Table Of Contents
PPP over Ethernet (PPPoE) for DSL
ISDN Network Side PRI for ETSI NET5
H.323 VOICE with SS7 Connectivity
Broadband Wirleess Point to Point
FR/ATM Service Interworking (FRF.8)
FR/ATM Network Interworking (FRF.5)
Network Side PRI Signaling, Trunking, and Switching
AS5800 TCP-Clear Performance Optimization
TDM Potent: MIX-enabled 2/4/8 Port Multichannel T1/E1 Port Adapter with CSU/DSU
MC3810-V3 and MC3810-HCM6/MC3810-HCM2
Virtual Private Network (VPN) Module for the Cisco 1700 Series Routers
DS3 and E3 ATM Network Modules for the Cisco 2600 and 3600 multiservice access routers
Virtual Private Network (VPN) Modules for the Cisco 2600 and 3600 Series Routers
Port Enhanced ESCON Channel Port Adapter
TN3270 Server Connectivity Enhancements
Cisco IOS DHCP Relay Support for Unnumbered Interfaces
Cisco IOS DHCP Server Import Capability
Cisco IOS Dynamic Host Control Protocol Client
NAT—Support for PPTP in an Overload (Port Address Translation) Configuration
Cisco IOS DHCP Client—Phase II
MPLS Traffic Engineering and Enhancements
AutoInstall Using DHCP for LAN Interfaces
NAT—Support of IP Phone to Cisco CallManager
NAT—Support of H.323 v2 Call Signaling (FastConnect)
NAT—Support for NetMeeting Directory (Internet Locator Service-ILS)
Catalyst 4000 Access Gateway Module
Trace Route Enhancement for MPLS
MPLS Scalability Enhancement for LSC and ATM LSR
Cisco 2600/3600 10/100 Ethernet/Token Ring Mixed Media NMs
Frame Relay Switching Enhancements
Service Assurance Agent Enhancements
Virtual Switch Interface Master MIB
Trace Route Support in a MPLS Network
Monitoring Resource Availability on Cisco AS5x00 Universal Access Servers
Bidirectional Protocol-Independent Multicast (PIM)
Source Specific Multicast (SSM)
Router-Port Group Management Protocol (RGMP)
Conference Manager with Voice Gateway image with RSVP to ATM SVC
Express Resource Transport Protocol and TCP Header Compression (CRTP)
Common Open Policy Service for Resource Reservation Protocol
DOCSIS 1.0+ Quality-of-Service Enhancements
Distributed Low Latency Queuing
Network-Based Application Recognition (NBAR)
Class Based Policer for the DiffServ AF PHB
Pragmatic General Multicast (PGM)
Cisco 7500 Single Line Card Reload
Preauthentication with ISDN PRI and Channel Associated Signaling
Secure Shell Version 1 Integrated Client
MGCP (Media Gateway Control Protocol)
Inter-Autonomous System for MPLS VPNs
Cisco Signaling Link Terminal (SLT)
Settlement with Roaming for VoIP
H.323 Version 2 Support Phase 2
VoiceTelephony over ATM AAL2—Narrowband Trunking
VoFR/VoIP/VoATM feature parity for 26XX,36XX and 3810
Caller ID for Analog Voice Interfaces
MGCP Residential Gateway Support
Enhancements to the Session Initiation Protocol for VoIP on Cisco Access Platforms
AS5800/Voice Gateway Service Enhancement
PSTN Fallback Call Admissions Control
Advanced Voice Busyout Monitor (AVBO)
Fax Relay Packet Loss Concealment
Cisco TCL IVR Enhancements (IVR version 2.0)
Link Fragmentation and Interleaving (LFI) for FR and ATM PVCs
Enhanced Voice Services for Japan for Cisco 800 Series Routers
ISDN Progress Indicator Support for SIP Using 183 Session Progress
ICS 7750 Multiservice Route Processor 200
FR Fragmentation with Hardware Compression
CEF Switching for Routed Bridge Encapsulation
CUG Selection Facility Suppress Option
Frame Relay Switching Diagnostics
PPPoE RADIUS Port Identification
Mica PIAFS Support for PHS Wireless Networks
GSR Enhanced Full Rate Codec (EFR)
Product Bulletin No.1065
Cisco IOS Software 12.1T Early Deployment Release Series
New Features Overview
The Cisco IOS® Software Release 12.1 T series is an Early Deployment (ED) release delivering the latest Cisco IOS Software functionality and platform support. Cisco IOS Software ED Release 12.1 T series, unlike the 12.1 Major Release series, will integrate new features with every maintenance release.
This Early Deployment Technology (T) release will reach End of Engineering when the next Cisco IOS Software major release becomes available. Ongoing support for the functionality introduced in Release 12.1 T series will be carried forward into the next ED release.
Use the matrix below to quickly locate a feature of interest.
Table 3
VPN WAN Optimization WAN Services Web Scaling and Reliability WirelessWCCP Input Feature 12.1(3)T
PPP Over Ethernet Client 12.1(5)T
Frame Relay Switching Diagnostics 12.1(5)T
GSR Enhanced Full Rate Codec (EFR) 12.1(5)T
PPP Over Fast Ethernet 802.1Q 12.1(5)T
PPPoE RADIUS Port Identification 12.1(5)T
Connectivity and Scalability
Async Call Queuing
Description
Async Call Queuing solves problems that most telcos face when users vie for the same voice switch craft port resources within the telcos management network TMN's Data Communication Network (DCN). Voice switches such as the DMS have asynchronous management interface ports that need to be accessed by multiple engineers from various data centers in the network. Typically, there are three to four async ports but seven to eight users may contend for them at the same time.
Until now, telcos typically used a device from Infotron, a small niche player in the United Kingdom. This device allows rotary access to these ports. Users who tried to access these ports after initial ports were busy were put into a queue to wait for a free port. In other words, they couldn't get cleared because of busy ports; they were suspended and had to wait in queue. Cisco Async Call Queuing solves this issue for central offices.
Figure 1 Async Call Queuing
Benefits
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Telcos don't have to implement small black box solutions that lower overall operating costs
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Telcos can implement one platform, such as the 3662-DC-CO or 2600 class platforms for DCN connectivity
Platforms/Considerations
First appearance in a Cisco IOS Software "T" release: 12.1(1)T
Marketing Contact
Sanjay Bhardwaj
sbhardwa@cisco.com
PPP over Ethernet (PPPoE) for DSL
Description
PPP over Ethernet (PPPoE) provides network service providers a fast and cost-efficient means of increasing infrastructure to DSL and cable environments. PPP enables service providers to use existing RADIUS infrastructures to provide functions such as accounting, authentication, and authorization to all network users.
PPPoE is a client-initiated connection. On the CPE side, a PC with an Ethernet NIC and the PPPoE client code are established. This and other PCs are connected to the DSL or cable modem provided Ethernet segment, where ATM is applied if it is DSL, or DOCSIS is applied if it is cable.
Figure 2 PPPoE Termination and Aggregation (PTA)
Benefits
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Provides a quick and inexpensive DSL or cable solution
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PPP is terminated on a Cisco 6400 aggregation platform
Platforms/Considerations
First appearance in a Cisco IOS Software "T" release: 12.1(1)T
Marketing Contact
Ruben Rios
rurios@cisco.com
ISDN Network Side PRI for ETSI NET5
Description
The ISDN Network-Side PRI for ETSI NET5 feature enables Cisco IOS Software to replicate the public switched network interface to a PBX that is compatible with ETSI NET5.
Routers and PBXs are both traditionally CPE with respect to the public switched network interfaces. For voice over IP (VoIP) applications, it is desirable to interface routers to PBX's with the router representing the public switched network.
Enterprise organizations use the VoIP feature with Cisco products as a method to reduce long-distance costs for phone calls within and outside of their organizations. However, sometimes a call cannot be placed over VoIP, and the call needs to be placed using the PSTN. The customer then needs two devices connected to a PBX to allow some calls to be placed using VoIP and some calls to be placed over the PSTN.
The ISDN Network-Side PRI will allow Cisco ISDN-enabled routers to switch calls across interfaces as legacy phone switches do today and mimic the behavior of the legacy phone switches. As shown in Figure 3, when a call arrives at the router, it is either passed to VoIP or terminated on the router.
Figure 3 ISDN Network Side PRI
Benefits
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For Voice-over-IP applications, a customer PBX may be directly connected to a Cisco router rather than to the PSTN so that PBX station calls may be automatically routed to the IP network
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Bypassing the PSTN tariffed services, such as trunks and administration, increases cost savings when using Voice-over-IP in the network
Platforms/Considerations
First appearance in a Cisco IOS Software "T" release: 12.1(1)T
Marketing Contact
Anita Freeman
anfreema@cisco.com
Cable Interface Bundling
Description
Interface bundling aids in the distribution of IP addressing across RF line cards in a single UBR7200 chassis without adding the disadvantages of a bridged CMTS. Current software consumes an IP subnet for each RF line card, while lessening IP subnet consumption is considered critical for several customers. A solution to reduce the number of subnets consumed per UBR7200 is therefore required. Cable interface bundling allows a service provider to share one IP subnet across multiple cable interfaces that are grouped into a cable interface bundle. This eliminates the need to designate a separate IP subnet for each individual cable interface. This in turn avoids the performance, memory, and security problems that would result if a bridging solution were used to manage the subnets, especially for a large number of subscribers.
Interface bundling allows the conservation of IP subnets, while retaining its routing functionality between cable interfaces and to the egress port. Routed networks provide:
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Security—access to applications can be filtered and bandwidth limited based on Layer-3 attributes.
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Stability—broadcast and other network anomalies can be isolated to a single subnet.
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Scalability—IP addresses have a structure that is mapped onto the network topology. Layer 2 (bridged) networks have no such structure; therefore, each bridge must know every MAC address on the network; eventually network size will be limited by the size of the MAC address forwarding tables in the bridges.
Benefits
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Offers IP address conservation with the routing capabilities of the CMTS, thereby eliminating the bridged CMTS competitors' argument that a routed CMTS complicates IP addressing issues
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Simplifies IP addressing across multiple modem cards in the UBR7200 chassis
Platforms/Considerations
First appearance in a Cisco IOS Software "T" release: 12.1(1)T.
Marketing Contact
Maribea Berry
Mberry@cisco.com
H.323 VOICE with SS7 Connectivity
Description
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COT Support—COT (continuity testing) is used by the SS7 network to verify the operational status of the bearer channels it intends to use. For the H.323 VoIP with SS7 Connectivity project, the MICA modems provide tone generation and detection to support COT. A similar capability will be provided for the voice DSPs. COT support of outgoing calls requires tone generation and detection capabilities. Incoming call COT support requires loopback and transponder capability (generation of a tone in response to a received tone). H.323 VoIP with SS7 supports all four of these capabilities. COT operations are initiated (and test results reported) via special commands implemented on the NI2 ISDN interface from the signaling controller for voice and dialup Internet applications.
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Caller ID Presentation Support—In SS7 networks, caller ID (ANI) information is normally present in call setup (IAM) messages, even when a caller wants this information to remain private. The indication that the ANI is not to be displayed to the called ("B") party is contained in a field known as the presentation indicator, which is sent along with the ANI. The H.225 standard does not permit the presentation indicator to be sent in the SETUP message.
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External Interface—A new protocol type can be selected by a VoIP peer for use where presentation indicator transmission is desired.
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Destination Wild Cards—At present, the destination-pattern of a VIP peer has a limited wild-carding capability. Wild cards are limited to single digits, and must appear only at the end of the phone number. Because these VoIP peers are used to define a dialing plan under the H.323 VoIP with SS7 architecture, several VoIP peers are currently required and may limit the complexity of the dialing plan that can be achieved. While this issue is not specific to SS7 applications, such applications often have complex dialing plans.
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External Interface—The external interface will be an extension of the VoIP peer destination-pattern command.
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Caller ID Screening—As noted above, SS7 networks typically transmit restricted caller ID information. Because some H.323 endpoints may not be trusted to receive restricted ANI information, a VoIP-peer parameter will need to be added to cause the gateway to block ANI to those endpoints. This had not been a significant problem with PRI-connected gateways, because they rarely receive restricted ANI information.
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External Interface—A new VoIP-peer parameter will need to be added to control the transmission of ANI to H.323 endpoints.
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Numbering Type Support—The SS7 network deals with telephone numbers in many formats, with the format being specified by the nature of address (NOA) field. This corresponds to the numbering type field in ISDN calling and called party number information elements.
In the H.323 VoIP with SS7 system, the signaling controller maps the NOA in an incoming IAM message into the numbering type field in the SETUP message it presents to the gateway.
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External Interface—The destination-pattern command will be modified to the form: destination-pattern string [type] where string is now a more general regular expression, with the "*" function changed to another character in order to support the existing usage of "*" as an ordinary character. Type is a numbering designation such as international, national, subscriber, abbreviated, or unknown. If present, the number to be matched must be of the specified type.
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Number Rewriting Support—A rule-based number rewriting feature will provide greater flexibility in reformatting calling and called party numbers. Many regular expression substitutions can be organized into a rule set, which can be applied to a calling or called party number. These rules can be applied to outgoing calls from a VoIP peer or to incoming calls from a voice port.
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External Interface—The external interface for this feature is described in ENG-34585.
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Bearer Capability Mapping—The bearer capability in the progress indicator IE of the SETUP message is currently not faithfully carried end to end through the H.323 network. For example, similar bearer capabilities such as 3.1 kHz audio and speech are sometimes interchanged. Similarly, the progress indicator IE of the CONNECT message is not forwarded.
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These problems were noted in earlier tested and documented in CSCdk57885 and CSCdk62123 and fixed in previous Cisco IOS images. Since the impact of these bugs is almost entirely limited to H.323 VoIP w/SS7, these bug fixes were considered a part of the H.323 VoIP with SS7 feature. Bearer capability support will remain the same. Support for additional bearer capabilities, for example, 64K unrestricted data, is not included.
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External Interface—These bug fixes have no CLI or control requirement and are only visible under protocol analysis.
Benefits
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COT Support—Achieve critical part of SS7 interconnection. These are described in the H.323 VoIP with SS7 project, ENG-23686.
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Caller ID Presentation Support—Allows additional flexibility and robustness in SS7 message mapping. In order to avoid possible interoperability issues, a new H.323- "plus" variant will be defined that will support this and possibly future extensions to the H.225 protocol.
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Destination Wild Cards—Allows additional flexibility and robustness in dial peer building and functionality. A more general pattern matching scheme will be employed, including the general wildcard "*", and the use of "[]" constructs for ranges and multiple choices. In addition, in connection with numbering type support (described below), an optional parameter will be added to the destination pattern to allow it to match destinations with a specific numbering type.
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Caller ID Screening—Allows additional flexibility and robustness in call screening on the gateways.
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Numbering Type Support—Allows additional flexibility and robustness in numbering types supported. To use this information for call routing, an optional parameter will be added to the destination-pattern VoIP peer parameter to allow selection of the VoIP peer based on numbering type as well as destination number.
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Number Rewriting Support—Allows additional flexibility and robustness in number re-writing, allowing a carrier or service provider to "normalize" a number in different ways.
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Bearer Capability Mapping—Allows additional flexibility and robustness in bearer capability translations, correcting bugs identified in specific inter-working situations.
Platforms/Considerations
First appearance in a Cisco IOS Software "T" release: 12.1(1)T
Marketing Contact
Don Pitchford
dpitchfo@cisco.com
Broadband Wirleess Point to Point
Description
Point-to-point wireless is defined as a fixed dedicated wireless link from one site to another.
Broadband wireless is a system that delivers tens of megabits of data, using microwave or millimeter-wave radio technology.
This product delivers 44-Mbps full-duplex point-to-point (PTP) fixed-site data in a 12-MHz RF channel in the MMDS band (2.500 to 2.690 GHz) and the U-NII band (5.725 to 5.825 GHz). Future phases will deliver lower data rates in other frequency bands.
The PTP wireless router is be an integrated solution, which will consist of a base router (Cisco universal broadband router 7246), a wireless modem line card, an RF head unit, a power-feed panel, cables, and antenna subsystems. This PTP wireless router will be deployed in a service provider's "network cloud" environment, as well as in enterprise campus locations. Thus, the target customers for the PTP wireless router will be both the service provider channel and the enterprise end user.
The PTP wireless router system will be positioned as another option in the broadband access portfolio of Cisco solutions. Currently, these solutions consist of Digital Subscriber Lines (DSL) and cable technologies.
The PTP link is a "last-mile-access" link to reach local customers. It is not targeted for a "long-haul" or "backbone" application. The distinction is in the performance requirements normally assigned to a "last-mile" link in a telecommunications network. Of course this does not prevent it from being used in a "backbone" or "backhaul" application when the performance meets specified needs.
Benefits
•
The first broadband wireless solution to provide high speed wireless connections in obstructed environments
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Cisco IOS integrated router solution, seamless interconnect between the wireless path and other router connections; manage one network
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Fiber quality performance, 10^-11 bit-error-rate (BER) performance for data, 10^-8 for voice plus forward error correction (FEC) and automatic repeat request (ARQ) capability
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Software modem: configurable radio settings through remote interface
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Full simple network management protocol (SNMP) support with on-board diagnostics and path monitoring
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(1)T.
Marketing Contact
Mitch Taylor
Mittaylo@cisco.com
FR/ATM Service Interworking (FRF.8)
Description
Frame Relay (FR)/ATM service interworking (IW) is a technique to allow communication between a Frame Relay end user and an ATM end user. The technique is based upon the FRF.8 (Frame Relay Forum) Implementation Agreement (IA). FRF.8 specifies that a Frame Relay end station may communicate with an ATM end station provided that there is a router that is performing FRF.8 in software between the two end stations. FRF.8 essentially allows protocol translation of Frame Relay traffic to ATM traffic. It is for permanent virtual circuit (PVC)-based networks only and requires a one-to-one mapping of FR PVCs to ATM PVCs.
Benefits
•
Allows communication between a FR end station and an ATM end station
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Based upon the FRF.8 standard.
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Requires a one-to-one mapping of FR PVCs to ATM PVCs
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(2)T.
Marketing Contact
Sanjay Bhardwaj
sbhardwa@cisco.com
Figure 4 FR/ATM Service Interworking (FRF.8)
FR/ATM Network Interworking (FRF.5)
Description
FR/ATM IW is a technique to allow transport of Frame Relay traffic through an ATM network. The technique is based upon the FRF.5 implementation agreement specifies that two Frame Relay end stations may communicate with each other through an ATM network between the end stations. This works provided that there are routers at each side of the end stations that are performing FRF.5 in software between the FR and ATM ports. FRF.5 essentially allows tunneling of Frame Relay traffic through an ATM network. It is for based networks only and allows multiple FR PVCs to be multiplexed onto a single ATM PVC.
Benefits
•
Allows communication between FR end stations through an ATM network
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Based upon the FRF.5 standard
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Allows multiple FR PVCs to be multiplexed onto a single ATM PVC
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(2)T.
Marketing Contact
Sanjay Bhardwaj
sbhardwa@cisco.com
Figure 5 FR/ATM Network Interworking (FRF.5)
AAA Server Group Dead Timer
Description
In previous Cisco IOS Software releases, a limited configuration may be applied to a AAA server group that includes the list of AAA servers in the server group. The AAA Server Group feature allows users to select a subset of servers from a global host list and use the server group for a particular service. The currently supported server host types are RADIUS and TACACS+ server hosts.
A global configuration sets the deadtime in minutes when to stop waiting for a non-responding server.
With the AAA Server Group Dead Timer feature the deadtime configuration will be configured per server group. The deadtime attribute is supported only for RADIUS hosts. If the deadtime is defined globally, the local server group deadtime configuration will override the global configuration.
Benefits
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Before the introduction of the AAA Server Group Dead Timer feature, the server deadtime attribute could be configured only as a unique, global attribute in Cisco IOS AAA.
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This feature allows you to fully configure a server in the server group. And it allows you to configure each dead server timer per server group. Thus, you are no longer limited to a global configuration when configuring a server group.
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(2)T.
Marketing Contact
Anita Freeman
anfreema@cisco.com
Network Side PRI Signaling, Trunking, and Switching
Description
The Network side PRI Signaling, Trunking, and Switching feature enables Cisco IOS Software to replicate the public switched network interface to a PBX. Network Side PRI enables the NAS to provide a standard ISDN PRI network-side interface to the PBXs and mimic the behavior of legacy phone switches. To a PBX, the NAS functions as a National ISDN PRI switch type or an ETSI PRI Net5 switch. No change in PBX capability or behavior is required.
Call switching using dial peers enables Cisco (voice-over-IP) VoIP gateways to switch voice and data calls between different interfaces based on the dial peer matching. An incoming call is matched against configured dial peers, and based on the configured called number, the outgoing interface is selected. Any call that arrives from an ISDN PRI network side on a supported platform is either terminated on the NAS, switched to an IP network, or switched to the PSTN, depending on the configuration. A dial peer is an addressable call endpoint identified, for example, by a phone number or a port number. In VoIP, there are two kinds of dial peers: POTS and VoIP. The Cisco AS5800 access server switches both voice and data calls. The Cisco AS5300 access server switches only voice calls.
The Trunk Group Resource Manager (TGRM) supports the logical grouping, configuration, and joint management of one or more PRI interfaces. The TGRM is used to store configuration information and to accept or select an interface from a trunk group when requested. A trunk group is provisioned as the target of a dial peer, and the TGRM transparently selects the specific PRI interface and channels to use for incoming or outgoing calls. A trunk group can include any number of PRI interfaces, but all the interfaces in a trunk group must use the same type of signaling.
The Class of Restrictions (COR) functionality provides the ability to deny certain call attempts based on the incoming and outgoing class of restrictions provisioned on the dial peers. This functionality provides flexibility in network design, allows users to block calls (for example, to 900 numbers), and applies different restrictions to call attempts from different originators. COR is used to specify which incoming dial peer can use which outgoing dial peer to make a call.
In Cisco IOS Software Release 12.1(2)XH, the trunking and COR parts of this feature are available only on the Cisco AS5800. The remainder of the feature is platform independent.
Benefits
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The Network Side PRI Signaling, Switching, and Trunking feature allows you to bypass PSTN tariffed services such as trunking and administration, extending the cost savings of VoIP.
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It allows your PBXs to be connected directly to a Cisco NAS, so PBX station calls can be routed automatically to the IP network without the need for special IP telephones.
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It provides flexibility in network design.
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It also enables you to block calls selectively based on the called number or the calling number.
Platforms/Considerations
First appearance in a Cisco IOS Software Release: 12.1(3)T.
Marketing Contact
Anita Freeman
anfreema@cisco.com
NTT PRI NFAS
Description
NTT PRI NFAS adds the NTT switch type to the existing ISDN PRI NFAS with D Channel Backup feature.
ISDN Non-Facility Associated Signaling (NFAS) allows multiple ISDN Primary Rate Interfaces (PRIs) to be controlled by a single D channel. The NTT switch type for NFAS does not use an associated D channel for backup.
Once the channelized T1 controllers 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.
Benefits
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Addition of the NTT switch type makes NFAS available in geographic areas where NTT switches are available.
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Use of a single D channel to control multiple PRI interfaces can free one B channel on each interface to carry other traffic.
Platforms/Considerations
First appearance in a Cisco IOS Software Release: 12.1(3)T.
Marketing Contact
Anita Freeman
anfreema@cisco.com
PPP over ATM SVC's
Description
As DSL deployments scale to beyond the trial stage of <1000 customers to upwards of millions of customers, efficiencies must be brought about in the provisioning of the service. Currently, DSL service has been standardized on PPP over ATM. In provisioning a customer, PVC's are "nailed up" between the subscriber and the Network Access Provider (NAP) or, in some cases, all the way to the Network Service Provider (NSP). The cost of provisioning these subscribers with PVC's, even as GUI based provisioning alternatives are developed, are untenable from the service provider's (esp. NAP) viewpoint. Enabling service providers with PPP over ATM SVC's will reduce this cost as well as enhance the manageability of DSL services.
Support for PPP over ATM SVC's will also help NAP's provide services to end-users by providing a method of destination selection as well as QoS selection. Based upon the invoked address, a VC will be set up that sets up a PPP session for a particular level of bandwidth and service (VBR, UBR etc.) as well as to the particular network of interest (e.g. ISP1, ISP2, Corp1, Corp2). With telecommuting and home networking becoming more prolific, support for multiple destinations will become a standard requirement.
From an end-to-end perspective, the Cisco DSL product line is able to leverage PPP over ATM SVC's from either the NAP or the NSP perspective. While, the CPE and DSLAM will have very little to offer, the aggregation device in the NAP and the termination and service delivery device in the NSP (either the Cisco 6400 or 7200 series in both cases) will need to have SVC functionality. On the client side, Microsoft is building SVC capability into Windows 98 an extending it in Win2000 (includes ILMI support). With this feature, Cisco leads the market in enabling end-to-end SVC's. At the SVC termination point, layer 2 service selection of this type can then lead to layer 3 selection at the NSP (provided for by Vulcan/SSG functionality in the Cisco 6400).
Benefits
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PVC's no longer need to be "nailed up" between the subscriber and the Network Access Provider (NAP)
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Network Scaling-Ease of management with no PVC definitions SVC's allocate resources more dynamically
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Can reduce network operating costs.
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Help's the NAP's provide services to end-users by providing a method of destination selection as well as QoS selection.
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Window 98 and Win2000 compatible
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(3)T.
Marketing Contact
Sanjay Bhardwaj
sbhardwa@cisco.com
AS5800 TCP-Clear Performance Optimization
Description
The feature provides inbound and outbound performance optimization for service providers who provide ports to America Online (AOL) using the Cisco AS5800. This feature enables an AS5800 to support 1344 (2 x T3) clear-channel Telnet sessions, otherwise known as "TCP-Clear". Each TCP-Clear session carries data for a single AOL user.
Outbound TCP-Clear traffic handling is now event-driven and processed at interrupt level. In addition, the flow control algorithm is enhanced to handle the higher volume of traffic and to eliminate some out-of-resource conditions that could result in abnormal termination of the session.
Inbound TCP-Clear traffic has the same optimizations as outbound, and scanning for special characters is also eliminated. This scanning is required on a normal Telnet session to process Telnet control sequences but is CPU intensive. In TCP-Clear, no Telnet control sequences are ever sent. In addition, a Nagle algorithm is used to form the inbound data stream into larger packets, minimizing packet-processing overhead.
Benefits
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Wholesale dial service providers with AS5800s can now run handle AOL user loads to 100 percent of system capacity
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Each TCP-Clear/AOL session requires far less CPU thereby leaving more processing capability for other traffic types running on the same system
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Any mixture of PPP/AOL traffic can be handled by the AS5800
Platforms/Considerations
First appearance in a Cisco IOS Software Release: 12.1(3)T.
Marketing Contact
Mathew Lodge
mlodge@cisco.com
AS58-324UPC-CC
Description
The current AS5800 universal access server can support up to 1344 modems. With the addition of the 324 port UPC card, the AS5800 can be populated with more than 2000 modems. In the US market this represents a change in density from two CT3s to three CT3s. In E1 countries, the AS5800 can support up to 64 E1 trunks.
The UPC modem card can be installed in an AS5800 that already includes the current 144 port modem cards or the older 72 port modem cards. Current features such as modem pooling, port level diagnostics, and hot-swap capability are available on the UPC card. The UPC card uses the Cisco Nextport modem technology which is shared with the new AS5400 access server platform.
Benefits
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It is interoperable with current modem cards protects customer investment.
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Increased density per card allows higher port densities in the AS5800 dial shelf.
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The addition of higher-density per modem cards allows more voice cards to be installed in the dial shelf when using the AS5800 as a combined voice and data platform.
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Operation of the UPC is consistent with the operation of previous modem cards, which reduces operator retraining and eases troubleshooting in mixed modem card configurations.
Platforms/Considerations
First appearance in a Cisco IOS Software Release: 12.1(3)T.
Marketing Contact
Tony Gerbic
tgerbic@cisco.com
General Packet Radio Service
Description
General Packet Radio Service (GPRS) is defined and standardized by the European standards body ETSI. GPRS is a packet-based (IP/X.25) data service for GSM networks. The GPRS network essentially consists of two major elements, the serving GPRS support node (SGSN) and gateway GPRS support node (GGSN).
Cisco GGSN is the gateway to IP-based network and services, whether it is the public Internet or a corporate intranet. The GGSN provides authenticated access for a mobile subscriber for resources in the IP domain, whether it is to connect to a corporate network or an ISP, for data services such as e-mail, Web browsing, or other Internet-based applications. Cisco GGSN is a software solution on standard Cisco IOS router platform. Thus, the Cisco GGSN leverages all the benefits of Cisco IOS technology such as support for routing protocols and features such as DHCP, NAT, VPN, IPSec., and so on.
Benefits
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Fewer routers required to support the same number of areas-this feature makes it possible for one Cisco router to support multiple Level 1 areas, as opposed to the single Level 1 area previously supported by each router
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Network scaling-because a single Cisco router is capable of supporting up to 29 Level 1 areas plus one Level 2 area, expansion of networks using multi-area IS-IS routing is simpler
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Connectivity for local Level 1 areas on the same router-this feature also provides connectivity between Level 1 areas local to the router, previously, Level 1 areas could be connected only by using the Level 2 backbone
Platforms/Considerations
Platforms supported are 7200 VXR,
75xx, and 36XX platforms will be supported in subsequent releases.
First appearance in a Cisco IOS Software "T" release: 12.1(3)T
Marketing Contact
Manoj Goel
mgoel@cisco.com
Distributed FRF.11/.12
Description
FRF.11 provides a standards-base voice transport, and FRF.12 provides a standards-based data fragmentation mechanism over Frame Relay. When voice and data frames are interleaved in a Frame Relay network using low-speed links or low CIR values, fragmentation of large data frames becomes necessary to avoid excessive delays experienced by real-time traffic such as voice.
Distributed FRF.11/12 provides the above functionality for the VIP on the Cisco 7500 series router. This feature moves the voice encapsulation and the data fragmentation function into the distributed CEF path in the VIPs, increasing the overall performance and scalability of the system. This occurs by offloading the central processor from the memory and computing intensive tasks such as fragmentation and reassembly of data frames, as well as voice encapsulation. Another important advantage of confining the handling of voice and data packets to the VIP is the reduced latency by shortening the processing path inside the system
With Distributed FRF11/12 feature, the Cisco 7500 series router becomes a strong play in the multiservice and voice aggregation solution space.
Benefits
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Provides a standards-based solution for transporting voice over Frame Relay and data fragmentation in a Frame Relay network
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Improves voice quality for voice over Frame Relay, by reducing end-to-end delay
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Provides investment protection to the Cisco 7500 customers interested in migrating from TDM network to packet networks
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(5)T.
Marketing Contact
Harbans Kaur
harbkaur@cisco.com
TDM Potent: MIX-enabled 2/4/8 Port Multichannel T1/E1 Port Adapter with CSU/DSU
Description
The MIX-enabled 2/4/8 port multichannel T1/E1 port adapter supports 2, 4, 8, T1/E1 ports on the same port adapter. With a maximum of 48 ports supported on a single Cisco 7206VXR, this port adapter provides industry-leading density voice aggregation. This port adapter provides an ideal solution for New World voice applications for customers migrating from TDM networks to packet networks.
Multiservice Interchange (MIX) adds TDM connection capabilities for the Cisco 7200VXR series routers. The adapter takes advantage of the TDM connection capabilities of the Cisco 7200 VXR and provides voice, data integration on the same physical interface, Drop and Insert (Cross-Connect) TDM functionality, and DSP Farming.
Drop and Insert of DS0s and NxDS0s time-slots are provided between T1/E1 interfaces in the port adapter, providing efficient aggregation of different traffic types using fewer interfaces.
This port adapter makes efficient use of DSP farm supported on the 2 T1E1 digital voice port adapter, PA-VXC-2TE1. The DSP farm on the PA-VXC-2TE1 can now be shared with the MIX-enabled 2/4/8 port multichannel T1/E1 port adapter, providing a maximum of 20 T1 or 16 E1 digital voice interfaces. Various signaling protocols are supported, including T1 and E1 CAS, E1 R2, ISDN PRI, and QSIG.
Benefits
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High-Density TDM Trunk Aggregation—a maximum of 48 T1/E1 interfaces are supported
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Drop and Insert Capability (D & I)—DS0 D & I between any interface within the port adapter provides efficient traffic aggregation and reduction in WAN links
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DSP Farm—the adapters share the DSP resources on PA-VXC-2TE1 port adapters in the Cisco 7200 VXR router series, providing low-cost, high-density voice termination; 20 T1 or 16 E1 worth of voice traffic can be terminated on a single Cisco 7206 VXR router
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(5)T.
Marketing Contact
Harbans Kaur
harbkaur@cisco.com
Hardware
Cisco 3660 Router
Description
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All network modules on the Cisco 2600/3600 family are now supported on the Cisco 3660 with Cisco IOS Software Release 12.0(07)XK.
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NEW—E1 high density voice
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NEW—Fast Ethernet mixed media (NM-1FE2W, NM-2FE2W, NM-1FE1R2W, NM-2W)
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Digital and analog modems for dial access
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Channelized T1/E1/PRI
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ATM-25
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VIC-2BRI
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WIC-2A/S
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WIC-2T
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See the following URL for additional details and updated documentation on all above items:
http://www.cisco.com/en/US/products/hw/routers/ps274/index.html
Benefits
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With the added support of all the existing 2600/3600 network modules, VICs and WICs, the Cisco 3660 offers customers a greater feature set on a higher availability, higher-density, and higher-performance platform
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With the introduction of the new E1 digital voice network module, and support for QSIG signaling, advanced PBX connectivity options are now available on the Cisco 2600/3600 product family
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With the introduction of the new mixed media Fast Ethernet network modules, customers have greater flexibility in their LAN/WAN connectivity options
Platforms/Considerations
First appearance in a Cisco IOS Software "T" release: 12.1(1)T.
Marketing Contact
Sarat Khilnani
sarat@cisco.com
408-525-7514
MC3810-V3 and MC3810-HCM6/MC3810-HCM2
Description
The 12.1(2)T release is the first early deployment (ED) release which supports the next generation MC3810, the MC3810-V3 and it's accompanying next generation voice compression module, the MC3810-HCM6 or MC3810-HCM2.
The MC3810-V3 introduces a higher performance processor that insures high quality, high-density voice over packet technologies. The MC3810-V3 includes 64MB of SDRAM and 32MB of Flash for long-term investment protection.
The MC3810-HCM (High performance Compression Module) provides greater voice call density for the MC3810 series and may be used on either the MC3810-V3 or the classic MC3810-V. There are two models for the MC3810-HCM, the HCM2, which supports a maximum of 8 voice channels and the HCM6 with supports up to 24 channels. With a single MC3810-HCM6, the MC3810-V3 supports up to 24 channels of G.729a, G.711 voice or fax. Likewise, the MC3810-V or V3 supports 12 channels of G.723.1 or G.726.
Benefits
Improved investment protection with latest technology of processing, digital signal processor and memory headroom.
Robust architecture for support of existing and emerging voice networking technologies.
Backward compatibility for voice compression hardware means that installed base customers benefit from the newest in voice technology without a major disruption to their network.
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(2)T.
Marketing Contact
Teresa Newell
tnewell@cisco.com
Virtual Private Network (VPN) Module for the Cisco 1700 Series Routers
Description
The VPN module handles VPN security by implementing IP Security (IPsec)—an industry-wide standard for assuring the privacy, integrity, and authenticity of information crossing public IP networks. The VPN module, which fits in a slot inside the Cisco 1720 or 1750 chassis, encrypts data using Digital Encryption Standard (DES) and 3DES algorithms at speeds suitable for a single full-duplex T1/E1 serial connection (4 megabits per second for 1514-byte packets). The module together with the platform supports as many as 100 IPsec tunnels (400 security associations) for concurrent sessions with mobile users or other sites.
The Cisco 1700 series together with the VPN module and IOS Firewall Feature Sets is the perfect IPsec VPN solution for connecting small offices to other remote offices, mobile users, central-office intranets, or partner extranets.
Additional information on the VPN Module may be found at:
http://www.cisco.com/en/US/products/hw/routers/ps221/index.htmlBenefits
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Enables the secure use of public switched networks and the Internet for wide area networking
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Increases overall encryption performance over software encryption methods
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Significantly reduces the system costs, management complexity, and deployment effort over multiple box solutions
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Enables deployment of VPNs to up to 100 mobile users or sites
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Reserves critical processing resources for other services such as routing, firewall, and voice
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(2)T.
Marketing Contact
Russell Ruthen
ruthen@cisco.com
OC-3/STM-1 ATM Circuit Emulation Services (CES) Network Modules for the Cisco 3600 Multiservice Access Routers
Description
Three new single-port OC-3/STM-1 ATM Circuit Emulation Services (CES) network modules for the Cisco 3600 series provide ATM Forum-compliant CES. All three ATM network modules support STS-3c and STM-1 framing standards over multimode, single-mode intermediate reach, or single-mode long reach fiber-optic interfaces. Up to two T1 or E1 trunk ports are available for private branch exchange (PBX) or video-initiated traffic using the multiflex voice/WAN interface cards (VWICs). These ATM CES network modules provide a cost-effective solution that can be deployed as service-provider customer premises equipment (CPE) for consolidating multiservice data, voice, and video services over a single ATM link.
Based on the ATM Forum Circuit Emulation specification, the Cisco 3600 CES capability provides both structured and unstructured CES for transparent channel associated signaling (CAS) and common channel signaling (CCS) support. The OC-3/STM-1 network modules provide constant bit-rate (CBR) capabilities for voice and video applications that require guaranteed bandwidth across the ATM network. Integrated echo-cancellation features can also be enabled for up to 30 DS0s in a single T1 or E1 trunk.
Benefits
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Multimode, singlemode intermediate reach and single-mode long reach fiber support
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ATM Forum Standards ATM Adaption layer 1 (AAL1) and AAL5
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ATM Forum Traffic Management with User-Network Interface (UNI) 3.0, 3.1, and 4.0
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Support for all multiflex VWICs, with the exception of the 1- and 2-port G.703 drop-and-insert VWICs
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Integrated echo cancellation with configurable parameters for voice transport beyond the metropolitan area
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Support for either 1- or 2-port T1/E1 PBX trunk or video coder/decoder (codec) connections
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Multiple clocking modes, including global, synchronous residual time stamp (SRTS), and adaptive clocking
Platforms/Considerations
First appearance in a Cisco IOS Software release: 12.1(2)T.
Marketing Contact
Bill Massung
massung@cisco.com
DS3 and E3 ATM Network Modules for the Cisco 2600 and 3600 multiservice access routers
Description
Two new single-port DS3 and E3 ATM network modules are now available for the Cisco 2600 and 3600 multiservice access routers. Both versions provide a single ATM connection of either 44 Mbps for DS3, or 34 Mbps for E3 using 75-ohm BNC connectors. Both the DS3 and E3 ATM network modules support ATM Forum-compliant framing standard AAL5. ATM traffic management classes include Unspecified Bit Rate (UBR), UBR+, Variable bit rate-realtime (VBR-rt), VBR non-realtime (VBR-nrt), available bit rate (ABR), and CBR.
The DS3 and E3 ATM network modules complete the ATM family of interfaces that include the ATM25, 4-and 8-port ATM with inverse multiplexing over ATM (IMA), and OC-3/STM-1 network modules. These DS3/E3 ATM network modules provide a cost-effective solution that can be deployed as service-provider CPE or enterprise branch offices for consolidating multiservice data, voice and video services over a single ATM link.
Benefits
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Support for ATM classes of service: UBR, UBR+, VBR-rt, VBR-nrt, ABR, and CBR (data only)
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ATM UNI 3.0, 3.1, and 4.0 traffic management
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RFC 1483 and 1577 support
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1024 simultaneous virtual connections-(virtual path identifier [VPI] range 0-256, virtual channel identifier [VCI] range 0-1024)
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Permanent virtual circuits (PVCs) and switched virtual circuits (SVCs)
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Physical layer convergence procedure (PLCP) and header error control (HEC) cell delineation support
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Operations and management (F5 Operation Administration and Maintenance [OAM]) cell support
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LAN Emulation (LANE) 2.0
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Integrated Local Management Interface (ILMI) 1.0
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Internet Engineering Task Force (IE TF) Point-to-Point Protocol ([PPP] over ATM)
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IP-to-ATM class-of-service (CoS) mapping feature
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Multiprotocol Label Switching /virtual private network (MPLS/VPN)
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Multiprotocol over ATM (MPOA) Client and Server
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Next Hop Resolution Protocol (NHRP)
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