Table Of Contents
New Voice and Telephony Features in Cisco IOS Release 12.4T
Finding Support Information for Platforms and Cisco IOS Software Images
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release
New Voice and Telephony Features in Cisco IOS Release 12.4T
This document lists new Cisco IOS voice and telephony features in Cisco IOS Release 12.4T, and the location in the Cisco IOS Voice Configuration Library where each feature is documented. This information is presented in two tables:
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
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Note
Finding Support Information for Platforms and Cisco IOS Software Images
Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
Table 1 lists in alphabetical order new voice and telephony features in Cisco IOS Releases 12.4T.
Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
ANI Suppression During L2TP Setup
12.4(6)T
Provides the ability to suppress all or some part of the calling number field in the Layer 2 Tunneling Protocol (L2TP) setup process through RADIUS attribute functionality. The Calling Number Suppression for L2TP Setup feature feature allows you to make part or all of the calling number anonymous.
Busyout Monitor Gatekeeper
12.4(6)T
Simplifies monitoring of a large number of voice ports by adding busyout monitor gatekeeper command under voice class busyout mode.
Call Detail Records (CDR) Feature Correlation ID for Supplementary Features
12.4(9)T
Captures additional information in CDRs for voice calls that are transferred or forwarded on phones controlled by Cisco Unified CallManager Express (Cisco Unified CME) or Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST). It includes a unique correlation ID that identifies a given call feature across all legs in a call. CDR information can be output in RADIUS vendor-specific attributes (VSAs) or system log (syslog) messages.
Call Type Detection Feature in an IP-to-IP Gateway
12.4(4)T
Enables Cisco H.323 VoIP gateways to report the call type (voice/fax/modem) to a Cisco IOS gatekeeper at the end of each call.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
CDRs for Alternate Endpoints Tried in an IP-to-IP Gateway
12.4(4)T
Controls alternate endpoint hunting based on call disconnect cause codes.
"Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
Cisco CallManager Express (Cisco CME) 3.4
12.4(4)T
Cisco CallManager Express (Cisco CME) 3.4 adds station-side RFC3261 standard-based support for Session Initiation Protocol (SIP) phones directly into Cisco CME. This enables Cisco IP phones to place calls across SIP networks in the same way that the current Skinny Client Control Protocol (SCCP) phones do.
For full information aboutCisco CME 3.4, see the Cisco CallManager Express 3.4 Configuration Guide.
Cisco CallManager Express (CME) 4.0(1)
12.4(9)T
Delivers a number of key telephony features for customers including: Q.SIG integration with TDM PBX's, remote teleworker phone support, feature access codes for call handling, IP phone authentication, second Cisco CME for redundancy, hunt group login, fax pass-though with SCCP, and support for Cisco IP Phone models 7911G, 7941G/GE and 7961G/GE.
Cisco IOS VoiceXML 2.0
12.4(11)T
Provides support for the VoiceXML Version 2.0 W3C Recommendation (March 16, 2004) on Cisco IOS voice gateway VoiceXML browsers which enables interaction with VoiceXML applications.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide
"Cisco VoiceXML Features" and "Cisco VoiceXML Elements: Reference Table" chapters of the Cisco VoiceXML Programmer's Guide
Cisco IOS VoiceXML Browser Update to W3C Voice XML 2.1
12.4(15)T
Provides support for the VoiceXML 2.1 W3C Candidate Recommendation (June 13, 2005) on Cisco IOS voice gateway VoiceXML browsers which enables interaction with VoiceXML applications.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide, Release 12.3(14)T and later.
"Cisco VoiceXML Features" and "Cisco VoiceXML Elements: Reference Table" chapters of the Cisco VoiceXML Programmer's Guide
Cisco Modem Relay
12.4(4)T
Implements non-negotiated, bearer switched modem relay (gateway-controlled) on select gateways, enabling V.34 modem traffic to be reliably transported. Cisco Modem Relay supports H.323, SIP and MGCP signaling types, and because it is gateway-controlled, all call agents, including Cisco CallManager, Cisco CallManager Express, Cisco PGW Softswitch and Cisco BTS Softswitch are supported.
"Configuring Cisco Modem Relay" chapter of the Cisco IOS Fax and Modem Services over IP Application Guide.
This new combined guide replaces the previous Cisco IOS Fax Services over IP Application Guide and Modem Support for VoIP in the VCL.
Cisco Text Relay for Baudot Text Phones
12.4(6)T
Implements a mechanism for transporting Text Telephone (TTY) signals over Voice over IP (VoIP) calls in a highly reliable and robust manner. This feature supports Baudot 45.45 and 50 bps text (TTY) phones.
Cisco Unified CallManager Express 4.0(3)
12.4(15)T
Delivers two key features: Extension Assigner which allows for easy deployment or replacement of phones on site using a TCL IVR application and new IP Phone localizations for Asia and Eastern Europe.
Cisco Unified Communications Manager Express System Administrator Guide
Cisco Unified CallManager Express SIP Station-Side Enhancements
12.4(15)T
Includes music on hold (MoH), MoH with transcoding, dialplan-pattern, KPML and dialplan, speed dial, caller ID and status line update, phone directories button, line status subscription providing presence with authorization and authentication, and busy lamp field (BLF) for speed dial and missed call lists. Adds provisioning for Cisco 7970G, 7971GE, 7941G/GE, 7961G/GE, and 7911G 3951 SIP phones. Adds CLI to disallow SIP supplementary services. Line status subscription is for registered/in service, idle, in-use, and busy.
Cisco Unified Communications Manager Express System Administrator Guide
Cisco Unified Communications Manager Express Release 4.2
12.4(15)T
Introduces interoperability with a session server, such as Cisco Unified Contact Center Express (Unified CCX). This interface guide details the configuration, registration, and subscription portions of the call and line monitoring functions on the Cisco Unified CME.
Cisco Unified Communications Manager Express Call Monitoring Interface Guide
Customizable PSTN Tones and H.323 Call-Disconnect Cause Codes
12.4(9)T
Enables you to customize the following PSTN tones and H.323 call-disconnect cause codes for certain disconnect scenarios:
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You can also specify the mechanism for detecting media inactivity (silence) on a voice call: RTP, RTCP, or both.
Customizable PSTN Tones and H.323 Call-Disconnect Cause Codes
DSP Voice-Quality Statistics in DLCX Messages
12.4(4)T
Added new voice quality parameters, and two new keywords to mgcp voice-quality-stats. Provides a method to trace a Media Gateway Control Protocol (MGCP) call between a Cisco PGW 2200 and the Cisco IOS gateway by including the MGCP call ID and the DS0 and digital signal processor (DSP) channel ID in call-active and call-history records.
Enhanced MF for FGD and Analog CAMA Trunks
12.4(9)T
Enhances the 911 interconnect capabilities of Cisco IOS based gateways. This document describes new E911 support requirements, which includes support for Enhanced Multi-frequency (MF) signaling for Feature Group D (FGD) and Analog Centralized Automated Message Accounting (CAMA) signaling protocols per National Emergency Number Association standards. This feature supports 20-digit ANI requirements and mapping of remote party IDs (RPID) to PANI.
Enhancing CISCO-H225-MIB with Disconnect Cause Codes
12.4(4)T
The CISCO-H225-MIB was enhanced with the Q.931 disconnect cause codes that the H.323 subsystem can receive. A disconnect can originate from the far-end gateway or from the opposite call leg on the local gateway. This enhancement to the MIB allows you to report disconnect cause code information, including the cause code type and the number of cause code disconnects received from either H.323 peer. The enhancement corresponds to the usage of the show h323 gateway command. See the show h323 gateway command for an example of the disconnect cause code display.
The Enhancing CISCO-H225-MIB with Disconnect Cause Codes feature provides SNMP MIB enhancements on the following platforms:
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The MIB contains objects that represent active H.323 calls and also includes call details. For definitions of the H.323 MIB objects, see the following MIBs:
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To locate and download MIBs, use Cisco MIB Locator found at the following URL:
Extending Dynamic Zone Prefix Registration to Include Gateway Priority
12.4(9)T
Simplifies the H.323 zone configuration process by defining the zone prefix and the corresponding gateway priorities together on the gateway.
"Configuring H.323 Gateways" chapter of the Cisco IOS H.323 Configuration Guide
Fax Relay Support for SG3 Fax Machines at G3 Speeds
12.4(4)T
Introduces a fax machine spoofing mechanism on select gateways to force Super Group 3 (SG3) fax machines to automatically fall back to Group 3 (G3) speeds. This enables faxes to be sent between 2 SG3 fax machines over T.38 Fax Relay and Cisco Fax Relay at the supported G3 speeds (14.4 kbps).
"Configuring Cisco Fax Relay" and "Configuring T.38 Fax Relay" chapters of the Cisco IOS Fax and Modem Services over IP Guide.
This new combined guide replaces the previous Cisco IOS Fax Services over IP Application Guide and Modem Support for VoIP in the VCL.
Final Flag notification from the GKTMP Server
12.4(4)T
Enables a control field in the Gatekeeper Transaction Message Protocol (GKTMP) that allows an external application to halt normal alternate routing procedures at the gatekeeper, to reduce call setup times and reject calls quickly during peak traffic periods in the wholesale provider's network.
No configuration is required.
" Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
H.323 Standard Based Hopcount Field in LRQ
12.4(4)T
Support for H.225 version 4 standard hopCount field in LocationRequest RAS message.
No configuration is required.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
H.323 VoIP Call Preservation Enhancements for WAN Link Failures
12.4(9)T
"Configuring H.323 Gateways" chapter of the Cisco IOS H.323 Configuration Guide
High-Density Packet Voice Feature Card for Cisco AS5350XM and AS5400XM Universal Gateways
12.4(9)T
Supports up to six high-density packet voice/fax digital signal processor (DSP) modules (product number AS5X-PVDM2-64), providing scalability from 64 to 384 channels.
High-Density Packet Voice Feature Card for Cisco AS5350XM and AS5400XM Universal Gateways
iLBC Codec Support
12.4(11)T
Supports the internet Low Bitrate Codec (iLBC), a standard, high-complexity speech codec that is suitable for robust voice communication over IP. iLBC has built-in error correction functionality that helps the codec perform in networks with a high-packet loss.
"Dial Peer Overview" chapter and "Dial Peer Features and Configuration" chapter in Dial Peer Configuration on Voice Gateway Routers
iLBC Support for SIP and H.323
12.4(15)T
Supports the internet Low Bitrate Codec (iLBC), a standard, high-complexity speech codec that is suitable for robust voice communication over IP. This codec is supported on both SIP and H.323.
"Dial Peer Overview" chapter and "Dial Peer Features and Configuration" chapter in Dial Peer Configuration on Voice Gateway Routers
In-Service Updates to Gatekeeper Zone Prefix Configuration
12.4(6)T
Increases the availability of H.323 VoIP networks by allowing changes to a gatekeeper zone prefix while the gatekeeper is running and managing active E.164 registrations.
" Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
Integrated Data Primary Rate Interface (PRI) Services
12.4(9)T
Enables PRI interfaces that were previously only capable of TDM voice to also be simultaneously capable of handling PRI Data channels.
Integrating Data and Voice Services for ISDN PRI Interfaces on Multiservice Access Routers.
Interoperability Enhancements to the Cisco Multiservice IP-IP Gateway
12.4(4)T
Enables operation of IP-to-IP gateway features concurrently on the same router with H.323 gatekeeper and TDM-IP voice-gateway features.
Land Mobile Radio (LMR) over IP Enhancement
12.4(2)T
Allows Cisco multiservice routers to transport LMR traffic over IP networks by modifying voice gateway functionality. LMR over IP enables LMR systems to extend beyond their traditional geographic limitations created by transmitter signal strength and enables interoperability, allowing public safety personnel in different agencies or jurisdictions to communicate with each other by radio on demand, in real time.
Media and Signaling Authentication and Encryption Feature for Cisco IOS H.323 Gateways
12.4(6)T1
Provides authentication, integrity, and encryption of voice media and call control signaling for H.323 protocol-based voice gateways. New secure voice call capabilities between gateways include:
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Media and Signaling Authentication and Encryption Feature for Cisco IOS H.323 Gateways
Media Resource Control Protocol (MRCP) version 2
12.4(15)T
Provides support for MRCP v2 for use with specified Cisco IOS voice gateways' VoiceXML Browser. The MRCP v2 protocol allows a client device to control media processing resources on the network. Media processing resources include speech recognition engines, speech synthesis engines, speaker verification and speaker identification engines. MRCP v2 enables the implementation of distributed Interactive Voice Response (IVR) platforms using VoiceXML browsers or other client applications while maintaining separate back-end speech processing capabilities on specialized speech processing servers. MRCP v2 is based on the earlier MRCP developed by Cisco Systems, Inc., Nuance Communications, Inc. and Speechworks, Inc.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide, Release 12.3(14)T and later.
"Cisco VoiceXML Features" and "Cisco VoiceXML Elements: Reference Table" chapters of the Cisco VoiceXML Programmer's Guide
MGCP Call Centric Debug
12.4(4)T
Enables the filtering of MGCP debug output based on selected criteria and standardizes the format of the MGCP debug header. All MGCP debug output for a single call can be identified and correlated across the various layers in IOS software. Filtering debug output reduces extraneous information, making it easier to locate the correct information and reducing the impact to platform performance.
"Filtering Troubleshooting Output" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide.
MGCP CAS MD Package
12.4(4)T
Introduces support for Feature Group D (FGD) Exchange Access North American (EANA) protocol signaling. The MD package adds support for the reporting of automatic number identification (ANI) and dialed number identification service (DNIS) digits to enable the MGCP call agent to better handle customer billing.
"Configuring MGCP CAS MD Package" chapter in the Cisco IOS MGCP and Related Protocols Configuration Guide.
MGCP Controlled Backhaul of BRI Signaling
12.4(2)T
Extends support for the MGCP-Controlled Backhaul of BRI Signaling in Conjunction with the Cisco CallManager feature to the NM-HD, NM-HDV2, EVM-HD, and Cisco 2800/3800 series with a BRI HWIC interface.
"Configuring MGCP-Controlled Backhaul of BRI Signaling in Conjunction with Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
MGCP Endpoint Range Support
12.4(4)T
Extends the mgcp behavior command by adding the rsip-range keyword. The rsip-range keyword controls whether the gateway can generate ReStart In Progress (RSIP) messages with endpoint ranges for versions other than Trunking Gateway Control Protocol (TGCP).
MGCP Layer 2 Teardown for IUA DPNSS Trunks
12.4(9)T
Stops voice calls from being lost during a WAN failure by tearing down all Layer-2 calls and notifying the PBX of the out-of-service trunk.
MGCP NAS Package LAPB-TA
12.4(6)T
Implements autodetection for the MGCP NAS package, as supported in Cisco IOS Release 12.3(9) under ISDN serial interfaces.
No Retry on User Busy in an IP-to-IP Gateway
12.4(4)T
Changes the default behavior of the gateway to not retry alternate endpoints when the release complete reason is user busy.
" Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
Outbound Proxy Support for the SIP Gateway
12.4(15)T
Configures an outbound-proxy server that receives all initiating request (INVITE and SUBSCRIBE) messages and routes them to the designated destination.
"Configuring SIP Message, Timer, and Response Features" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
R2 Call Blocking for Brazil
12.4(2)T
Provides incoming collect call block support. Collect calls will be blocked based on a specific category. For example, in Brazil, collect calls arrive with a category II-8 for which the Cisco access router sends B-7 as response instead of an answer signal.
For an incoming collect call, the gateway answers the call with a clear-back after 1 second and re-answers the call after 2 seconds. This causes the collect call to be dropped and normal calls to stay connected. This is implemented as a CLI option.
RAS retry and timer
12.4(4)T
Allows service providers the ability to control transmit time margins on Cisco gatekeepers by changing RAS message timeout LRQ value and message retry counter values.
" Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
RFC 2833 DTMF MTP Passthrough
12.4(11)T
Passes DTMF tones transparently between SIP endpoints that require either transcoding or use of the RSVP Agent feature.
"Configuring SIP DTMF Features" chapter in the Cisco IOS SIP Configuration Guide.
"Configuring Voice Mail Integration for Cisco Unified CME for SIP Phones " section of Cisco Unified CME Configuration Guide for SIP Phones at Cisco Unified CallManager Express: All Versions
RSVP Agent
12.4(6)T
Implements a Resource Reservation Protocol (RSVP) agent on Cisco IOS voice gateways that support Cisco Unified CallManager 5.0.
SCCP Analog (FXS) Ports
12.4(2)T
Enables Skinny Client Control Protocol (SCCP) supplementary features on analog FXS ports on a Cisco VG 224 voice gateway under the control of Cisco CallManager or Cisco CallManager Express (Cisco CME).
SCCP PLAR with DTMF Out Pulse Digits for FXS Analog Phones
12.4(6)T
Provides private line automatic ring-down (PLAR) support and enhanced speed-dial capabilities for Skinny Client Control Protocol (SCCP) analog ports on a Cisco IOS voice gateway under the control of Cisco CallManager or a Cisco CallManager Express (Cisco CME) system.
SCCP Controlled Analog (FXS) Ports with Supplementary Features in IOS Gateways
SCTP Show/Clear CLI Enhancements
12.4(11)T
Provides access to additional SCTP information that can help with troubleshooting potential problems. These enhancements also make the updated SCTP show and clear commands consistent with the CLI of other transport protocols.
Cisco IOS Voice Command Reference
Secure Communication Between IP-STE Endpoint and Trunkside STE Endpoint
12.4(2)T
Supports encrypted and decrypted calls from an IP secure terminal equipment (STE) controlled by Cisco CallManager through a voice gateway to an STE in the Defense Switch Network (DSN). This feature implements a subset of the V.150.1 modem relay standard, allowing users to operate US Department of Defense-compliant (Type-1 encryption) devices across a VoIP network, and between VoIP networks and the Defense Switching Network.
The mgcp modem relay voip mode, mgcp modem relay voip mode sse, mgcp modem relay voip sprt v.14, mgcp package-capability, show call active voice, show mgcp, show mgcp connection, and show modem relay statistics commands in the Cisco IOS Voice Command Reference and the debug modem relay v14 command in the Cisco IOS Debug Command Reference
Secure Communication BetweenIP-STE Endpoint and Line-Side STE Endpoint
12.4(4)T
Provides the following Cisco IOS gateway capabilities:
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Secure Communication BetweenIP-STE Endpoint and Line-Side STE Endpoint
Secure HTTP client (SSL) for Cisco IOS VxML Browser
12.4(15)T
Provides secure communications between the Cisco IOS VoiceXML browser and VoiceXML servers that also support HTTP over SSL. The Cisco IOS VoiceXML Browser enables interaction with VoiceXML application servers.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide, Release 12.3(14)T and later releases
Sequential LRQ timer
12.4(4)T
Defines the time window during which the gatekeeper collects responses from the gateway before resending a RAS message to a gatekeeper, and the number of times to resend the RAS message after the timeout period expires.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
SIP: Busy Out Support
12.4(6)T
Introduces, at the SIP level, a generic keepalive mechanism that allows the SIP gateway to monitor the status of the SIP servers and provide the option of busying-out the associated voice ports upon total keepalive failure.
SIP: Cisco IOS Gateway Signaling Support Over TLS Transport
12.4(6)T
Implements the Transport Layer Security (TLS) protocol on the Transmission Control Protocol (TCP) transport for Cisco IOS SIP gateways. The feature leverages the existing gateway's support of the public-key infrastructure (PKI) (for certificate management) and Open Secure Socket Layer-Transport Layer Security (OPSSL-TLS) application program interfaces (APIs) in order to provide the necessary functionality. The use of PKI on Cisco IOS software requires that the clock on the session initiation protocol (SIP) gateway be synchronized with the network time to ensure proper validation of certificates.
SIP: Cisco IOS SIP Gateway Signaling Support Over TLS Transport
SIP: CLI for Caller ID When Privacy Exists
12.4(4)T
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SIP: Domain Name Support in SIP Headers
12.4(2)T
Adds a command-line interface (CLI) switch to provide a host or domain name in the host portion of the locally generated SIP headers (for example, From, RPID, and Call-ID). The SIP: Domain Name Support in SIP Headers feature also affects the outgoing dialog initiating SIP requests (for example, INVITE and SUBSCRIBE message requests).
"Configuring SIP Message, Timer, and Response Features" of the Cisco IOS SIP Configuration Guide
SIP: Multilevel Precedence and Priority Support
12.4(2)T
Enables Cisco IOS gateways to interoperate with other multilevel-precedence and preemption (MLPP)-capable circuit-switched networks.
An MLPP-enabled call has an associated priority level that applications that handle emergencies and congestions use to determine which lower-priority call to preempt in order to dedicate their end-system resources to high-priority communications. This feature addresses the aspect of preemption when interworking with defense-switched networks (DSNs) that are connected through the Cisco IOS gateway.
"Configuring SIP Connection-Oriented Media, Forking, and MLPP Features" of the Cisco IOS SIP Configuration Guide
SIP MWI NOTIFY - QSIG MWI Translation
12.4(11)T
Enhances MWI functionality to include SIP-MWI-Notify-to-QSIG-MWI translation between gateways and routers.
"Configuring SIP MWI Features" chapter in the Cisco IOS SIP Configuration Guide.
"Configuring Voice Mail Integration for Cisco Unified CME for SIP Phones " section of Cisco Unified CME Configuration Guide for SIP Phones at Cisco Unified CallManager Express: All Versions.
SIP:SIP Gateway OOB DTMF Support with KPML
12.4(9)T
Provides a command-line interface (CLI) option that forwards DTMF tones using KeyPad Markup Language (KPML) by way of SIP SUBSCRIBE and NOTIFY messages.
"Configuring SIP DTMF Features" in the Cisco IOS SIP Configuration Guide.
SIP:SIP Gateway Session Timer Support
12.4(9)T
Enhances session timer support for gateways to comply with IETF Session Timer RFC 4028.
"Configuring SIP Message, Timer, and Response Features" in the Cisco IOS SIP Configuration Guide.
SIP:SIP Gateway Support for SDP Session Information and Permit Hostname CLI
12.4(9)T
Adds support for Session Protocol Description (SDP) session information to comply with IETF SDP RFC 2327. Adds support for validating up to 10 hostnames for incoming initial INVITE messages.
"Configuring SIP Message, Timer, and Response Features" in the Cisco IOS SIP Configuration Guide.
SIP: SIP Support for Hookflash
12.4(11)T
Configures IP Centrex supplementary services on SIP-enabled, Foreign Exchange Station (FXS) lines.
"Configuring SIP Support for Hookflash" chapter in the Cisco IOS SIP Configuration Guide.
SIP: Support for Asymmetric SDP
12.4(15)T
Configures SIP gateways to send and receive Dual Tone Multi-Frequency (DTMF) and dynamic codec Real Time Protocol (RTP) packets with different payloads.
"Configuring SIP DTMF Features" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
SIP: Support for PAI
12.4(15)T
Provides support for RFC 3323 and RFC 3325 that allow you to enable either P-Asserted-Identity (PAI) or P-Preferred-Identity (PPI) privacy headers in outgoing SIP request or response messages to assert the identity of authenticated users in trusted domains.
"Configuring SIP Message, Timer, and Response Features" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
SIP: Support for SRTP
12.4(15)T
Ensures the integrity of RTP and Real-Time Control Protocol (RTCP) packets providing authentication, integrity, and encryption of media packets between two SIP endpoints.
"Configuring SIP Support for SRTP" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
SIP Stack Portability
12.4(2)T
Implements the following capabilities to the SIP gateway Cisco IOS stack:
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It sends an INVITE message request without Session Definition Protocol (SDP) and provides SDP in either the PRACK or ACK message request for both initial call establishment and mid-call re-INVITE message requests.
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The stack applies certain rules and restrictions for a subset of headers and for some content types (such as SDP) to protect the integrity of the stack's functionality and to maintain backward compatibility. When receiving SIP message requests, it reads the SIP header and any attached body without any restrictions.
SIP: User Agent MIB Enhancements
12.4(2)T
Provides SNMP MIB object enhancements to the CISCO-SIP-UA-MIB and informational updates to the CISCO-SIP-UA-CAPABILITY file. The CISCO-SIP-UA-CAPABILITY file provides information such as the Cisco IOS release number and the capabilities of the CISCO-SIP-UA-MIB. The SNMP MIB object updates provide configuration and counter support that are equivalent to command-line interface additions introduced in several SIP features.
In addition, the SIP: User Agent MIB Enhancements feature provides two new MIB objects. In Release 12.3(4)T, the SIP Gateway Support Enhancements to the bind Command feature extended the Cisco IOS bind command by adding the media keyword. The media keyword allows multiple instances of the bind command. One instance defines the control address, and one instance defines the media address. Because the original CISCO-SIP-UA-MIB was defined with a single instance of the bind command, the MIB objects that support the bind command are replaced with two new MIB objects that support multiple instances.
SIP: User Agent MIB Enhancements (Continued)
12.4(2)T
Any SNMP applications that SET or GET the following objects from the CISCO-SIP-UA-MIB need to refer to the new objects:
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While the objects above have not been removed and are still accessible with Cisco IOS Release 12.4(2)T, they will be removed in a future release. Users must upgrade any affected application to the following new objects:
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You can specify pairs of cSipCfgBindSourceAddrInterface and cSipCfgBindSourceAddrScope. Specifying pairs allows you to associate one interface address with control traffic and another interface address with media traffic. Please note that "Src" in the prior objects has been replaced with "Source" in the new objects.
For full definitions of the SIP MIB objects, see the CISCO-SIP-UA-MIB. To locate and download MIBs, use Cisco MIB Locator found at the following URL:
SIP-to-H.323 Extended Call Interworking
12.4(4)T
Enables the IP-to-IP gateway to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323).
"Features Supported by the Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
SIP-to-SIP Basic Call Interworking
12.4(4)T
Enables the IP-to-IP gateway to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323).
" Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
SIP-to-SIP Extended Feature Functionality for Session Border Controller (SBC)
12.4(6)T
Enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP UAs. New SIP-to-SIP features available include:
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The "Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
SIP-to-SIP Supplementary Services for Session Border Controller(SBC)
12.4(9)T
Enhances terminating and re-originating both signaling and media between VoIP and Video networks by supporting supplementary features such as Message Waiting Indication, Call Waiting, Call Transfer, Call Forward, Distinctive Ringing, Call Hold/Resume, Music on Hold.
"Configuring SIP-to-SIP Connections in a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
SIP REFER
12.4(15)T
Allows remote applications to establish calls by sending a REFER message to Cisco Unified CME, Cisco Unified SRST, or a SIP gateway without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application.
Cisco Unified Communications Manager Express System Administrator Guide
Support for IP-to-IP Gateway and Gatekeeper Features on the Cisco 2801
12.4(4)T
Provides integrated voice and video services on the Cisco 2801.
"Configuring a Cisco Multiservice IP-to-IP Gateway " chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
Survivable Remote Site Telephony (SRST) 3.4
12.4(4)T
Cisco SIP SRST Version 3.4 describes SRST functionality for Session Initiation Protocol (SIP) networks. Cisco SIP SRST Version 3.4 provides backup to an external SIP proxy server by providing basic registrar and back-to-back user agent (B2BUA) services. These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy.
Cisco SIP SRST Version 3.4 can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco SIP SRST Version 3.4, SIP phones can place calls across SIP networks in the same way as Skinny Client Control Protocol (SCCP) phones.
Survivable Remote Site Telephony Version 4.0
12.4(9)T
Cisco Unified Survivable Remote Site Telephony (SRST): All Versions
Test Call
12.4(4)T
Provides the ability for a remote station or gateway to establish a call to any destination address from a Test Call station located at a network operations center and to audibly verify the voice path.
"Troubleshooting H.323 Interfaces to the IP Network" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide.
Unique Calling Party Information with Alternate Endpoints
12.4(6)T
Enables alternate endpoint capabilities of the Cisco IOS H.323 gatekeeper and voice gateway to associate a unique calling party number automatic number identification (ANI) with each alternate endpoint using the GKTMP.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
Video Support for SCCP-Based Endpoints
12.4(9)T
Provides the capability to send H.320 encapsulated Audio/Video calls over TDM voice interfaces.
"Video Support for SCCP-Based Endpoints" chapter in the Cisco Unified CallManager Express System Adminstrator Guide, 4.0
Voice Call Debug Filtering on H.323 Gatekeepers
12.4(4)T
Enables selected debugging traces for voice calls. This feature allows you to filter and trace voice call debug messages based on selected filtering criteria, reducing the volume of output for more efficient troubleshooting.
"Filtering Troubleshooting Output" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guidee
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release
Table 2 lists new voice and telephony features in Cisco IOS Release 12.4T by the maintenance release in which each feature was added. The most recent release is listed first.
Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release
12.4(15)T
Cisco IOS VoiceXML Browser Update to W3C Voice XML 2.1
Provides support for the VoiceXML 2.1 W3C Candidate Recommendation (June 13, 2005) on Cisco IOS voice gateway VoiceXML browsers which enables interaction with VoiceXML applications.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide, Release 12.3(14)T and later.
"Cisco VoiceXML Features" and "Cisco VoiceXML Elements: Reference Table" chapters of the Cisco VoiceXML Programmer's Guide
12.4(15)T
Cisco Unified CallManager Express 4.0(3)
Delivers two key features: Extension Assigner which allows for easy deployment or replacement of phones on site using a TCL IVR application and new IP Phone localizations for Asia and Eastern Europe.
Cisco Unified Communications Manager Express System Administrator Guide
12.4(15)T
Cisco Unified CallManager Express SIP Station-Side Enhancements
Includes music on hold (MoH), MoH with transcoding, dialplan-pattern, KPML and dialplan, speed dial, caller ID and status line update, phone directories button, line status subscription providing presence with authorization and authentication, and busy lamp field (BLF) for speed dial and missed call lists. Adds provisioning for Cisco 7970G, 7971GE, 7941G/GE, 7961G/GE, and 7911G 3951 SIP phones. Adds CLI to disallow SIP supplementary services. Line status subscription is for registered/in service, idle, in-use, and busy.
Cisco Unified Communications Manager Express System Administrator Guide
12.4(15)T
Cisco Unified Communications Manager Express Release 4.2
Introduces interoperability with a session server, such as Cisco Unified Contact Center Express (Unified CCX). This interface guide details the configuration, registration, and subscription portions of the call and line monitoring functions on the Cisco Unified CME.
Cisco Unified Communications Manager Express Call Monitoring Interface Guide
12.4(15)T
iLBC Support for SIP and H.323
Supports the internet Low Bitrate Codec (iLBC), a standard, high-complexity speech codec that is suitable for robust voice communication over IP. This codec is supported on both SIP and H.323.
"Dial Peer Overview" chapter and "Dial Peer Features and Configuration" chapter in Dial Peer Configuration on Voice Gateway Routers
12.4(15)T
Media Resource Control Protocol (MRCP) version 2
Provides support for MRCP v2 for use with specified Cisco IOS voice gateways' VoiceXML Browser. The MRCP v2 protocol allows a client device to control media processing resources on the network. Media processing resources include speech recognition engines, speech synthesis engines, speaker verification and speaker identification engines. MRCP v2 enables the implementation of distributed Interactive Voice Response (IVR) platforms using VoiceXML browsers or other client applications while maintaining separate back-end speech processing capabilities on specialized speech processing servers. MRCP v2 is based on the earlier MRCP developed by Cisco Systems, Inc., Nuance Communications, Inc. and Speechworks, Inc.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide, Release 12.3(14)T and later.
"Cisco VoiceXML Features" and "Cisco VoiceXML Elements: Reference Table" chapters of the Cisco VoiceXML Programmer's Guide
12.4(15)T
Outbound Proxy Support for the SIP Gateway
Configures an outbound-proxy server that receives all initiating request (INVITE and SUBSCRIBE) messages and routes them to the designated destination.
"Configuring SIP Message, Timer, and Response Features" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
12.4(15)T
Secure HTTP client (SSL) for Cisco IOS VxML Browser
Provides secure communications between the Cisco IOS VoiceXML browser and VoiceXML servers that also support HTTP over SSL. The Cisco IOS VoiceXML Browser enables interaction with VoiceXML application servers.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide, Release 12.3(14)T and later releases
12.4(15)T
SIP: Support for Asymmetric SDP
Configures SIP gateways to send and receive Dual Tone Multi-Frequency (DTMF) and dynamic codec Real Time Protocol (RTP) packets with different payloads.
"Configuring SIP DTMF Features" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
12.4(15)T
SIP: Support for PAI
Provides support for RFC 3323 and RFC 3325 that allow you to enable either P-Asserted-Identity (PAI) or P-Preferred-Identity (PPI) privacy headers in outgoing SIP request or response messages to assert the identity of authenticated users in trusted domains.
"Configuring SIP Message, Timer, and Response Features" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
12.4(15)T
SIP: Support for SRTP
Ensures the integrity of RTP and Real-Time Control Protocol (RTCP) packets providing authentication, integrity, and encryption of media packets between two SIP endpoints.
"Configuring SIP Support for SRTP" chapter of the Cisco IOS SIP Configuration Guide, Release 12.4T.
12.4(15)T
SIP REFER
Allows remote applications to establish calls by sending a REFER message to Cisco Unified CME, Cisco Unified SRST, or a SIP gateway without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application.
Cisco Unified Communications Manager Express System Administrator Guide
12.4(11)T
Cisco IOS VoiceXML 2.0
Provides support for the VoiceXML Version 2.0 W3C Recommendation (March 16, 2004) on Cisco IOS voice gateway VoiceXML browsers which enables interaction with VoiceXML applications.
"Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter of the Cisco Tcl IVR and VoiceXML Application Guide
"Cisco VoiceXML Features" and "Cisco VoiceXML Elements: Reference Table" chapters of the Cisco VoiceXML Programmer's Guide
12.4(11)T
iLBC Codec Support
Supports the internet Low Bitrate Codec (iLBC), a standard, high-complexity speech codec that is suitable for robust voice communication over IP. iLBC has built-in error correction functionality that helps the codec perform in networks with a high-packet loss.
"Dial Peer Overview" chapter and "Dial Peer Features and Configuration" chapter in Dial Peer Configuration on Voice Gateway Routers
12.4(11)T
RFC 2833 DTMF MTP Passthrough
Passes DTMF tones transparently between SIP endpoints that require either transcoding or use of the RSVP Agent feature.
"Configuring SIP DTMF Features" chapter in the Cisco IOS SIP Configuration Guide.
"Configuring Voice Mail Integration for Cisco Unified CME for SIP Phones " section of Cisco Unified CME Configuration Guide for SIP Phones at Cisco Unified CallManager Express: All Versions
12.4(11)T
SCTP Show/Clear CLI Enhancements
Provides access to additional SCTP information that can help with troubleshooting potential problems. These enhancements also make the updated SCTP show and clear commands consistent with the CLI of other transport protocols.
Cisco IOS Voice Command Reference
12.4(11)T
SIP MWI NOTIFY - QSIG MWI Translation
Enhances MWI functionality to include SIP-MWI-Notify-to-QSIG-MWI translation between gateways and routers.
"Configuring SIP MWI Features" chapter in the Cisco IOS SIP Configuration Guide.
"Configuring Voice Mail Integration for Cisco Unified CME for SIP Phones " section of Cisco Unified CME Configuration Guide for SIP Phones at Cisco Unified CallManager Express: All Versions.
12.4(11)T
SIP: SIP Support for Hookflash
Configures IP Centrex supplementary services on SIP-enabled, Foreign Exchange Station (FXS) lines.
"Configuring SIP Support for Hookflash" chapter in the Cisco IOS SIP Configuration Guide.
12.4(9)T
Call Detail Records (CDR) Feature Correlation ID for Supplementary Features
Captures additional information in CDRs for voice calls that are transferred or forwarded on phones controlled by Cisco Unified CallManager Express (Cisco Unified CME) or Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST). It includes a unique correlation ID that identifies a given call feature across all legs in a call. CDR information can be output in RADIUS vendor-specific attributes (VSAs) or system log (syslog) messages.
12.4(9)T
Cisco CallManager Express (CME) 4.0(1)
Delivers a number of key telephony features for customers including: Q.SIG integration with TDM PBX's, remote teleworker phone support, feature access codes for call handling, IP phone authentication, second Cisco CME for redundancy, hunt group login, fax pass-though with SCCP, and support for Cisco IP Phone models 7911G, 7941G/GE and 7961G/GE.
12.4(9)T
Cisco IOS H.320 Video Gateway
Provides the capability to send H.320 encapsulated Audio/Video calls over TDM voice interfaces.
"Video Support for SCCP-Based Endpoints" chapter in the Cisco Unified CallManager Express System Adminstrator Guide, 4.0
12.4(9)T
Customizable PSTN Tones and H.323 Call-Disconnect Cause Codes
Enables you to customize the following PSTN tones and H.323 call-disconnect cause codes for certain disconnect scenarios:
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You can also specify the mechanism for detecting media inactivity (silence) on a voice call: RTP, RTCP, or both.
Customizable PSTN Tones and H.323 Call-Disconnect Cause Codes
12.4(9)T
Enhanced MF for FGD and Analog CAMA Trunks
Enhances the 911 interconnect capabilities of Cisco IOS based gateways. This document describes new E911 support requirements, which includes support for Enhanced Multi-frequency (MF) signaling for Feature Group D (FGD) and Analog Centralized Automated Message Accounting (CAMA) signaling protocols per National Emergency Number Association standards. This feature supports 20-digit ANI requirements and mapping of remote party IDs (RPID) to PANI.
12.4(9)T
Extending Dynamic Zone Prefix Registration to Include Gateway Priority
Simplifies the H.323 zone configuration process by defining the zone prefix and the corresponding gateway priorities together on the gateway.
"Configuring H.323 Gateways" chapter of the Cisco IOS H.323 Configuration Guide
12.4(9)T
H.323 VoIP Call Preservation Enhancements for WAN Link Failures
"Configuring H.323 Gateways" chapter of the Cisco IOS H.323 Configuration Guide
12.4(9)T
High-Density Packet Voice Feature Card for Cisco AS5350XM and AS5400XM Universal Gateways
Supports up to six high-density packet voice/fax digital signal processor (DSP) modules (product number AS5X-PVDM2-64), providing scalability from 64 to 384 channels.
High-Density Packet Voice Feature Card for Cisco AS5350XM and AS5400XM Universal Gateways
12.4(9)T
Integrated Data Primary Rate Interface (PRI) Services
Enables PRI interfaces that were previously only capable of TDM voice to also be simultaneously capable of handling PRI Data channels.
Integrating Data and Voice Services for ISDN PRI Interfaces on Multiservice Access Routers.
12.4(9)T
MGCP Layer 2 Teardown for IUA DPNSS Trunks
Stops voice calls from being lost during a WAN failure by tearing down all Layer-2 calls and notifying the PBX of the out-of-service trunk.
12.4(9)T
SIP:SIP Gateway OOB DTMF Support with KPML
Provides a command-line interface (CLI) option that forwards DTMF tones using KeyPad Markup Language (KPML) by way of SIP SUBSCRIBE and NOTIFY messages.
"Configuring SIP DTMF Features" in the Cisco IOS SIP Configuration Guide.
12.4(9)T
SIP:SIP Gateway Session Timer Support
Enhances session timer support for gateways to comply with IETF Session Timer RFC 4028.
"Configuring SIP Message, Timer, and Response Features" in the Cisco IOS SIP Configuration Guide.
12.4(9)T
SIP:SIP Gateway Support for SDP Session Information and Permit Hostname CLI
Adds support for Session Protocol Description (SDP) session information to comply with IETF SDP RFC 2327. Adds support for validating up to 10 hostnames for incoming initial INVITE messages.
"Configuring SIP Message, Timer, and Response Features" in the Cisco IOS SIP Configuration Guide.
12.4(9)T
SIP-to-SIP Supplementary Services for Session Border Controller(SBC)
Enhances terminating and re-originating both signaling and media between VoIP and Video networks by supporting supplementary features such as Message Waiting Indication, Call Waiting, Call Transfer, Call Forward, Distinctive Ringing, Call Hold/Resume, Music on Hold.
"Configuring SIP-to-SIP Connections in a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(9)T
Survivable Remote Site Telephony Version 4.0
Cisco Unified Survivable Remote Site Telephony (SRST): All Versions
12.4(6)T1
Media and Signaling Authentication and Encryption Feature for Cisco IOS H.323 Gateways
Provides authentication, integrity, and encryption of voice media and call control signaling for H.323 protocol-based voice gateways. New secure voice call capabilities between gateways include:
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Media and Signaling Authentication and Encryption Feature for Cisco IOS H.323 Gateways
12.4(6)T
ANI Suppression During L2TP Setup
Provides the ability to suppress all or some part of the calling number field in the Layer 2 Tunneling Protocol (L2TP) setup process through RADIUS attribute functionality. The Calling Number Suppression for L2TP Setup feature feature allows you to make part or all of the calling number anonymous.
12.4(6)T
Busyout Monitor Gatekeeper
Simplifies monitoring of a large number of voice ports by adding busyout monitor gatekeeper command under voice class busyout mode.
12.4(6)T
Cisco Text Relay for Baudot Text Phones
Implements a mechanism for transporting Text Telephone (TTY) signals over Voice over IP (VoIP) calls in a highly reliable and robust manner. This feature supports Baudot 45.45 and 50 bps text (TTY) phones.
12.4(6)T
In-Service Updates to Gatekeeper Zone Prefix Configuration
Increases the availability of H.323 VoIP networks by allowing changes to a gatekeeper zone prefix while the gatekeeper is running and managing active E.164 registrations.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(6)T
MGCP NAS Package LAPB-TA
Implements autodetection for the MGCP NAS package, as supported in Cisco IOS Release 12.3(9) under ISDN serial interfaces.
12.4(6)T
RSVP Agent
Implements a Resource Reservation Protocol (RSVP) agent on Cisco IOS voice gateways that support Cisco Unified CallManager 5.0.
12.4(6)T
SCCP PLAR with DTMF Out Pulse Digits for FXS Analog Phones
Provides private line automatic ring-down (PLAR) support and enhanced speed-dial capabilities for Skinny Client Control Protocol (SCCP) analog ports on a Cisco IOS voice gateway under the control of Cisco CallManager or a Cisco CallManager Express (Cisco CME) system.
SCCP Controlled Analog (FXS) Ports with Supplementary Features in IOS Gateways
12.4(6)T
SIP: Busy Out Support
Introduces, at the SIP level, a generic keepalive mechanism that allows the SIP gateway to monitor the status of the SIP servers and provide the option of busying-out the associated voice ports upon total keepalive failure.
12.4(6)T
SIP: Cisco IOS Gateway Signaling Support Over TLS Transport
Implements the Transport Layer Security (TLS) protocol on the Transmission Control Protocol (TCP) transport for Cisco IOS SIP gateways. The feature leverages the existing gateway's support of the public-key infrastructure (PKI) (for certificate management) and Open Secure Socket Layer-Transport Layer Security (OPSSL-TLS) application program interfaces (APIs) in order to provide the necessary functionality. The use of PKI on Cisco IOS software requires that the clock on the session initiation protocol (SIP) gateway be synchronized with the network time to ensure proper validation of certificates.
SIP: Cisco IOS SIP Gateway Signaling Support Over TLS Transport
12.4(6)T
SIP-to-SIP Extended Feature Functionality for Session Border Controller (SBC)
Enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP UAs. New SIP-to-SIP features available include:
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The "Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(6)T
Unique Calling Party Information with Alternate Endpoints
Enables alternate endpoint capabilities of the Cisco IOS H.323 gatekeeper and voice gateway to associate a unique calling party number automatic number identification (ANI) with each alternate endpoint using the GKTMP.
The "Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
Call Type Detection Feature in an IP-to-IP Gateway
Enables Cisco H.323 VoIP gateways to report the call type (voice/fax/modem) to a Cisco IOS gatekeeper at the end of each call.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
CDRs for Alternate Endpoints Tried in an IP-to-IP Gateway
Controls alternate endpoint hunting based on call disconnect cause codes.
"Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
Cisco CallManager Express (Cisco CME) 3.4
Cisco CallManager Express (Cisco CME) 3.4 adds station-side RFC3261 standard-based support for Session Initiation Protocol (SIP) phones directly into Cisco CME. This enables Cisco IP phones to place calls across SIP networks in the same way that the current Skinny Client Control Protocol (SCCP) phones do.
For full information aboutCisco CME 3.4, see the Cisco CallManager Express 3.4 Configuration Guide.
12.4(4)T
Cisco Modem Relay
Cisco Modem Relay implements non-negotiated, bearer switched modem relay (gateway-controlled) on select gateways, enabling V.34 modem traffic to be reliably transported. Cisco Modem Relay supports H.323, SIP and MGCP signaling types, and because it is gateway-controlled, all call agents, including Cisco CallManager, Cisco CallManager Express, Cisco PGW Softswitch and Cisco BTS Softswitch are supported.
"Configuring Cisco Modem Relay" chapter of the Cisco IOS Fax and Modem Services over IP Application Guide.
This new combined guide replaces the previous Cisco IOS Fax Services over IP Application Guide and Modem Support for VoIP in the VCL.
12.4(4)T
DSP Voice-Quality Statistics in DLCX Messages
Added new voice quality parameters, and two new keywords to mgcp voice-quality-stats. Provides a method to trace a Media Gateway Control Protocol (MGCP) call between a Cisco PGW 2200 and the Cisco IOS gateway by including the MGCP call ID and the DS0 and digital signal processor (DSP) channel ID in call-active and call-history records.
12.4(4)T
Enhancing CISCO-H225-MIB with Disconnect Cause Codes
The CISCO-H225-MIB was enhanced with the Q.931 disconnect cause codes that the H.323 subsystem can receive. A disconnect can originate from the far-end gateway or from the opposite call leg on the local gateway. This enhancement to the MIB allows you to report disconnect cause code information, including the cause code type and the number of cause code disconnects received from either H.323 peer. The enhancement corresponds to the usage of the show h323 gateway command. See the show h323 gateway command for an example of the disconnect cause code display.
The Enhancing CISCO-H225-MIB with Disconnect Cause Codes feature provides SNMP MIB enhancements on the following platforms:
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The MIB contains objects that represent active H.323 calls and also includes call details. For definitions of the H.323 MIB objects, see the following MIBs:
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To locate and download MIBs, use Cisco MIB Locator found at the following URL:
12.4(4)T
Fax Relay Support for SG3 Fax Machines at G3 Speeds
The Fax Relay Support for SG3 Fax Machines at G3 Speeds feature introduces a fax machine spoofing mechanism on select gateways to force Super Group 3 (SG3) fax machines to automatically fall back to Group 3 (G3) speeds. This enables faxes to be sent between 2 SG3 fax machines over T.38 Fax Relay and Cisco Fax Relay at the supported G3 speeds (14.4 kbps).
Configuring Cisco Fax Relay" and "Configuring T.38 Fax Relay" chapters of the Cisco IOS Fax and Modem Services over IP Guide.
This new combined guide replaces the previous Cisco IOS Fax Services over IP Application Guide and Modem Support for VoIP in the VCL.
12.4(4)T
Final Flag notification from the GKTMP Server
Enables a control field in the Gatekeeper Transaction Message Protocol (GKTMP) that allows an external application to halt normal alternate routing procedures at the gatekeeper, to reduce call setup times and reject calls quickly during peak traffic periods in the wholesale provider's network.
No configuration is required.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
H.323 Standard Based Hopcount Field in LRQ
Support for H.225 version 4 standard hopCount field in LocationRequest RAS message.
No configuration is required.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
Interoperability Enhancements to the Cisco Multiservice IP-IP Gateway
Enables operation of IP-to-IP gateway features concurrently on the same router with H.323 gatekeeper and TDM-IP voice-gateway features.
12.4(4)T
MGCP Call Centric Debug
Enables the filtering of MGCP debug output based on selected criteria and standardizes the format of the MGCP debug header. All MGCP debug output for a single call can be identified and correlated across the various layers in IOS software. Filtering debug output reduces extraneous information, making it easier to locate the correct information and reducing the impact to platform performance.
"Filtering Troubleshooting Output" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide.
12.4(4)T
MGCP CAS MD Package
Introduces support for Feature Group D (FGD) Exchange Access North American (EANA) protocol signaling. The MD package adds support for the reporting of automatic number identification (ANI) and dialed number identification service (DNIS) digits to enable the MGCP call agent to better handle customer billing.
"Configuring MGCP CAS MD Package" chapter in the Cisco IOS MGCP and Related Protocols Configuration Guide.
12.4(4)T
MGCP Endpoint Range Support
Extends the mgcp behavior command by adding the rsip-range keyword. The rsip-range keyword controls whether the gateway can generate ReStart In Progress (RSIP) messages with endpoint ranges for versions other than Trunking Gateway Control Protocol (TGCP).
12.4(4)T
No Retry on User Busy in an IP-to-IP Gateway
Changes the default behavior of the gateway to not retry alternate endpoints when the release complete reason is user busy.
" Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
RAS retry and timer
Allows service providers the ability to control transmit time margins on Cisco gatekeepers by changing RAS message timeout LRQ value and message retry counter values.
" Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
Secure Communication BetweenIP-STE Endpoint and Line-Side STE Endpoint
Provides the following Cisco IOS gateway capabilities:
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Secure Communication BetweenIP-STE Endpoint and Line-Side STE Endpoint
12.4(4)T
Sequential LRQ timer
Defines the time window during which the gatekeeper collects responses from the gateway before resending a RAS message to a gatekeeper, and the number of times to resend the RAS message after the timeout period expires.
"Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
SIP: CLI for Caller ID When Privacy Exists
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12.4(4)T
SIP-to-H.323 Extended Call Interworking
Enables the IP-to-IP gateway to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323).
"Features Supported by the Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
SIP-to-SIP Basic Call Interworking
Enables the IP-to-IP gateway to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323).
" Configuring a Cisco Multiservice IP-to-IP Gatekeeper" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
12.4(4)T
Support for IP-to-IP Gateway and Gatekeeper Features on the Cisco 2801
Provides integrated voice and video services on the Cisco 2801.
"Configuring a Cisco Multiservice IP-to-IP Gateway " chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.4(4)T
Survivable Remote Site Telephony (SRST) 3.4
Cisco SIP SRST Version 3.4 describes SRST functionality for Session Initiation Protocol (SIP) networks. Cisco SIP SRST Version 3.4 provides backup to an external SIP proxy server by providing basic registrar and back-to-back user agent (B2BUA) services. These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy.
Cisco SIP SRST Version 3.4 can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco SIP SRST Version 3.4, SIP phones can place calls across SIP networks in the same way as Skinny Client Control Protocol (SCCP) phones.
12.4(4)T
Test Call
Provides the ability for a remote station or gateway to establish a call to any destination address from a Test Call station located at a network operations center and to audibly verify the voice path.
"Troubleshooting H.323 Interfaces to the IP Network" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide
12.4(4)T
Voice Call Debug Filtering on H.323 Gatekeepers
Enables selected debugging traces for voice calls. This feature allows you to filter and trace voice call debug messages based on selected filtering criteria, reducing the volume of output for more efficient troubleshooting.
"Filtering Troubleshooting Output" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide
12.4(2)T
Land Mobile Radio (LMR) over IP Enhancement
The Land Mobile Radio (LMR) over IP Enhancement feature allows Cisco multiservice routers to transport LMR traffic over IP networks by modifying voice gateway functionality. LMR over IP enables LMR systems to extend beyond their traditional geographic limitations created by transmitter signal strength and enables interoperability, allowing public safety personnel in different agencies or jurisdictions to communicate with each other by radio on demand, in real time.
12.4(2)T
MGCP Controlled Backhaul of BRI Signaling
Extends support for the MGCP-Controlled Backhaul of BRI Signaling in Conjunction with the Cisco CallManager feature to the NM-HD, NM-HDV2, EVM-HD, and Cisco 2800/3800 series with a BRI HWIC interface.
"Configuring MGCP-Controlled Backhaul of BRI Signaling in Conjunction with Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.4(2)T
R2 Call Blocking for Brazil
E1 R2 Collect Call Blocking provides incoming collect call block support. Collect calls will be blocked based on a specific category. For example, in Brazil, collect calls arrive with a category II-8 for which the Cisco access router sends B-7 as response instead of an answer signal.
For an incoming collect call, the gateway answers the call with a clear-back after 1 second and re-answers the call after 2 seconds. This causes the collect call to be dropped and normal calls to stay connected. This is implemented as a CLI option.
12.4(2)T
SCCP Analog (FXS) Ports
Enables Skinny Client Control Protocol (SCCP) supplementary features on analog FXS ports on a Cisco VG 224 voice gateway under the control of Cisco CallManager or Cisco CallManager Express (Cisco CME).
12.4(2)T
Secure Communication Between IP-STE Endpoint and Trunkside STE Endpoint
Supports encrypted and decrypted calls from an IP secure terminal equipment (STE) controlled by Cisco CallManager through a voice gateway to an STE in the Defense Switch Network (DSN). This feature implements a subset of the V.150.1 modem relay standard, allowing users to operate US Department of Defense-compliant (Type-1 encryption) devices across a VoIP network, and between VoIP networks and the Defense Switching Network.
The mgcp modem relay voip mode, mgcp modem relay voip mode sse, mgcp modem relay voip sprt v.14, mgcp package-capability, show call active voice, show mgcp, show mgcp connection, and show modem relay statistics commands in the Cisco IOS Voice Command Reference and the debug modem relay v14 command in the Cisco IOS Debug Command Reference
12.4(2)T
SIP: Domain Name Support in SIP Headers
The SIP: Domain Name Support in SIP Headers feature adds a command-line interface (CLI) switch to provide a host or domain name in the host portion of the locally generated SIP headers (for example, From, RPID, and Call-ID). The SIP: Domain Name Support in SIP Headers feature also affects the outgoing dialog initiating SIP requests (for example, INVITE and SUBSCRIBE message requests).
"Configuring SIP Message, Timer, and Response Features" of the Cisco IOS SIP Configuration Guide
12.4(2)T
SIP: Multilevel Precedence and Priority Support
The SIP: Multilevel Precedence and Priority Support feature enables Cisco IOS gateways to interoperate with other multilevel-precedence and preemption (MLPP)-capable circuit-switched networks.
An MLPP-enabled call has an associated priority level that applications that handle emergencies and congestions use to determine which lower-priority call to preempt in order to dedicate their end-system resources to high-priority communications. This feature addresses the aspect of preemption when interworking with defense-switched networks (DSNs) that are connected through the Cisco IOS gateway.
"Configuring SIP Connection-Oriented Media, Forking, and MLPP Features" of the Cisco IOS SIP Configuration Guide
12.4(2)T
SIP Stack Portability
The SIP Stack Portability feature implements the following capabilities to the SIP gateway Cisco IOS stack:
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It sends an INVITE message request without Session Definition Protocol (SDP) and provides SDP in either the PRACK or ACK message request for both initial call establishment and mid-call re-INVITE message requests.
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The stack applies certain rules and restrictions for a subset of headers and for some content types (such as SDP) to protect the integrity of the stack's functionality and to maintain backward compatibility. When receiving SIP message requests, it reads the SIP header and any attached body without any restrictions.
12.4(2)T
SIP: User Agent MIB Enhancements
The SIP: User Agent MIB Enhancements feature provides SNMP MIB object enhancements to the CISCO-SIP-UA-MIB and informational updates to the CISCO-SIP-UA-CAPABILITY file. The CISCO-SIP-UA-CAPABILITY file provides information such as the Cisco IOS release number and the capabilities of the CISCO-SIP-UA-MIB. The SNMP MIB object updates provide configuration and counter support that are equivalent to command-line interface additions introduced in several SIP features.
In addition, the SIP: User Agent MIB Enhancements feature provides two new MIB objects. In Release 12.3(4)T, the SIP Gateway Support Enhancements to the bind Command feature extended the Cisco IOS bind command by adding the media keyword. The media keyword allows multiple instances of the bind command. One instance defines the control address, and one instance defines the media address. Because the original CISCO-SIP-UA-MIB was defined with a single instance of the bind command, the MIB objects that support the bind command are replaced with two new MIB objects that support multiple instances.
12.4(2)T
SIP: User Agent MIB Enhancements (Continued)
Any SNMP applications that SET or GET the following objects from the CISCO-SIP-UA-MIB need to refer to the new objects:
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While the objects above have not been removed and are still accessible with Cisco IOS Release 12.4(2)T, they will be removed in a future release. Users must upgrade any affected application to the following new objects:
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You can specify pairs of cSipCfgBindSourceAddrInterface and cSipCfgBindSourceAddrScope. Specifying pairs allows you to associate one interface address with control traffic and another interface address with media traffic. Please note that "Src" in the prior objects has been replaced with "Source" in the new objects.
For full definitions of the SIP MIB objects, see the CISCO-SIP-UA-MIB. To locate and download MIBs, use Cisco MIB Locator found at the following URL:
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