Cisco AS5350XM and Cisco AS5400XM Universal Gateways Software Configuration Guide
Voice Enhancement Features for Cisco IOS Release 12.4(24)T
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Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Table Of Contents

Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Contents

Prerequisites for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Restrictions for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Information About Voice Enhancement Features for Cisco IOS Release 12.4(24)T

SIP Support for Media Forking

Automatic Switch to G.Clear Codec when ISDN Bearer Capability IE Indicates 64K Unrestricted Bearer

Private Line Automatic Ringdown for Trading Turrets Connection

Connection Trunk and Transparent CCS for Trading Turrets

Modem Pass-through Using SIP re-invite

Voice Port Bit Conditioning

Voice Trunk Conditioning and Trunk Conditioning OOS

Enhanced Multifrequency Signaling for Feature Group D (E911)

Call Progress Analysis

T.38 Fax Relay Statistics

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Command Reference

Feature Information for Voice Enhancement Features for Cisco IOS Release 12.4(24)T


Voice Enhancement Features for Cisco IOS Release 12.4(24)T


First Published: February 27, 2009
Last Updated: February 27, 2009

This chapter describes feature support that has been added in Cisco IOS Release 12.4(24)T on Cisco VGD 1T3, Cisco AS5350XM, and AS5400XM voice gateways. These features enhance performance of the voice gateways.

Finding Feature Information in This Module

Your Cisco IOS software release may not support all of the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To reach links to specific feature documentation in this chapter and to see a list of the releases in which each feature is supported, use the "Information About Voice Enhancement Features for Cisco IOS Release 12.4(24)T" section.

Finding Support Information for Platforms and Cisco IOS and Catalyst OS Software Images

Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Contents

Prerequisites for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Restrictions for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Information About Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Additional References

Command Reference

Feature Information for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Prerequisites for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

The prerequisites defined in the following sections apply to the configuration of the Cisco VGD 1T3, Cisco AS5350XM, and Cisco AS5400XM voice gateways:

You must have Cisco IOS Release 12.4(24)T or a later release installed on the Cisco VGD 1T3, Cisco AS5350XM, and Cisco AS5400XM voice gateways.

To enable Secure Real-Time Transport Protocol (SRTP) on a Cisco VGD 1T3, Cisco AS5350XM, or Cisco AS5400XM configured as a Media Gateway Control Protocol (MGCP) gateway, first establish an IPsec connection between the Cisco Unified Communications Manager and the MGCP gateway before using the MGCP SRTP package. Otherwise, media keys are sent in clear text and your voice call is not secure.

Restrictions for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

For the Private Line Automatic Ringdown (PLAR) Connection feature, only E&M immediate start signaling configured on the T1 within the CT3 line is supported. This feature does not support Foreign Exchanger Service (FXS) loop-start signaling.

The Media Forking feature is not supported on the Cisco Unified Border Element, and SRTP is not supported on the forked stream.

Information About Voice Enhancement Features for Cisco IOS Release 12.4(24)T

To configure these enhancement features, you should understand the following:

SIP Support for Media Forking

Automatic Switch to G.Clear Codec when ISDN Bearer Capability IE Indicates 64K Unrestricted Bearer

Private Line Automatic Ringdown for Trading Turrets Connection

Connection Trunk and Transparent CCS for Trading Turrets

Modem Pass-through Using SIP re-invite

Voice Port Bit Conditioning

Voice Trunk Conditioning and Trunk Conditioning OOS

Enhanced Multifrequency Signaling for Feature Group D (E911)

Call Progress Analysis

T.38 Fax Relay Statistics

SIP Support for Media Forking

The Session Initiation Protocol (SIP) Support for Media Forking feature provides the ability to create midcall multiple streams (or branches) of audio associated with a single call and then send those streams of data to different destinations. This feature allows service providers to use technologies such as speech recognition, voice authentication, speech recording, and text-to-speech conversion to provide sophisticated services to their end-user customers. An example is a web-browsing application that uses voice recognition and text-to-speech (TTS) technology to make reservations, verify shipments, or order products.

To enable the media forking feature on the Cisco AS5350XM, Cisco AS5400XM, and Cisco VGD 1T3 voice gateways, enter the media forking command in dial-peer configuration mode, voice-service voip configuration mode or global configuration mode:

dial-peer voice xxx voip
media forking

or

voice-service voip
media forking

or

voice class media tag
media forking

For more information, refer to the "SIP Support for Media Forking" section in "Configuring SIP Connection-Oriented Media Forking and MLPP Features" in the Cisco IOS SIP Configuration Guide.

Automatic Switch to G.Clear Codec when ISDN Bearer Capability IE Indicates 64K Unrestricted Bearer

Switching automatically to the G.Clear codec when the ISDN bearer capability information element (IE) indicates 64K unrestricted digital information is required to ensure consistency between Cisco AS5350XM, Cisco AS5400XM, and Cisco VGD 1T3 voice gateways and Cisco integrated services router (ISR) voice gateways. Without this feature, the gateway rejects a call with 64K unrestricted bearer capability on a voice interface because it is considered a data call, not a voice call.

There are no modifications to the command-line interface for this feature. The capability is added by default when you are configuring voice ports on the gateway platforms.

Private Line Automatic Ringdown for Trading Turrets Connection

The Private Line Automatic Ringdown for Trading Turrets feature improves connection service for turrets in the financial industry—primarily for corporations and enterprises that use turrets and POTS telephones for trading. Implementation of this feature ensures that a call between traders on a PLAR connection will be maintained if one of the traders goes on-hook or on-hold. This new capability also ensures that bandwidth is used only when needed.

For more information about this feature, refer to the Private Line Automatic Ringdown for Trading Turrets document.

Connection Trunk and Transparent CCS for Trading Turrets

The Connection Trunk feature creates a permanent VoIP call between two endpoints. This permanent call allows the connection trunk to pass a supplemental signal, such as hookflash or point-to-point hoot-n-holler, between the two endpoints.

This feature sets up permanent trunks, bridging PBX networks over a WAN cloud. The gateways transparently switch the data to the other end of the trunk without interpreting the data in the bearer or signaling channels. Each DS0 under the controllers must be configured either to one of the channel-associated signaling (CAS) variants or to ext-sig for T-CCS. The voice ports have to be configured with the correct connection number and the dial peers have to be configured to direct the setup message that is generated by the router internally to the other end.

For more information about this feature, refer to the Private Line Automatic Ringdown for Trading Turrets feature guide.

Modem Pass-through Using SIP re-invite

The Modem Pass-through Using SIP re-invite feature brings modem pass-through into compliance with Session Initiation Protocol (SIP) standards. Upon detection of a modem tone, a SIP re-invite is sent in order to adjust the speed to G.711 codec mode on both gateways (instead of sending a Named Signaling Event (NSE) across to the peer gateway). This enables the Cisco AS5350XM, Cisco AS5400XM, and Cisco VGD 1T3 voice gateways to interoperate with other gateways.

As part of the re-invite, the gateway also sends an additional attribute to indicate that the call is modem pass-through. If the call is a fax pass-through, the additional attribute is not sent. If the terminating gateway is a Cisco gateway, then all of the re-invite would function as described. If the terminating gateway is not a Cisco gateway, the re-invite may not indicate that it is a modem. In this case, the originating gateway assumes that it is a fax pass-through call.

Figure 1 shows a topology and sequence for modem pass-through call flow through a protocol.

Figure 1

Modem Pass-through Call Flow through a Protocol

The following steps describe the progress of a call based on the initial recognition of a 2100-Hz tone (± 15 Hz) (as shown in Figure 1):

1. Switch the active codec in use on the call to G.711 (if a codec other than G.711 was previously in use).

2. Disable the high pass filter.

3. Disable voice activity detection (VAD) and comfort noise generation (CNG).

4. Switch from any adaptive or dynamic jitter buffer in use to a fixed-length jitter buffer. (A depth of 200 ms is recommended when switching to a fixed-length jitter buffer.)

To enable the modem pass-through feature, use the modem passthrough command in voice service configuration mode:

modem passthrough {nse | protocol codec {g711alaw | g711ulaw [redundancy]}}

Select the nse keyword to use NSEs

Select the protocol keyword to use SIP/H.323 protocol to signal modem pass-through

Select the codec g711alaw keyword to indicate G.711 a-law (64,000 bps) for E1

Select the codec g711ulaw keyword to indicate G.711 mu-law (64,000 bps) for T1

Select the codec g711 ulaw redundancy keywords to use packet redundancy for modem traffic (in compliance with RFC 2198)

For more information about modem pass-through, refer to the "Configuring Modem Passthrough" chapter in the Cisco IOS Fax, Modem, and Text Support over IP Configuration Guide.

Voice Port Bit Conditioning

To customize signaling parameters for a particular E1 or T1 channel group on a channelized line, use the cas-custom command in controller configuration mode. To disable the signaling customization, use the no form of this command.

You can configure the system to deviate from a country's default settings as defined by Cisco. To do this, choose invert-abcd as an optional keyword:

cas-custom channel {define | ignore | invert-abcd}

The invert-abcd keyword inverts the ABCD bits before TX and after RX. This feature is disabled by default, which is the ITU default.

Voice Trunk Conditioning and Trunk Conditioning OOS

The voice trunk conditioning feature enables you to create a voice class, configure specific signaling attributes to the voice class, and then map the attributes in the voice class to either a Voice over Frame Relay, Voice over ATM, or a Voice over HDLC dial peer. Using the voice class, you can define the keepalive-signaling packet interval and the signal pattern (ABCD) bit pattern for Cisco-trunk (private-line) calls.

Trunk-conditioning signaling attributes apply to permanent point-to-point voice connections (private lines and tie lines) that you create using the connection trunk command.

Trunk conditioning enables control over Cisco private-line calls that are sent over Frame Relay or ATM networks. When private-line or tie-line calls are sent between two PBXs, fault indications are sent to the sending PBX. If the call fails, the PBX can select an alternate path to route the calls. Selecting an alternate path applies to analog connections or digital T1/E1 using channel-associated signaling (CAS) ABCD signaling. It does not apply to common-channel signaling (CCS).

When T1/E1 CAS is carried in transparent pass-through mode for arbitrary, unknown, or unsupported CAS protocols, you must define on-hook or idle patterns so that the digital signal processor (DSP) code can sense the idle call state and shut off the flow of voice packets when no active call is in progress. This mode provides an additional idle bandwidth-saving mechanism for those cases when voice activity detection (VAD) is not desired.

For more information about this feature, refer to the Trunk Management Features document.


Note An out-of-service (OOS) condition can be signaled using an ABCD bit pattern that is different from the busy or seized state. The difference enables the PBX to distinguish between OOS and congestion. For more information, refer to the Trunk Management Features document.


Enhanced Multifrequency Signaling for Feature Group D (E911)

The Enhanced Multifrequency for Feature Group D and Analog CAMA Trunks feature enhances the 911 interconnect capabilities of Cisco IOS-based gateways. This document describes E911 support requirements, which include support for Enhanced Multifrequency signaling for Feature Group D and Analog Centralized Automated Message Accounting (CAMA) signaling protocols per National Emergency Number Association standards. This feature supports 20-digit Automatic Number Identification (ANI) requirements and mapping of remote party IDs (RPID) to PANI.

For more information, refer to the Enhanced MF for FGD and Analog CAMA Trunks document.

Call Progress Analysis

To enable answering machine detection for contact center applications, an enhanced call progress analysis (CPA) function for the Outbound Option Dialer is implemented in Cisco IOS Release 12.4(24)T. The answer machine and answer machine terminating tone detection algorithm is ported from the IPCC Dialer into the PVDM2 DSP voice feature card (VFC) and ISR PVDM2 DSP. The fax/modem detection portion uses existing logic in the PVDM2 DSP.

This feature is enabled by the following Cisco IOS commands in voice service voip configuration mode:

cpa [threshold | timing]

The basic command enables the call progress analysis algorithm for all outbound VoIP calls in the voice gateway. If you select one of the optional keywords, you can specify other parameters for CPA threshold values and CPA timing values.

cpa threshold:

cpa threshold active signal {9db | 12db | 15db | 18db | 21db}

This command sets the threshold of an active signal. If a signal is greater than the measured noise floor level specified in this command, a signal is considered active. The default value is 15 db.

cpa threshold noise-level max {-45dBm0 | -50dBm0 | -55dBm0 | -60dBm0}

This command sets the maximum threshold that the CPA algorithm uses to measure the noise floor level. If the measured noise floor level is greater that the vlaue specified in this command, the measured noise floor level is set to the configured maximum noise floor level. The default value is -50 dBm0.

cpa threshold noise-level min {-55dBm0 | -60dBm0 | -65dBm0 | -70dBm0}

This command sets the minimum threshold that the CPA algorithm uses to measure the noise floor level. If the measured noise floor level is less than the value specified in this command, the measured noise floor level is set to the configured minimum noise floor level. The default value is -60 dBm0.

cpa timing (in milliseconds):

cpa timing live-person 1 ... 60000

This command sets the timing that the CPA algorithm uses to determine whether or not a call is answered by a live person. The default value is 2500 ms.

cpa timing noise-period 1 ... 60000

This command sets the maximum time that the CPA algorithm uses to meaure the noise floor level at the beginning of a call. The default value is 100 ms.

cpa timing silent 1 ... 60000

This command sets the minimum silent duration after active speech is detected in order for the CPA algorithm to determine that a call is answered by a live person. The default value is 375 ms.

cpa timing term-tone 1 ... 60000

This command sets the maximum time that the CPA algorithm waits for the answering machine terminating tone after the answering machine is detected. The default value is 15000 ms.

cpa timing timeout 1 ... 60000

This command sets the maximum time that the CPA algorithm should wait before timing out if it doesn't detect a voice signal. The default value is 3000 ms.

cpa timing valid-speech 1 ... 60000

This command sets the minimum voice duration for the CPA algorithm to consider it a valid speech signal. The default value is 112 ms.

For more information about the use of these commands, refer to the Cisco IOS Voice Command Reference.

T.38 Fax Relay Statistics

The Internet Engineering Task Force (IETF) RADIUS standards (RFCs 2865 and 2866) specify that you can use attribute 26 for communicating vendor-specific information between a network access server (NAS with RADIUS client) and an authentication/accounting server (RADIUS server). RADIUS attribute 26 is the vendor-specific attribute (VSA). The vendor ID for Cisco VSAs is 9.

Beginning with Cisco IOS Release 12.4(24)T, T.38 fax relay statistics are available for RADIUS accounting with new vendor-specific attributes added to the call detail record (CDR) via the "accounting-stop" RADIUS message. The fax relay VSAs can be individually added when you use an accounting template. If no template is used, all fax relay VSAs are sent when gateway accounting is enabled. The proposed VSAs are presented to the AAA server as formatted text strings. Prior to Cisco IOS Release 12.4(24)T, these attributes were available for NextPort DSPs. Beginning with Cisco IOS Release 12.4(24)T, these attributes are added for HPI-based 5510 DSPs.

The following T.38 statistics require VSAs to be added:

Maximum jitter buffer depth

Number of jitter buffer overflow errors

Number of transmitted fax packets

Number of received fax packets

Initial high-speed modulation

Most recent high-speed modulation

Number of fax pages (sum of transmitted and received pages)

Nonstandard frame (NSF) country code

Fax success indicator (derived value in Cisco IOS software)

The statistics can be displayed using the command-line interface by entering the show call [active | history] [fax | voice] command.

The statistics are also available for authentication, authorization, and accounting (AAA) by adding VSAs to the RADIUS "accounting-stop" message when gateway accounting is enabled. Gateway accounting is enabled with the gw-accounting command in global configuration mode. The fax relay VSAs can be individually enabled by using an accounting template. If no template is used, all fax relay VSAs will be sent when gateway accounting is enabled.

Additional References

The following sections provide references related to the Voice Enhancements for Cisco IOS Release 12.4(24)T features.

Related Documents

Related Topic
Document Title

Basic Overview and Configuration Information for Session Initiation Protocol.

Cisco IOS SIP Configuration Guide

Basic Overview and Configuration Information for Fax and Modem Support.

Cisco IOS Fax, Modem, and Text Support over IP Configuration Guide

Basic Overview and Configuration Information for Trunk Connections.

Trunk Management Features

Reference information for Cisco IOS commands.

Cisco IOS Voice Command Reference


Standards

Standard
Title

 

MIBs

MIB
MIBs Link

To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs


RFCs

RFC
Title

RFC 2198

RTP Payload for Redundant Audio Data

RFC 2865

Remote Authentication Dial In User Service (RADIUS)

RFC 2866

RADIUS Accounting


Technical Assistance

Description
Link

The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

http://www.cisco.com/techsupport


Command Reference

The following commands are introduced or modified in the feature or features documented in this module. For information about these commands, see the Cisco IOS Voice Command Reference at http://www.cisco.com/en/US/docs/ios/voice/command/reference/
vr_book.html
. For information about all Cisco IOS commands, use the Command Lookup Tool at http://tools.cisco.com/Support/CLILookup or the Cisco IOS Master Command List, All Releases, at http://www.cisco.com/en/US/docs/ios/mcl/allreleasemcl/all_book.html.

cas-custom channel {define | ignore | invert-abcd}

cpa [threshold | timing]

modem passthrough {nse | protocol codec {g711alaw | g711ulaw [redundancy]}}

Feature Information for Voice Enhancement Features for Cisco IOS Release 12.4(24)T

Table 1 lists the release history for this feature.

Not all commands may be available in your Cisco IOS software release. For release information about a specific command, see the command reference documentation.

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.


Note Table 1 lists only the Cisco IOS software release that introduced support for a given feature in a given Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS software release train also support that feature.


Table 1 Feature Information for Voice Enhancement Features for Cisco IOS Release 12.4(24)T 

Feature Name
Releases
Feature Information

Voice Enhancement Features for Cisco IOS Release 12.4(24)T

12.4(24)T

The Voice Enhancement features were integrated into Release 12.4(24)T.