposed to general policy applied across multiple users.
With the authentication proxy feature, users can log in to the network or access the Internet via HTTP, and their specific access profiles are automatically retrieved and applied from a CiscoSecure ACS, or other RADIUS, or TACACS+ authentication server. The user profiles are active only when there is active traffic from the authenticated users.
The authentication proxy is compatible with other Cisco IOS security features such as Network Address Translation (NAT), Context-based Access Control (CBAC), IP Security (IPSec) encryption, and Cisco Secure VPN Client (VPN client) software.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/secur_c/scprt3/scdauthp.htm.
Four SS7 Link Support on the Cisco Signaling Link Terminal
The Four SS7 Link Support on the Cisco Signaling Link Terminal feature introduces support for up to four Cisco SS7 links on a new platform for the Cisco SLT, the Cisco 2651 Multiservice Access Router. All existing Cisco 2611-based Cisco SLT functionality is supported on the new platform, and both Cisco SLT platforms use the same Cisco IOS software image.
The Cisco 2651-based Cisco SLT supports up to four SS7 A-links and F-links, and each SS7 link can support up to 0.4 erlangs of signaling traffic during normal operation. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_4lnk.htm.
Frame Relay 64-Bit Counters
The Frame Relay 64-Bit Counters feature provides 64-bit counter support on Frame Relay interfaces and subinterfaces. This feature enables the gathering of statistics through Simple Network Management Protocol (SNMP) for faster interfaces operating at OC-3, OC-12, and OC-48 speeds.
The following counters are supported by this feature: Bytes In, Bytes Out, Packets In, and Packets Out.
The show frame-relay pvc command has been modified to display the 64-bit counters. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft64bits.htm.
Frame Relay MIB Enhancements
The Cisco Frame Relay MIB describes managed objects that enable users to remotely monitor Frame Relay operations using Simple Network Management Protocol (SNMP). The Frame Relay MIB Enhancements feature extends the Cisco Frame Relay MIB by adding MIB objects to monitor the following Frame Relay functionality:
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The Frame Relay MIB enhancements also modify the load-interval command to enable you to configure the load interval per permanent virtual circuit (PVC). Before the introduction of this feature, the load interval could be configured only for the interface. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftfrmibe.htm.
High-Performance Gatekeeper
The Cisco High-Performance Gatekeeper feature introduces new gatekeeper functionality and modifications for facilitating carrier class reliability, security, and performance into the Cisco Voice Network solution portfolio. These H.323 standard-based features have carrier grade reliability and performance characteristics with a robust open application protocol interface to enable development of enhanced applications like voice Virtual Private Networks (VPNs) and wholesale voice solutions.
This feature addresses the scalability, redundancy, and performance aspects of the gatekeeper as part of the Cisco Multimedia Conference Manager (MCM) to present a complete Cisco solution. The Cisco H.323 MCM provides the network administrator with the ability to identify H.323 traffic and to apply appropriate policies.
iBGP Multipath Load Sharing
When a Border Gateway Protocol (BGP) speaker router with no local policy configured receives multiple network layer reachability information (NLRI) from the internal BGP for the same destination, the router will choose one internal BGP path as the best path. The best path is then installed in the IP routing table of the router.
The Internal BGP Multipath Load Sharing feature enables the BGP speaker router to select multiple internal BGP paths as the best paths to a destination. The best paths or multipaths are then installed in the IP routing table of the router.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft11bmls.htm
Note
ICMP ECHO-Based RTT Probing by DRP Agents
DistributedDirector users can now control Director Response Protocol (DRP) agents to send both TCP and ICMP packets for round-trip time (RTT) measurement.The RTT measurement is used to dynamically direct Internet customers to the closest regional web proxy based on response time.
In the original implementation, some Internet DNS servers did not respond when the DRP agents sent them a query to measure the RTT.
This feature introduces the new ip director drp rttprobe tcp | icmp command that enables DistributedDirector to instruct a DRP agent to send ICMP-echo packets to measure the RTT.
When both ICMP and TCP are enabled, DistributedDirector will instruct DRP agents to send both TCP and ICMP packets for RTT probing. The returned RTT from a DRP agent will be the RTT collected from either the TCP or ICMP mechanism, which ever becomes available first.
IGMP MIB Support Enhancements for SNMP
The Internet Group Management Protocol (IGMP) is used by IP hosts to report their multicast group memberships to neighboring multicast routers. The IGMP MIB describes objects that enable users to remotely monitor and configure IGMP using Simple Network Management Protocol (SNMP). It also allows users to remotely subscribe and unsubscribe from multicast groups. The IGMP MIB Support Enhancements for SNMP feature adds full support of RFC 2933 (Internet Group Management Protocol MIB) in Cisco IOS software. There are no new or modified Cisco IOS commands associated with this feature.
For complete details on the IGMP MIB, see the IGMP-STD-MIB.my file available from the Cisco MIB website on Cisco.com at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
Inter-Domain Gatekeeper Security Enhancement
The Inter-Domain Gatekeeper Security Enhancement provides a means of authenticating and authorizing H.323 calls between the administrative domains of Internet Telephone Service Providers (ITSPs).
An interzone ClearToken (IZCT) is generated in the originating gatekeeper when a location request (LRQ) is initiated or an admission confirmation (ACF) is about to be sent for an intrazone call within an ITSP administrative domain. As the IZCT traverses through the routing path, each gatekeeper stamps the IZCT destination gatekeeper ID with its own ID. This identifies when the IZCT is being passed over to another ITSP domain. The IZCT is then sent back to the originating gateway in the location confirmation (LCF) message. The originating gateway passes the IZCT to the terminating gateway in the SETUP message. The terminating gatekeeper forwards the IZCT in the admission request (ARQ) answerCall field to the terminating gatekeeper, which then validates it.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft_ctoke.htm.
Interesting Traffic PPP and Customer Profile Idle Timer
The Interesting Traffic PPP and Customer Profile Idle Timer feature supports a PPP idle timer based on interesting traffic for dialer interfaces.
Interface Index Display
The Interface Index (IfIndex) is a user-specified identification number for an interface used in SNMP network management. The IfIndex is an object in the Interfaces Group MIB (IF-MIB), which can be set by a network manager to consistently identify an interface. A new Cisco IOS software command, show snmp mib ifmib ifindex, allows the user to display the IfIndex identification numbers assigned to interfaces and subinterfaces using the CLI. The IFIndex provides a way to display these values without the need for a Network Management Station.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftshowif.htm.
IP to ATM Class of Service Mapping for SVC Bundles
The IP to ATM Class of Service Mapping for SVC Bundles feature supports multiple switched virtual circuits (SVCs) to the same NSAP destination for different types of service (ToS). This feature is an extension to the feature described in the chapter "Configuring IP to ATM Class of Service" in the Cisco IOS Quality of Service Solutions Configuration Guide. The original feature was limited to permanent virtual circuits (PVCs) only. This feature is an extension because it applies to SVCs.
The PVC bundle feature requires that the user configure PVCs for different IP ToS. The PVCs have to be set up throughout the ATM network between endpoints. The IP to ATM Class of Service Mapping for SVC Bundles feature needs configuration only at the endpoints. The user does not configure SVCs; the software sets up SVCs in a bundle between endpoints. When the router receives the first IP packet for the destination that is configured in the SVC bundle, that event triggers the creation of the SVC.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftsvbund.htm.
IPSec MIB Support for VPN Management
The IPSec MIB Support for VPN Management feature allows the monitoring of IP Security (IPSec) and Internet Key Exchange (IKE) protocols using SNMP. IPSec and IKE monitoring is especially useful in Virtual Private Networks (VPNs) supporting gateway devices and customer premises equipment (CPE).
For IPSec MIB implementation details, see the CISCO-IPSEC-MIB.my, CISCO-IPSEC-POLICY-MAP-MIB.my, and CISCO-IPSEC-FLOW-MONITOR-MIB.my files, available through the Cisco.com MIB site.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121e/121e4/dtipmib.htm.
IPv6 for Cisco IOS Software
IPv6, formerly called IPng (next generation), is a replacement for the current version of IP (version 4). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/index.htm.
IRR Triggers for GKTMP
The IRR Triggers for GKTMP feature allows a Cisco Gateway to send an information request response (IRR) to the Gatekeeper (GK) containing the details of a particular call after a successful connect. The feature also allows a back end application to set triggers for this message and the GK to deliver the IRR information to the application.
ISIS: Allows BGP to Control the Configuration of the Overload Bit
The Intermediate-System to Intermediate-System (IS-IS) protocol defines a special bit in each link-state packet (LSP) called the overload-bit. IS-IS uses the overload bit to "tell" other routers to ignore this router in their shortest path first (SPF) calculations. This function prevents transit traffic from passing through the router before the routing table has converged, and transit traffic is not lost.
L2TP IPSec
The L2TP IPSec feature provides enhanced security for tunneled PPP frames between the L2TP Access Concentrator (LAC) and the L2TP Network Server (LNS). Previous releases of the Cisco IOS provided only a one time, optional mutual authentication during tunnel setup with no authentication of subsequent data packets or control messages. In situations where L2TP is used to tunnel PPP sessions over an untrusted infrastructure such as the internet, the security attributes of L2TP and PPP are inadequate. PPP provides no protection of the L2TP tunnel, and current PPP encryption protocols provide inadequate key management and no authentication or integrity mechanisms. The L2TP IPSec feature allows the robust security features of IPSec to protect the L2TP tunnel and the PPP sessions within the tunnel. In addition, the L2TP IPSec feature provides built in keepalives and standardized interfaces for user authentication and accounting to AAA servers.
The deployment of Windows 2000 demands the integration of IPSec with L2TP as this is the default Virtual Private Dialup Network (VPDN) networking scenario. This integration of protocols is also used for LAN-to-LAN VPDN connections in Windows 2000. The L2TP IPSec feature provides integration of IPSec with L2TP in a solution that is scalable to large networks with minimal configuration.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftl2tsec.htm.
L2TP Large-Scale Dial-Out
The L2TP Large-Scale Dial-Out feature enables the router to dial multiple Layer 2 Tunnel Protocol (L2TP) access concentrators (LACs) from a single L2TP network server (LNS). The LACs are signaled through the LNS and use L2TP to establish the dial sessions. User-defined profiles can be configured on an authentication, authorization, and accounting (AAA) server and retrieved by the LNS when dial-out occurs.The L2TP Large-Scale Dial-Out feature also supports multiple LACs bound into one stack group, call traffic load balancing, and outbound call congestion management.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftl2lsdo.htm.
Leased and Switched BRI Interfaces for ETSI NET3
The Leased and Switched BRI Interfaces for ETSI NET3 feature allows one BRI B channel on an ETSI NET3 switch to be configured as a leased line, and the second B channel to be configured as a standard ISDN or dial interface and used as a switched channel to the Public Switched Telephone Network (PSTN). When the Leased and Switched BRI Interfaces for ETSI NET3 feature is configured, one B channel functions as a point-to-point 64 kbps leased line and the other B channel functions as a circuit-switched channel using the D channel to provide the signaling features available for the ETSI NET3 signaling protocol.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftlswbri.htm.
Location Confirmation Enhancements for Alternate Endpoints
The Location Confirmation (LCF) Enhancements for Alternate Endpoints feature allows a Cisco IOS Gatekeeper (GK) to collect additional routes to endpoints that are indicated by multiple LCF responses from remote GKs, and convey a collection of those routes to the requesting (calling) endpoint. Currently, the originating GK sends Location Request (LRQ) messages to multiple remote zones. Remote GKs in the zones return LCF responses to the originating GK. The LCF responses indicate alternate routes to the remote GK endpoints.
The LCF Enhancements for Alternate Endpoints feature allows the originating GK to discover and relay more possible terminating endpoints to the requesting endpoint, therefore providing alternate routes to endpoints that can be used if the best route is busy or does not provide any alternate routes. The endpoint receiving the list of alternate endpoints tries to reach them in the order in which the alternate endpoints were received. The LCF Enhancements for Alternate Endpoints feature can be used on GKs that originate LRQs and directory GKs that forward LRQ messages.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_lcfep.htm.
Low Latency Queueing
Low Latency Queueing is now supported on Cisco 820 routers. The Low Latency Queueing feature brings strict priority queueing to Class-Based Weighted Fair Queueing (CBWFQ). Strict priority queueing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic. Information about LLQ is provided in the Quality of Service Solutions Configuration Guide. For overview information, refer to the following chapter:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fqos_c/fqcprt2/qcfconmg.htm#xtocid1239530.
For configuration instructions, refer to the following chapter:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fqos_c/fqcprt2/qcfwfq.htm#xtocid2836441.
MD5 File Validation
The MD5 File Validation feature allows you to check the integrity of a Cisco IOS software image by comparing its MD5 checksum value against a known MD5 checksum value for the image. MD5 values are now made available on Cisco.com for all Cisco IOS software images for comparison against local system image values.
To perform the MD5 integrity check, execute the verify command using the new "/md5" keyword. For example, executing the verify flash:c7200-is-mz.122-2.T.bin /md5 command will calculate and display the MD5 value for the software image. Compare this value with the value available on Cisco.com for this image.
Alternatively, you can get the MD5 value from Cisco.com first, then specify this value in the command syntax. For example, executing the verify flash:c7200-is-mz.122-2.T.bin /MD5 8b5f3062c4caeccae72571440e962233 command will display a message verifying that the MD5 values match or that there is a mismatch.
A mismatch in MD5 values means that either the image is corrupt or the wrong MD5 value was entered.
MGCP 1.0 Including NCS 1.0 and TGCP 1.0 Profiles
This feature implements the following MGCP protocols on Cisco media gateways:
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The MGCP 1.0 specification and the NCS and TGCP profiles support new packages, endpoints, and event definitions. In addition, the specifications provide more detail regarding error recovery. In general, the latest edition of the MGCP specification provides guidelines for more reliable implementations of the protocol.
Media Gateway Control Protocol (MGCP) 1.0 is a protocol for the control of Voice over IP (VoIP) calls by external call-control elements known as media gateway controllers (MGCs) or call agents (CAs). It is described in the informational RFC2705, published by the Internet Engineering Task Force (IETF). MGCP 1.0 provides interoperability with a wide variety of call agents, thus enabling an extensive range of solutions.
The NCS and TGCP protocol specifications were developed through PacketCable, an industry-wide initiative to develop interoperability standards for multimedia services over cable facilities using packet technology that is led by CableLabs, an industry consortium. In Europe, the EuroPacketCable working group is ensuring that packet cable standards are available to meet European requirements and equipment characteristics.
NCS and TGCP protocol specifications contain extensions and modifications to MGCP while preserving basic MGCP architecture and constructs. NCS 1.0 is designed for use with analog, single-line user equipment on residential gateways, while TGCP 1.0 is intended for use in VoIP-to-PSTN trunking gateways in a cable environment. TGCP and NCS allow participation in packet cable solutions, but the specifications do not preclude their use in non-cable environments.
Media gateway platforms supported for this feature include:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_lcfep.htm.
MGCP Voice Gateway Interoperability with Cisco CallManager
MGCP voice gateway interoperability with Cisco CallManager allows modular access routers to act as redundant failover MGCP gateways. You can enable IP telephony and Cisco CallManager solutions using Cisco 2600 and Cisco 3600 series routers as voice gateways. This allows you to use the Cisco 2600 and 3600 platforms already in your networks as MGCP gateways within an IP telephony architecture.
An MGCP gateway handles the translation between audio signals and the packet network. The gateways interact with a call agent (also called a Media Gateway Controller or MGC) that performs signal and call processing on gateway calls.
In the MGCP configurations that Cisco IOS supports, the gateway can be any of the following:
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The call agent is either of the following:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft_mgccm.htm.
Mobile IP MIB Support for SNMP
The Mobile IP MIB Support for SNMP feature adds a MIB module which expands network monitoring capabilities of Foreign Agent (FA) and Home Agent (HA) Mobile IP Entities. Mobile IP management using SNMP is defined in two MIBs: the RFC2006-MIB and the CISCO-MOBILE-IP-MIB. The Cisco Mobile IP MIB is a Cisco enterprise-specific extension to IETF RFC 2006 MIB module which allows you to monitor the total number of HA Mobile bindings and the total number of FA visitor bindings. This release also adds support for RFC 2006 Set operations and a SNMP notification. Set operations (performed from a Network Management System) are supported for starting and stopping the mobile IP service, configuring security associations, modifying advertisement parameters, and configuring "care-of addresses" for foreign agents. An SNMP notification (trap or inform) for security violations can be enabled on supported routing devices using the snmp-server enable traps ipmobile and snmp-server host global configuration CLI commands. As this feature affects security, use of SNMPv3 is strongly recommended.
Mobile Networks
The Cisco Mobile Networks feature enables a Mobile Router and its subnets to be mobile and maintain all IP connectivity, transparent to the IP hosts connecting through this Mobile Router.
Mobile IP, as defined in standard RFC 2002, provides the architecture that enables the Mobile Router to connect back to its home network. Mobile IP allows a device to roam while appearing to be at its home network. Such a device is called a mobile node. A mobile node is a node, for example, a personal digital assistant, a laptop computer, or a data-ready cellular phone, that can change its point of attachment from one network or subnet to another. This mobile node can travel from link to link and maintain ongoing communications while using the same IP address.
The Mobile Router functions similarly to the mobile node with one key difference—the Mobile Router allows entire networks to roam. For example, a plane with a Mobile Router can fly around the world while passengers stay connected to the Internet. This communication is accomplished by Mobile IP aware routers tunneling packets, which are destined to hosts on the mobile networks, to the location where the Mobile Router is visiting. The Mobile Router then forwards the packets to the destination device.
These devices can be mobile nodes running Mobile IP client software or nodes without the software. The Mobile Router eliminates the need for a Mobile IP client. In fact, the nodes on the mobile network are not aware of any IP mobility at all. The Mobile Router "hides" the IP roaming from the local IP nodes so that the local nodes appear to be directly attached to the home network.
The Cisco Mobile Networks feature is a static network implementation that supports stub routers only. The Mobile Router avoids convergence problems by statically defining which networks it can address. The Mobile Router can do the following:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftmbrout.htm.
Mobile Networks MIB Support
The Cisco Mobile Networks MIB Support feature implements mobile node MIB groups for the monitoring and management of Cisco Mobile Network activity. Data from managed objects is returned through the use of the "show" commands described in the documentation for the "Cisco Mobile Networks" 12.2(4)T feature, or can be retrieved from a Network Management System using SNMP.
The Cisco Mobile Networks MIB Support feature implements the following mobile node (mn) groups in the Mobile IP MIB (RFC2006-MIB): the mnSystem group, the mnDiscovery group, and the mnRegistrationGroup.
For further details, refer to the RFC2006-MIB.my file, available through Cisco.com at ftp://ftp.cisco.com/pub/mibs/v2/, and RFC 2206, "The Definitions of Managed Objects for IP Mobility Support using SMIv2."
MPLS Label Switch Controller and Enhancements
The Multiprotocol Label Switching (MPLS) Label Switch Controller (LSC), combined with a slave ATM switch, supports scalable integration of IP services over an ATM network. The MPLS LSC enables the slave ATM switch to:
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This feature was originally introduced in Cisco IOS Release 11.1CT as the Tag Switch Controller. Cisco IOS Release 12.2(4)T adds support for the following changes and additions:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftmpls.htm.
MPLS Traffic Engineering (TE)—Automatic Bandwidth Adjustment for TE Tunnels
Traffic engineering automatic bandwidth adjustment provides the means to automatically adjust the bandwidth allocation for traffic engineering tunnels based on their measured traffic load.
Traffic engineering autobandwidth samples the average output rate for each tunnel marked for automatic bandwidth adjustment. For each marked tunnel, it periodically (for example, once per day) adjusts the tunnel's allocated bandwidth to be the largest sample for the tunnel since the last adjustment.
The frequency with which tunnel bandwidth is adjusted and the allowable range of adjustments is configurable on a per-tunnel basis. In addition, the sampling interval and the interval over which to average tunnel traffic to obtain the average output rate is user-configurable on a per-tunnel basis.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftbwadjm.htm.
MPLS Traffic Engineering (TE)—IP Explicit Address Exclusion
The MPLS traffic engineering Internet Protocol (IP) explicit address exclusion feature provides a means to exclude a link or node from the path for an MPLS traffic engineering label-switched path (LSP).
The feature is accessible by the ip explicit-path command that allows you to create an IP explicit path and enter a configuration submode for specifying the path. The feature adds to the submode commands the exclude-address command for specifying addresses to exclude from the path.
If the exclude-address for an MPLS traffic engineering LSP identifies a flooded link, the constraint-based shortest path firs (CSPF) routing algorithm does not consider that link when computing paths for the LSP. If the exclude-address specifies a flooded MPLS traffic engineering router ID, the CSPF routing algorithm does not allow paths for the LSP to traverse the node identified by the router ID.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftaddexc.htm.
Multiservice Interchange (MIX) Support
On the Cisco 2600 series router, the Cisco 3620 router, and the Cisco 3640 router, MIX features are software only. On the Cisco 3660 router, MIX requires the installation of a multiservice interchange card, also called a MIX module (MIX-3660-64), which provides additional functionality.
MIX features support applications that are sensitive to time delay, such as voice and video. MIX enables the combination of different types of calls on a single T1 or E1 connection, giving customers the flexibility to manage traffic through their routers efficiently, as either traditional TDM connections or in packet-based format.
On Cisco 2600 series router, MIX allows connection of TDM streams between two voice/WAN interface cards (VWICs) on the same zero-LAN 2-slot network module (NM-2W).
On all Cisco 3600 series routers, MIX allows connection of TDM streams between two voice/WAN interface cards (VWICs) on the same Fast Ethernet network module (NM-xFE2W).
On the Cisco 3660 router, the MIX module also enables the following features:
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The MIX feature also enhances extended availability drop and insert (EADI) functionality to ensure that TDM connections across slots survive a software reload if they have been saved in NVRAM. This means that the data or voice connections carried over TDM will survive even if the router goes down and comes back up again. No separate configuration is necessary for EADI, but to ensure that the TDM connections are not interrupted, their connect commands must be saved to NVRAM by writing the configuration. Other types of MIX connections, such as circuit emulation service (CES) connections and voice connections that terminate on the router, will not survive a software reboot or reload.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_24mix.htm.
NAT Support of H.323 RAS
The Cisco IOS NAT feature supports all H.225 and H.245 message types, including Registration, Admission, and Status (RAS). RAS provides a number of messages that are used by software clients and VoIP devices to register their location, request assistance in call setup, and control bandwidth. The RAS messages are directed toward an H.323 gatekeeper.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftnatras.htm.
NAT—Ability to Use Route Maps with Static Translations
The NAT—Ability to Use Route Maps with Static Translations feature provides support for NAT multihoming capability with static address translations. Support has been added for 1-to-1 static address translation only. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftnatrt.htm.
NAT—Static Mapping Support with HSRP for High Availability
The NAT—HSRP VMAC with NAT ARP Response feature allows NAT to use the HSRP Virtual MAC for ARPs. Failover is ensured without having to time out and repopulate upstream ARP caches. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftnthsrp.htm.
NAT—Translation of External IP Addresses Only
The NAT—Translation of External IP Addresses Only feature allows the configuration of Cisco IOS NAT to ignore all embedded IP addresses for any application and traffic type. It cannot be configured on a per application/traffic type basis. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftnatxip.htm.
NetFlow Multiple Export Destinations
The NetFlow Multiple Export Destinations feature enables configuration of multiple destinations of the NetFlow data. With this feature enabled, two identical streams of NetFlow data are sent to the destination host. Currently, the maximum number of export destinations allowed is two.
The NetFlow Multiple Export Destinations feature improves the chances of receiving complete NetFlow data by providing redundant streams of data. Because the same export data is sent to more than one NetFlow collector, fewer packets will be lost.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dtnfdest.htm.
NetFlow ToS-Based Router Aggregation
The NetFlow ToS-Based Router Aggregation feature provides the ability to enable limited router-based type of service (ToS) aggregation of NetFlow Export data, which results in summarized NetFlow Export data to be exported to a collection device. The result is lower bandwidth requirements for NetFlow Export data and reduced platform requirements for NetFlow data collection devices.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120s/120s15/dtnfltos.htm.
Offload Server Accounting Enhancement
The Offload Server Accounting Enhancement feature allows users to configure their access servers (NAS) to synchronize authentication and accounting information— NAS-IP-Address (attribute 4) and Class (attribute 25)—with the offload server.
An offload server interacts with an access server via Virtual Private Network (VPN) to perform required Point-to-Point Protocol (PPP) negotiation for calls. The NAS performs call preauthentication, while the offload server performs user authentication. Thus, this feature allows the authentication and accounting data of the NAS to synchronize with the offload server as follows:
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The offload server will include the new, unique session-id in user access requests and user session accounting requests. The Class attribute that was passed from the NAS will be included in the user access request, but a new Class attribute will be received in the user access reply; this new Class attribute should be included in user session accounting requests.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftoffact.htm.
Optimized PPP Negotiation
The Optimized PPP Negotiation feature optimizes the time needed for PPP negotiation when a connection is made. PPP negotiation can include several cycles before the negotiation options are acknowledged. These negotiation cycles can cause a significant user-perceived delay, especially in networks with slow links such as a wireless data connection. Additionally, the PPP negotiation time can add significantly to the total time the user stays connected in these types of connections. Changes to the PPP link control protocol (LCP) and PPP Internet Protocol Control Protocol (IPCP) negotiation strategies as part of Cisco IOS Release 12.2(4)T make a reduction in the negotiation time possible.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftcphneg.htm.
OSPF ABR Type 3 LSA Filtering
The OSPF ABR Type 3 link-state advertisement (LSA) Filtering feature extends the ability of an ABR that is running the OSPF protocol to filter type 3 LSAs between different OSPF areas. This feature allows only specified prefixes to be sent from one area to another area and restricts all other prefixes.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftabrt3f.htm.
OSPF Stub Router Advertisement
The OSPF Stub Router Advertisement feature allows you to bring a new router into a network without immediately routing traffic through the new router and allows you to gracefully shut down or reload a router without dropping packets that are destined for other networks. This feature introduces three configuration options that allow you to configure a router that is running the Open Shortest Path First (OSPF) protocol to advertise a maximum or infinite metric to all neighbors.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftospfau.htm.
OSPF Update Packet-Pacing Configurable Timers
The OSPF Update Packet-Pacing Configurable Timers feature allows you to configure the rate at which Open Shortest Path First (OSPF) link-state advertisement (LSA) flood pacing, group pacing, and retransmission pacing updates occur. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftospfct.htm.
PIM MIB Extension for IP Multicast
Protocol Independent Multicast (PIM) is an IP Multicast routing protocol used for routing multicast data packets to multicast groups. RFC 2934 defines the Protocol Independent Multicast for IPv4 MIB, which describes managed objects that enable users to remotely monitor and configure PIM using Simple Network Management Protocol (SNMP).
The PIM MIB Extension for IP Multicast feature introduces support in Cisco IOS software for the CISCO-PIM-MIB, which is an extension of RFC 2934 and an enhancement to the existing Cisco implementation of the PIM MIB. This feature introduces the following new classes of PIM notifications:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftpimmib.htm.
PPPoA/PPPoE Autosense for ATM PVCs
The PPPoA/PPPoE Autosense for ATM PVCs feature enables the router to distinguish between incoming PPP over ATM (PPPoA) and PPP over Ethernet (PPPoE) over ATM sessions and to create virtual access based on demand for both PPP types.
The PPPoA/PPPoE Autosense for ATM PVCs feature is supported on LLC-encapsulated ATM PVCs only.
This new feature also adds support for precloning of virtual access interfaces for PPPoA and PPPoE over ATM. Precloning is the allocation of a specified number of virtual access interfaces at system start. Precloning significantly reduces the load on the system during call setup. When precloning is used, the virtual-access interface is attached to the permanent virtual circuit (PVC) upon receipt of the first PPP packet from the client on the PVC. The virtual-access interface is detached from the PVC on termination of the PPP session.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftp_auto.htm.
PPPoE over Gigabit Ethernet
The PPPoE over Gigabit Ethernet feature enhances PPP over Ethernet (PPPoE) functionality by adding support for PPPoE and PPPoE over IEEE 802.1Q VLANs on Gigabit Ethernet interfaces. The PPPoE over Gigabit Ethernet feature is supported on Cisco 7200 series routers with Gigabit Ethernet line cards.
PPPoE Session Limit
The PPPoE Session Limit feature enables you to limit the number of PPP over Ethernet (PPPoE) sessions that can be created on a router or on an ATM permanent virtual circuit (PVC), PVC range, or virtual circuit (VC) class.
This new feature introduces a new command and a modification to an existing command that enable you to specify the maximum number of PPPoE sessions that can be created. The new pppoe limit max-sessions command limits the number of PPPoE sessions that can be created on the router. The modified pppoe max-sessions command limits the number of PPPoE sessions that can be created on an ATM PVC, PVC range, VC class, or Ethernet subinterface.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftppoesl.htm.
PRI Backhaul Using the Stream Control Transmission Protocol and the ISDN Q.921 User Adaptation Layer
The PRI Backhaul Using the Stream Control Transmission Protocol and the ISDN Q.921 User Adaptation Layer feature fulfills the need for a standards based PRI Signaling backhaul that works with third party Call Agents to enable solutions like Integrated Access, IP PBX, and Telecommuter.
This feature provides the following:
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These features do the following:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_0546.htm.
PRI/Q.931 Signaling Backhaul for Call Agent Applications
This feature implements PRI/Q.931 signaling backhaul support for call agent applications on the Cisco 2600 and Cisco 3600 series routers and Cisco MC3810 series access concentrators. PRI/Q.931 signaling backhaul is the transport of PRI signaling (Q.931 and above layers) between a media gateway (such as a Cisco access server, router, or concentrator) and a media gateway controller (Cisco VSC3000).
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_bhaul.htm.
PSTN Fallback for Cisco 7200 and 7500 Series Routers
The PSTN Fallback feature monitors congestion in the IP network and redirects calls to the PSTN or rejects calls on the basis of network congestion. The fallback subsystem has a network traffic cache that maintains the Calculated Planning Impairment Factor (ICPIF) or delay/loss values for various destinations. Performance is improved because each new call to a well-known destination does not have to wait on a probe to be admitted and the value is usually cached from a previous call.
This feature was originally introduced in Cisco IOS Release 12.1(3)T. With this release, support is added for the Cisco 7200 and 7500 series routers; the call fallback command is added, and the call fallback reject cause code command is added.
Refer to the following document for additional information:
http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvftrunk.html#wp1105908
RADIUS Attribute Screening
The RADIUS Attribute Screening feature allows users to configure a list of "accept" or "reject" RADIUS attributes on the network access server (NAS) for purposes such as authorization or accounting.
If a NAS accepts and processes all RADIUS attributes received in an Access-Accept packet, unwanted attributes may be processed, creating a problem for wholesale providers who do not control their customers' authentication, authorization, and accounting (AAA) servers. For example, there may be attributes that specify services to which the customer has not subscribed, or there may be attributes that may degrade service for other wholesale dial users. The ability to configure the NAS to restrict the use of specific attributes has therefore become a requirement for many users.
The RADIUS Attribute Screening feature should be implemented in one of the following ways:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftras.htm.
RADIUS Attribute 82: Tunnel Assignment ID
The RADIUS Attribute 82: Tunnel Assignment ID feature allows the Layer 2 Transport Protocol access concentrator (LAC) to group users from different per-user or domain RADIUS profiles into the same active tunnel. Previously, Cisco IOS software assigned a separate virtual private dialup network (VPDN) tunnel for each per-user or domain RADIUS profile, even if tunnels with identical endpoints already existed. The RADIUS Attribute 82: Tunnel Assignment ID feature defines a new AV pair, Tunnel-Assignment-ID, which allows the LAC to group users from different RADIUS profiles into the same tunnel if the chosen endpoint, tunnel type, and Tunnel-Assignment-ID are identical. This feature introduces new software functionality. No new commands are introduced with this feature.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftrada82.htm.
RADIUS Tunnel Preference for Load Balancing and Fail-Over
Tunnel servers may be load balanced or failed-over from a single tunnel initiator, as selected by the RADIUS Tunnel Preference for Load Balancing and Fail-Over attribute. There is no configuration associated with this feature. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftradtun.htm.
Redial Enhancements
The Redial Enhancements feature improves the performance of redial and provides greater control over redial behavior. The dialer will now cycle through all matching dialer strings or dialer maps before applying the redial interval, and may select a different physical dialer on each redial attempt. New dial-out attempts will not be initiated if a redial to the same destination is pending. The dialer can now be configured to apply a disable timer without performing any redial attempts, and a disable time can be applied to a dialer profile interface and to a serial dialer.
By default, the Cisco IOS software considers a call successful if it connects at the physical layer (Layer 1 of the Open System Interconnection [OSI] reference model). However, problems such as poor quality telco circuits or peer misconfiguration can cause dial-out failure even though a connection is made at the physical layer. The Redial Enhancements feature introduces a new command that allows the router to be configured to wait a specific amount of time for a line protocol to come up before considering a dial-out attempt successful. If the timer runs out or the call is dropped before the line protocol comes up, the call is considered unsuccessful. Unsuccessful dial-out attempts will trigger redial if the redial options have been configured.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/dialenhc.htm.
RSVP Support for Low Latency Queueing
Resource Reservation Protocol (RSVP) is a network-control protocol that provides a means for reserving network resources—primarily bandwidth—to guarantee that applications transmitting end-to-end across networks achieve the desired quality of service (QoS).
RSVP enables real-time traffic (which includes voice flows) to reserve resources necessary for low latency and bandwidth guarantees.
Voice traffic has stringent delay and jitter requirements. It must have very low delay and minimal jitter per hop to avoid degradation of end-to-end QoS. This calls for an efficient queueing implementation, such as Low Latency Queueing (LLQ), that can service voice traffic at almost strict priority in order to minimize delay and jitter.
RSVP uses weighted fair queueing (WFQ) to provide fairness among flows and to assign a low weight to a packet to attain priority. However, the preferential treatment provided by RSVP is insufficient to minimize the jitter because of the nature of the queueing algorithm itself. As a result, the low latency and jitter requirements of voice flows might not be met in the prior implementation of RSVP and WFQ.
RSVP provides admission control. However, to provide the bandwidth and delay guarantees for voice traffic and get admission control, RSVP must work with LLQ. The RSVP support for LLQ feature allows RSVP to classify voice flows and queue them into the priority queue (PQ) within the LLQ system while simultaneously providing reservations for nonvoice flows by getting a reserved queue.
Sequential LRQ Enhancement
The Sequential LRQ Enhancement feature enhances the existing sequential location request (LRQ) feature in the Cisco IOS Gatekeeper (GK) to provide a potentially faster LRQ response to the originator of the request when a location reject (LRJ) response is received while the GK is sending sequential LRQs. In the current sequential LRQ implementation on the gateway, the GK sends an LRQ to the next zone only after the sequential delay timer expires. The Sequential LRQ Enhancement feature introduces a fixed delay for the GK to send sequential LRQs to successive zones even when a negative response or an LRJ is received from the current zone. If an LRJ is received from the current zone, the GK assumes that the current zone cannot satisfy the request and immediately sends an LRQ to the next zone. This feature works for both typical and directory GKs.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftseqlrq.htm.
SNMPv3 Community MIB Support
The SNMPv3 Community MIB Support feature implements support for the SNMP Community MIB (SNMP-COMMUNITY-MIB) module, defined in RFC 2576, in Cisco IOS software.
The SNMPv1/v2c Message Processing Model and Security Model require mappings between parameters used in SNMPv1 and SNMPv2c messages and the version independent parameters used in the Simple Network Management Protocol (SNMP) architecture. The SNMP Community MIB contains objects for mapping between these community strings and version-independent SNMP message parameters.
The mapped parameters consist of the SNMPv1/v2c community name and the SNMP securityName and contextEngineID/contextName pair. This MIB provides mappings in both directions, that is, a community name may be mapped to a securityName, contextEngineID, and contextName, or the combination of securityName, contextEngineID, and contextName may be mapped to a community name. This MIB also augments the snmpTargetAddrTable with a transport address mask value and a maximum message size value.
For implementation details, refer to the SNMP-COMMUNITY-MIB.my file, available through Cisco.com at ftp://ftp.cisco.com/pub/mibs/v2/.
SS7 Four-Link Support for Cisco Signaling Link Terminal
The SS7 Four-Link Support for Cisco Signaling Link Terminal feature introduces support for up to four Cisco SS7 links on a new platform for the Cisco SLT, the Cisco 2651 Multiservice Access Router. All existing Cisco 2611-based Cisco SLT functionality is supported on the new platform, and both Cisco SLT platforms use the same Cisco IOS software image.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_4lnk.htm.
Stream Control Transmission Protocol (SCTP), Release 1
Stream Control Transmission Protocol (SCTP) is a reliable datagram-oriented IP transport protocol, specified by RFC 2960. It provides the layer between an SCTP user application and an unreliable end-to-end datagram service such as IP. The basic service offered by SCTP is the reliable transfer of user datagrams between peer SCTP users. It performs this service within the context of an association between two SCTP hosts. SCTP is connection-oriented, but SCTP association is a broader concept than the Transmission Control Protocol (TCP) connection, for example.
SCTP provides the means for each SCTP endpoint to provide its peer with a list of transport addresses, such as address and UDP port combinations, for example. This list is provided during association startup and shows the transport addresses through which the endpoint can be reached and from which messages originate. The SCTP association includes transfer over all of the possible source and destination combinations that might be generated from the two endpoint lists (also known as multihoming).
SCTP is not explicitly configured on routers, but it underlies several Cisco applications. The commands described in this document are useful for troubleshooting when SCTP issues are suspected as the cause of problems.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft_sctp.htm.
T.38 Fax Services for Cisco 1750 Access Routers
When the Cisco 1750 access router is equipped with a VFC that has one or more slots for voice interface cards (VICs), the Cisco 1750 access router supports carrier-class Voice over IP (VoIP) and fax over IP services. The VIC has Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), and BRI interfaces.
Since the Cisco 1750 access router is H.323 compliant, it supports a family of industry-standard voice codecs and provides echo cancellation and voice activity detection (VAD) and silence suppression. There is an interactive voice response (IVR) application that provides voice prompts and digit collection in order to authenticate the user and identify the call destination.
The VIC is a coprocessor card with a powerful Reduced Instruction Set Computer (RISC) engine and dedicated, high-performance digital signal processors (DSPs) modules to ensure predictable, real-time voice processing. The design enables streamlined packet forwarding. The Cisco 1750 access router supports one VIC with two voice ports.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftfaxrly.htm.
Timer and Retry Enhancements for L2TP and L2F
The L2TP & L2F Timer/Retry Enhancement feature allows the user to configure certain adjustable timers for the Layer 2 Tunnel Protocol (L2TP) and Layer 2 Forwarding (L2F) protocols. For L2F, the settings for control packet retries and control packet timeouts are now both configurable. Initial tunnel packet retries and initial tunnel packet timeouts are now configurable for both the L2F and L2TP protocols.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftretreh.htm.
Two-Rate Policer
Networks police traffic by limiting the input or output transmission rate of a class of traffic based on user-defined criteria. Policing traffic allows you to control the maximum rate of traffic sent or received on an interface, and to partition a network into multiple priority levels or class of service (CoS).
The Two-Rate Policer performs the following functions:
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With the Two-Rate Policer, you can enforce traffic policing according to two separate rates—committed information rate (CIR) and peak information rate (PIR). You can specify the use of these two rates, along with their corresponding values, by using two keywords, cir and pir, of the police command. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ft2rtplc.htm.
TX Ring Adjustment
Each permanent virtual circuit (PVC) has a hardware transmit queue, or TX ring. It is a simple FIFO queue, and on the c820 it has a default size of 16 packets. This feature allows adjustment of the size of the TX ring. If both voice and data packets are transmitted on the same PVC, the length of the TX ring must be reduced to a value of about 3 packets. This reduces delay and jitter for voice packets by decreasing the maximum number of data packets or fragments that can be in front of a voice packet inside the TX ring.
Using 31-Bit Prefixes on IPv4 Point-to-Point Links
The Using 31-Bit Prefixes on IPv4 Point-to-Point Links feature allows 31-bit prefixes to be used on IP version 4 point-to-point links. The number of IP addresses is reduced by 50 percent and the number of denial of service (DoS) attacks is also reduced. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft31addr.htm.
VPDN Group Session Limiting
Before the introduction of the VPDN Group Session Limiting feature, you could only globally limit the number of Virtual Private Dialup Network (VPDN) sessions on a router with limits applied equally to all VPDN groups. Using the VPDN-Group Session Limiting feature, you can limit the number of VPDN sessions allowed per VPDN group. This feature is implemented with the introduction of the session-limit number command in VPDN group configuration mode. VPDN group session limiting is applied after the global VPDN session limiting (which is configured via the vpdn session-limit session command in configuration mode) is enforced.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t4/ftvpdngs.htm.
Hardware Platforms and Modules Newly Supported in Cisco IOS Release 12.2(2)T
The following hardware platforms and modules are now supported in Cisco IOS Release 12.2(2)T. These platforms and modules were first introduced in earlier Cisco IOS software releases.
1-Port ADSL WAN Interface Card
The ADSL WAN interface card is a 1-port WAN interface card (WIC) for the Cisco 1700 series of modular access routers. The card provides asymmetric digital subscriber line (ADSL) high-speed digital data transfer between a single customer premises equipment (CPE) subscriber and the central office.
The ADSL WIC is compatible with the Alcatel Digital Subscriber Loop Access Multiplexer (DSLAM), the Cisco 6260 DSLAM with Flexi-line cards, and the Cisco 6130 DSLAM with Flexi-line cards. It supports ATM adaptation layer (AAL5) and various classes of quality of service (QoS) for both voice and data service.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis3600/wan_mod/index.htm.
ADSL over ISDN
Cisco 826 routers connect corporate telecommuters and small offices via Internet Service Providers (ISPs) over asymmetric digital subscriber lines (ADSLs) to corporate LANs and the Internet. The router can provide bridging and multiprotocol routing between LAN and WAN ports. Cisco 826 routers provide connectivity to an ISDN network through an ADSL port.
Cisco uBR905 Cable Access Router
The Cisco uBR905 Cable Access Router features a single F-connector interface to the cable system, four RJ-45 (10BASE-T Ethernet) hub ports, and one RJ-45 console port to connect to a laptop computer/console terminal for local Cisco IOS configuration. The Cisco uBR905 Cable Access Router also provides an onboard IPSec hardware accelerator, which provides high-performance encryption that is substantially faster than software-based encryption.
Small Office, Home Office ADSL Router
Cisco IOS Release 12.2 T supports the following Cisco SOHO series routers:
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The SOHO 76 and SOHO 77 are small office, home office (SOHO) asymmetric digital subscriber line (ADSL) routers, each with one Ethernet interface for connection to service provider networks.
The SOHO routers also provide the following key hardware features:
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WT-2750 Multipoint Broadband Wireless System
The Cisco broadband fixed wireless point-to-multipoint system is an integrated solution consisting of one headend (WT-2751 Multipoint Headend Line Card) and multiple subscriber units (WT-2755 Multipoint Subscriber Network Module). The fixed wireless point-to-multipoint subscriber unit is designed to receive radio frequency (RF) signals from the headend. It also transmits a return signal to the headend. This return signal is a point-to-point signal, so a properly installed subscriber antenna must be correctly oriented with the headend antenna to which it is transmitting.
For more information about the fixed wireless point-to-multipoint headend feature, see Point-to-Multipoint Support for the Cisco uBR7200 Series Universal Broadband Router at the following location:
http://www.cisco.com/univercd/cc/td/doc/product/wireless/bbfw/p2mp/index.htm.
The fixed wireless multipoint system incorporates Vector Orthogonal Frequency Division Multiplexing (VOFDM), so it does not always depend on line-of-sight (LOS) deployment. With VOFDM, the system allows wireless operation in obstructed, non-line-of-sight (non-LOS) environments by taking advantage of multipath signals. This can be particularly useful in urban and suburban environments.
Wireless Network Module
The NM-WMDA wireless network module installs in the network module slot of a Cisco 2600 series router. Installing a wireless network module enables the Cisco 2600 series router to act as a subscriber unit (SU) in a point-to-multipoint wireless network. It is configured through the router's system console or via the CiscoView network management system. The network module provides the control and data interface between the Cisco 2600 series digital motherboard and the radio frequency (RF) subsystem in the wireless transverter. It also provides the up/down conversion from baseband to intermediate frequency (IF). One network module supports one or two wireless transverters (main and diversity).
Microcode software images ship in Flash memory along with the system software image. When the router starts, the system software unpacks the microcode software bundle and loads the proper software on all the interface line cards.
It is possible to use a later version of microcode software than the one shipped with the Cisco IOS software from the factory. The microcode software in Flash memory is mapped to the line cards. Unless you fully understand how Cisco IOS software uses microcode software, it is important to keep the factory configuration.
The multipoint wireless modem card requires external microcode software. Information about this microcode software is available (with a Cisco.com login) at the following location:
http://www.cisco.com/cgi-bin/tablebuild.pl/rsu.
For further information regarding the network module, refer to the Cisco Network Modules Hardware Installation Guide (for Cisco 2600 series routers) for detailed installation instructions, and the Software Configuration Guide (for Cisco 2600 series routers) for an overview of network module configuration procedures and information on configuring specific network modules.
New Software Features in Cisco IOS Release 12.2(2)T
The following new features are supported in Cisco IOS Release 12.2(2)T. Some of these features may have been introduced on other hardware platforms in earlier Cisco IOS software releases.
56K CSU Support for the Cisco Signaling Link Terminal
This feature module verifies support for the WIC-1DSU-56K4 WAN interface card for support of DS0 interconnect by the Cisco Signaling Link Terminal (SLT).
The addition of the WIC-1DSU-56K4 support to the Cisco SLT provides support for DS0 interconnect to the SS7 network without the need for an external CSU/ DSU. The WIC-1DSU-56K4 interface card is a single-port serial interface card providing a 4-wire, 56/64-kbps Kb/s interface with an integrated onboard CSU/DSU. This card is a standard option for the Cisco 2600 series routers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftsltwic.htm.
Analog DID for Cisco 2600 and Cisco 3600 Series Routers
Direct Inward Dialing (DID) is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant. This service makes use of DID trunks, which forward only the last three to five digits of a phone number to the PBX. If, for example, a company has a PBX with extensions 555-1000 to 555-1999, and a caller dials 555-1234, the local CO would forward 234 to the PBX. The PBX would then ring extension 234. This entire process is transparent to the caller.
When this feature is configured, a voice-enabled Cisco 2600 and Cisco 3600 series router can receive calls from a DID trunk and connect them to the appropriate extensions. The DID state machine is identical to the E&M state machine and uses one of the following signaling types:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dt_did.htm.
ATM PVC Range and Routed Bridge Encapsulation Subinterface Grouping
In a digital subscriber line (DSL) environment, many applications require the configuration of a large number of ATM permanent virtual circuits (PVCs). The ATM PVC Range and Routed Bridge Encapsulation Subinterface Grouping feature enables you to group a number of PVCs into a PVC range in order to configure them all at once.
For applications that use multipoint subinterfaces, such as PPP over Ethernet and PPP over ATM, the PVC range is on a single multipoint subinterface. For applications that use point-to-point subinterfaces, such as routed bridge encapsulation (RBE), a point-to-point subinterface is created for each PVC in the range.
A PVC range is defined by two VPI-VCI pairs. The two virtual path identifiers (VPIs) define a VPI range, and the two virtual channel identifiers (VCIs) define a VCI range. The number of PVCs in the PVC range equals the number of VPIs in the VPI range multiplied by the number of VCIs in the VCI range.
Once the PVC range is defined, you can configure the range by using the existing interface-ATM-VC configuration commands that are also supported in ATM PVC range configuration mode. The shutdown ATM PVC range configuration mode command can be used to deactivate the range without deleting the configuration.
The ATM PVC Range and Routed Bridge Encapsulation Subinterface Grouping feature also introduces the pvc-in-range command, which allows you to explicitly configure an individual PVC within the defined range of PVCs on a multipoint subinterface. The shutdown ATM PVC-in-range configuration mode command allows you to deactivate an individual PVC within a range.
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtatmpvr.htm.
BGP Link Bandwidth
The Border Gateway Protocol (BGP) Link Bandwidth feature is used to advertise the bandwidth of an autonomous system exit link as an extended community. The BGP Link Bandwidth feature is supported by the internal BGP (iBGP) and external BGP (eBGP) multipath features. The link bandwidth extended community indicates the preference of an autonomous system exit link in terms of bandwidth. The link bandwidth extended community attribute may be propagated to all iBGP peers and used with the BGP multipath features to configure unequal cost load balancing. When a router receives a route from a directly connected external neighbor and advertises this route to iBGP neighbors, the router may advertise the bandwidth of that link.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftbgplb.htm.
Circuit Interface Identification Persistence for SNMP
The Circuit Interface MIB (CISCO-CIRCUIT-INTERFACE-MIB) provides a MIB object (cciDescr) that can be used to identify individual circuit-based interfaces for SNMP monitoring. The Circuit Interface Identification Persistence for SNMP feature maintains this user-defined name of the circuit across reboots, allowing the consistent identification of circuit interfaces. Circuit Interface Identification Persistence is enabled using the snmp mib persist circuit global configuration command.
Cisco High-Performance Gatekeeper
The Cisco High-Performance Gatekeeper feature introduces new gatekeeper functionality and modifications for facilitating carrier class reliability, security, and performance into Cisco's Voice Network solution portfolio. These H.323 standard-based features have carrier grade reliability and performance characteristics with a robust open application protocol interface to enable development of enhanced applications like voice Virtual Private Networks (VPNs) and wholesale voice solutions.
The new gatekeeper is characterized by the following:
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This feature addresses the scalability, redundancy, and performance aspects of the gatekeeper as part of the Cisco Multimedia Conference Manager (MCM) to present a complete Cisco solution. The Cisco H.323 MCM provides the network administrator with the ability to identify H.323 traffic and to apply appropriate policies. The Cisco H.323 Multimedia Conference Manager is implemented on Cisco IOS software and enables a network manager to do the following:
Limit the H.323 traffic on the LAN and WAN.
Provide user accounting for records based on the service utilization.
Inject quality of service (QoS) parameters for the H.323 traffic generated by applications such as VoIP, and data and video conferencing.
Provide the mechanism to implement security for H.323 communications.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xm/121xm_5/ft_0394.htm.
Cisco IOS Server Load Balancing
The IOS SLB feature is a Cisco IOS-based solution that provides IP server load balancing. Using the IOS SLB feature, the network administrator defines a virtual server that represents a group of real servers in a cluster of network servers known as a server farm. In this environment the clients are configured to connect to the IP address of the virtual server. The virtual server IP address is configured as a loopback address, or secondary IP address, on each of the real servers. When a client initiates a connection to the virtual server, the IOS SLB function chooses a real server for the connection based on a configured load-balancing algorithm.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/iosslb5t.htm.
Cisco Signaling Link Terminal G.732 Support
The addition of ITU-T G.732 support to the Cisco Signaling Link Terminal (SLT) is a fundamental requirement for passing homologation in many European countries. As an integral part of the Cisco Signaling Controller 2200 (SC2200) and the Cisco VSC3000 Virtual Switch Controller (VSC3000) architecture, the Cisco SLT provides the Cisco Signaling System 7 (SS7) connectivity into the SC or VSC node.
The Cisco SLT enables service providers to reliably transport Signaling System 7 (SS7) protocols across an IP network. The Cisco SLT uses the Cisco IOS SS7 SLT feature set, providing reliable interoperability with the Cisco SC2200 or the Cisco VSC3000. The Cisco SLT is responsible for terminating the Message Transfer Part (MTP) 1 and MTP 2 layers of the SS7 protocol stack. Using the Cisco Reliable User Datagram Protocol (RUDP), the Cisco SLT backhauls, or transports, upper-layer SS7 protocols across an IP network to the Cisco SC2200 or the Cisco VSC3000. The Cisco SLT is supported only on the Cisco 2611 router.
ITU-T G.732 is an extract from the ITU-T blue book describing characteristics of primary Pulse Code Modulation (PCM) multiplex equipment operating at 2048 kbit/s (E1). The requirements describing excessive bit error ratios detected by monitoring the frame alignment signal (loss of frame alignment fault conditions) and subsequent alarming actions relate to the Cisco SLT.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_g732.htm.
Cisco Quality of Service Device Manager 2.0 Support for Cisco 1700 Series Routers
QDM is now supported on Cisco 1700 series routers.
Cisco Quality of Service Device Manager (QDM) is a web-based Java application with which users can configure and monitor advanced IP-based Quality of Service (QoS) functionality within Cisco routers using a graphical user interface (GUI).
QDM 2.0 is available as a separate product download and is free of charge. If you would like to install or reinstall QDM, refer to the Release and Installation Notes for Cisco Quality of Service Device Manager 2.0 on Cisco.com for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/rtrmgmt/qdm/qdmrn20.htm.
Class-Based Marking
The Class-Based Packet Marking feature provides users with a user-friendly command-line interface (CLI) for efficient packet marking by which users can differentiate packets based on the designated markings. The Class-Based Packet Marking feature allows users to perform the following tasks:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/cbpmark2.htm
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Classifying VoIP Signaling and Media with DSCP for QoS
The Classifying VoIP Signaling and Media with DSCP for QoS feature introduces the ip qos dscp command. The ip precedence command in dial-peer configuration mode, was originally designed to allow the prioritizing of H.323 traffic and the priority used, typically higher than that of IP data traffic. There was no means, however, for the end user to configure prioritization of H.245, H.225, and SIP signaling packets, which resulted in a delay when a call was set up over a congested network.
In order to provide finer tuning of priorities, the ip precedence command has been replaced by the ip qos dscp command. If a non zero value is specified for a particular type of traffic stream, this value is stored in the DSCP (Differentiated Services Code Point) before the gateway sends the packet out its WAN interface.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_dscp.htm.
CNS Configuration Agent
CNS is a foundation technology for linking users to network services. CNS SDK accomplishes this by making applications network-aware and increasing the intelligence of the network elements. CNS SDK provides building blocks to a range of customers in market segments such as Enterprise, service provider, independent software vendors, and system integrators.
The CNS Configuration Agent supports routing devices by providing:
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Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcns_ca.htm.
CNS Event Agent
CNS is a foundation technology for linking users to network services. CNS SDK accomplishes this by making applications network-aware and increasing the intelligence of the network elements. CNS SDK provides building blocks to a range of customers in market segments such as Enterprise, service provider, independent software vendors, and system integrators.
The CNS Event Agent is part of the Cisco IOS infrastructure that allows Cisco IOS applications, for example CNS Configuration Agent, to publish and subscribe to events on a CNS Event Bus. CNS Event Agent works in conjunction with CNS Configuration Agent.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcns_ea.htm.
Control Plane DSCP for RSVP
The Control Plane DSCP Support for RSVP feature allows you to set the priority value in the type of service (ToS) byte/differentiated services (DiffServ) field in the IP header for RSVP signaling messages. The IP header functions with resource providers such as weighted fair queueing (WFQ), so that voice frames have priority over data fragments and data frames. When packets arrive in a router output queue, the voice packets are placed ahead of the data frames.
There is one new command:
ip rsvp signalling dscp [value]—Specifies the DSCP to be used on all RSVP messages sent on an interface.
There is one modified command:
show ip rsvp interface detail—The detail keyword, was added to display information about RSVP interface parameters.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dscprsvp.htm.
DF Bit Override Functionality with IPSec Tunnels
The DF Bit Override Functionality with IPSec Tunnels feature allows customers to configure the setting of the DF bit when encapsulating tunnel mode IPSec traffic on a global or per-interface level. Thus, if the DF bit is set to clear, routers can fragment packets regardless of the original DF bit setting. Refer to the following document for additional information:
http:/www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftdfipsc.htm.
DFP Support in DistributedDirector
DistributedDirector can obtain load information from Cisco LocalDirector, Catalyst 4840g, and other clients using Dynamic Feedback Protocol (DFP). This protocol allows the user to configure the DistributedDirector to communicate with various DFP agents. The DistributedDirector tells the DFP agents how often they should report load information; then the DFP agent can tell the DistributedDirector which LocalDirector cluster to remove from providing service.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/iaabu/distrdir/dtdddfp.htm.
DHCP Option 82 Support for Routed Bridge Encapsulation
The DHCP Option 82 Support for Routed Bridge Encapsulation feature provides support for the DHCP relay agent information option when ATM routed bridge encapsulation (RBE) is used.
This feature enables the DHCP relay agent to communicate information to the DHCP server using a suboption of the DHCP relay agent information option called agent remote ID. The information sent in agent remote ID includes an IP address identifying the relay agent and information about the ATM interface and the PVC over which the DHCP request came in. The DHCP server can use this information to make IP address assignments and security policy decisions.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftrbeo82.htm.
Distributed Time-Based Access Lists
Cisco IOS software allows implementation of access lists based on the time of day. To do the implementation, you create a time range that defines specific times of the day and week. The time range is identified by a name and then referenced by a function, so that those time restrictions are imposed on the function itself.
Before the introduction of the Distributed Time-Based Access Lists feature, time-based access lists were not supported on line cards for the Cisco 7500 series routers. If time-based access lists were configured, they behaved as normal access lists. If an interface on a line card was configured with access lists, the packets switched into the interface were not distributed switched through the line card but forwarded to the Route Processor for processing.
The Distributed Time-Based Access Lists feature allows packets destined for an interface configured with time-based access lists to be distributed switched through the line card.
The Distributed Time-Based Access Lists feature gives network administrators more control over permitting or denying a user access to resources. Customers can now take advantage of the performance benefits of distributed switching and the flexibility given by time-based access lists.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftdistac.htm.
DNS Server Support for NS Records
DistributedDirector has improved server load-balancing capacity with the Domain Name System (DNS) Server Support for Name Server (NS) Records feature. This feature adds support for NS records to the Cisco IOS DNS server. With this feature, the DistributedDirector can distribute the server-selection process to multiple DistributedDirectors, improving overall server capacity.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftddns.htm.
Enhanced Multilingual Support for Cisco IOS Integrated Voice Response
This feature releases the infrastructure to support Tool Command Language (TCL)-based script interpreters, which allow you to easily add new languages to your router or access server. You can add a new language by creating a TCL script that interprets prompts into a sequence of audio files or silences. The underlying Cisco IOS dynamic prompting code interfaces with the TCL script to translate the message into a sequence of URLs that point to audio files. Then, the Cisco IOS software plays the sequence of audio files as a dynamic prompt. New TCL-script language interpreters operate simultaneously with the current built-in languages: Spanish, Chinese/Mandarin, and English. Adversely, new TCL-script language interpreters can replace one or more of the built-in languages by overwriting the built-in language functionality.
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmultil.htm.
Firewall Feature Set for Cisco 820 Series Routers
The Cisco IOS Firewall feature set is available on the Cisco 820 series routers. This feature set provides the following capabilities:
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The Cisco IOS Firewall Feature Set feature module provides several sample firewall configurations, including the following examples for small-office environments:
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Frame Relay Discard Eligibility Bit Setting
The Modular QoS CLI in Cisco IOS Release 12.2(2)T has been enhanced to include matching and marking based on the Frame Relay Discard Eligibility (DE) bit. Frame Relay DE bit Matching and Marking is documented as part of the Class-Based Marking feature module.
The DE bit in the address field of a Frame Relay frame is used as a method for prioritizing the discarding of frames in congested frame relay networks. The Frame Relay DE bit has only one bit and can therefore only have two settings, 0 or 1. If congestion occurs in a Frame Relay network, frames with the DE bit set at 1 are discarded before frames with the DE bit set at 0. Therefore, important traffic should have the DE bit set at 0 and less important traffic should be forwarded with the DE bit set at 1.
The default DE bit setting is 0. The Class-Based Packet Marking feature allows users to change the DE bit setting to 1 for various traffic, giving users the option of keeping the default value of 0 or changing the value to 1. Users can therefore use the Frame Relay DE bit marking to prioritize frames in a Frame Relay network.
Frame Relay Point-Multipoint Wireless
This feature provides an end-to-end frame relay network for customers using wireless interfaces in their frame relay network. Several new commands are used to establish a virtual frame relay interface, then link it to a specific multipoint destination mac address. The configuration information is associated with a new interface type, virtual frame relay and new interface commands, interface virtual-framerelay and frame relay over radio.
Using the new interface enables Cisco uBR7200 series, Cisco 3600 and Cisco 2600 routers to provide a seamless transition from a serial interface to a multipoint frame relay interface. By implementing RFC 1315, Frame Relay DTE MIB, a virtual frame relay interface can be linked to a specific multipoint radio interface and destination MAC address. The headend (HE) router acts as a frame relay switch, receiving radio frequency signals from subscriber units. Once received, the multipoint link is switched to a serial link and then to an upstream router.
Functionality Changed for the tunnel mpls traffic-eng autoroute metric Command
The default behavior of the tunnel mpls traffic-eng autoroute metric interface configuration command has been changed in Cisco IOS Release 12.2(2)T. This command now combines the costs of all Intermediate-System to Intermediate-System (IS-IS) routes that are downstream from a Traffic Engineering (TE) tunnel into an additive path metric. IS-IS uses the additive path metric to set the metric of the TE tunnel.
FXO Answer and Disconnect Supervision
The FXO Answer and Disconnect Supervision feature enables analog FXO ports to monitor call-progress tones, and to monitor voice and fax transmissions returned from a PBX or from the PSTN.
You can configure voice ports to detect either the standard call-progress tones that are preconfigured for certain countries, or you can configure custom call-progress tone detection. Tone detection is performed by the digital signal processor (DSP) and causes a DSP event to be reported to the host software.
Answer supervision can be accomplished in two ways: by detecting battery reversal, or by detecting voice, fax, or modem tones. If an FXO voice port is connected to the PSTN, and battery reversal is supported, use the battery reversal method. Voice ports that do not support battery reversal must use the answer supervision method, in which answer supervision is triggered when the DSP detects voice, modem, or fax transmissions. Configuring answer supervision automatically enables disconnect supervision; however, you can configure disconnect supervision separately if answer supervision is not configured.
Disconnect supervision can be configured to detect call-progress tones sent by the PBX or PSTN (for example, busy, reorder, out-of-service, number-unavailable), or to detect any tone received (for example, busy tone or dial tone). When an incoming call ends, the DSP detects the associated call-progress tone, causing the analog FXO voice port to go on-hook.
You can configure disconnect tones to be detected either continuously during calls or only during call setup (before calls are answered). Detection of any tone operates only during call setup. If you configure detection of any tone, you must also enable echo cancellation to prevent disconnection due to detection of the ringback tone of the router.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_ansds.htm.
H.323 Call Redirection Enhancements
The user-to-user information element (UUIE) of the Facility message is used primarily for call redirection. The UUIE contains a field, facilityReason, that indicates the nature of the redirection. The H.323 Call Redirection Enhancements feature adds support for two of the reasons: routeCallToGatekeeper and callForwarded. It also provides a nonstandard method for using the Facility message to effect call transfer.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftcallrd.htm.
H.323 Version 2 Phase 2
Cisco H.323 Version 2 Phase 2 upgrades Cisco IOS software by adding the following optional features, and facilitates customized extensions to the Cisco gatekeeper:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t1/h323v2p2.htm
High-Performance Gatekeeper
The Cisco High-Performance Gatekeeper feature introduces new gatekeeper functionality and modifications for facilitating carrier class reliability, security, and performance into the Cisco voice network solution portfolio. These H.323 standard-based features have carrier grade reliability and performance characteristics with a robust open application protocol interface to enable development of enhanced applications like voice VPNs and wholesale voice solutions.
This feature addresses the scalability, redundancy, and performance aspects of the gatekeeper as part of the Cisco Multimedia Conference Manager (MCM) to present a complete Cisco solution. The Cisco H.323 MCM provides the network administrator with the ability to identify H.323 traffic and to apply appropriate policies.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xm/121xm_5/ft_0394.htm.
iBGP Multipath Load Sharing
When a Border Gateway Protocol (BGP) speaker router with no local policy configured receives multiple network layer reachability information (NLRI) from the internal BGP for the same destination, the router will choose one internal BGP path as the best path. The best path is then installed in the IP routing table of the router.
The Internal BGP Multipath Load Sharing feature enables the BGP speaker router to select multiple internal BGP paths as the best paths to a destination. The best paths or multipaths are then installed in the IP routing table of the router.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftbgpls.htm.
Interactive Voice Response Version 2.0 on Cisco VoIP Gateways
IVR Version 2.0 is the fourth release of IVR and TCL scripting on Cisco IOS VoIP gateways. The Cisco IVR feature (first made available in Cisco IOS Release 12.0(3)T and 12.0(7)T) provides IVR capabilities using TCL scripts.
IVR is a term that is used to describe systems that provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words, or more commonly dual tone multifrequency (DTMF) signaling. For example, when a user makes a call with a debit card, an IVR application is used to prompt the caller to enter a specific type of information, such as a PIN. After playing the voice prompt, the IVR application collects the predetermined number of touch tones (digit collection), forwards the collected digits to a server for storage and retrieval, and then places the call to the destination phone or system. Call records can be kept and a variety of accounting functions performed.
The IVR application (or script) is a voice application designed to handle calls on a voice gateway, which is a router that is equipped with Voice over IP (VoIP) features and capabilities.
The IVR feature allows an IVR script to be used during call processing. The scripts interact with the IVR software to perform the various functions. Typically, IVR scripts contain both executable files and audio files that interact with the system software.
IVR Version 2.0 is made up of several separate components in the section that follows. These new features include:
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These features add scalability and enable the IVR scripting functionality on VoIP call legs. In addition, support for RTSP enables VoIP gateways to play messages from RTSP-compliant announcement servers.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5800/12_2t/pulskynx.htm.
Interface Alias Long Name Support
The Interface Alias (ifAlias) is a user-specified description of an interface used for SNMP network management. The ifAlias is an object in the Interfaces Group MIB (IF-MIB), which can be set by a network manager to "name" an interface. The ifAlias value for an interface or subinterface can be set using the description command in interface configuration mode, or by using a Set operation from a Network Management System.
Prior to the Cisco IOS Release 12.2(2)T, ifAlias descriptions for subinterfaces were limited to 64 characters. A new Cisco IOS software command, snmp ifmib ifalias long, configures the system to handle ifAlias descriptions of up to 256 characters. IfAlias descriptions appear in the output of the show interfaces CLI command.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftshowif.htm.
Interface Index Display
The Interface Index (IfIndex) is a user-specified identification number for an interface used in SNMP network management. The IfIndex is an object in the Interfaces Group MIB (IF-MIB), which can be set by a network manager to consistently identify an interface. A new Cisco IOS software command, show snmp mib ifmib ifindex, allows the user to display the IfIndex identification numbers assigned to interfaces and subinterfaces using the CLI. The IFIndex provides a way to display these values without the need for a Network Management Station.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftshowif.htm.
IP Header Compression Enhancement—PPPoATM and PPPoFR Support
In Cisco IOS Release 12.2(2)T, IP header compression (TCP and IP/UDP/RTP) is now supported on PPP-over-ATM interfaces and PPP-over-Frame Relay interfaces. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fqos_c/fqcprt6/qcflem.htm.
IPSec and 3DES Feature Set for Cisco 820 Series Routers
The Internet Protocol Security (IPSec) feature is available on the Cisco 820 series routers. IPSec is a framework of open standards developed by the Internet Engineering Task Force (IETF) that provides security for transmission of sensitive information over unprotected networks such as the Internet. It acts at the network level and implements the following standards:
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IPSec services are similar to those provided by Cisco Encryption Technology (CET), a proprietary security solution introduced in Cisco IOS Release 11.2. (The IPSec standard was not yet available at Release 11.2.) It provides network data encryption at the IP packet level and implements the following standards:
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IPSec provides a more robust security solution and is standards-based. IPSec also provides data authentication and antireplay services in addition to data confidentiality services, and CET provides only data-confidentiality services.
The following component technologies are implemented for IPSec:
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Triple Data Encryption Standard Feature Set for Cisco 820 Series Routers
The Triple Data Encryption Standard (3DES) Cisco IOS feature is available on Cisco 820 series routers. This feature encrypts packet data. Cisco IOS software implements the mandatory 56-bit DES-Cipher Block Chaining (CBC) with an Explicit initialization vector (IV).
IPv6 for Cisco IOS Software
IPv6, formerly called IPng (next generation), is the latest version of IP that offers many benefits, such as a larger address space, over the previous version of IP (version 4). Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ipv6/index.htm.
Low Latency Queueing with Priority Percentage Support
This feature allows you to configure bandwidth as a percentage within low latency queueing (LLQ). Specifically, you can designate a percentage of the bandwidth to be allocated to an entity (such as a physical interface, a shaped ATM permanent virtual circuit (PVC), or a shaped Frame Relay PVC) to which a policy map is attached. Traffic associated with the policy map will then be given priority treatment. This feature also allows you to specify the percentage of bandwidth to be allocated to nonpriority traffic classes.
This feature modifies two existing commands—bandwidth and priority. This feature adds a new keyword to the bandwidth command—remaining percent. The feature also changes the functionality of the existing percent keyword. These changes result in the following commands for bandwidth: bandwidth percent and bandwidth remaining percent. The bandwidth percent command configures bandwidth as an absolute percentage of the total bandwidth on the interface. The bandwidth remaining percent command allows you to allocate bandwidth as a relative percentage of the total bandwidth available on the interface. This command allows you to specify the relative percentage of the bandwidth to be allocated to the classes of traffic.
This feature also adds the percent keyword to the priority command. The priority percent command indicates that the bandwidth will be allocated as a percentage of the total bandwidth of the interface. You can then specify the percentage (that is, a number from 1 to 100) to be allocated by using the percentage argument with the priority percent command.
Unlike the bandwidth command, the priority command provides a strict priority to the traffic class, which ensures low latency to high priority traffic classes. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftllqpct.htm.
Note
MGCP CAS PBX and PRI Backhaul on Cisco 7200 Series Routers
The MGCP CAS PBX and PRI Backhaul on Cisco 7200 Series Routers features extend the earlier Simple Gateway Control Protocol (SGCP) channel-associated signaling (CAS) and AAL2 support onto the merged SGCP/MGCP software base to enable various service provider solutions.
PRI/Q.931 Signaling Backhaul is the ability to reliably transport the signaling (Q.931 and above layers) from a PRI trunk that is physically connected to a media gateway (for example, a Cisco 7200 series router) to a media gateway controller (Cisco VSC3000) for processing.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_mg7xx.htm
MGCP CAS PBX and AAL2 PVC with Basic CLASS and Operator Services
The MGCP CAS PBX and AAL2 PVC software package is a solutions-oriented program that focuses on several customer gateway scenarios. These scenarios require features that address residential, business, and trunking gateway needs on a variety of hardware platforms:
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To answer these needs, the MGCP CAS PBX and AAL2 PVC feature combines and expands existing feature sets on the merged Simple Gateway Control Protocol (SGCP)/MGCP software platform as follows:
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Refer to the following documents for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmgcptk.htm.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftmgcpgr.htm.
MGCP VoIP Signaling for 1750 Series
The MGCP CAS PBX and AAL2 PVC features extend the earlier Simple Gateway Control Protocol (SGCP) Channel Associated Signaling (CAS) and AAL2 support onto the merged SGCP/MGCP software base to enable various service provider solutions. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121limit/121x/121xm/121xm_5/ftmgcpba.htm.
Mobile IP MIB Support for SNMP
The Mobile IP MIB Support for SNMP feature adds a MIB module that expands network monitoring capabilities of foreign agent (FA) and home agent (HA) mobile IP entities. Mobile IP management using SNMP is defined in two MIBs: the RFC2006-MIB and the CISCO-MOBILE-IP-MIB. The Cisco Mobile IP MIB is a Cisco enterprise-specific extension to IETF RFC 2006 MIB module that allows you to monitor the total number of HA Mobile bindings and the total number of FA visitor bindings. Cisco IOS Release 12.2(2)T also adds support for RFC 2006 Set operations and a SNMP notification. Set operations (performed from a Network Management System) are supported for starting and stopping the mobile IP service, configuring security associations, modifying advertisement parameters, and configuring "care-of addresses" for foreign agents. An SNMP notification (trap or inform) for security violations can be enabled on supported routing devices using the snmp-server enable traps ipmobile and snmp-server host global configuration CLI commands. Because this feature affects security, use of SNMPv3 is strongly recommended.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft1mip.htm.
Modem Script and System Script Support in Large-Scale Dial-Out
Modem connection and system login chat scripts are often used when asynchronous dial-on-demand routing (DDR) is configured. Currently, however, the large-scale dial-out network architecture does not allow chat scripts for a particular session to be passed through the network. Cisco IOS Release 12.2(2)T allows modem and system chat scripts to pass through large-scale dial-out networks by allocating two new authentication, authorization, and accounting (AAA) attributes for outbound service.
The AAA attributes define specific AAA elements in a user profile. Large-scale dial-out supports Cisco attribute-value (AV) pairs and TACACS+ attributes. The Modem Script and System Script Support in Large-Scale Dial-Out feature provides two new outbound service attributes for passing chat scripts: modem-script and system-script.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftlschat.htm.
MPLS Label Distribution Protocol
The Cisco MPLS label distribution protocol (LDP) allows the construction of highly scalable and flexible IP Virtual Private Networks (VPNs) that support multiple levels of services.
LDP provides a standard methodology for hop-by-hop distribution of labels in an MPLS network by assigning labels to routes that have been chosen by the underlying Interior Gateway Protocol (IGP) routing protocols. The resulting label switch paths (LSPs) forward label traffic across an MPLS backbone to particular destinations. These capabilities enable service providers to implement the Cisco MPLS-based IP VPNs and IP+ATM services across multivendor MPLS networks.
LDP enables label switching routers (LSRs) to request, distribute, and release label prefix binding information to peer routers in a network. Thus, LSRs can discover potential peers and establish LDP sessions with those peers to exchange label binding information.
LDP is a superset of the Cisco prestandard Tag Distribution Protocol (TDP), which also supports MPLS forwarding along normally routed paths. For the features that LDP and TDP share in common, the pattern of protocol exchange between network routing platforms is identical. The differences between LDP and TDP for those features supported by both protocols are largely embedded in their respective implementation details, such as the encoding of protocol messages.
This release of LDP supports both the LDP and TDP protocols and provides the means for changing an existing network from a TDP environment to an LDP environment. Thus, you can run LDP and TDP simultaneously on any router platform. The routing protocol that you select can be configured on a per-interface basis for directly connected neighbors and on a per-session basis for nondirectly connected (targeted) neighbors. In addition, an LSP across an MPLS network can be supported by LDP on some hops and by TDP on other hops.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ldp_221t.htm.
MPLS Label Distribution Protocol MIB
The MPLS label distribution protocol (LDP) MIB is an idealized label switching database that provides an effective management infrastructure for using LDP in an MPLS network.
The notation used in the MPLS LDP MIB adheres to the conventions defined in the Abstract System Notation One (ASN.1) standard, which defines an Open System Interconnection (OSI) language used in describing data types independently from particular computer structures and presentation techniques.
Each object in the MPLS LDP MIB incorporates a DESCRIPTION field that describes the meaning and usage of the object, which, together with other object characteristics, provides information that enables network administrators to monitor and control network devices, measure network performance, and collect network statistics.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ldpmib2t.htm.
MPLS Label Switching Router MIB
The MPLS Label Switching Router MIB allows you to use the Simple Network Management Protocol (SNMP) to remotely monitor a label switching router (LSR) that is using the Multiprotocol Label Switching (MPLS) technology. The MPLS-LSR-MIB mirrors the Cisco Label Switching subsystem, specifically the LSR management information that is provided by the label forwarding information base (LFIB).
The MPLS-LSR-MIB contains managed objects that support the retrieval of label switching information from a router and is based on Revision 05 of the IEFT MPLS-LSR-MIB. This implementation enables a network administrator to get information on the status, character, and performance of the following:
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In addition, the network manager can retrieve the status of cross-connect entries that associate MPLS segments together. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/lsrmibt.htm.
MPLS QoS Multi-VC Mode for PA-A3
MPLS QoS Multi-VC Mode functionality substantially enhances MPLS quality of service (QoS) capabilities. This new MPLS QoS feature enables users to map the experimental (EXP) field value of an MPLS label to an ATM virtual circuit (VC) to create "bundles" of labeled virtual circuits (LVCs). Each bundle consists of multiple LVCs, and each LVC is treated as a member of the bundle.
Each member of a bundle can be associated with any pair of ATM-connected routers in the networking environment of the user, and each member of a bundle can have a QoS different from other members of the bundle.
By means of virtual circuit bundles, differentiated services can be provided to users of MPLS-enabled service provider networks. This service differentiation is accomplished by setting an appropriate value in the EXP field in the header of each incoming packet as it is received by the provider edge (PE) router in the service provider network.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/cos1221t.htm.
MPLS Traffic Engineering MIB
SNMP agent code operating in conjunction with the MPLS TE MIB enables a standardized, SNMP-based approach to be used in managing the MPLS traffic engineering features in Cisco IOS software.
The MPLS TE MIB is based on the IETF draft MIB entitled draft-ietf-mpls-te-mib-05.txt, which includes objects describing features that support MPLS traffic engineering. This IETF draft MIB, which undergoes revisions from time to time, is being evolved toward becoming a standard. Accordingly, the Cisco implementation of the MPLS TE MIB is expected to track the evolution of the IETF draft MIB.
Slight differences between the IETF draft MIB and the implementation of the traffic engineering capabilities within Cisco IOS software require some minor translations between the MPLS TE MIB and the internal data structures of Cisco IOS software. These translations are accomplished by means of the SNMP agent code that is installed and operating on various hosts within the network. This SNMP agent code, running in the background as a low priority process, provides a management interface to Cisco IOS software.
The SNMP objects defined in the MPLS TE MIB can be displayed using any standard SNMP utility. All MPLS TE MIB objects are based on the IETF draft MI, which means that no specific Cisco SNMP application is required to support the functions and operations pertaining to the MPLS TE MIB.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/te_mib12.htm.
NAT Support of H.323 RAS
The Cisco IOS NAT feature supports all H.225 and H.245 message types, including Registration, Admission, and Status (RAS). RAS provides a number of messages that are used by software clients and VoIP devices to register their location, request assistance in call setup, and control bandwidth. The RAS messages are directed toward an H.323 gatekeeper.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftnatras.htm.
NetFlow Multiple Export Destinations
The NetFlow Multiple Export Destinations feature enables configuration of multiple destinations of the NetFlow data. With this feature enabled, two identical streams of NetFlow data are sent to the destination host. Currently, the maximum number of export destinations allowed is two.
The NetFlow Multiple Export Destinations feature improves the chances of receiving complete NetFlow data by providing redundant streams of data. Because the same export data is sent to more than one NetFlow collector, fewer packets will be lost.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dtnfdest.htm.
Network-Based Application Recognition
Network-Based Application Recognition is now supported on Cisco 1700 series routers.
As IP quality of service (QoS) technology matures and customers begin QoS deployment in production networks, new requirements for packet classification have emerged. The applications require high performance to ensure competitiveness in an increasingly fast-paced business environment. Networks provide a variety of services to ensure that mission-critical applications receive the required bandwidth for high performance. Internet-based and client/server applications make it difficult for networks to identify packets and provide the proper level of control.
Network-Based Application Recognition (NBAR) solves this level of control by adding intelligent network classification to network infrastructures. NBAR is a new classification engine that recognizes a wide variety of applications, including web-based and other difficult-to-classify protocols that utilize dynamic TCP/UDP port assignments. When an application is recognized and classified by NBAR, a network can invoke services for that specific application. NBAR ensures that network bandwidth is used efficiently by working with QoS features to provide the following features:
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NBAR introduces several new classification features as follows:
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NBAR can also classify static port protocols. Although access control lists (ACLs) can also be used for this purpose, NBAR is easier to configure and can provide classification statistics that are not available when using ACLs.
NBAR provides a special Protocol Discovery feature that determines which application protocols are traversing a network at any given time. The Protocol Discovery feature captures key statistics associated with each protocol in a network. These statistics can be used to define traffic classes and QoS policies for each traffic class.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtnbar.htm.
PPP over Ethernet Client
The PPP over Ethernet Client feature provides PPP over Ethernet (PPPoE) client support on routers or digital subscriber line (DSL) modems on customer premises.
PPPoE client is supported on ATM permanent virtual circuits (PVCs) using a dialer interface for cloning virtual access. One PVC will support one PPPoE client. Multiple PPPoE clients can run concurrently on different PVCs, but each PPPoE client must use a separate dialer interface and a separate dialer pool.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftpppoec.htm.
Preauthentication with ISDN PRI and Channel-Associated Signaling Enhancements
Preauthentication allows a Cisco network access server (NAS) to decide—on the basis of the Dialed Number Identification Service (DNIS) number—whether to answer an incoming call. When an incoming call arrives from the public network switch but before it is answered, the NAS sends the DNIS number to a RADIUS server for authorization.
The Preauthentication with ISDN PRI and Channel-Associated Signaling Enhancements feature provides additional support for preauthentication, which was introduced in a previous Cisco IOS release. For more information about preauthentication, refer to the Cisco IOS Release 12.1(3)T feature module titled Preauthentication with ISDN PRI and Channel-Associated Signaling.
This feature supports the use of attribute 44 by the RADIUS server application, which allows user authentication on the basis of the Calling Line Identification (CLID) number in the same transaction. For more information about attribute 44 and how it works with preauthentication, refer to the Cisco IOS Release 12.0(7)T feature module titled RADIUS Attribute 44 (Accounting Session ID) in Access Requests.
This feature also supports the use of new RADIUS attributes. These RADIUS attributes are configured in the RADIUS preauthentication profiles to specify preauthentication behavior. They may also be used, for instance, to specify whether subsequent authentication should occur and, if so, what authentication method should be used.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtdt1.htm.
Prefix Dial for 800 Series Routers
Cisco 803 and Cisco 804 routers now support prefix dialing. You can add a telephone prefix and create a prefix filter to the dialed number for analog telephone calls. When a telephone number is dialed through the telephone port, the router checks for prefix filters. If the router finds a match, no prefix is added to the dialed number. If no filter match is found, the router adds the user-defined prefix to the called number.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_vs800.htm.
Quality of Service for Virtual Private Networks
When packets are encapsulated by tunnel or encryption headers, Quality of Service (QoS) features are unable to examine the original packet headers and correctly classify the packets. Packets traveling across the same tunnel have the same tunnel headers, so the packets are treated identically if the physical interface is congested.
With the growing popularity of Virtual Private Networks (VPNs), the need to classify traffic within a traffic tunnel is gaining importance. QoS features have historically been unable to classify traffic within a tunnel. With the introduction of the Quality of Service for Virtual Private Networks (QoS for VPNs) feature, packets can now be classified before tunneling and encryption occur. The process of classifying features before tunneling and encryption is called preclassification.
The QoS for VPNs feature is designed for tunnel interfaces. When the new feature is enabled, the QoS features on the output interface classify packets before encryption, allowing traffic flows to be adjusted in congested environments. The end result is more effective packet tunneling.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtqosvpn.htm.
RADIUS Attribute 66 (Tunnel-Client-Endpoint) Enhancements
Virtual private networks (VPNs) use Layer 2 Forwarding (L2F) or Layer 2 Tunnel Protocol (L2TP) tunnels to tunnel the link layer of high-level protocols (for example, PPP) or asynchronous High-Level Data Link Control (HDLC)). Internet service providers (ISPs) configure their network access servers (NASs) to receive calls from users and forward the calls to the customer tunnel server. Usually, the ISP maintains only information about the tunnel server—the tunnel endpoint. The customer maintains the IP addresses, routing, and other user database functions of the tunnel server users.
The RADIUS Attribute 66 (Tunnel-Client-Endpoint) Enhancements feature adds the ability to specify the host name of the NAS—rather than the IP address of the NAS—in RADIUS attribute 66 (Tunnel-Client-Endpoint).
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t5/dtdt4.htm.
RSVP Scalability Enhancements
RSVP typically performs admission control, classification, policing, and scheduling of data packets on a per-flow basis and keeps a database of information for each flow. RSVP scalability enhancements let you select a resource provider (formerly called a quality of service (QoS) provider) and disable data-packet classification so that RSVP performs admission control only. These enhancements facilitate integration with service provider (differentiated services) networks and enables scalability across enterprise networks.
Class-based weighted fair queueing (CBWFQ) provides the classification, policing, and scheduling functions. CBWFQ puts packets into classes based on the differentiated services code point (DSCP) value in the IP header of the packet, thereby eliminating the need for per-flow state and per-flow processing.
There are two new commands:
ip rsvp data-packed classifications none—Disables data packet classification.
ip rsvp resource-provider {none | wfq interface | wfq pvc}—Configures a resource provider for an aggregate flow.
There is one modified command:
show ip rsvp interface detail—The detail keyword was added to display information about RSVP interface parameters.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/rsvpscal.htm
RSVP Support for ATM/PVCs
The RSVP Support for ATM/PVCs feature allows RSVP to function with per-PVC queueing for voice-like flows. Specifically, RSVP can install reservations on PVCs defined at the interface and subinterface levels. There is no limit to the number of PVCs that can be configured per interface or subinterface.
There are two new commands:
ip rsvp layer2 overhead [h c n]—Controls the overhead accounting performed by RSVP/WFQ when a flow is admitted onto an ATM PVC.
ip rsvp resource-provider {none | wfq interface | wfq pvc}—Configures a resource provider for an aggregate flow.
There is one modified command:
show ip rsvp interface detail—The detail keyword was added to display information about RSVP interface parameters.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/rsvp_atm.htm.
SA Agent Support for Application Monitoring, Frame Relay, VoIP, and MPLS VPN
The Cisco Service Assurance Agent (SA Agent) is a Cisco IOS software network monitoring solution. This enhancement to the Cisco SA Agent provides the following features: Application Performance Monitoring, Frame Relay Monitoring, Path Jitter, and MPLS VPN awareness.
SA Agent Application Performance Monitor (APM) operations allow the user to monitor performance of applications over a network. Monitoring the performance of network-hosted applications gives service providers and IT departments the ability to verify that applications are performing as needed and to implement improvements as necessary.
SA Agent Frame Relay Monitor (FRM) operations allow the user to monitor key performance metrics (round trip latency, packet loss, and data integrity) over Frame Relay PVCs. Proactively monitoring the performance of Frame Relay networks is essential for service providers that offer Frame Relay services.
SA Agent path echo operations have been enhanced to provide hop-by-hop jitter measurement using ICMP packets for VoIP monitoring. The Cisco SA Agent has also been enhanced to allow monitoring within MPLS Virtual Private Networks (VPNs).
Refer to the following document for additional information about the SA Agent Application Performance Monitor:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft2_apm.htm.
Refer to the following document for additional information about the SA Agent Support for Frame Relay, VoIP, and MPLS VPN Monitoring:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft1csaa.htm.
Secure Copy
The Secure Copy (SCP) feature provides a secure and authenticated method for copying router configuration or router image files. SCP relies on Secure Shell (SSH), an application and a protocol that provides a secure replacement for the Berkeley r-tools.
The behavior of SCP is similar to that of remote copy (rcp), which comes from the Berkeley r-tools suite, except that it is reliant upon SSH for security. In addition, SCP requires that AAA authorization be configured so the router can determine whether the user has the correct privilege level.
SCP allows a user logged in to Cisco IOS software to copy anything that exists in the Cisco IOS File System (IFS) to and from a router by using the copy command. A user using a remote workstation cannot perform this task.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftscp.htm.
Secure Shell Terminal-Line Access
Although Cisco IOS supports reverse Telnet, which allows users to Telnet to a certain port range that connects them to tty (asynchronous) lines, Telnet provides no security because all Telnet traffic goes over the network in the clear. The SSH Terminal-Line Access feature replaces reverse Telnet with secure shell (SSH), thereby, allowing users to configure their Cisco IOS routers securely.
The SSH Terminal-Line Access feature enables users to configure their router with secure access and perform the following tasks:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftrevssh.htm.
Shell-Based Authentication of VPDN Users
The Shell-Based Authentication of VPDN Users feature provides terminal services for VPDN users to support rollout of wholesale dial networks. Terminal services (shell login or exec login) on the network access server (NAS) provide the following capabilities:
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With the terminal services, user authentication methods other than PAP and CHAP can be applied to PPP users. With the Shell-Based Authentication of VPDN Users feature, PPP authentication data is preconfigured or entered before PPP starts. Authentication is completed without any further input from the user.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftexvpnt.htm.
SIP Diversion Header Implementation for Redirecting Number
SIP is a new protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to the ITU-T H.323 specification. SIP is defined by RFC 2543 and is used for multimedia call session setup and control over IP networks. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/sipcf2.htm.
SIP Gateway Support for Third-Party Call Control
SIP is a new protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to the ITU-T H.323 specification. SIP is defined by RFC 2543 and is used for multimedia call session setup and control over IP networks. Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/sipcf2.htm.
SLT Dual Ethernet
The Cisco SLT Dual Ethernet feature adds Cisco SLT dual Ethernet support to the virtual switch controller (VSC). This enhanced Cisco SLT support provides two IP networks and two additional Session Manager sessions (for a total of four Session Manager sessions) for improved backhaul communication. These additions increase the resilience of Cisco SLT/VSC communications by supporting two RUDP sessions from each Ethernet interface to each VSC. These VSC enhancements contribute to determining when to switch Ethernets and when to switch VSC activity.
The Cisco SLT, which is based on the Cisco 2611 Multi-Service Access Router, is shipped with two Ethernet interfaces. Until this feature was released, the Cisco SLT/VSC solution supported only one of the two Ethernet interfaces. Both Session Manager sessions needed to travel over this single Ethernet interface: This Ethernet was a single-point failure. The Cisco SLT Dual Ethernet feature supports the second Ethernet, which improves the resilience of the backhaul IP communications.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftsltdes.htm.
SLT G.732 Support
The Cisco SLT enables service providers to reliably transport Signaling System 7 (SS7) protocols across an IP network. The Cisco SLT uses the Cisco IOS SS7 SLT feature set, providing reliable interoperability with the Cisco SC2200 or the Cisco VSC3000 device. The Cisco SLT is responsible for terminating the Message Transfer Part (MTP) 1 and MTP 2 layers of the SS7 protocol stack. Using the Cisco Reliable User Datagram Protocol (RUDP), the Cisco SLT backhauls, or transports, upper-layer SS7 protocols across an IP network to the Cisco SC2200 or VSC3000 device. The Cisco SLT is supported only on the Cisco 2611 router.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_g732.htm.
SNMP Support over VPN
The SNMP Support over VPN feature allows the sending and receiving of SNMP notifications using VPN Routing Forwarding table (VRF).
SNMP is an application-layer protocol that provides a message format for communication between SNMP managers and agents.
A VPN is a network that provides high connectivity transfers on a shared system with the same usage guidelines as a private network. A VPN can be built on the Internet or on the service provider IP, Frame Relay, or ATM system.
A VRF stores per-VPN routing data. It defines the VPN membership of a customer site attached to the network access server (NAS). A VRF consists of an IP routing table, a derived Cisco Express Forwarding (CEF) table, guidelines, and routing protocol parameters that control the information that is included in the routing table.
The SNMP Support over VPN feature provides configuration commands that allow users to associate SNMP agents and managers with specific VRFs. The specified VRF is used for the sending of SNMP notifications (traps and informs) and responses between agents and managers. If a VRF is not specified, the default routing table for the VPN is used.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftnm_vpn.htm
SNMP Trap Support for the Virtual Switch Interface Master MIB
The VSI Master MIB allows you to manage and monitor the activities of the VSI components, including controllers, sessions, logical interfaces, and cross-connects. The MIB provides notifications in the form of traps when any of the VSI components change operational state, violate configured thresholds, or are added or removed.
The MIB allows you to specify which VSI components can send traps. To enable the traps for certain VSI components, you can use the MIB objects or Cisco IOS commands.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/mstrmib.htm.
Supplementary Telephone Services for the Euro-ISDN Switch
The Cisco 800 series routers now support the following plain old telephone service (POTS) features for the European Telecommunications Standards Institute (ETSI) Euro-ISDN switch type:
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Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_vs800.htm.
TCL IVR disconnect cause-code Manipulation
The leg disconnect command disconnects one or more call legs that are not part of any connection. The cause_code argument, which has been added in Cisco IOS Release 12.2(1)T, is an integer ISDN cause code for the disconnect. It is of the form di-xxx or just xxx, where xxx is the ISDN cause code. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2.htm.
Traffic Policing
The Traffic Policing feature performs the following functions:
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Traffic policing allows you to control the maximum rate of traffic transmitted or received on an interface. The Traffic Policing feature is applied when you attach a traffic policy contain the Traffic Policing configuration to an interface. A traffic policy is configured using the Modular Quality of Service Command-Line Interface (Modular QoS CLI).
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftpoli.htm.
Note
Trimble Palisade NTP Synchronization Driver for the Cisco 7200 Series Routers
The Trimble Palisade Smart Antenna can provide a signal that can by used for NTP time-synchronization of a network. The Trimble Palisade NTP Synchronization Kit can be connected to the auxiliary port of a Cisco 7200 router. The refclock (reference clock) driver provided by this feature provides the ability to receive an RTS time-stamp signal on the auxiliary port of the router.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t1/dtrimble.htm.
Using 31-bit Prefixes on IPv4 Point-to-Point Links
The Using 31-bit Prefixes on IPv4 Point-to-Point Links feature allows 31-bit prefixes to be used on IP version 4 point-to-point links. The number of IP addresses is reduced by 50 percent and the number of denial of service (DoS) attacks is also reduced. Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft31addr.htm.
Voice over ATM with AAL2 Trunking on Cisco 7200 Series Routers
Voice over ATM (VoATM)
This feature enables Cisco 7200 series routers to carry voice traffic (for example, telephone calls and faxes) over ATM networks using AAL2. AAL2 is the most bandwidth-efficient standards-based trunking method for transporting compressed voice, voice-band data, circuit-mode data, and frame-mode data over ATM infrastructures.
Transparent Common Channel Signaling (T-CCS)
The Transparent Common Channel Signaling (T-CCS) feature provides a way to interconnect PBX, key systems (KTs), and central office (CO) switches when the private integrated services network exchange (PINX) does not support Q (point of the ISDN model) Signaling (QSIG), or when the PINX uses a proprietary solution. T-CCS allows the connection of two PBXs with PRI interfaces that use one CCS protocol without the need for interpretation of CCS signaling for call processing. A PBX PRI group is transported transparently through the data network, and the feature preserves proprietary signaling. From the PBX standpoint, this signaling is accomplished through a point-to-point connection. Calls from the PINXs are not routed, but follow a preconfigured route to the destination. Frame forwarding, used with T-CCS, forwards High-Level Data Link Control (HDLC) frames over a preconfigured interface running HDLC, Frame Relay, or ATM encapsulation.
Additional Information
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ft_aal72.htm.
X.25 Annex G Session Status Change Reporting
The X.25 Annex G Session Status Change Reporting feature introduces the logging event frame-relay x25 interface configuration command, which provides console or system log notification of X.25 Annex G session status changes when an X.25 Annex G session carried over Frame Relay changes state. Before this feature was introduced, there was no notification.
This feature detects changes in session status using an X.25 Link Access Procedure, Balanced (LAPB) N2 counter. The LAPB N2 counter is the number of unsuccessful transmit attempts that are made before the link is declared down. After the N2 consecutive polled commands have not been answered, a notification is generated, indicating that the X.25 profile or context associated with the data-link connection identifier (DLCI) that is running across the failed radio link has gone down. A message is generated to the console or system log when the link goes down. A message is also generated to the console or system log when the link comes back up. The notification response time is contingent on the values assigned to the LAPB N1 counter and the LAPB T1 timer.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/ftanxg.htm.
MIBs
To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
Deprecated and Replacement MIBs
Old Cisco MIBs will be replaced in a future release. Currently, OLD-CISCO-* MIBs are being converted into more scalable MIBs without affecting existing Cisco IOS products or network management system (NMS) applications. You can update from deprecated MIBs to the replacement MIBs as shown in Table 32.
Limitations and Restrictions
Cisco 2620XM
When the traffic is sent with rate 100pps (256 bytes size), some cells are lost on the router where VWIC-1MFT-E1 is configured as ATM port. There is no workaround to this limitation. For a detailed description, see Traffic Shaping on Cisco 3810 Routers at the following URL:
http://www.cisco.com/warp/public/121/traff_shape3810.pdf
SNMP Version 1 BGP4-MIB Limitations
You may notice incorrect BGP trap OID output when using the SNMP version 1 BGP4-MIB that is available for download at ftp://ftp.cisco.com/pub/mibs/v1/BGP4-MIB-V1SMI.my. When a router sends out BGP traps (notifications) about state changes on an SNMP version 1 monitored BGP peer, the enterprise OID is incorrectly displayed as .1.3.6.1.2.1.15 (bgp) instead of .1.3.6.1.2.1.15.7 (bgpTraps). The problem is not due to any error with Cisco IOS software. This problem occurs because the BGP4-MIB does not follow RFC 1908 rules regarding version 1 and version 2 trap compliance. This MIB is controlled by IANA under the guidance of the IETF, and work is currently in progress by the IETF to replace this MIB with a new version that represents the current state of the BGP protocol. In the meantime, we recommend that you use the SNMP version 2 BGP4-MIB or the CISCO-BGP4-MIB to avoid an incorrect trap OID.
Important Notes
The following sections contain important notes about Cisco IOS Release 12.2T.
Field Notices and Bulletins
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Important Notes for Cisco IOS Release 12.2(15)T9
The following information applies to Cisco IOS Release 12.2(15)T9.
Cisco Images Deferred Because of Caveat CSCec46250
In Cisco IOS Release 12.2(15)T9, images for three Cisco platforms have been deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCec46250 (Headline:Format problem in saving DS power-level info onto nvram). The affected platforms are as follows:
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The software solution for these deferred images is Cisco IOS Release 12.2(15)T7.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveat CSCec46250
In Cisco IOS Release 12.2(15)T9, two images have been deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCec46250 (Headline:Format problem in saving upstream power-level info into NVRAM). The affected images are as follows:
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The software solution for these deferred images is Cisco IOS Release 12.2(15)T7.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(15)T8
The following information applies to Cisco IOS Release 12.2(15)T8.
Cisco Images Deferred Because of Caveats CSCec46250 and CSCin50865
In Cisco IOS Release 12.2(15)T8, images for three Cisco platforms have been deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCec46250 (Headline: Format problem in saving DS power-level info onto nvram) and Cisco caveat ID CSCin50865 (Headline: Called part hangs during the H.323 call). The affected platforms are as follows:
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The software solution for these deferred images is Cisco IOS Release 12.3(5).
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveat CSCec46250
In Cisco IOS Release 12.2(15)T8, images for three Cisco platforms have been deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCec46250 (Headline:Format problem in saving DS power-level info onto nvram). The affected platforms are as follows:
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The software solution for these deferred images is Cisco IOS Release 12.2(15)T7.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveat CSCec46250
In Cisco IOS Release 12.2(15)T8, two images have been deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCec46250 (Headline:Format problem in saving upstream power-level info into NVRAM). The affected images are as follows:
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The software solution for these deferred images is Cisco IOS Release 12.2(15)T7.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(15)T5
The following information applies to Cisco IOS Release 12.2(15)T5.
Cisco Images Deferred Because of Caveat CSCea91464
In Cisco IOS Release 12.2(15)T5, four images have been deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCea91464 (Headline: IP Packet send out 5850 was not DCEF-ed). The affected images are as follows:
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The software solution for these deferred images is Cisco IOS Release 12.2(15)T7.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(15)T4
The following information applies to Cisco IOS Release 12.2(15)T4.
Images Deferred Because of Caveats CSCea21186, CSCeb07534, CSCeb07595, and CSCeb10053
In Cisco IOS Release 12.2(15)T4, five images have been deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCea21186, CSCeb07534, CSCeb07595, and CSCeb10053. The affected images are as follows:
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With caveat CSCea21186, TACACS server host command causes reload. With caveat CSCeb07534, reset of dual LSC in node-a results in tailend LVCs created on PE in node-b. With caveat CSCeb07595, provider edge (PE) box may reload after modifying MPLS partition VCIi range on an ATM interface. With caveat CSCeb10053, RPM runs out of buffers causing SAR no_buffer errors. The software solution for these deferred images is Cisco IOS Release 12.2(15)T5.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(15)T3
The following information applies to Cisco IOS Release 12.2(15)T3.
Images Deferred Because of Caveats CSCdx08292, CSCea57593, CSCea63209, CSCea67430, CSCea72272, CSCea73441, CSCea74222, CSCea75235, CSCea78687, CSCea84387, CSCea91135, CSCeb02097, and CSCeb02520
In Cisco IOS Release 12.2(15)T3, four images have been deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdx08292, CSCea57593, CSCea63209, CSCea67430, CSCea72272, CSCea73441, CSCea74222, CSCea75235, CSCea78687, CSCea84387, CSCea91135, CSCeb02097, and CSCeb02520. The affected images are as follows:
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With caveat CSCdx08292, auto-summary and sync not turned on by default under addr-fam VRF. With caveat CSCea57593, a Cisco RPM-PR router may reload with a bus error at 0x600ED128. With caveat CSCea63209, with dual LSCs and 1:N redundancy configured, one might experience a 10+ sec data disruption when a resetcd is issued for the active/primary LSC. With caveat CSCea67430, SNMP MIB variables are accessible to VRF interfaces on the RPM. With caveat CSCea72272, configuration file goes corrupt with multiple simultaneous VTY write memory. With caveat CSCea73441, RPM Path-Check causes router reset. With caveat CSCea74222, IGP label rewrite information for remote PE is lost from CEF table on a local PE. With caveat CSCea75235, during LSC switchover, a second outage found. With caveat CSCea78687, LSNT: LDP goes up/down under congestion situation. With caveat CSVea84387, two simultaneous policy map displays cause problems. With caveat CSCea91135, RPM may stay in error state (auto recovery disabled/heartbeat going). With caveat CSCeb 02097, LSNT: saving configuration took long time. With caveat CSCeb02520, RPM-PR router configured as eLSR might reset upon execution of the show queue command where interface is of MPLS type. The software solution for these deferred images is Cisco IOS Release 12.2(15)T5.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(15)T1
The following information applies to Cisco IOS Release 12.2(15)T1.
Images Deferred Because of Caveat CSCin40652
In Cisco IOS Release 12.2(15)T1, 352 images have been deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCin40652. The affected images are as follows:
With caveat CSCin40652, Media Gateway Control Protocol (MGCP) channel-associated signaling (CAS) does not recieve path confirmation from terminating gateway. The software solution for these deferred images is Cisco IOS Release 12.2(15)T2.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(15)T
The following information applies to Cisco IOS Release 12.2(15)T.
Cisco Images Deferred Because of Caveat CSCdy01600
All ICS7700 images in Cisco IOS Release 12.2(15)T, 12.2(15)T1, 12.2(15)T2, 12.2(15)T3, 12.2(15)T4, 12.2(15)T4a, 12.2(15)T5, 12.2(15)T6, and 12.2(15)T7 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy01600 (Headline: Router fails to recognize voice cards or load running config). This caveat affects all ICS7700 images supported in the affected releases.
The software solution for these deferred images is Cisco IOS Release 12.3(2)XA.
In order to increase network availability, Cisco recommends that you upgrade affected IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco, with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveats CSCdv82735, CSCea08727, CSCea11340, CSCea17465, CSCea35454, and CSCin38050
Six images in Cisco IOS Release 12.2(15)T were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdv82735, CSCea08727, CSCea11340, CSCea17465, CSCea35454, and CSCin38050. These caveats affect the following images:
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With caveat CSCdv82735, speed/duplex cannot be hard set on FE ports connected to IP phone. With caveat CSCea08727, local-address broken in Cisco Easy VPN configuration. With caveat CSCea11340, Cisco Easy VPN web interface is broken on Cisco uBR925. With caveat CSCea17465, input queue size may go negative leading to the Cisco Easy VPN connections getting stuck on the Cisco uBR925. With caveat CSCea35454, the c2500-is-l image size is too large for maximum memory. With CSCin38050, there is wrong accounting for PPPoX SSG users. The software solution for these deferred images is Cisco IOS Release 12.2(15)T1.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(13)T1
The following information applies to Cisco IOS Release 12.2(13)T1.
Cisco 1600 Series Router Images Deferred Because of Caveat CSCdz38371
Two images in Cisco IOS Release 12.2(13)T1 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdz38371. This caveat affects the following images:
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With caveat CSCdz38371, the c1600-bk8nor2sy-1 and c1600-bk8nor2sy-mz images are too large for maximum router flash. The software solution for these deferred images is Cisco IOS Release 12.2(11)T3.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(13)T
The following information applies to Cisco IOS Release 12.2(13)T.
Configuring MD5 Authentication for BGP Peering Sessions
This document provides general information about deploying MD5 authentication for a BGP session. You can configure MD5 authentication between two BGP peers, meaning that each segment sent on the TCP connection between the peers is verified. MD5 authentication must be configured with the same password on both BGP peers; otherwise, the connection between them will not be made. Configuring MD5 authentication causes the Cisco IOS software to generate and check the MD5 digest of every segment sent on the TCP connection. If authentication is invoked and a segment fails authentication, then an error message will be displayed in the console.
Old Behavior
In previous versions of Cisco IOS software, configuring MD5 authentication for a BGP peering session was generally considered to be difficult because the initial configuration and any subsequent MD5 configuration changes required the BGP neighbor to be reset.
New Behavior
This behavior has been changed in current versions of Cisco IOS software. CSCdx23494 (integrated in Cisco IOS Release 12.2(13)T) introduced a change to MD5 authentication for BGP peering sessions. The BGP peering session does not need to be reset to maintain or establish the peering session for initial configuration or after the MD5 configuration has been changed. However, the configuration must be completed on both the local and remote BGP peer before the BGP hold timer expires. If the hold down timer expires before the MD5 configuration has been completed on both BGP peers, the BGP session will time out.
The following example enables the authentication feature between this router and the BGP neighbor at 10.108.1.1. The password that must also be configured for the neighbor is bla4u00=2nkq. The remote peer must be configured before the holddown timer expires.
router bgp 109neighbor 10.108.1.1 password bla4u00=2nkq
When the password has been configured, the MD5 key is applied to the tcp session immediately. If one peer is configured before the other, the TCP segments will be discarded on both the local and remote peers due to an authentication failure. The peer that is configured with the password will print an error message in the console similar to the following:
00:03:07: %TCP-6-BADAUTH: No MD5 digest from 10.0.0.2(179) to 10.0.0.1(11000)The time period in which the password must changed is typically the life time of a stale BGP session. When the password or MD5 key is configured, incoming TCP segments will only be accepted if the key is known. If the key is unknown on both the remote and local peer, the TCP segments will be dropped, and the BGP session will time out when the holddown timer expires.
If the BGP session has been preconfigured with a hold time of 0 seconds, no keepalive messages will be sent. The BGP session will stay up until one of the peers, on either side, tries to transmit a message (For example, a prefix update).
Note
Cisco 1600 Series Router Images Deferred Because of Caveat CSCdz38371
Two images in Cisco IOS Release 12.2(13)T were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdz38371. This caveat affects the following images:
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With caveat CSCdz38371, the c1600-bk8nor2sy-1 and c1600-bk8nor2sy-mz images are too large for maximum router flash. The software solution for these deferred images is Cisco IOS Release 12.2(11)T3.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco 3620 Series Router Images Deferred Because of Caveat CSCdz45923
Two images in Cisco IOS Release 12.2(13)T were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdz45923. This caveat affects the following images:
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With caveat CSCdz45923, Appletalk is missing from Cisco 3620 images. The software solution for this deferred image is Cisco IOS Release 12.2(15)T.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco AS5800 Images Deferred Because of Caveats CSCdz04856, CSCdz09639, CSCdz26779, and CSCdy87529
Three images in Cisco IOS Release 12.2(13)T were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdz04856, CSCdz09639, CSCdz26779, and CSCdy87529. These caveats affect the following images:
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With caveat CSCdz04856, a Cisco UPC324 dial feature card may stop accepting analog calls after running for about two hours. With caveat CSCdz09639, RS reloads at rs_set_debounce_timer after sh run.With caveat CSCdz26779, CRM shows resource active even after calls are disconnected. With caveat CSCdy87529, the Simple Network Management Protocol (SNMP) counters of a Cisco AS5800 may begin to deviate and may no longer reflect the actual number of calls when random analog and digital calls are received. The software solution for these deferred images is Cisco IOS Release 12.2(13)Tl.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Catalyst 4000 Access Gateway Module Images Deferred Because of Caveat CSCdz27525
Cisco Catalyst 4000 Access Gateway Module images were deferred in Cisco IOS Release 12.2(13)T because of a severe defect. This defect has been assigned Cisco caveat ID CSCdz27525. This caveat affects the following images:
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With caveat CSCdz27525, a Cisco Catalyst 4000 Gateway Module may experience a reload from overtemperature. The software solution for these deferred images is Cisco IOS Release 12.2(13)T2.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveat CSCdy01600
All ICS7700 images in Cisco IOS Release 12.2(13)T, 12.2(13)T1, 12.2(13)T2, 12.2(13)T3, 12.2(13)T4, 12.2(13)T5, 12.2(15)T6, 12.2(15)T7, and 12.2(15)T8 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy01600 (Headline: Router fails to recognize voice cards or load running config). This caveat affects all ICS7700 images supported in the affected releases.
The software solution for these deferred images is Cisco IOS Release 12.3(2)XA.
In order to increase network availability, Cisco recommends that you upgrade affected IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco, with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(11)T9
The following information applies to Cisco IOS Release 12.2(11)T9.
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Four images in Cisco IOS Release 12.2(11)T5, Release 12.2(11)T6, Release 12.2(11)T8, and Release 12.2(11)T9 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCec06547 (with headline: seeing MIPC timer error after boot and PIF-3-GIGE_DISABLE_GMAC_ERROR). This caveat affects the following images:
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The software solution for these deferred images is Cisco IOS Release 12.2(11)T10.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(11)T8
The following information applies to Cisco IOS Release 12.2(11)T8.
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Four images in Cisco IOS Release 12.2(11)T5, Release 12.2(11)T6, Release 12.2(11)T8, and Release 12.2(11)T9 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCec06547 (with headline: seeing MIPC timer error after boot and PIF-3-GIGE_DISABLE_GMAC_ERROR). This caveat affects the following images:
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The software solution for these deferred images is Cisco IOS Release 12.2(11)T10.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(11)T6
The following information applies to Cisco IOS Release 12.2(11)T6.
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Four images in Cisco IOS Release 12.2(11)T5, Release 12.2(11)T6, Release 12.2(11)T8, and Release 12.2(11)T9 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCec06547 (with headline: seeing MIPC timer error after boot and PIF-3-GIGE_DISABLE_GMAC_ERROR). This caveat affects the following images:
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The software solution for these deferred images is Cisco IOS Release 12.2(11)T10.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(11)T5
The following information applies to Cisco IOS Release 12.2(11)T5.
Cisco AS5850 Images Deferred Because of Caveat CSCec06547
Four images in Cisco IOS Release 12.2(11)T5, Release 12.2(11)T6, Release 12.2(11)T8, and Release 12.2(11)T9 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCec06547 (with headline: seeing MIPC timer error after boot and PIF-3-GIGE_DISABLE_GMAC_ERROR). This caveat affects the following images:
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The software solution for these deferred images is Cisco IOS Release 12.2(11)T10.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(11)T3
The following information applies to Cisco IOS Release 12.2(11)T3.
Cisco IAD2420 Images Deferred Because of Caveat CSCdz62759
Two images in Cisco IOS Release 12.2(11)T3 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdz62759. This caveat affects the following images:
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With caveat CSCdz62759, no ring-back tone when making hairpin calls between ports. The software solution for these deferred images is Cisco IOS Release 12.2(11)T4.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Important Notes for Cisco IOS Release 12.2(11)T2
The following information applies to Cisco IOS Release 12.2(11)T2.
Update to the mgcp fax t38 Command
Some Media Gateway Control Protocol (MGCP) call agents do not properly pass those portions of Session Description Protocol (SDP) messages that advertise T.38 and named service event (NSE) capabilities. As a result, gateways that are controlled by these call agents are unable to use NSEs to signal T.38 fax relay to other gateways that use NSEs. The new syntax for the mgcp fax t38 command provides a way to enable gateway-controlled T.38 fax relay between an MGCP gateway and another gateway even if the capability to use T.38 and NSEs cannot be negotiated by the MGCP call agent at call setup time. The other gateway can be H.323, Session Initiation Protocol (SIP), or MGCP.
Important Notes for Cisco IOS Release 12.2(11)T
The following information applies to Cisco IOS Release 12.2(11)T.
Cisco Catalyst 4000 Access Gateway Module Images Deferred Because of Caveat CSCdy17203
Cisco Catalyst 4000 Access Gateway Module images were deferred in Cisco IOS Release 12.2(11)T because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy17203. This caveat affects the following images:
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With caveat CSCdy17203, a Cisco Catalyst 4000 Gateway Module may experience failure to reboot. The software solution for these deferred images is Cisco IOS Release 12.2(11)T1.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco H.235 Accounting and Security Enhancements for Cisco Gateways
With the Cisco H.235 Accounting and Security Enhancements for Cisco Gateways feature, Cisco H.323 gateways support three levels of authentication:
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Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5800/sw_conf/ios_122/pul0242x.htm.
Note
Cisco Images Deferred Because of Caveat CSCdy01600
All ICS7700 images in Cisco IOS Release 12.2(11)T, 12.2(11)T1, 12.2(11)T2, 12.2(11)T3, 12.2(11)T4, 12.2(11)T5, 12.2(11)T6, 12.2(11)T7, 12.2(11)T8, and 12.2(11)T9 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy01600 (Headline: Router fails to recognize voice cards or load running config). This caveat affects all ICS7700 images supported in the affected releases.
The software solution for these deferred images is Cisco IOS Release 12.3(2)XA.
In order to increase network availability, Cisco recommends that you upgrade affected IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco, with respect to the deferred images will apply to the replacement images.
Detecting Carrier Sense Errors on the Cisco uBR905 and Cisco uBR925 Cable Access Routers
The Cisco uBR905 and Cisco uBR925 cable access routers cannot detect carrier sense errors on the four Ethernet ports that connect the router to the subscriber's local area network. This is because the four Ethernet ports are provided by an internal hub that always provides a carrier sense signal to the Cisco IOS software, even if no Ethernet devices are connected to the external ports.
In particular, this means that the dot3StatsCarrierSenseErrors attribute in ETHERLIKE-MIB (RFC 2665) will never indicate any drops in carrier of the Ethernet interface.
Dialing Number Enhancement
The Dialing Number Enhancement feature removes previous restrictions on the number of dialed digits accepted as a valid telephone number in the Called Party number information element (IE) by an interface configured for the National or International numbering types.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftdilnme.htm
Displaying Alarm Settings on the Cisco AS5800
The show vrm vdevices command displays detailed information for digital signal processors (DSPs) or a brief summary for all voice feature cards (VFCs). The display provides information such as the following: the number of channels, channels per DSP, bitmap of digital signal processor modules (DSPMs), DSP alarm statistics, and version numbers. This information is useful in monitoring the current state of the VFCs on a Cisco AS5800 Universal Access Server. In Cisco IOS Release 12.2(11)T, the alarms keyword and vfc-slot-number-for-alarms argument have been added for the show vrm vdevices command.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ft_svv.htm.
Fine-Grain Address Segmentation in Dial Peers
The Fine-Grain Address Segmentation in Dial Peers feature applies to dial plans in universal gateways that use universal ports to handle simultaneous voice and modem calls. It enables you to indicate any numbers within the range that the peer normally handles that should be rejected because they go to modems.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/fx_dpsgw.htm.
Gatekeeper Alias Registration and Address Resolution Enhancements
The Gatekeeper Alias Registration and Address Resolution Enhancements feature allows you to configure multiple prefixes for a local zone and register an endpoint belonging to multiple zone prefixes. With this feature, gatekeepers can accept a registration request (RRQ) message that has multiple E.164 aliases that use different prefixes.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftgkar.htm.
MICA and NextPort Modem Tech-Support Commands for the AS5xxx Platforms
New show tech-support modem and show tech-support spe commands are useful to the Cisco customer and Cisco customer support personnel alike. For example, when quality assurance technicians gather troubleshooting information, rather than typing in a series of commands, the technicians can simply add the output of the show tech-support modem and show tech-support spe commands to their report. Development engineers can then have a consistent output to look at when troubleshooting problems.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftmodsho.htm.
OSP Client Performance Improvement
The url command change for the OSP Client Performance Improvement feature is a minor modification in the way that the settlement providers are configured. As a result of minimum command-line interface (CLI) change in the architecture, the Open Settlement Protocol (OSP) process must be shut down before any URL change is performed.
RADIUS Debug Enhancements
The RADIUS Debug Enhancements feature provides enhanced RADIUS output. RADIUS is a distributed client/server system that secures networks against unauthorized access. In the Cisco implementation, RADIUS clients run on Cisco routers and send authentication requests to a central RADIUS server that contains all user authentication and network service access information.
The new feature provides enhanced RADIUS output display including the following:
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There is one modified command: debug radius—displays information associated with RADIUS in enhanced formats.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftdebug.htm.
SIP Media Inactivity Timer
The SIP Media Inactivity Timer feature enables Cisco gateways to monitor and disconnect Voice over IP (VoIP) calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
When RTCP reports are not received by a Cisco gateway, the SIP Media Inactivity Timer feature releases the hung session and its network resources in an orderly manner. These network resources include the gateway digital signal processor (DSP) and time-division multiplexing (TDM) channel resources that are utilized by the hung sessions. Because call signaling is sent to tear down the call, any stateful Session Initiation Protocol (SIP) proxies involved in the call are also notified to clear the state that they have associated with the hung session. The call is also cleared back through the TDM port so that any attached TDM switching equipment also clears its resources.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftsiprtp.htm.
Note
SS7 Interconnect to Lucent 1AESS Switches
The Lucent 1AESS local exchange telephone switching system was widely deployed in the 1970s across what was then the Bell System. During the past two decades, most 1AESS switches have been replaced by the next-generation digital switches, such as the Lucent 5ESS and Nortel DMS-100. While few 1AESSs remain, those still in service are generally heavily built out—about 2 to 5 percent of lines are on 1AESS switches.
Service providers that offer wholesale dial, Internet/intranet, and access Virtual Private Networks (VPNs) require remote access and expect to provide widely available service at the lowest cost. To do so, they must have Signaling System 7 (SS7) trunks to each local exchange in a service area. And for the Internet service provider (ISP) or competitive local exchange carrier (CLEC) that wants 100 percent dial coverage, interfacing to the remaining 1AESSs is mandatory.
Using SS7 signaling avoids investment in central office circuit switches, which must be used as concentration points for dial-in traffic to the access servers when ISDN PRI or in-band trunk signaling is used.
This feature provides 1AESS support for the Cisco AS5400. The configuration is on a T1 basis: one or several T1 lines are designated to support 1AESS, but no fractional T1s (FT1s) can be configured.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xa/122xa_2/ft1aessv.htm.
T1 CAS for VoIP
The T1 CAS for VoIP feature adds support for T1 channel-associated signaling (CAS) and limited support for E1 R2 signaling to the Cisco AS5800 and Cisco AS5850 with the Voice Feature Card (VFC).
CAS is the transmission of signaling information within the voice channel. Various types of CAS signaling are available in the T1 world. The most common forms of CAS signaling are loop-start, ground-start, and recEive and transMit (E&M). The biggest disadvantage of CAS signaling is its use of user bandwidth to perform signaling functions. CAS signaling is often referred to as robbed-bit-signaling because user bandwidth is being "robbed" by the network for other purposes. In addition to receiving and placing calls, CAS signaling also processes the receipt of Dialed Number Identification System (DNIS) and automatic number identification (ANI) information, which is used to support authentication and other functions.
T1 CAS capabilities have been implemented on the Cisco AS5800 and Cisco AS5850 VFC to enhance and integrate T1 CAS capabilities on common central office (CO) and PBX configurations for voice calls. The service provider application for T1 CAS includes connectivity to the public network using T1 CAS from the Cisco AS5800 or Cisco AS5850 to the end office switch. In this configuration, the Cisco AS5800 or Cisco AS5850 captures the dialed-number or called-party-number information and passes it along to the upper-level applications for interactive voice response (IVR) script selection, modem pooling, and other applications. Service providers also require access to calling party number, ANI, for user identification, for the billing account number, and in the future, for more complicated call routing.
Service providers who implement Voice over IP (VoIP) include traditional voice carriers, new voice and data carriers, and existing Internet service providers. Some of these service providers might use subscriber side lines for their VoIP connectivity to the Public Switched Telephone Network (PSTN); others will use tandem-type service provider connections.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5800/sw_conf/ios_122/ft_t1cas.htm.
Important Notes for Cisco IOS Release 12.2(8)T2
The following information applies to Cisco IOS Release 12.2(8)T2.
Use 12.2(8)T1 Version of c7200-kboot-mz Image
For Cisco IOS Release 12.2(8)T2, please use the c7200-kboot-mz image from Cisco IOS Release 12.2(8)T1. This image is available on Cisco.com.
Important Notes for Cisco IOS Release 12.2(8)T1
The following information applies to Cisco IOS Release 12.2(8)T1.
ATM OC-3 Network Modules
The ATM OC-3 Network Modules are not currently supported on the Cisco 2691, Cisco 3725, and Cisco 3745 platforms.
Cisco IGX 8400 Series URM Images Deferred Because of Caveat CSCdx41149
Three images in Cisco IOS Release 12.2(8)T1 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdx41149. This caveat affects the following images:
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With caveat CSCdx41149, a CiscoIGX84 series URM may experience an IPC failure. The software solution for these deferred images is Cisco IOS Release 12.2(8)T4.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco 7200 Series Router Limitation
The maximum number of Modular QoS CLI (MQC) Quality of Service (QoS) policy maps on a Cisco 7200 series router is limited to 256.
Important Notes for Cisco IOS Release 12.2(8)T
The following information applies to Cisco IOS Release 12.2(8)T.
Changes to Feature Support with Cisco IOS Release 12.2(8)T
Starting with Cisco IOS Release 12.2(8)T, the following features are removed from all features and feature sets for the Cisco 2600 series, Cisco 3640, and Cisco 3660 platforms:
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Starting with Cisco IOS Release 12.2(8)T, the following features are removed or support is not included on all feature sets for the Cisco 3620:
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Starting with Cisco IOS Release 12.2(8)T, the following features were removed from all "IP Plus" images and incorporated into the "Enterprise Plus" images on the Cisco 2600 only:
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Cisco IGX 8400 Series URM Images Deferred Because of Caveat CSCdx41149
Three images in Cisco IOS Release 12.2(8)T were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdx41149. This caveat affects the following images:
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With caveat CSCdx41149, a CiscoIGX84 series URM may experience an IPC failure. The software solution for these deferred images is Cisco IOS Release 12.2(8)T4.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveat CSCdy01600
All ICS7700 images in Cisco IOS Release 12.2(8)T, 12.2(8)T0a, 12.2(8)T0b, 12.2(8)T1, 12.2(8)T2, 12.2(8)T3, 12.2(8)T4, 12.2(8)T4a, 12.2(8)T5, 12.2(8)T6, 12.2(8)T7, 12.2(8)T8, 12.2(8)T9, and 12.2(8)T10 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy01600 (Headline: Router fails to recognize voice cards or load running config). This caveat affects all ICS7700 images supported in the affected releases.
The software solution for these deferred images is Cisco IOS Release 12.3(2)XA.
In order to increase network availability, Cisco recommends that you upgrade affected IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco, with respect to the deferred images will apply to the replacement images.
Enhanced Gigabit Ethernet Interface Processor Support on Cisco 7500/RSP Series
The Enhanced Gigabit Ethernet Interface Processor (GEIP+) is a single-port interface processor that, when combined with the appropriate optical fiber cable and a Gigabit Interface Converter (GBIC), provides one Gigabit Ethernet (GE) interface that is compliant with the IEEE 802.3z specification. The GE interface on aGEIP+ operates in full-duplex mode.
Refer to the following document for further information:
http://www.cisco.com/univercd/cc/td/doc/product/core/cis7505/vip1/vip4/10699dwg/index.htm.
HSRP Restructure
The output of the show standby command has been revised, making the output clearer and easier to use.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t8/fthsrp.htm.
MPLS Defects in Cisco IOS Release 12.2(8)T
Because of the following caveats, it is recommended that you do not enable MPLS and/or Tag Switching on Cisco IOS Release 12.2(8)T. There are no workarounds. You can use the current version of the software release or wait until the next renumber build.
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Feature module documentation for new MPLS features that appear in Cisco IOS Release 12.2(8)T are not supported due to the above-listed caveats.
Refer to the Field Notice at the following location for additional information:
http://www.cisco.com/warp/customer/770/fn18286.shtml.
SIP Media Inactivity Timer
The SIP Media Inactivity Timer feature enables Cisco gateways to monitor and disconnect Voice over IP (VoIP) calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
When RTCP reports are not received by a Cisco gateway, the SIP Media Inactivity Timer feature releases the hung session and its network resources in an orderly manner. These network resources include the gateway digital signal processor (DSP) and time-division multiplexing (TDM) channel resources that are utilized by the hung sessions. Because call signaling is sent to tear down the call, any stateful Session Initiation Protocol (SIP) proxies involved in the call are also notified to clear the state that they have associated with the hung session. The call is also cleared back through the TDM port so that any attached TDM switching equipment also clears its resources.
Refer to the following document for additional information:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xb/122xb_2/ftsiprtp.htm.
Important Notes for Cisco IOS Release 12.2(4)T
The following information applies to Cisco IOS Release 12.2(4)T.
Cisco 7500 Series Images Deferred Because of Caveat CSCdu01272
Twenty images in Cisco IOS Release 12.2(4)T were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdu01272. This caveat affects the following images:
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With caveat CSCdu01272, a Cisco 7500 series with a PA-MC-T3 port adapter may experience a Versatile Interface Processor (VIP) reload. The software solution for these deferred images is Cisco IOS Release 12.2(2)T1, which is available on Cisco.com.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco 15104 Optical Networking System Image Deferred
The regen-i6-mz image for the Cisco 15104 Optical Networking System has been deferred in Cisco IOS Release 12.2(4)T.
Cisco Images Deferred Because of Caveat CSCdy01600
All ICS7700 images in Cisco IOS Release 12.2(4)T, 12.2(4)T1, 12.2(4)T3, 12.2(4)T4, 12.2(4)T5, and 12.2(4)T6 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy01600 (Headline: Router fails to recognize voice cards or load running config). This caveat affects all ICS7700 images supported in the affected releases.
The software solution for these deferred images is Cisco IOS Release 12.3(2)XA.
In order to increase network availability, Cisco recommends that you upgrade affected IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco, with respect to the deferred images will apply to the replacement images.
MPLS VPN with TE and MPLS InterAS Advisory on Cisco IOS Software
Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN) functionality is compromised for the following platforms in Cisco IOS Release 12.2(4)T:
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Refer to the advisory notice at the following location:
http://www-tac.cisco.com/Support_Library/field_alerts/fn15911.html.
Important Notes for Cisco IOS Release 12.2(2)T
The following information applies to Cisco IOS Release 12.2(2)T.
Addition of the squeeze Command for Cisco 2600 and Cisco 3600 Series Routers
The squeeze command, which is used to erase all files marked for deletion on a Flash file system, is now available on Cisco 2600 and Cisco 3600 series routers.
Changes to the output attenuation Command
In Cisco IOS Release 12.2(2), the range of the output attenuation command for voice ports has changed from 0-14 to -6-14.
Cisco 820 and SOHO 70 Router Images Deferred Because of Caveat CSCds69577
Six images in Cisco IOS Release 12.2(2)T and 12.2(2)T1 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCds69577. This caveat affects the following images:
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With caveat CSCds69577, connectivity to some web sites is lost when the router terminates PPP over Ethernet. The software solution for these deferred images is Cisco IOS Release 12.2(1)XD1, which is available on Cisco.com.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Catalyst 4000 Gateway Images Deferred Because of Caveats CSCdu59093 and CSCdu63022
Three images in Cisco IOS Release 12.2(2)T and 12.2(2)T1 were deferred because of severe defects. These defects have been assigned Cisco caveat ID CSCdu59093 and CSCdu63022. These caveats affect the following images:
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With caveat CSCdu59093, a Catalyst 4000 Gateway may reload when a conference call is made. With caveat CSCdu63022, a Cisco Catalyst 4000 Gateway may not be able to be used as a conference bridge. The software solution for these deferred images is Cisco IOS Release 12.1(5)T9, which is available on Cisco.com.
To increase network availability, Cisco recommends that you upgrade affected Cisco IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected Cisco IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco with respect to the deferred images will apply to the replacement images.
Cisco Images Deferred Because of Caveat CSCdy01600
All ICS7700 images in Cisco IOS Release 12.2(2)T, 12.2(2)T1, 12.2(2)T3, and 12.2(2)T4 were deferred because of a severe defect. This defect has been assigned Cisco caveat ID CSCdy01600 (Headline: Router fails to recognize voice cards or load running config). This caveat affects all ICS7700 images supported in the affected releases.
The software solution for these deferred images is Cisco IOS Release 12.3(2)XA.
In order to increase network availability, Cisco recommends that you upgrade affected IOS images with the suggested replacement software images. Cisco will discontinue manufacturing shipment of affected IOS images. Any pending order will be substituted by the replacement software images.
Note
The terms and conditions that governed your rights and obligations and those of Cisco, with respect to the deferred images will apply to the replacement images.
Caveats for Cisco IOS Release 12.2T
Caveats describe unexpected behavior in Cisco IOS software releases. Severity 1 caveats are the most serious caveats; severity 2 caveats are less serious. Severity 3 caveats are moderate caveats, and only select severity 3 caveats are included in the caveats document.
For information on caveats in Cisco IOS Release 12.2 T, refer to the Caveats for Cisco IOS Release 12.2 T document, which lists severity 1 and 2 caveats and select severity 3 caveats for Cisco IOS Release 12.2 T and is located on Cisco.com.
The Dictionary of Internetworking Terms and Acronyms contains definitions of acronyms that are not defined in this document:
http://www.cisco.com/univercd/cc/td/doc/cisintwk/ita/index.htm
Note
Troubleshooting
The following documents provide assistance with troubleshooting your Cisco hardware and software:
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http://www.cisco.com/warp/public/108/index.shtml
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http://www.cisco.com/en/US/products/sw/iosswrel/ps1831/products_tech_note09186a00800cdd51.shtml
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http://www.cisco.com/warp/public/63/lose_config_6201.html
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http://www.cisco.com/warp/public/63/why_hang.html
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http://www.cisco.com/warp/public/63/mallocfail.shtml
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http://www.cisco.com/warp/public/63/highcpu.html
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http://www.cisco.com/warp/public/122/crashes_router_troubleshooting.shtml
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http://www.cisco.com/warp/public/63/car_rate_limit_icmp.html
