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:
• New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order, page 2
• New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release,
page 25

Note For information about the full set of Cisco IOS voice features, see the entire Cisco IOS Voice
Configuration Library—including library preface, glossary, and other documents—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

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 New Voice and Telephony Features in Cisco IOS Release 12.4T
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

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

Feature
ANI Suppression During L2TP
Setup
Busyout Monitor Gatekeeper
Call Detail Records (CDR)
Feature Correlation ID for
Supplementary Features
Call Type Detection Feature in an 12.4(4)T
IP-to-IP Gateway
CDRs for Alternate Endpoints
Tried in an IP-to-IP Gateway
First
Supported
Release Feature Description Where Documented
12.4(6)T Provides the ability to suppress all or ANI Suppression During
some part of the calling number field L2TP Setup
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 Simplifies monitoring of a large Trunk-Management
number of voice ports by adding Features
busyout monitor gatekeeper
command under voice class busyout
mode.
12.4(9)T Captures additional information in RADIUS VSA Voice
CDRs for voice calls that are Implementation Guide
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.
Enables Cisco H.323 VoIP gateways “Configuring a Cisco
to report the call type Multiservice IP-to-IP
(voice/fax/modem) to a Cisco IOS Gatekeeper” chapter in the
gatekeeper at the end of each call. Cisco Multiservice IP-to-IP
Gateway Application Guide
12.4(4)T Controls alternate endpoint hunting “Configuring a Cisco
based on call disconnect cause codes. Multiservice IP-to-IP
Gateway” chapter in the
Cisco Multiservice IP-to-IP
Gateway Application Guide

2
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
Cisco CallManager Express 12.4(4)T Cisco CallManager Express (Cisco Cisco CallManager Express
(Cisco CME) 3.4 CME) 3.4 adds station-side RFC3261 3.4 Configuration Guide
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 12.4(9)T Delivers a number of key telephony Cisco Unified CallManager
(CME) 4.0(1) features for customers including: Express Roadmap: All
Q.SIG integration with TDM PBX's, Versions
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 “Configuring Basic
Version 2.0 W3C Recommendation Functionality for Tcl IVR
(March 16, 2004) on Cisco IOS voice and VoiceXML
gateway VoiceXML browsers which Applications” chapter of the
enables interaction with VoiceXML Cisco Tcl IVR and
applications. VoiceXML Application
Guide
“Cisco VoiceXML Features”
and “Cisco VoiceXML
Elements: Reference Table”
chapters of the Cisco
VoiceXML Programmer’s
Guide

3
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
Cisco IOS VoiceXML Browser
Update to W3C Voice XML 2.1
Cisco Modem Relay
Cisco Text Relay for Baudot Text 12.4(6)T
Phones
Cisco Unified CallManager
Express 4.0(3)
First
Supported
Release Feature Description
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.
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.
Implements a mechanism for
transporting Text Telephone (TTY) Text Phones
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(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.
Where Documented
"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
“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
Cisco Unified
Communications Manager
Express System
Administrator Guide

4
 New Voice and Telephony Features in Cisco IOS Release 12.4T
Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)
Feature
Cisco Unified CallManager
Express SIP Station-Side
Enhancements
Cisco Unified Communications
Manager Express Release 4.2
Customizable PSTN Tones and
H.323 Call-Disconnect Cause
Codes
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
First
Supported
Release Feature Description Where Documented
12.4(15)T Includes music on hold (MoH), MoH Cisco Unified
with transcoding, dialplan-pattern, Communications Manager
KPML and dialplan, speed dial, Express System
caller ID and status line update, Administrator Guide
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.
12.4(15)T Introduces interoperability with a Cisco Unified
session server, such as Cisco Unified Communications Manager
Contact Center Express (Unified Express Call Monitoring
CCX). This interface guide details Interface Guide
the configuration, registration, and
subscription portions of the call and
line monitoring functions on the
Cisco Unified CME.
12.4(9)T Enables you to customize the Customizable PSTN Tones
following PSTN tones and H.323 and H.323 Call-Disconnect
call-disconnect cause codes for Cause Codes
certain disconnect scenarios:
• PSTN tones that are applicable
to FXS, PRI, and BRI calls and
IP phones
• Q.850 call-disconnect cause
codes for H.323 gateways
You can also specify the mechanism
for detecting media inactivity
(silence) on a voice call: RTP, RTCP,
or both.
5
 New Voice and Telephony Features in Cisco IOS Release 12.4T
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Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
DSP Voice-Quality Statistics in 12.4(4)T Added new voice quality parameters, DSP Voice-Quality Statistics
DLCX Messages and two new keywords to in DLCX Messages
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 12.4(9)T Enhances the 911 interconnect Enhanced MF for FGD and
Analog CAMA Trunks capabilities of Cisco IOS based Analog CAMA Trunks
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.

6
 New Voice and Telephony Features in Cisco IOS Release 12.4T
Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)
Feature
Enhancing CISCO-H225-MIB
with Disconnect Cause Codes
Extending Dynamic Zone Prefix
Registration to Include Gateway
Priority
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
First
Supported
Release Feature Description Where Documented
12.4(4)T The CISCO-H225-MIB was Release Notes
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:
• Cisco AS5350 series universal
gateways
• Cisco AS5400 series universal
gateways
• Cisco AS5850 universal
gateways
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:
• CISCO-H225-MIB
To locate and download MIBs, use
Cisco MIB Locator found at the
following URL:
http://www.cisco.com/go/mibs
12.4(9)T Simplifies the H.323 zone “Configuring H.323
configuration process by defining the Gateways” chapter of the
zone prefix and the corresponding Cisco IOS H.323
gateway priorities together on the Configuration Guide
gateway.
7
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Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
Fax Relay Support for SG3 Fax
Machines at G3 Speeds
Final Flag notification from the
GKTMP Server
H.323 Standard Based Hopcount
Field in LRQ
H.323 VoIP Call Preservation
Enhancements for WAN Link
Failures
First
Supported
Release Feature Description Where Documented
12.4(4)T Introduces a fax machine spoofing “Configuring Cisco Fax
mechanism on select gateways to Relay” and “Configuring
force Super Group 3 (SG3) fax T.38 Fax Relay” chapters of
machines to automatically fall back the Cisco IOS Fax and
to Group 3 (G3) speeds. This enables Modem Services over IP
faxes to be sent between 2 SG3 fax Guide.
machines over T.38 Fax Relay and This new combined guide
Cisco Fax Relay at the supported G3 replaces the previous Cisco
speeds (14.4 kbps). IOS Fax Services over IP
Application Guide and
Modem Support for VoIP in
the VCL.
12.4(4)T Enables a control field in the “ Configuring a Cisco
Gatekeeper Transaction Message Multiservice IP-to-IP
Protocol (GKTMP) that allows an Gatekeeper” chapter in the
external application to halt normal Cisco Multiservice IP-to-IP
alternate routing procedures at the Gateway Application Guide
gatekeeper, to reduce call setup times
and reject calls quickly during peak
traffic periods in the wholesale
provider’s network.
No configuration is required.
12.4(4)T Support for H.225 version 4 standard “Configuring a Cisco
hopCount field in LocationRequest Multiservice IP-to-IP
RAS message. Gatekeeper” chapter in the
Cisco Multiservice IP-to-IP
No configuration is required.
Gateway Application Guide
12.4(9)T Sustains connectivity for H.323 topol- “Configuring H.323
ogies where signaling is handled by an Gateways” chapter of the
entity that is different from the other Cisco IOS H.323
endpoint, such as a gatekeeper that Configuration Guide
provides routed signaling or a call
agent, such as the Cisco BTS 10200
Softswitch, Cisco PGW 2200, or
Cisco Unified CallManager, that bro-
kers signaling between the two con-
nected parties.
Call preservation is useful when a
gateway and the other endpoint (typi-
cally a Cisco Unified IP phone) are
collocated at the same site and the call
agent is remote and therefore more
likely to experience connectivity fail-
ures.

8
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
High-Density Packet Voice
Feature Card for Cisco
AS5350XM and AS5400XM
Universal Gateways
iLBC Codec Support
iLBC Support for SIP and H.323
In-Service Updates to Gatekeeper 12.4(6)T
Zone Prefix Configuration
Integrated Data Primary Rate
Interface (PRI) Services
Interoperability Enhancements to 12.4(4)T
the Cisco Multiservice IP-IP
Gateway
First
Supported
Release Feature Description
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.
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.
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.
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.
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.
Enables operation of IP-to-IP
gateway features concurrently on the Gateway Application Guide
same router with H.323 gatekeeper
and TDM-IP voice-gateway features.
Where Documented
High-Density Packet Voice
Feature Card for Cisco
AS5350XM and AS5400XM
Universal Gateways
“Dial Peer Overview”
chapter and “Dial Peer
Features and Configuration”
chapter in Dial Peer
Configuration on Voice
Gateway Routers
“Dial Peer Overview”
chapter and “Dial Peer
Features and Configuration”
chapter in Dial Peer
Configuration on Voice
Gateway Routers
“ Configuring a Cisco
Multiservice IP-to-IP
Gatekeeper” chapter in the
Cisco Multiservice IP-to-IP
Gateway Application Guide
Integrating Data and Voice
Services for ISDN PRI
Interfaces on Multiservice
Access Routers.
Cisco Multiservice IP-to-IP

9
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
Land Mobile Radio (LMR) over 12.4(2)T Allows Cisco multiservice routers to Land Mobile Radio over IP
IP Enhancement transport LMR traffic over IP Enhancement
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 12.4(6)T1 Provides authentication, integrity, Media and Signaling
Authentication and Encryption and encryption of voice media and Authentication and
Feature for Cisco IOS H.323 call control signaling for H.323 Encryption Feature for
Gateways protocol-based voice gateways. New Cisco IOS H.323 Gateways
secure voice call capabilities
between gateways include:
• Gateway to gateway call control
authentication and encryption
using IPSec.
• Media encryption and
authentication of voice streams
using SRTP.
• Exchange of RTP Control
Protocol (RTCP) information
using Secure RTCP.
• SRTP to RTP fallback for calls
between secure and nonsecure
endpoints. You can configure
secure call fallback either
globally or by dial peer.
• Cisco IOS IP-to-IP gateway
interoperation with secure Cisco
IOS H.323 gateways.

10
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
Media Resource Control Protocol 12.4(15)T Provides support for MRCP v2 for "Configuring Basic
(MRCP) version 2 use with specified Cisco IOS voice Functionality for Tcl IVR
gateways' VoiceXML Browser. The and VoiceXML
MRCP v2 protocol allows a client Applications" chapter of the
device to control media processing Cisco Tcl IVR and
resources on the network. Media VoiceXML Application
processing resources include speech Guide, Release 12.3(14)T
recognition engines, speech and later.
synthesis engines, speaker “Cisco VoiceXML Features”
verification and speaker and “Cisco VoiceXML
identification engines. MRCP v2 Elements: Reference Table”
enables the implementation of chapters of the Cisco
distributed Interactive Voice VoiceXML Programmer’s
Response (IVR) platforms using Guide
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.
MGCP Call Centric Debug 12.4(4)T Enables the filtering of MGCP debug “Filtering Troubleshooting
output based on selected criteria and Output” chapter in the
standardizes the format of the MGCP Cisco IOS Voice
debug header. All MGCP debug Troubleshooting and
output for a single call can be Monitoring Guide.
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.
MGCP CAS MD Package 12.4(4)T Introduces support for Feature Group “Configuring MGCP CAS
D (FGD) Exchange Access North MD Package” chapter in the
American (EANA) protocol Cisco IOS MGCP and
signaling. The MD package adds Related Protocols
support for the reporting of Configuration Guide.
automatic number identification
(ANI) and dialed number
identification service (DNIS) digits
to enable the MGCP call agent to
better handle customer billing.

11
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
MGCP Controlled Backhaul of
BRI Signaling
MGCP Endpoint Range Support
MGCP Layer 2 Teardown for IUA 12.4(9)T
DPNSS Trunks
MGCP NAS Package LAPB-TA
No Retry on User Busy in an
IP-to-IP Gateway
Outbound Proxy Support for the
SIP Gateway
First
Supported
Release Feature Description
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.
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).
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(6)T Implements autodetection for the MGCP NAS Package
MGCP NAS package, as supported in LAPB-TA
Cisco IOS Release 12.3(9) under
ISDN serial interfaces.
12.4(4)T Changes the default behavior of the
gateway to not retry alternate
endpoints when the release complete
reason is user busy.
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.
Where Documented
“Configuring
MGCP-Controlled Backhaul
of BRI Signaling in
Conjunction with
Cisco CallManager” chapter
in the Cisco CallManager
and Cisco IOS
Interoperability Guide.
Cisco IOS Voice Command
Reference.
MGCP Layer 2 Teardown for
IUA DPNSS Trunks
“ Configuring a Cisco
Multiservice IP-to-IP
Gateway” chapter in the
Cisco Multiservice IP-to-IP
Gateway Application Guide
“Configuring SIP Message,
Timer, and Response
Features” chapter of the
Cisco IOS SIP
Configuration Guide,
Release 12.4T.

12
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
R2 Call Blocking for Brazil 12.4(2)T Provides incoming collect call block Release Notes
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 “ Configuring a Cisco
to control transmit time margins on Multiservice IP-to-IP
Cisco gatekeepers by changing RAS Gatekeeper” chapter in the
message timeout LRQ value and Cisco Multiservice IP-to-IP
message retry counter values. Gateway Application Guide
RFC 2833 DTMF MTP 12.4(11)T Passes DTMF tones transparently “Configuring SIP DTMF
Passthrough between SIP endpoints that require Features” chapter in the
either transcoding or use of the RSVP Cisco IOS SIP
Agent feature. 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 RSVP Agent
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 SCCP Analog (FXS) Ports
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).

13
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New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description
SCCP PLAR with DTMF Out 12.4(6)T Provides private line automatic
Pulse Digits for FXS Analog ring-down (PLAR) support and
Phones 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.
SCTP Show/Clear CLI 12.4(11)T Provides access to additional SCTP Cisco IOS Voice Command
Enhancements 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.
Secure Communication Between 12.4(2)T Supports encrypted and decrypted
IP-STE Endpoint and Trunkside calls from an IP secure terminal
STE Endpoint 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.
Where Documented
SCCP Controlled Analog
(FXS) Ports with
Supplementary Features in
IOS Gateways
Reference
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

14
 New Voice and Telephony Features in Cisco IOS Release 12.4T
Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)
Feature
Secure Communication
BetweenIP-STE Endpoint and
Line-Side STE Endpoint
Secure HTTP client (SSL) for
Cisco IOS VxML Browser
Sequential LRQ timer
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order
First
Supported
Release Feature Description Where Documented
12.4(4)T Provides the following Cisco IOS Secure Communication
gateway capabilities: BetweenIP-STE Endpoint
and Line-Side STE Endpoint
• Support for establishing secure
calls between gateway-attached
secure terminal equipment
(STE) devices, which can be
foreign exchange station (FXS)
and BRI ports, and IP-STE
devices.
• Ability to configure modem
transport methods, and support
for the state signaling events
(SSE) protocol, allowing for
modem signaling end-to-end and
VoIP to modem over IP (MoIP)
transition and operation.
• Interoperation between line-side
and trunk-side gateways and
Cisco CallManager to determine
codec operation and V.150.1
negotiation to support either
modem relay, modem
pass-through, both modem
transport methods, or neither
method.
• Ability to tune V.150.1
modem-relay parameters to
address specific network
conditions.
12.4(15)T Provides secure communications "Configuring Basic
between the Cisco IOS VoiceXML Functionality for Tcl IVR
browser and VoiceXML servers that and VoiceXML
also support HTTP over SSL. The Applications" chapter of the
Cisco IOS VoiceXML Browser Cisco Tcl IVR and
enables interaction with VoiceXML VoiceXML Application
application servers. Guide, Release 12.3(14)T
and later releases
12.4(4)T Defines the time window during “Configuring a Cisco
which the gatekeeper collects Multiservice IP-to-IP
responses from the gateway before Gatekeeper” chapter in the
resending a RAS message to a Cisco Multiservice IP-to-IP
gatekeeper, and the number of times Gateway Application Guide
to resend the RAS message after the
timeout period expires.
15
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
SIP: Busy Out Support
SIP: Cisco IOS Gateway
Signaling Support Over TLS
Transport
SIP: CLI for Caller ID When
Privacy Exists
First
Supported
Release Feature Description Where Documented
12.4(6)T Introduces, at the SIP level, a generic SIP: Busy Out Support
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 Implements the Transport Layer SIP: Cisco IOS SIP Gateway
Security (TLS) protocol on the Signaling Support Over TLS
Transmission Control Protocol Transport
(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.
12.4(4)T The SIP: CLI for Caller ID When Cisco IOS SIP
Privacy Exists feature adds three Configuration Guide
command-line interface (CLI) options
that make the handling of caller ID
information more flexible.
Specifically, the SIP: CLI for Caller
ID When Privacy Exists feature
addresses the following situations:
• Passing along caller ID
information when privacy exists
• Handling the Display Name field
when no display name exists
• Allowing caller ID information
to be passed to ISDN as
network-provided

16
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
SIP: Domain Name Support in 12.4(2)T Adds a command-line interface “Configuring SIP Message,
SIP Headers (CLI) switch to provide a host or Timer, and Response
domain name in the host portion of Features” of the Cisco IOS
the locally generated SIP headers SIP Configuration Guide
(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).
SIP: Multilevel Precedence and 12.4(2)T Enables Cisco IOS gateways to “Configuring SIP
Priority Support interoperate with other Connection-Oriented
multilevel-precedence and Media, Forking, and MLPP
preemption (MLPP)-capable Features” of the Cisco IOS
circuit-switched networks. SIP Configuration Guide
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.
SIP MWI NOTIFY - QSIG MWI 12.4(11)T Enhances MWI functionality to “Configuring SIP MWI
Translation include Features” chapter in the
SIP-MWI-Notify-to-QSIG-MWI Cisco IOS SIP
translation between gateways and Configuration Guide.
routers.
“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.

17
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
SIP:SIP Gateway OOB DTMF
Support with KPML
SIP:SIP Gateway Session Timer
Support
SIP:SIP Gateway Support for
SDP Session Information and
Permit Hostname CLI
SIP: SIP Support for Hookflash
SIP: Support for Asymmetric
SDP
SIP: Support for PAI
SIP: Support for SRTP
First
Supported
Release Feature Description Where Documented
12.4(9)T Provides a command-line interface “Configuring SIP DTMF
(CLI) option that forwards DTMF Features” in the Cisco IOS
tones using KeyPad Markup SIP Configuration Guide.
Language (KPML) by way of SIP
SUBSCRIBE and NOTIFY
messages.
12.4(9)T Enhances session timer support for “Configuring SIP Message,
gateways to comply with IETF Timer, and Response
Session Timer RFC 4028. Features” in the Cisco IOS
SIP Configuration Guide.
12.4(9)T Adds support for Session Protocol “Configuring SIP Message,
Description (SDP) session Timer, and Response
information to comply with IETF Features” in the Cisco IOS
SDP RFC 2327. Adds support for SIP Configuration Guide.
validating up to 10 hostnames for
incoming initial INVITE messages.
12.4(11)T Configures IP Centrex “Configuring SIP Support
supplementary services on for Hookflash” chapter in
SIP-enabled, Foreign Exchange the Cisco IOS SIP
Station (FXS) lines. Configuration Guide.
12.4(15)T Configures SIP gateways to send and “Configuring SIP DTMF
receive Dual Tone Multi-Frequency Features” chapter of the
(DTMF) and dynamic codec Real Cisco IOS SIP
Time Protocol (RTP) packets with Configuration Guide,
different payloads. Release 12.4T.
12.4(15)T Provides support for RFC 3323 and “Configuring SIP Message,
RFC 3325 that allow you to enable Timer, and Response
either P-Asserted-Identity (PAI) or Features” chapter of the
P-Preferred-Identity (PPI) privacy Cisco IOS SIP
headers in outgoing SIP request or Configuration Guide,
response messages to assert the Release 12.4T.
identity of authenticated users in
trusted domains.
12.4(15)T Ensures the integrity of RTP and “Configuring SIP Support
Real-Time Control Protocol (RTCP) for SRTP” chapter of the
packets providing authentication, Cisco IOS SIP
integrity, and encryption of media Configuration Guide,
packets between two SIP endpoints. Release 12.4T.

18
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
SIP Stack Portability 12.4(2)T Implements the following Release Notes
capabilities to the SIP gateway Cisco
IOS stack:
• It receives inbound REFER
message requests both within a
dialog and outside of an existing
dialog from the user agents
(UAs).
• It sends and receives
SUBSCRIBE or NOTIFY
message requests via UAs.
• It receives unsolicited NOTIFY
message requests without having
to subscribe to the event that was
generated by the NOTIFY
message request.
• The portable stack supports
outbound delayed media.
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.
• It sets SIP headers and content
body in requests and responses.
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.

19
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
SIP: User Agent MIB 12.4(2)T Provides SNMP MIB object Release Notes
Enhancements 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.

20
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
SIP: User Agent MIB 12.4(2)T Any SNMP applications that SET or Release Notes
Enhancements (Continued) GET the following objects from the
CISCO-SIP-UA-MIB need to refer to
the new objects:
• cSipCfgBindSrcAddrScope
• cSipCfgBindSrcAddrInterface
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:
• cSipCfgBindSourceAddrScope.
This object can have a value of
either media (1) or control (2).
• cSipCfgBindSourceAddrInterface.
This object can have any value
of integer interface index.
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:
http://www.cisco.com/go/mibs
SIP-to-H.323 Extended Call 12.4(4)T Enables the IP-to-IP gateway to “Features Supported by the
Interworking bridge calls between networks that Cisco Multiservice IP-to-IP
support different VoIP call-signaling Gateway” chapter in the
protocols (SIP and H.323). Cisco Multiservice IP-to-IP
Gateway Application Guide

21
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
SIP-to-SIP Basic Call
Interworking
SIP-to-SIP Extended Feature
Functionality for Session Border
Controller (SBC)
SIP-to-SIP Supplementary
Services for Session Border
Controller(SBC)
First
Supported
Release Feature Description Where Documented
12.4(4)T Enables the IP-to-IP gateway to “ Configuring a Cisco
bridge calls between networks that Multiservice IP-to-IP
support different VoIP call-signaling Gatekeeper” chapter in the
protocols (SIP and H.323). Cisco Multiservice IP-to-IP
Gateway Application Guide
12.4(6)T Enables the SIP-to-SIP functionality The “Configuring a Cisco
to conform with RFC 3261 to Multiservice IP-to-IP
interoperate with SIP UAs. New Gateway” chapter in the
SIP-to-SIP features available Cisco Multiservice IP-to-IP
include: Gateway Application Guide
• Call Admission Control (based
on CPU, memory, total calls)
• Delayed Media Call
• Media Inactivity
• Modem passthrough
• TCP and UDP interworking
• Tcl scripts with SIP NOTIFY
VoiceXML with SIP-to-SIP
• Transport Layer Security (TLS)
• ENUM support
• Lawful Intercept
• Interoperability with Cisco
Unified CallManager 5.0 and
BroadSoft
12.4(9)T Enhances terminating and “Configuring SIP-to-SIP
re-originating both signaling and Connections in a Cisco
media between VoIP and Video Multiservice IP-to-IP
networks by supporting Gateway” chapter in the
supplementary features such as Cisco Multiservice IP-to-IP
Message Waiting Indication, Call Gateway Application Guide
Waiting, Call Transfer, Call Forward,
Distinctive Ringing, Call
Hold/Resume, Music on Hold.

22
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

Feature
SIP REFER
Support for IP-to-IP Gateway and 12.4(4)T
Gatekeeper Features on the
Cisco 2801
Survivable Remote Site
Telephony (SRST) 3.4
Survivable Remote Site
Telephony Version 4.0
First
Supported
Release Feature Description Where Documented
12.4(15)T Allows remote applications to Cisco Unified
establish calls by sending a REFER Communications Manager
message to Cisco Unified CME, Express System
Cisco Unified SRST, or a SIP Administrator Guide
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.
Provides integrated voice and video “Configuring a Cisco
services on the Cisco 2801. Multiservice IP-to-IP
Gateway ” chapter in the
Cisco CallManager and
Cisco IOS Interoperability
Guide.
12.4(4)T Cisco SIP SRST Version 3.4 Cisco IOS SIP SRST Version
describes SRST functionality for 3.4 System Administrator
Session Initiation Protocol (SIP) Guide
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(9)T Adds these key features; Support for Cisco Unified Survivable
IP Communicator Softphone, fax Remote Site Telephony
pass-though for ATA and VG 224/248 (SRST): All Versions
using SCCP mode, Cisco Unity at
remote site, and support for IP Phone
models 7911G, 7941G/GE and
7961G/GE

23
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order

Table 1 New Voice and Telephony Features in Cisco IOS Release 12.4T in Alphabetical Order (continued)

First
Supported
Feature Release Feature Description Where Documented
Test Call 12.4(4)T Provides the ability for a remote “Troubleshooting H.323
station or gateway to establish a call Interfaces to the IP
to any destination address from a Test Network” chapter in the
Call station located at a network Cisco IOS Voice
operations center and to audibly Troubleshooting and
verify the voice path. Monitoring Guide.
Unique Calling Party Information 12.4(6)T Enables alternate endpoint “Configuring a Cisco
with Alternate Endpoints capabilities of the Cisco IOS H.323 Multiservice IP-to-IP
gatekeeper and voice gateway to Gatekeeper” chapter in the
associate a unique calling party Cisco Multiservice IP-to-IP
number automatic number Gateway Application Guide
identification (ANI) with each
alternate endpoint using the GKTMP.
Video Support for SCCP-Based 12.4(9)T Provides the capability to send H.320 “Video Support for
Endpoints encapsulated Audio/Video calls over SCCP-Based Endpoints”
TDM voice interfaces. chapter in the Cisco Unified
CallManager Express
System Adminstrator Guide,
4.0
Voice Call Debug Filtering on 12.4(4)T Enables selected debugging traces “Filtering Troubleshooting
H.323 Gatekeepers for voice calls. This feature allows Output” chapter in the
you to filter and trace voice call Cisco IOS Voice
debug messages based on selected Troubleshooting and
filtering criteria, reducing the volume Monitoring Guidee
of output for more efficient
troubleshooting.

24
 New Voice and Telephony Features in Cisco IOS Release 12.4T
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

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

First
Supported
Release Feature Feature Description Where Documented
12.4(15)T Cisco IOS VoiceXML Browser Provides support for the VoiceXML 2.1 "Configuring Basic
Update to W3C Voice XML 2.1 W3C Candidate Recommendation Functionality for Tcl IVR
(June 13, 2005) on Cisco IOS voice and VoiceXML
gateway VoiceXML browsers which Applications" chapter of the
enables interaction with VoiceXML Cisco Tcl IVR and
applications. 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 Delivers two key features: Extension Cisco Unified
Express 4.0(3) Assigner which allows for easy Communications Manager
deployment or replacement of phones Express System
on site using a TCL IVR application Administrator Guide
and new IP Phone localizations for
Asia and Eastern Europe.
12.4(15)T Cisco Unified CallManager Includes music on hold (MoH), MoH Cisco Unified
Express SIP Station-Side with transcoding, dialplan-pattern, Communications Manager
Enhancements KPML and dialplan, speed dial, caller Express System
ID and status line update, phone Administrator Guide
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.

25
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(15)T Cisco Unified Communications Introduces interoperability with a Cisco Unified
Manager Express Release 4.2 session server, such as Cisco Unified Communications Manager
Contact Center Express (Unified CCX). Express Call Monitoring
This interface guide details the Interface Guide
configuration, registration, and
subscription portions of the call and
line monitoring functions on the Cisco
Unified CME.
12.4(15)T iLBC Support for SIP and H.323 Supports the internet Low Bitrate “Dial Peer Overview”
Codec (iLBC), a standard, chapter and “Dial Peer
high-complexity speech codec that is Features and Configuration”
suitable for robust voice chapter in Dial Peer
communication over IP. This codec is Configuration on Voice
supported on both SIP and H.323. Gateway Routers
12.4(15)T Media Resource Control Provides support for MRCP v2 for use "Configuring Basic
Protocol (MRCP) version 2 with specified Cisco IOS voice Functionality for Tcl IVR
gateways' VoiceXML Browser. The and VoiceXML
MRCP v2 protocol allows a client Applications" chapter of the
device to control media processing Cisco Tcl IVR and
resources on the network. Media VoiceXML Application
processing resources include speech Guide, Release 12.3(14)T
recognition engines, speech synthesis and later.
engines, speaker verification and
“Cisco VoiceXML Features”
speaker identification engines. MRCP
and “Cisco VoiceXML
v2 enables the implementation of
Elements: Reference Table”
distributed Interactive Voice Response
chapters of the Cisco
(IVR) platforms using VoiceXML
VoiceXML Programmer’s
browsers or other client applications
Guide
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.
12.4(15)T Outbound Proxy Support for the Configures an outbound-proxy server “Configuring SIP Message,
SIP Gateway that receives all initiating request Timer, and Response
(INVITE and SUBSCRIBE) messages Features” chapter of the
and routes them to the designated Cisco IOS SIP
destination. Configuration Guide,
Release 12.4T.

26
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(15)T Secure HTTP client (SSL) for Provides secure communications "Configuring Basic
Cisco IOS VxML Browser between the Cisco IOS VoiceXML Functionality for Tcl IVR
browser and VoiceXML servers that and VoiceXML
also support HTTP over SSL. The Applications" chapter of the
Cisco IOS VoiceXML Browser enables Cisco Tcl IVR and
interaction with VoiceXML application VoiceXML Application
servers. Guide, Release 12.3(14)T
and later releases
12.4(15)T SIP: Support for Asymmetric Configures SIP gateways to send and “Configuring SIP DTMF
SDP receive Dual Tone Multi-Frequency Features” chapter of the
(DTMF) and dynamic codec Real Time Cisco IOS SIP
Protocol (RTP) packets with different Configuration Guide,
payloads. Release 12.4T.
12.4(15)T SIP: Support for PAI Provides support for RFC 3323 and “Configuring SIP Message,
RFC 3325 that allow you to enable Timer, and Response
either P-Asserted-Identity (PAI) or Features” chapter of the
P-Preferred-Identity (PPI) privacy Cisco IOS SIP
headers in outgoing SIP request or Configuration Guide,
response messages to assert the identity Release 12.4T.
of authenticated users in trusted
domains.
12.4(15)T SIP: Support for SRTP Ensures the integrity of RTP and “Configuring SIP Support
Real-Time Control Protocol (RTCP) for SRTP” chapter of the
packets providing authentication, Cisco IOS SIP
integrity, and encryption of media Configuration Guide,
packets between two SIP endpoints. Release 12.4T.
12.4(15)T SIP REFER Allows remote applications to establish Cisco Unified
calls by sending a REFER message to Communications Manager
Cisco Unified CME, Cisco Unified Express System
SRST, or a SIP gateway without an Administrator Guide
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.

27
 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(11)T Cisco IOS VoiceXML 2.0 Provides support for the VoiceXML “Configuring Basic
Version 2.0 W3C Recommendation Functionality for Tcl IVR
(March 16, 2004) on Cisco IOS voice and VoiceXML
gateway VoiceXML browsers which Applications” chapter of the
enables interaction with VoiceXML Cisco Tcl IVR and
applications. 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 “Dial Peer Overview”
Codec (iLBC), a standard, chapter and “Dial Peer
high-complexity speech codec that is Features and Configuration”
suitable for robust voice chapter in Dial Peer
communication over IP. iLBC has Configuration on Voice
built-in error correction functionality Gateway Routers
that helps the codec perform in
networks with a high-packet loss.
12.4(11)T RFC 2833 DTMF MTP Passes DTMF tones transparently “Configuring SIP DTMF
Passthrough between SIP endpoints that require Features” chapter in the
either transcoding or use of the RSVP Cisco IOS SIP
Agent feature. 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 Provides access to additional SCTP Cisco IOS Voice Command
Enhancements information that can help with Reference
troubleshooting potential problems.
These enhancements also make the
updated SCTP show and clear
commands consistent with the CLI of
other transport protocols.

28
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New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(11)T SIP MWI NOTIFY - QSIG MWI Enhances MWI functionality to include “Configuring SIP MWI
Translation SIP-MWI-Notify-to-QSIG-MWI Features” chapter in the
translation between gateways and Cisco IOS SIP
routers. 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 “Configuring SIP Support
services on SIP-enabled, Foreign for Hookflash” chapter in
Exchange Station (FXS) lines. the Cisco IOS SIP
Configuration Guide.
12.4(9)T Call Detail Records (CDR) Captures additional information in RADIUS VSA Voice
Feature Correlation ID for CDRs for voice calls that are Implementation Guide
Supplementary Features 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 Delivers a number of key telephony Cisco Unified CallManager
(CME) 4.0(1) features for customers including: Express Roadmap: All
Q.SIG integration with TDM PBX's, Versions
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 “Video Support for
encapsulated Audio/Video calls over SCCP-Based Endpoints”
TDM voice interfaces. chapter in the Cisco Unified
CallManager Express
System Adminstrator Guide,
4.0

29
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(9)T Customizable PSTN Tones and Enables you to customize the following Customizable PSTN Tones
H.323 Call-Disconnect Cause PSTN tones and H.323 call-disconnect and H.323 Call-Disconnect
Codes cause codes for certain disconnect Cause Codes
scenarios:
• PSTN tones that are applicable to
FXS, PRI, and BRI calls and IP
phones
• Q.850 call-disconnect cause codes
for H.323 gateways
You can also specify the mechanism for
detecting media inactivity (silence) on
a voice call: RTP, RTCP, or both.
12.4(9)T Enhanced MF for FGD and Enhances the 911 interconnect Enhanced MF for FGD and
Analog CAMA Trunks capabilities of Cisco IOS based Analog CAMA Trunks
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 Simplifies the H.323 zone “Configuring H.323
Registration to Include Gateway configuration process by defining the Gateways” chapter of the
Priority zone prefix and the corresponding Cisco IOS H.323
gateway priorities together on the Configuration Guide
gateway.
12.4(9)T H.323 VoIP Call Preservation Sustains connectivity for H.323 topolo- “Configuring H.323
Enhancements for WAN Link gies where signaling is handled by an Gateways” chapter of the
Failures entity that is different from the other Cisco IOS H.323
endpoint, such as a gatekeeper that pro- Configuration Guide
vides routed signaling or a call agent,
such as the Cisco BTS 10200 Softswitch,
Cisco PGW 2200, or Cisco Unified Call-
Manager, that brokers signaling between
the two connected parties.
Call preservation is useful when a gate-
way and the other endpoint (typically a
Cisco Unified IP phone) are collocated at
the same site and the call agent is remote
and therefore more likely to experience
connectivity failures.

30
 New Voice and Telephony Features in Cisco IOS Release 12.4T
Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)
First
Supported
Release Feature
12.4(9)T High-Density Packet Voice
Feature Card for Cisco
AS5350XM and AS5400XM
Universal Gateways
12.4(9)T Integrated Data Primary Rate
Interface (PRI) Services
12.4(9)T MGCP Layer 2 Teardown for IUA Stops voice calls from being lost during MGCP Layer 2 Teardown for
DPNSS Trunks
12.4(9)T SIP:SIP Gateway OOB DTMF
Support with KPML
12.4(9)T SIP:SIP Gateway Session Timer Enhances session timer support for
Support
12.4(9)T SIP:SIP Gateway Support for
SDP Session Information and
Permit Hostname CLI
12.4(9)T SIP-to-SIP Supplementary
Services for Session Border
Controller(SBC)
12.4(9)T Survivable Remote Site
Telephony Version 4.0
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release
Feature Description Where Documented
Supports up to six high-density packet High-Density Packet Voice
voice/fax digital signal processor Feature Card for Cisco
(DSP) modules (product number AS5350XM and AS5400XM
AS5X-PVDM2-64), providing Universal Gateways
scalability from 64 to 384 channels.
Enables PRI interfaces that were Integrating Data and Voice
previously only capable of TDM voice Services for ISDN PRI
to also be simultaneously capable of Interfaces on Multiservice
handling PRI Data channels. Access Routers.
a WAN failure by tearing down all IUA DPNSS Trunks
Layer-2 calls and notifying the PBX of
the out-of-service trunk.
Provides a command-line interface “Configuring SIP DTMF
(CLI) option that forwards DTMF Features” in the Cisco IOS
tones using KeyPad Markup Language SIP Configuration Guide.
(KPML) by way of SIP SUBSCRIBE
and NOTIFY messages.
“Configuring SIP Message,
gateways to comply with IETF Session Timer, and Response
Timer RFC 4028. Features” in the Cisco IOS
SIP Configuration Guide.
Adds support for Session Protocol “Configuring SIP Message,
Description (SDP) session information Timer, and Response
to comply with IETF SDP RFC 2327. Features” in the Cisco IOS
Adds support for validating up to 10 SIP Configuration Guide.
hostnames for incoming initial INVITE
messages.
Enhances terminating and “Configuring SIP-to-SIP
re-originating both signaling and media Connections in a Cisco
between VoIP and Video networks by Multiservice IP-to-IP
supporting supplementary features Gateway” chapter in the
such as Message Waiting Indication, Cisco Multiservice IP-to-IP
Call Waiting, Call Transfer, Call Gateway Application Guide
Forward, Distinctive Ringing, Call
Hold/Resume, Music on Hold.
Adds these key features; Support for IP Cisco Unified Survivable
Communicator Softphone, fax Remote Site Telephony
pass-though for ATA and VG 224/248 (SRST): All Versions
using SCCP mode, Cisco Unity at
remote site, and support for IP Phone
models 7911G, 7941G/GE and
7961G/GE
31
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(6)T1 Media and Signaling Provides authentication, integrity, and Media and Signaling
Authentication and Encryption encryption of voice media and call Authentication and
Feature for Cisco IOS H.323 control signaling for H.323 Encryption Feature for
Gateways protocol-based voice gateways. New Cisco IOS H.323 Gateways
secure voice call capabilities between
gateways include:
• Gateway to gateway call control
authentication and encryption
using IPSec.
• Media encryption and
authentication of voice streams
using SRTP.
• Exchange of RTP Control Protocol
(RTCP) information using Secure
RTCP.
• SRTP to RTP fallback for calls
between secure and nonsecure
endpoints. You can configure
secure call fallback either globally
or by dial peer.
• Cisco IOS IP-to-IP gateway
interoperation with secure Cisco
IOS H.323 gateways.
12.4(6)T ANI Suppression During L2TP Provides the ability to suppress all or ANI Suppression During
Setup some part of the calling number field in L2TP Setup
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 Trunk-Management
number of voice ports by adding Features
busyout monitor gatekeeper
command under voice class busyout
mode.
12.4(6)T Cisco Text Relay for Baudot Implements a mechanism for Cisco Text Relay for Baudot
Text Phones transporting Text Telephone (TTY) Text Phones
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.

32
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(6)T In-Service Updates to Increases the availability of H.323 VoIP “Configuring a Cisco
Gatekeeper Zone Prefix networks by allowing changes to a Multiservice IP-to-IP
Configuration gatekeeper zone prefix while the Gatekeeper” chapter in the
gatekeeper is running and managing Cisco Multiservice IP-to-IP
active E.164 registrations. Gateway Application Guide
12.4(6)T MGCP NAS Package LAPB-TA Implements autodetection for the MGCP NAS Package
MGCP NAS package, as supported in LAPB-TA
Cisco IOS Release 12.3(9) under ISDN
serial interfaces.
12.4(6)T RSVP Agent Implements a Resource Reservation RSVP Agent
Protocol (RSVP) agent on Cisco IOS
voice gateways that support Cisco
Unified CallManager 5.0.
12.4(6)T SCCP PLAR with DTMF Out Provides private line automatic SCCP Controlled Analog
Pulse Digits for FXS Analog ring-down (PLAR) support and (FXS) Ports with
Phones enhanced speed-dial capabilities for Supplementary Features in
Skinny Client Control Protocol (SCCP) IOS Gateways
analog ports on a Cisco IOS voice
gateway under the control of Cisco
CallManager or a Cisco CallManager
Express (Cisco CME) system.
12.4(6)T SIP: Busy Out Support Introduces, at the SIP level, a generic SIP: Busy Out Support
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 Implements the Transport Layer SIP: Cisco IOS SIP Gateway
Signaling Support Over TLS Security (TLS) protocol on the Signaling Support Over TLS
Transport Transmission Control Protocol (TCP) Transport
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.

33
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(6)T SIP-to-SIP Extended Feature Enables the SIP-to-SIP functionality to The “Configuring a Cisco
Functionality for Session Border conform with RFC 3261 to interoperate Multiservice IP-to-IP
Controller (SBC) with SIP UAs. New SIP-to-SIP features Gateway” chapter in the
available include: Cisco Multiservice IP-to-IP
Gateway Application Guide
• Call Admission Control (based on
CPU, memory, total calls)
• Delayed Media Call
• Media Inactivity
• Modem passthrough
• TCP and UDP interworking
• Tcl scripts with SIP NOTIFY
VoiceXML with SIP-to-SIP
• Transport Layer Security (TLS)
• ENUM support
• Lawful Intercept
• Interoperability with Cisco Unified
CallManager 5.0 and BroadSoft
12.4(6)T Unique Calling Party Enables alternate endpoint capabilities The “Configuring a Cisco
Information with Alternate of the Cisco IOS H.323 gatekeeper and Multiservice IP-to-IP
Endpoints voice gateway to associate a unique Gatekeeper” chapter in the
calling party number automatic number Cisco Multiservice IP-to-IP
identification (ANI) with each alternate Gateway Application Guide
endpoint using the GKTMP.
12.4(4)T Call Type Detection Feature in Enables Cisco H.323 VoIP gateways to “Configuring a Cisco
an IP-to-IP Gateway report the call type (voice/fax/modem) Multiservice IP-to-IP
to a Cisco IOS gatekeeper at the end of Gatekeeper” chapter in the
each call. Cisco Multiservice IP-to-IP
Gateway Application Guide
12.4(4)T CDRs for Alternate Endpoints Controls alternate endpoint hunting “Configuring a Cisco
Tried in an IP-to-IP Gateway based on call disconnect cause codes. Multiservice IP-to-IP
Gateway” chapter in the
Cisco Multiservice IP-to-IP
Gateway Application Guide

34
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T Cisco CallManager Express Cisco CallManager Express (Cisco Cisco CallManager Express
(Cisco CME) 3.4 CME) 3.4 adds station-side RFC3261 3.4 Configuration Guide
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 “Configuring Cisco Modem
non-negotiated, bearer switched Relay” chapter of the Cisco
modem relay (gateway-controlled) on IOS Fax and Modem
select gateways, enabling V.34 modem Services over IP Application
traffic to be reliably transported. Cisco Guide.
Modem Relay supports H.323, SIP and This new combined guide
MGCP signaling types, and because it replaces the previous Cisco
is gateway-controlled, all call agents, IOS Fax Services over IP
including Cisco CallManager, Cisco Application Guide and
CallManager Express, Cisco PGW Modem Support for VoIP in
Softswitch and Cisco BTS Softswitch the VCL.
are supported.
12.4(4)T DSP Voice-Quality Statistics in Added new voice quality parameters, DSP Voice-Quality
DLCX Messages and two new keywords to Statistics in DLCX
mgcp voice-quality-stats. Provides a Messages
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.

35
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T Enhancing CISCO-H225-MIB The CISCO-H225-MIB was enhanced Release Notes
with Disconnect Cause Codes 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:
• Cisco AS5350 series universal
gateways
• Cisco AS5400 series universal
gateways
• Cisco AS5850 universal gateways
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:
• CISCO-H225-MIB
To locate and download MIBs, use
Cisco MIB Locator found at the
following URL:
http://www.cisco.com/go/mibs

36
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T Fax Relay Support for SG3 Fax The Fax Relay Support for SG3 Fax Configuring Cisco Fax
Machines at G3 Speeds Machines at G3 Speeds feature Relay” and “Configuring
introduces a fax machine spoofing T.38 Fax Relay” chapters of
mechanism on select gateways to force the Cisco IOS Fax and
Super Group 3 (SG3) fax machines to Modem Services over IP
automatically fall back to Group 3 (G3) Guide.
speeds. This enables faxes to be sent This new combined guide
between 2 SG3 fax machines over T.38 replaces the previous Cisco
Fax Relay and Cisco Fax Relay at the IOS Fax Services over IP
supported G3 speeds (14.4 kbps). Application Guide and
Modem Support for VoIP in
the VCL.
12.4(4)T Final Flag notification from the Enables a control field in the “Configuring a Cisco
GKTMP Server Gatekeeper Transaction Message Multiservice IP-to-IP
Protocol (GKTMP) that allows an Gatekeeper” chapter in the
external application to halt normal Cisco Multiservice IP-to-IP
alternate routing procedures at the Gateway Application Guide
gatekeeper, to reduce call setup times
and reject calls quickly during peak
traffic periods in the wholesale
provider’s network.
No configuration is required.
12.4(4)T H.323 Standard Based Hopcount Support for H.225 version 4 standard “Configuring a Cisco
Field in LRQ hopCount field in LocationRequest Multiservice IP-to-IP
RAS message. Gatekeeper” chapter in the
Cisco Multiservice IP-to-IP
No configuration is required.
Gateway Application Guide
12.4(4)T Interoperability Enhancements Enables operation of IP-to-IP gateway Cisco Multiservice IP-to-IP
to the Cisco Multiservice IP-IP features concurrently on the same Gateway Application Guide
Gateway 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 “Filtering Troubleshooting
output based on selected criteria and Output” chapter in the
standardizes the format of the MGCP Cisco IOS Voice
debug header. All MGCP debug output Troubleshooting and
for a single call can be identified and Monitoring Guide.
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.

37
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T MGCP CAS MD Package Introduces support for Feature Group D “Configuring MGCP CAS
(FGD) Exchange Access North MD Package” chapter in the
American (EANA) protocol signaling. Cisco IOS MGCP and
The MD package adds support for the Related Protocols
reporting of automatic number Configuration Guide.
identification (ANI) and dialed number
identification service (DNIS) digits to
enable the MGCP call agent to better
handle customer billing.
12.4(4)T MGCP Endpoint Range Support Extends the mgcp behavior command Cisco IOS Voice Command
by adding the rsip-range keyword. The Reference.
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 Changes the default behavior of the “ Configuring a Cisco
IP-to-IP Gateway gateway to not retry alternate endpoints Multiservice IP-to-IP
when the release complete reason is Gateway” chapter in the
user busy. Cisco Multiservice IP-to-IP
Gateway Application Guide
12.4(4)T RAS retry and timer Allows service providers the ability to “ Configuring a Cisco
control transmit time margins on Cisco Multiservice IP-to-IP
gatekeepers by changing RAS message Gatekeeper” chapter in the
timeout LRQ value and message retry Cisco Multiservice IP-to-IP
counter values. Gateway Application Guide

38
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T Secure Communication Provides the following Cisco IOS Secure Communication
BetweenIP-STE Endpoint and gateway capabilities: BetweenIP-STE Endpoint
Line-Side STE Endpoint and Line-Side STE Endpoint
• Support for establishing secure
calls between gateway-attached
secure terminal equipment (STE)
devices, which can be foreign
exchange station (FXS) and BRI
ports, and IP-STE devices.
• Ability to configure modem
transport methods, and support for
the state signaling events (SSE)
protocol, allowing for modem
signaling end-to-end and VoIP to
modem over IP (MoIP) transition
and operation.
• Interoperation between line-side
and trunk-side gateways and Cisco
CallManager to determine codec
operation and V.150.1 negotiation
to support either modem relay,
modem pass-through, both modem
transport methods, or neither
method.
• Ability to tune V.150.1
modem-relay parameters to
address specific network
conditions.
12.4(4)T Sequential LRQ timer Defines the time window during which “Configuring a Cisco
the gatekeeper collects responses from Multiservice IP-to-IP
the gateway before resending a RAS Gatekeeper” chapter in the
message to a gatekeeper, and the Cisco Multiservice IP-to-IP
number of times to resend the RAS Gateway Application Guide
message after the timeout period
expires.

39
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T SIP: CLI for Caller ID When The SIP: CLI for Caller ID When Cisco IOS SIP
Privacy Exists Privacy Exists feature adds three Configuration Guide
command-line interface (CLI) options
that make the handling of caller ID
information more flexible. Specifically,
the SIP: CLI for Caller ID When Privacy
Exists feature addresses the following
situations:
• Passing along caller ID
information when privacy exists
• Handling the Display Name field
when no display name exists
• Allowing caller ID information to
be passed to ISDN as
network-provided
12.4(4)T SIP-to-H.323 Extended Call Enables the IP-to-IP gateway to bridge “Features Supported by the
Interworking calls between networks that support Cisco Multiservice IP-to-IP
different VoIP call-signaling protocols Gateway” chapter in the
(SIP and H.323). Cisco Multiservice IP-to-IP
Gateway Application Guide
12.4(4)T SIP-to-SIP Basic Call Enables the IP-to-IP gateway to bridge “ Configuring a Cisco
Interworking calls between networks that support Multiservice IP-to-IP
different VoIP call-signaling protocols Gatekeeper” chapter in the
(SIP and H.323). Cisco Multiservice IP-to-IP
Gateway Application Guide
12.4(4)T Support for IP-to-IP Gateway Provides integrated voice and video “Configuring a Cisco
and Gatekeeper Features on the services on the Cisco 2801. Multiservice IP-to-IP
Cisco 2801 Gateway ” chapter in the
Cisco CallManager and
Cisco IOS Interoperability
Guide.

40
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(4)T Survivable Remote Site Cisco SIP SRST Version 3.4 describes Cisco IOS SIP SRST Version
Telephony (SRST) 3.4 SRST functionality for Session 3.4 System Administrator
Initiation Protocol (SIP) networks. Guide
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 “Troubleshooting H.323
or gateway to establish a call to any Interfaces to the IP
destination address from a Test Call Network” chapter in the
station located at a network operations Cisco IOS Voice
center and to audibly verify the voice Troubleshooting and
path. Monitoring Guide
12.4(4)T Voice Call Debug Filtering on Enables selected debugging traces for “Filtering Troubleshooting
H.323 Gatekeepers voice calls. This feature allows you to Output” chapter in the
filter and trace voice call debug Cisco IOS Voice
messages based on selected filtering Troubleshooting and
criteria, reducing the volume of output Monitoring Guide
for more efficient troubleshooting.
12.4(2)T Land Mobile Radio (LMR) over The Land Mobile Radio (LMR) over IP Land Mobile Radio over IP
IP Enhancement Enhancement feature allows Cisco Enhancement
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.

41
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(2)T MGCP Controlled Backhaul of Extends support for the “Configuring
BRI Signaling MGCP-Controlled Backhaul of BRI MGCP-Controlled Backhaul
Signaling in Conjunction with the of BRI Signaling in
Cisco CallManager feature to the Conjunction with
NM-HD, NM-HDV2, EVM-HD, and Cisco CallManager” chapter
Cisco 2800/3800 series with a BRI in the Cisco CallManager
HWIC interface. and Cisco IOS
Interoperability Guide.
12.4(2)T R2 Call Blocking for Brazil E1 R2 Collect Call Blocking provides Release Notes
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 Analog (FXS) Ports
(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).

42
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Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(2)T Secure Communication Between Supports encrypted and decrypted calls The mgcp modem relay
IP-STE Endpoint and Trunkside from an IP secure terminal equipment voip mode, mgcp modem
STE Endpoint (STE) controlled by relay voip mode sse, mgcp
Cisco CallManager through a voice modem relay voip sprt
gateway to an STE in the Defense v.14, mgcp
Switch Network (DSN). This feature package-capability, show
implements a subset of the V.150.1 call active voice, show
modem relay standard, allowing users mgcp, show mgcp
to operate US Department of connection, and show
Defense-compliant (Type-1 modem relay statistics
encryption) devices across a VoIP commands in the Cisco IOS
network, and between VoIP networks Voice Command Reference
and the Defense Switching Network. and the debug modem relay
v14 command in the Cisco
IOS Debug Command
Reference
12.4(2)T SIP: Domain Name Support in The SIP: Domain Name Support in SIP “Configuring SIP Message,
SIP Headers Headers feature adds a command-line Timer, and Response
interface (CLI) switch to provide a host Features” of the Cisco IOS
or domain name in the host portion of SIP Configuration Guide
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).
12.4(2)T SIP: Multilevel Precedence and The SIP: Multilevel Precedence and “Configuring SIP
Priority Support Priority Support feature enables Connection-Oriented
Cisco IOS gateways to interoperate Media, Forking, and MLPP
with other multilevel-precedence and Features” of the Cisco IOS
preemption (MLPP)-capable SIP Configuration Guide
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.

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 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(2)T SIP Stack Portability The SIP Stack Portability feature Release Notes
implements the following capabilities
to the SIP gateway Cisco IOS stack:
• It receives inbound REFER
message requests both within a
dialog and outside of an existing
dialog from the user agents (UAs).
• It sends and receives SUBSCRIBE
or NOTIFY message requests via
UAs.
• It receives unsolicited NOTIFY
message requests without having
to subscribe to the event that was
generated by the NOTIFY message
request.
• The portable stack supports
outbound delayed media.
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.
• It sets SIP headers and content
body in requests and responses.
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.

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 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(2)T SIP: User Agent MIB The SIP: User Agent MIB Release Notes
Enhancements 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.

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 New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

Table 2 New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release (continued)

First
Supported
Release Feature Feature Description Where Documented
12.4(2)T SIP: User Agent MIB Any SNMP applications that SET or Release Notes
Enhancements (Continued) GET the following objects from the
CISCO-SIP-UA-MIB need to refer to
the new objects:
• cSipCfgBindSrcAddrScope
• cSipCfgBindSrcAddrInterface
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:
• cSipCfgBindSourceAddrScope.
This object can have a value of
either media (1) or control (2).
• cSipCfgBindSourceAddrInterface.
This object can have any value of
integer interface index.
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:
http://www.cisco.com/go/mibs

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New Voice and Telephony Features in Cisco IOS Release 12.4T
New Voice and Telephony Features in Cisco IOS Release 12.4T Listed by First Supported Release

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