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Cisco IOS Software Releases 12.2 T

Modem Relay Support on VoIP Platforms

Table Of Contents

Modem Relay Support on VoIP Platforms

Feature Overview

Benefits

Restrictions

Related Features and Technologies

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Prerequisites

Configuration Tasks

Configuring Codec Complexity

Cisco 2600 and Cisco 3600 Series Routers

Cisco 7200 Series Routers

Configuring MGCP Modem Relay for VoIP

Configuring H.323 and SIP Modem Relay for VoIP Globally

Configuring H.323 and SIP Modem Relay for VoIP for a Specific Dial Peer

Verifying Modem Relay over VoIP

Troubleshooting Modem Relay over VoIP

Monitoring and Maintaining Modem Relay over VoIP

Configuration Examples

Modem Relay for a Cisco AS5300 Using H.323

Modem Relay for a Cisco 2650 Using H.323 CAS

Modem Relay for a Cisco 3810 Using H.323 and ISDN PRI

Modem Relay for a Cisco 3640 Using MGCP and CAS

Modem Relay on the Cisco AS5300 Using SIP (Originating Gateway)

Modem Relay on the Cisco AS5300 Using SIP (Terminating Gateway)

Command Reference

debug hpi all

debug modem relay errors

debug modem relay events

debug modem relay physical

debug modem relay packetizer

debug modem relay sprt

debug modem relay udp

debug modem relay v42

mgcp modem relay voip mode

mgcp tse payload

mgcp modem relay voip gateway-xid

mgcp modem relay voip latency

mgcp modem relay voip sprt retries

modem relay (dial-peer)

modem relay (voice-service)

modem relay gateway-xid

modem relay latency

modem relay sprt retries

show call active voice

show call history voice

show modem relay statistics

voice service voip

Glossary


Modem Relay Support on VoIP Platforms


Feature History

Release
Description

12.2(11)T

This feature was introduced on the Cisco 2600, Cisco 2691, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 3725, Cisco 3745, Cisco 7200, and Cisco AS5300.


This document describes the Modem Relay Support on VoIP Platforms feature and contains the following sections:

Feature Overview

Supported Platforms

Supported Standards, MIBs, and RFCs

Prerequisites

Configuration Tasks

Configuration Examples

Command Reference

Glossary

Feature Overview

The Modem Relay Support on VoIP Platforms feature provides support for modem connections across traditional time division multiplexing (TDM) networks. When service providers implement VoIP, they sometimes cannot separate fax or data traffic from voice traffic. These carriers that aggregate voice traffic over VoIP infrastructures require service offerings to carry fax and data as easily as voice.

Modem relay demodulates a modem signal at one voice gateway and passes it as packet data to another voice gateway where the signal is remodulated and sent to a receiving modem. On detection of the modem answer tone, the gateways switch into modem passthrough mode and then, if the call menu (CM) signal is detected, the two gateways switch into modem relay mode.

There are two ways to transport modem traffic over VoIP networks:

With modem passthrough, the modem traffic is carried between the two gateways in RTP packets, using an uncompressed voice codec—G.711 u-law or a-law. Although modem passthrough remains susceptible to packet loss, jitter, and latency in the IP network, packet redundancy may be used to mitigate the effects of packet loss in the IP network.

With modem relay, the modem signals are demodulated at one gateway, converted to digital form, and carried in Simple Packet Relay Transport (SPRT) protocol (which is a protocol running over User Datagram Protocol (UDP)) packets to the other gateway, where the modem signal is recreated and remodulated, and passed to the receiving modem.

In this implementation, the call starts out as a voice call, then switches into modem passthrough mode, and then into modem relay mode.

This feature significantly reduces the effects that dropped packets, latency and jitter have on the modem session. Compared to modem passthrough, it also reduces the amount of bandwidth used.

Primary applications for this feature are transport of modem dial-up traffic over IP networks.

Modem Tone Detection and Signaling

This implementation of modem relay supports V.34 modulation and the V.42 error correction and link layer protocol with maximum transfer rates of upto 33.6 kbps. It forces higher-rate modems to train down to the supported rates. Signaling support includes the session initiation protocol (SIP), MGCP/SGCP, and H.323:

For MGCP and SIP, during the call setup, the gateways negotiate the following:

To use or not use the modem relay mode

To use or not use the gateway-xid

The value of the payload type for Named Signaling Event (NSE) packets

For H.323, the gateways negotiate the following:

To use or not use the modem relay mode

To use or not use the gateway-xid

Relay Switchover

When the gateways detect a data modem, both the originating gateway and the terminating gateway switch to modem passthrough mode. This includes the following elements:

Switching to the G.711 codec

Disabling the high pass filter

Disabling voice activity detection (VAD)

Using special jitter buffer management algorithms

On detection of modem phase reversal tone, disabling the echo canceler

At the end of the modem call, the voice ports revert to the previous configuration and the digital signal processors (DSPs) switch back to the state before switchover. You can configure the codec by selecting the g711alaw or g711ulaw option of the codec command.

Figure 1 Modem Relay over VoIP Scenario

Controlled Redundancy

You can enable payload redundancy so that the modem passthrough over VoIP switchover causes the gateway to send redundant packets. Redundancy can be enabled in one or both of the gateways. When only a single gateway is configured for redundancy, the other gateway receives the packets correctly, but does not produce redundant packets.


Note By default, modem relay over VoIP capability and redundancy are disabled.


Packet Size

When redundancy is enabled, 10-ms sample-sized packets are sent. When redundancy is disabled, 20-ms sample-sized packets are sent.

Clock Slip Buffer Management

When the gateways detect a data modem, both the originating gateway and the terminating gateway switch from dynamic jitter buffers to static jitter buffers of 200-ms depth. The switch from dynamic to static is designed to compensate for Public Switched Telephone Network (PSTN) clocking differences at the originating and terminating gateways. When the modem call is concluded, the voice ports revert to dynamic jitter buffers.

Benefits

The Modem Relay Support on VoIP Platforms feature offers the following benefits:

Modem tone detection

Packetized modem signal transmission over the WAN

Significant reduction of dropped packet, latency, and jitter effects on modem sessions

Reduction of bandwidth used (as compared to modem passthrough)

Restrictions

Modem relay works only if it is configured and enabled on both gateways.

Modem relay works only if both modems are high-speed modems (such as V.34, V.90, V.92) using V.42bis bidirectional compression. For low-speed modems, gateways carrying traffic use modem passthrough.

Modem relay works only if both modems use V.42 error correction protocol and the error correction layer in both modems is enabled.

Cisco IOS Release 12.2(11)T must be running on the gateways for this feature to work.

MGCP, H.323, and SIP can be configured on the same gateway with some restrictions—all calls in a particular T1 or E1 must be handled by either MGCP, H.323, or SIP. If your gateway has multiple T1 or E1 facilities, then calls on some T1s or E1s can be managed by MGCP and others by H.323 or SIP.

Related Features and Technologies

For an overview of VoIP and information about Cisco VoIP features, refer to the Cisco IOS Voice, Video, and Fax Configuration Guide, Cisco IOS Release 12.2.

For a complete overview of the Service Assurance Agent (SAA), refer to the document, Network Monitoring Using Cisco Service Assurance Agent.

For an overview of the session initiation protocol, refer to the Guide to Cisco Systems VoIP Infrastructure Solution for SIP.

Related Documents

Cisco IOS Voice, Video, and Fax Configuration Guide, Cisco IOS Release 12.2
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fvvfax_c/index.htm

Cisco IOS Voice, Video, and Fax Command Reference, Cisco IOS Release 12.2
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/multi_r/index.htm

Modem Passthrough over Voice over IP
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtmodptr.htm

Cisco IOS Dial Services Quick Configuration Guide
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121sup/121dsqcg/index.htm

Media Gateway Control Protocol for the Cisco AS5300 Voice/Gateway
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120limit/120xr/mgcp1206.pdf

Enhancements to the Session Initiation Protocol for VoIP on Cisco Access Platforms
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtsipgv2.htm

Overview of the Session Initiation Protocol
http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsipov.htm

Supported Platforms

For Cisco IOS Release 12.2(11)T, the following platforms are suported:

Cisco AS5300 universal access server

Cisco 2600, Cisco 2691, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 3725, Cisco 3745, and Cisco 7200 series routers

Determining Platform Support Through Cisco Feature Navigator

Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.

Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.

To access Cisco  Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

http://www.cisco.com/register

Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:

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

Availability of Cisco IOS Software Images

Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.

Supported Standards, MIBs, and RFCs

Standards

No new or modified standards are supported by this feature.

MIBs

No new MIBs are supported by this feature.

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

http://tools.cisco.com/ITDIT/MIBS/servlet/index

If Cisco  MIB Locator does not support the MIB information that you need, you can also obtain a list of supported MIBs and download MIBs from the Cisco  MIBs page at the following URL:

http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml

To access Cisco MIB Locator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

http://www.cisco.com/register

RFCs

No new or modified RFCs are supported by this feature.

Prerequisites

VoIP enabled network.

Network suitability to relay modem traffic. The key attributes are packet loss, delay, and jitter. These characteristics of the network can be determined by using the Cisco IOS Service Assurance Agent (SAA) feature.

Configuration Tasks

This section provides the necessary tasks to configure modem relay for VoIP using MGCP, H.323, or SIP.


Note Prior to configuring modem relay, ensure that that you have configured high codec complexity and PRI backhaul for the originating and terminating gateways.


Configuring Codec Complexity (Required)

Configuring MGCP Modem Relay for VoIP

Configuring H.323 and SIP Modem Relay for VoIP Globally

Configuring H.323 and SIP Modem Relay for VoIP for a Specific Dial Peer


Tip You must configure modem relay in both the originating and terminating gateways for this feature to operate.


Configuring Codec Complexity

The following configuration task tables are for the Cisco 2600 , Cisco 3600, and Cisco 7200 routers.

For the Cisco AS5300 access server, codec complexity is determined by the VCWare code that is loaded on the voice feature card (VFC). For modem relay support, the VFC must be loaded with high-complexity code.

Cisco 2600 and Cisco 3600 Series Routers

To configure high codec complexity for the Cisco 2600 and Cisco 3600 series routers on the originating and terminating gateways, follow these steps:

 
Command
Purpose

Step 1 

Router(config)# voice-card slot

Enters voice-card mode. The argument slot specifies the voice-card slot location.

Step 2 

Router(config-voice-card)# codec complexity high

Sets codec complexity to high.

Cisco 7200 Series Routers

On the Cisco 7200 series routers, codec complexity is configured on the DSP interface.


Note Check the DSP voice channel activity using the show interfaces dspfarm command. If any DSP voice channels are in the busy state, changes can be made to the codec complexity selection. When all the DSP channels are in the idle state, changes can be made to the codec complexity selection.


To configure codec complexity on the DSP interface, follow the steps below, beginning in privileged EXEC mode:

 
Command
Purpose

Step 1 

Router# show interfaces dspfarm

Displays the DSP voice channel activity. If any DSP voice channels are in the busy state, codec complexity cannot be changed.

Step 2 

Roouter# configure terminal

Enters global configuration mode.

Step 3 

Router(config)# dspint dspfarm slot/port

Enters DSP interface configuration mode. The argument slot/port specifies the slot and port numbers of the interface.

Step 4 

Router(config-dspfarm)# codec {high|med}

Specifies codec complexity based on the codec standard being used.
The keyword high supports two voice channels encoded in any of the following formats: G.711, G.726, G.729, G.729 Annex B, G.728, and fax relay.
The keyword med is the default setting and supports up to four calls using the following codecs: G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay.

Step 5 

Router(config-dspfarm)# description

Enters a string to include descriptive text about this DSP interface connection. This information is displayed in the output for show commands and does not affect the operation of the interface in any way.

Step 6 

Router(config-dspfarm)# exit

Exits DSP interface configuration mode.

The following high complexity codecs are supported for modem relay using MGCP and H.323:

Clear channel: Clear channel at 64000 bps

g711alaw: G.711 a-law 64000 bps

g711ulaw: G.711 u-law 64000 bps

g723ar53: G.723.1 Annex-A 5300 bps

g723ar63: G.723.1 Annex-A 6300 bps

g723r53: G.723.1 5300 bps

g723r63: G.723.1 6300 bps

g726r16: G.726 16000 bps

g726r24: G.726 24000 bps

g726r32: G.726 32000 bps

g728: G.728 16000 bps

g729br8: G.729 Annex-B 8000 bps

g729r8: G.729 8000 bps

gsmefr: GSMEFR 12200 bps

gsmfr: GSMFR 13200 bps

The following high complexity codecs are supported for modem relay using SIP:

711alaw: G.711 a-law 64000 bps

g711ulaw: G.711 u-law 64000 bps

g723r63: G.723.1 6300 bps

g723r16: G.723.1 1600 bps

g728: G.728 16000 bps

g729r8: G.729 8000 bps

Configuring MGCP Modem Relay for VoIP

For MGCP, the parameters are negotiated between the two gateways and, if set differently, are adjusted for compatibility during negotiation.

Any MGCP command is applicable to the entire gateway. For MGCP calls, dial peers do not affect call handling because call agent takes care of the call routing. When configured, the following CLI commands affect MGCP calls only and not H.323 calls. H.323 and MGCP CLIs must be configured separately. Use the commands in the table below in global configuration mode.

To configure MGCP modem relay for VoIP using PRI, follow steps 1 to 4.

To configure MGCP modem relay for VoIP using CAS, follow steps 1 to 7.

To change modem relay parameters from their default values, follow steps 8 to 10.

: .

 
Command
Purpose

Step 1 

Router(config)# mgcp

Enables MGCP on the gateway.

Step 2 

Router(config)# mgcp call-agent x.x.x.x xxxx service-type mgcp version 0.1

Specifies the call agent information to the gateway and the protocol to use.

Step 3 

Router(config)# mgcp tse payload payload-value

Sets the payload value for the MGCP telephony signaling event (TSE). If not specifically configured with a value, the active default value is 100.

Step 4 

Router(config)# mgcp modem relay voip mode nse { [codec [g711alaw | g711ulaw]] [redundancy] }

Enables the modem relay feature with the mode nse, the appropriate codec—G.711 u-law for T1 or G.711 a-law for E1—and the optional use of redundancy for the modem passthrough phase.

Step 5 

Router(config)# dial-peer voice number pots

(Optional ) Enters dial peer configuration mode and defines a local dial peer that will connect to the POTS network. The argument number identifies the dial peer.

Step 6 

Router(config-dial-peer)# application MGCPAPP

(Optional ) Configures the MGCP application.

Step 7 

Router(congfig-dial-peer)# port controller number:D

(Optional ) Associates a destination number with a PRI span. The argument controller number specifies the T1 or E1 controller. The keyword D indicates the D channel associated with ISDN PRI.

Step 8 

Router(config)# mgcp modem relay voip gateway-xid {[ compress [backward | forward | both | no] } [ dictionary value ] [ string-length value ]


(Optional) Sets in-band negotiation of compression parameters between two VoIP gateways.

Step 9 

Router(config)# mgcp modem relay voip latency value

(Optional) Sets the estimated one-way delay across the IP network. This value is used to optimize the modem relay transport protocol. Default value is 200 ms.

Step 10 

Router(config)# mgcp modem relay voip sprt retries value

(Optional) Sets the maximum number of times for the SPRT protocol to carry modem relay traffic between the two gateways and send a packet before disconnecting.

This example shows configuring MGCP modem relay for VoIP:

mgcp
mgcp modem relay voip mode nse codec g711ulaw redundancy
mgcp call-agent 209.165.200.225 2000 service-type mgcp version 0.1
mgcp tse payload 98
mgcp modem relay voip latency 100
mgcp modem relay voip sprt retries 15
mgcp modem relay voip gw-xid compress forward dictionary 1020   string-length 30

Configuring H.323 and SIP Modem Relay for VoIP Globally


Note For H.323 and SIP modem relay over VoIP to operate, you must configure modem relay over VoIP in the originatingand terminating gateway.


H.323 cannot negotiate the value of payload-type for NSE packets. Configure the same value on both gateways, 100 by default.

For H.323 and SIP configurations, modem relay can be configured at two levels:

Under voice-service configuration mode for VoIP: This configuration is the global or system-wide configuration that can be applied to any VoIP call on the gateway.

Under dial-peer voice configuration mode for VoIP dial peers: This configuration applies only to calls that match a specific dial peer.

The two configuration tasks can be used separately or together. If both are configured, the dial-peer configuration takes precedence over and thus overrides the global configuration. Consequently, a call matching a specific dial peer first tries to apply the modem relay configuration on the dial peer. Then, if a specific dial peer is not configured, the router use the global configuration. This can ease the configuration burden for users. The configuration for modem relay can be made just once on a global basis. Of course, if finer granularity is desired—so that different modem relay or passthrough configurations are used for different calls on the gateway—you must use dial-peer configuration. The default dial-peer configuration is modem relay system, which tells the gateway to use the parameters configured at the global level. Because this is the default behavior, modem relay system does not show up in show running configuration command output.

When using the voice service voip and modem relay nse commands on a terminating gateway to globally set up fax or modem relay with NSEs, you must also ensure that each incoming call will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate calls with dial peers by using the incoming called-number command to specify a sequence of digits that incoming calls can match. You can ensure that all calls will match at least one dial peer by using the following commands:

Router(config)# dial-peer voice tag voip
Router(config-dial-peer)# incoming called-number .

To configure modem relay over VoIP for all connections (global) of a Cisco AS5300 gateway, use the following commands beginning in global configuration mode:

 
Command
Purpose

Step 1 

Router(config)# voice service voip

Enters voice service configuration mode.

Configures voice service for all the connections for the gateways.

Step 2 

router(conf-voi-serv)# h323 call start slow

(For H.323 only) Forces the H.323 gateway to use slow connect procedures for H.323 calls.

Step 3 

Router(conf-voi-serv)# modem relay nse [payload-type number] codec {g711ulaw | g711alaw}
[redundancy] [maximum-sessions value]

Configures modem relay over VoIP for the Cisco AS5300. The default behavior is no modem relay.

The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type number can be set from 98 to 120. If you do not specify the payload-type number, the number defaults to 100. When the payload-type is 100, and you use the show running-config command, the payload-type parameter does not appear.

Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.

The redundancy keyword is an optional parameter for sending redundant packets for modem traffic during the passthrough phase. By default, redundancy is disabled.

The maximum-sessions keyword is an optional parameter for the modem relay command in the voice-service configuration mode. This parameter determines the maximum number of redundant, simultaneous modem passthrough sessions. The recommended value for the maximum-sessions keyword is 16. The value can be set from 1 to 26. Changing this value would not affect the number of modem relay sessions that can be simultaneously supported. The maximum-sessions keyword applies only if the redundancy keyword is used.

Step 4 

Router(conf-voi-serv)# exit

Exits voice-service configuration mode.

Step 5 

Router(config)# exit

Exits global configuration mode.

Configuring H.323 and SIP Modem Relay for VoIP for a Specific Dial Peer

You can configure H.323 and SIP modem relay over VoIP on a specific dial peer in the following two ways:

Globally in voice-service configuration mode

Individually in dial-peer configuration mode on a specific dial peer

The default behavior for the voice-service configuration mode is no modem relay. This default behavior implies that modem relay is disabled for all dial peers on the gateway by default.

To enable modem relay on the VoIP dial peers on both the originating and terminating gateway, configure modem relay globally or explicitly on the dial peer.

For modem relay to operate, you must define VoIP dial peers on both gateways to match the call, for example, by using a destination pattern. The modem relay parameters associated with those dial peers then will apply to the call.


Note When modem relay is configured individually for a specific dial peer, that configuration for the specific dial peer takes precedence over the global configuration.


To configure modem relay over VoIP for a specific dial peer, use the following commands beginning in global configuration mode:

 
Command
Purpose

Step 1 

Router(config)# dial-peer voice number voip

Enters dial-peer configuration mode and configures a specific dial peer.

Step 2 

Router(config-dial-peer)# modem relay {system | nse [payload-type number] codec {g711ulaw | g711alaw}[redundancy]}

Configures modem relay over VoIP for a specific dial peer. The default behavior for modem relay for VoIP in dial-peer configuration mode is modem relay system. This means use the global modem relay configuration.

The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type number can be set from 98 to 120. If you do not specify the payload-type number, the number defaults to 100. When the payload-type is 100, and you use the show running-configuration command, the payload-type parameter does not appear.

Use the same codec type for both the originating and terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.

The redundancy keyword is an optional parameter for sending redundant packets for modem traffic during passthrough phase.

When the system keyword is enabled, the following parameters are not available: nse, payload-type, codec, and redundancy. Instead, the values from the global configuration are used.

Step 3 

Router(config-dial-peer)# exit

Exits dial-peer configuration mode and returns to global configuration mode.

Step 4 

Router(config)# exit

Exits global configuration mode.

The example here shows configuring modem relay in dial peer mode.

voice service voip
modem relay nse codec g711ulaw redundancy
dial-peer voice 12 voip
modem relay nse codec g711ulaw redundancy

Verifying Modem Relay over VoIP

To verify that modem relay over VoIP is enabled, perform the following steps:


Step 1 Enter the show running-configuration command to verify the configuration.

Step 2 Enter the show dial-peer voice command to verify that modem relay over VoIP is enabled.


Troubleshooting Modem Relay over VoIP

You might be able to determine why the modem relay is not working correctly by viewing the gateway digital signal processor (DSP) modem-relay termination codes that display when the debug hpi all command is used. The DSP-to-host messages for the modem relay termination indicate the modem relay session termination time, physical or link layer, and other probable causes of disconnection. On receiving this indication from the DSP, the host can disconnect the call or place the channel in modem passthrough state.

Table 1 Modem Relay Termination Cause Codes 

Modem Relay Termination Cause Code
Description
0x65

SPRT—Channel 1 max retransmit count exceeded on DSP

0x66

SPRT—Channel 1 invalid transport frame type in transmit queue

0x67

SPRT—Channel 2 max retransmit count exceeded on DSP

0x68

SPRT—Channel 2 invalid transport frame type in transmit queue

0x69

SPRT—Channel 1 invalid base sequence number received by DSP from remote host

0x6A

SPRT—Channel 2 invalid base sequence number received by DSP from remote host

0x6B

SPRT—Received RELEASE request from peer

0x6C

SPRT—Channel 1 invalid transmit sequence number

0x6D

SPRT—Channel 2 invalid transmit sequence number

0x6E

SPRT—Invalid transmit t_frame type

0x6F

SPRT—Requested to transmit null (zero length) info t_frame

0x71

V42—Unexpected SABME received

0x72

V42—Client modem capability appears incompatible with V42bis capability on originating leg gateway

0x73

V42—Client modem capability appears incompatible with V42bis capability on terminating leg gateway

0x74

V42—Exceeded max XID retransmit count

0x77

V42—Exceeded max SABME retransmit count

0x78

V42—NR sequence exception

0x79

V42—Invalid acknowledgement received

0x7A

V42—Exceeded N401 retransmit count

0x7B

SPRT—Requested to transmit info t_frame that exceeds max allowed size

0x7C

V42—Received V42 DISC packet from client modem

0x7D

V42—Received V42 FRMR packet from client modem

0x82

V42—Failed to add packet to V42 transmit queue

0x8C

V42—Invalid "VA"

0x8D

PHYSICAL—Modem data pump terminated/failed

0xC9

SPRT—Channel 1 max retransmit count exceeded on line card

0xCA

SPRT—Channel 2 max retransmit count exceeded on line card

0xCD

SPRT—Channel 1 invalid base sequence number received by line card from DSP

0xCE

SPRT—Channel 2 invalid base sequence number received by line card from DSP

0xCF

SPRT—Channel 1 invalid base sequence number received by line card from remote host

0xD0

SPRT—Channel 2 invalid base sequence number received by line card from remote host



TipTo troubleshoot modem relay over VoIP:

Ensure that you can make a voice call.

Ensure that modem relay over VoIP is configured on the originating and terminating gateway.

Ensure that both the originating and terminating gateway have the same named signaling event (NSE) payload-type number.

Use the debug vtsp dsp, debug vtsp session, and debug hpi all commands to debug a problem


Monitoring and Maintaining Modem Relay over VoIP

To monitor and maintain modem relay over VoIP, use the following commands in privileged EXEC mode:

Command
Purpose

Router# show call active {voice | fax}[brief]

Displays information for the active call table or displays the active call. The brief option displays a truncated version of either option.

Router# show dial-peer voice [number | summary]

Displays configuration information for dial peers. The number argument specifies a specific dial peer from 1 to 32767. The summary option displays a summary of all dial peers.


Configuration Examples

This section provides the following configuration examples:

Modem Relay for a Cisco AS5300 Using H.323

Modem Relay for a Cisco 2650 Using H.323 CAS

Modem Relay for a Cisco 3810 Using H.323 and ISDN PRI

Modem Relay for a Cisco 3640 Using MGCP and CAS

Modem Relay on the Cisco AS5300 Using SIP (Originating Gateway)

Modem Relay on the Cisco AS5300 Using SIP (Terminating Gateway)

Modem Relay for a Cisco AS5300 Using H.323

The following example shows a sample configuration for H.323 modem relay over VoIP for the Cisco AS5300 universal access server:

version 12.1
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
voice service voip
	 modem relay nse codec g711ulaw redundancy maximum-session 5
!
resource-pool disable
!
ip subnet-zero
ip ftp source-interface Ethernet0
ip ftp username lab
ip ftp password lab
no ip domain-lookup
!
isdn switch-type primary-5ess
cns event-service server
!
mta receive maximum-recipients 0
!
controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 1
 shutdown
 clock source line secondary 1
!
controller T1 2
 shutdown
!
controller T1 3
 shutdown
!
interface Ethernet0
 ip address 10.165.201.159
 no ip route-cache
 no ip mroute-cache
!
interface Serial0:23
 no ip address
 encapsulation ppp
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-5ess
 isdn incoming-voice modem
 no peer default ip address
 no fair-queue
 no cdp enable
 no ppp lcp fast-start
!
interface FastEthernet0
 ip address 10.165.201.159 255.0.0.0
 no ip route-cache
 no ip mroute-cache
 load-interval 30
 duplex full
 speed auto
 no cdp enable
!
ip classless
ip route 10.165.201.159.255.0.0 1.1.1.1
no ip http server
!
voice-port 0:D
!
dial-peer voice 1 pots
 incoming called-number 55511..
 destination-pattern 020..
 direct-inward-dial
 port 0:D
 prefix 020
!
dial-peer voice 2 voip
 incoming called-number 020..
 destination-pattern 55511..
 modem relay nse codec g711ulaw redundancy
 session target ipv4:26.0.0.2
!
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 login
!
end

Modem Relay for a Cisco 2650 Using H.323 CAS

version 12.2
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname k-262
!
no logging buffered
logging rate-limit console 10 except errors
!
voice-card 1
  codec complexity high
  dspfarm
!
ip subnet-zero
!
no ip dhcp-client network-discovery
lcp max-session-starts 0
!
voice service voip
  h323
   call start slow
  modem relay nse codec g711ulaw redundancy
!
dial-peer voice 4000 voip
destination-pattern 4......
session target ipv4:10.2.00.86
playout-delay maximum 300

controller T1 1/0
  framing esf
  linecode b8zs
  cablelength long gain36 -15db
  ds0-group 1 timeslots 1 type e&m-wink-start
!
controller T1 1/1
  shutdown
  framing esf
  clock source internal
  linecode ami
!
interface FastEthernet0/0
  ip address 10.165.201.159 255.255.0.0
  duplex auto
  speed auto
interface Serial0/0
  no ip address
  shutdown

ip classless
ip route 0.0.0.0 0.0.0.0 1.2.0.1
ip http server

voice-port 1/0:1
!
no mgcp timer receive-rtcp
!
mgcp profile default
!
dial-peer cor custom
!
!
dial-peer voice 4201 pots
  destination-pattern 4201000
  port 1/0:1
!
!
line con 0
  transport input none
line aux 0
line vty 0 4
  login
line vty 5 15
  login
!
end

Modem Relay for a Cisco 3810 Using H.323 and ISDN PRI

version 12.2
no service single-slot-reload-enable
no service pad
no service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname k-gw1
!
no logging buffered
logging rate-limit console 10 except errors
!
network-clock base-rate 56k
ip subnet-zero
!
no ip finger
!
no ip dhcp-client network-discovery
isdn switch-type primary-5ess
isdn voice-call-failure 0
!
voice service voip
  h323
   call start slow
  modem relay nse codec g711ulaw
!
voice-card 0
  codec complexity high
!
controller T1 0
  mode atm
  framing esf
  linecode b8zs
controller T1 1
  framing esf
  clock source internal
  linecode b8zs
  pri-group timeslots 1-24
!
interface Ethernet0
  ip address 209.165.200.225 255.255.0.0

interface Serial1:23
  no ip address
  no logging event link-status
  isdn switch-type primary-5ess
  isdn incoming-voice voice
  no cdp enable
  slarp_ip_addr retry 1
!
...
ip classless
ip route 0.0.0.0 0.0.0.0 1.1.0.1
no ip http server

call rsvp-sync
!
voice-port 1:23
!
dial-peer voice 3001 pots
  destination-pattern 30.....
  direct-inward-dial
  port 1:23
!
dial-peer voice 400 voip
  destination-pattern 4......
  session target ipv4:1.2.00.86
  playout-delay maximum 300
line con 0
  transport input none
line aux 0
line 2 3
line vty 0 4
  login

Modem Relay for a Cisco 3640 Using MGCP and CAS

!
version 12.2
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname uut1-3640
!
logging rate-limit console 10 except errors
!
voice-card 3
  codec complexity high
!
ip subnet-zero
!
no ip dhcp-client network-discovery
lcp max-session-starts 0
!
controller T1 3/0
  framing esf
  clock source internal
  linecode b8zs
  ds0-group 0 timeslots 1 type e&m-immediate-start
  ds0-group 1 timeslots 2 type e&m-immediate-start
controller T1 3/1
  framing sf
  linecode ami
!
interface Ethernet0/0
  ip address 10.165.200.225. 255.255.0.0
  half-duplex
!
interface Serial0/0
  no ip address
  shutdown
  no fair-queue
!
ip classless
ip route 0.0.0.0 0.0.0.0 1.2.0.1
ip http server
snmp-server manager
call rsvp-sync
!
voice-port 3/0:0
!
voice-port 3/0:1

mgcp
mgcp call-agent 10.3.64.1 service-type mgcp version 0.1
mgcp modem relay voip mode nse
mgcp modem relay voip gateway-xid
no mgcp timer receive-rtcp
!
mgcp profile default
!
dial-peer cor custom
!
dial-peer voice 1001 pots
  application mgcpapp
  port 3/0:0
!
dial-peer voice 1002 pots
  application mgcpapp
  port 3/0:1
!
line con 0
  exec-timeout 0 0
  transport input none
line aux 0
line vty 0 4
  login
!

Modem Relay on the Cisco AS5300 Using SIP (Originating Gateway)

show running-config
Building configuration...

Current configuration : 31658 bytes
!
version 12.2
resource-pool disable
!
ip subnet-zero
no ip domain-lookup
ip host tftps 10.165.200.225
ip host tftp 10.165.200.225
ip host dir 10.165.200.225
!
no ip dhcp-client network-discovery
isdn switch-type primary-5ess
!

voice service voip
modem relay nse codec g711ulaw redundancy
!
fax interface-type modem
mta receive maximum-recipients 1
!
controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 pri-group timeslots 1-24
!
interface Ethernet0
 ip address 10.165.201.159.255.0.0
!
interface FastEthernet0
 ip address 10.0.0.4 255.255.255.0
 load-interval 30
 duplex full
 speed auto
 no cdp enable
!
voice-port 0:D
!
voice-port 1:D
!
no mgcp timer receive-rtcp
!
mgcp profile default
!
dial-peer voice 1 pots
 incoming called-number 5551111
 destination-pattern 6661111
 direct-inward-dial
 port 0:D
 prefix 6661111
!
dial-peer voice 101 voip
 description SIP-testing
 application session
 incoming called-number 6661111
 destination-pattern 5551111
 modem relay nse payload-type 101 codec g711ulaw
 modem relay gateway-xid compress no
 session protocol sipv2
 session target ipv4:10.0.1.3
 codec g711ulaw
 fax-relay ecm disable
 fax rate disable
!
exec-timeout 0 0
end

Modem Relay on the Cisco AS5300 Using SIP (Terminating Gateway)

hostname uut-ter
!
voice service voip
 modem relay nse codec g711ulaw redundancy
!
fax interface-type modem
mta receive maximum-recipients 1
!
controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 1
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
!
interface Ethernet0
 ip address 10.165.200.225 255.255.0.0
 no ip route-cache
 no ip mroute-cache
 load-interval 30
!
interface Serial0:23
 no ip address
 encapsulation ppp
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-5ess
 isdn incoming-voice modem
 no fair-queue
 no cdp enable
!
interface FastEthernet0
 ip address 10.0.1.3 255.255.255.0
 load-interval 30
 duplex full
 speed auto
!
ip default-gateway 1.6.0.1
ip classless
ip route 10.0.0.0 155.255.255.0 10.0.1.1
ip route 10.165.200.225 255.255.255.255 1.6.0.1
no ip http server
!
call rsvp-sync
!
voice-port 0:D
!
voice-port 1:D
!
voice-port 2:D
!
no mgcp timer receive-rtcp
!
mgcp profile default
!
dial-peer voice 1 pots
 incoming called-number 6661111
 destination-pattern 5551111
 direct-inward-dial
 port 0:D
 prefix 5551111
!
dial-peer voice 101 voip
 description SIP-testing
 application session
 incoming called-number 5551111
 destination-pattern 6661111
 modem relay nse payload-type 101 codec g711ulaw
 modem relay gateway-xid compress no
 session protocol sipv2
 session target ipv4:10.0.0.4
 codec g711ulaw
 fax-relay ecm disable
 fax rate disable
!
line con 0
 exec-timeout 0 0
line aux 0
line vty 0 4
 exec-timeout 0 0
 password 7 09404F0B
 login local
!
no scheduler max-task-time
scheduler interval 1000
end

Command Reference

This section documents new and modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.2 command reference publications.

New Commands:

debug hpi all

debug modem relay errors

debug modem relay events

debug modem relay physical

debug modem relay packetizer

debug modem relay v42

debug modem relay sprt

debug modem relay udp

mgcp modem relay voip mode

mgcp modem relay voip gateway-xid

mgcp modem relay voip sprt retries

modem relay (dial-peer)

modem relay (voice-service)

modem relay gateway-xid

modem relay latency

modem relay sprt retries

show modem relay statistics

Modified Commands:

mgcp tse payload

show call active voice

show call history voice

voice service voip

debug hpi all

To view the gateway DSP modem relay termination codes, use the debug hpi all command in privileged EXEC mode. To disable the debugging output, use the no form of this command.

debug hpi all

no debug hpi all

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.(2)11T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this command to view gateway DSP modem relay termination codes. The DSP to host messages for the modem relay termination indicate to the host the modem relay session termination time, physical or link layer, and other probable causes for disconnection. On receiving this indication from the DSP, the host can disconnect the call or place the channel in the modem passthrough state.

Examples

A sample output of the debug hpi all command is shown below.

Jan 11 05:29:28.795:			packet_len=14 channel_id=609 packet_id=193
*Jan 11 05:29:28.795:hpi [0:D:11] hpi_receive_message:tone_detect - Modem CM:
*Jan 11 05:29:28.795:			packet_len=14 channel_id=609 packet_id=193
*Jan 11 05:29:28.795:hpi [0:D:11] hpi_inband_det_ctrl:
*Jan 11 05:29:28.795:			packet_len=16 channel_id=8801 packet_id=111 signal_type=2 mode=0
*Jan 11 05:29:28.795:hpi [0:D:11] hpi_gen_peer_to_peer:rtp:
*Jan 11 05:29:28.795:			packet_len=22 channel_id=8801 packet_id=103
		event 203, volume 0x0, duration 0x0
		disable_redundancy 0, redundancy_interval 20
		ssrc_hi 0, ssrc_lo 0, payload_type 100
*Jan 11 05:29:28.835:hpi [0:D:11] hpi_receive_message:sent peer-to-peer message
*Jan 11 05:29:28.947:hpi [0:D:11] hpi_idle_service:
*Jan 11 05:29:28.947:			packet_len=8 channel_id=8801 packet_id=68
*Jan 11 05:29:28.947:hpi [0:D:11] hpi_open_service:setting codec 34
*Jan 11 05:29:29.499:hpi [0:D:11] hpi_modemrelay_mode:Role 0, Debug_flags 0x0 latency=200 
retries=12 gw_xid=1 dict_size=1024 string_len=32 compress_dir=3
*Jan 11 05:29:39.559:hpi [0:D:11] hpi_receive_message:modem_relay_msg:
*Jan 11 05:29:39.559:			packet_len=34 channel_id=609 packet_id=123
	function_code=128 CONNECTED. phys=1, ec=1, Modem dict. size=2048, string length=32, 
compression dir=3,Negotiated dict. size=1024, string length=32,compression dir=3,local 
Rx/Tx =26400/28800, remote Rx/Tx =33600/31200

Related Commands

Command
Description

debug modem relay errors

To view modem relay network errors.

debug modem relay events

To view the events that may cause failure of the modem relay network.


debug modem relay errors

To view modem relay network errors, use the debug modem relay errors command in privileged EXEC mode. To disable the debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] errors

no debug modem relay [call-identifier call-setup-time call-index] errors

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

In a stable modem relay network, this command produces little output.

Examples

A sample output of the debug modem relay errors command is shown below.

The output shows the sequence number of the packet, timestamp, direction, layer, and payload-bytes, followed by each byte of the payload in hexadecimal.

Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 28 tm 11944 OUT ERR, pb=12, payload: 00 06 
00 00 00 00 00 07 00 00 01 DE
*Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 29 tm 11944 OUT ERR, pb=12, payload: 00 
06 00 00 00 00 00 04 00 00 00 BE
*Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 30 tm 11944 OUT ERR, pb=12, payload: 00 
06 00 00 00 00 00 05 FF FF FF FD

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay events

To view the events that may cause failure of the modem relay network.


debug modem relay events

To view the events that may cause failure of the modem relay network, use the debug modem relay events command in privileged EXEC mode. To disable debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] events

no debug modem relay [call-identifier call-setup-time call-index] events

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

In a stable modem relay network, this command produces little output.

Examples

A sample output of the debug modem relay events command is shown below.

The output shows the sequence number of the packet, timestamp, direction, layer, and payload-bytes, followed by each byte of the payload in hexadecimal.

Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 28 tm 11944 OUT EVNT, pb=12, payload: 00 
06 00 00 00 00 00 07 00 00 01 DE
*Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 29 tm 11944 OUT EVNT, pb=12, payload: 00 
06 00 00 00 00 00 04 00 00 00 BE
*Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 30 tm 11944 OUT EVNT, pb=12, payload: 00 
06 00 00 00 00 00 05 FF FF FF FD

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay errors

To view modem relay network errors.


debug modem relay physical

To view modem relay physical layer packets, use the debug modem relay physical command in privileged EXEC mode. To disable debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] physical

no debug modem relay [call-identifier call-setup-time call-index] physical

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Disable console logging and use buffered logging before using this command. Using this command generates a large volume of debugs, which can affect router performance.

Examples

A sample output of the debug modem relay physical command is shown below.

The output shows the sequence number of the packet, timestamp, direction, layer, and payload-bytes, followed by each byte of the payload in hexadecimal.

Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 28 tm 11944 OUT PHYS, pb=12, payload: 00 
06 00 00 00 00 00 07 00 00 01 DE
*Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 29 tm 11944 OUT PHYS, pb=12, payload: 00 
06 00 00 00 00 00 04 00 00 00 BE
*Jan 11 05:35:09.119:ModemRelay pkt[0:D:11]. sqn 30 tm 11944 OUT PHYS, pb=12, payload: 00 
06 00 00 00 00 00 05 FF FF FF FD

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay errors

To view modem relay network errors.


debug modem relay packetizer

To view events occuring in the modem relay packetizer module, use the debug modem relay packetizer command in privileged EXEC mode. To disable debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] packetizer

no debug modem relay [call-identifier call-setup-time call-index] packetizer

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Disable console logging and use buffered logging before using this command. Using this command generates a large volume of debugs, which can affect router performance.

Examples

A sample output of the debug modem relay packetizer command is shown below.

The output shows the sequence number of the packet, timestamp, direction, layer, and payload-bytes, followed by each byte of the payload in hexadecimal.

Jan 11 05:33:33.715:ModemRelay pkt[0:D:11]. sqn 8 tm 47610 IN PKTZR, pb=7, payload: 82 38 
00 18 03 01 87
*Jan 11 05:33:33.727:ModemRelay pkt[0:D:11]. sqn 9 tm 47616 OUT PKTZR, pb=7, payload: 82 
20 00 18 03 01 47
*Jan 11 05:33:35.719:ModemRelay pkt[0:D:11]. sqn 10 tm 49614 IN PKTZR, pb=7, payload: 82 
39 00 18 03 01 87

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay errors

To view modem relay network errors.


debug modem relay sprt

To view modem relay Simple Packet Relay Transport (SPRT) protocol packets, use the debug modem relay sprt command in privileged EXEC mode. To disable debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] sprt

no debug modem relay [call-identifier call-setup-time call-index] sprt

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced for Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Disable console logging and use buffered logging before using this command. Using this command generates a large volume of debugs, which can affect router performance.

Examples

A sample output of the debug modem relay sprt command is shown below.

The output shows the sequence number of the packet, timestamp, direction, layer, and payload-bytes, followed by each byte of the payload in hexadecimal.

Jan 11 05:37:16.151:ModemRelay pkt[0:D:11]. sqn 34 tm 7910 OUT SPRT, pb=4, payload: 02 00 
03 71
*Jan 11 05:37:16.295:ModemRelay pkt[0:D:11]. sqn 35 tm 8048 IN SPRT, pb=13, payload: 02 00 
01 F1 F7 7E FD F5 90 F3 3E 90 55
*Jan 11 05:37:16.303:ModemRelay pkt[0:D:11]. sqn 36 tm 8060 IN SPRT, pb=6, payload: 02 00 
01 41 04 00

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay errors

To view modem relay network errors.


debug modem relay udp

To view events occurring in the User Datagram Protocol (UDP) stack, use the debug modem relay udp command in privileged EXEC mode. To disable debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] udp

no debug modem relay [call-identifier call-setup-time call-index] udp

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Disable console logging and use buffered logging before using this command. Using this command generates a large volume of debugs, which can affect router performance.

Examples

A sample output of the debug modem relay udp command is shown below.

The output shows three UDP packets related to modem relay. In the sample output, OUT or IN represent packet direction, UDP indicates the specific layer that reported the packet.

Jan 1 03:39:29.407:ModemRelay pkt[0:D (4)]. sqn 61 tm 3060 OUT UDP, pb=6, payload: 80 00 
00 00 00 00
*Jan 1 03:39:29.471:ModemRelay pkt[0:D (4)]. sqn 62 tm 3120 IN UDP, pb=6, payload: 40 00 
00 00 00 00
*Jan 1 03:39:29.471:ModemRelay pkt[0:D (4)]. sqn 63 tm 3120 IN UDP, pb=6, payload: 80 00 
00 00 00 00

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay errors

To view modem relay network errors.


debug modem relay v42

To view events occuring in the V.42 layer, use the debug modem relay v42 command in privileged EXEC mode. To disable debugging output, use the no form of this command.

debug modem relay [call-identifier call-setup-time call-index] v42

no debug modem relay [call-identifier call-setup-time call-index] v42

Syntax Description

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 10.


Defaults

Disabled

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Disable console logging and use buffered logging before using this command. Using this command generates a large volume of debugs, which can affect router performance.

Examples

A sample output of the debug modem relay v42 command is shown below.

The output shows the sequence number of the packet, timestamp, direction, layer, and payload-bytes, followed by each byte of the payload in hexadecimal.

Jan 11 05:42:08.715:ModemRelay pkt[0:D:13]. sqn 3 tm 10104 OUT V42, pb=43, payload: 03 AF 
82 80 00 13 03 03 8A 89 00 05 02 03 E0 06 02 03 E0 07 01 08 08 01 08 F0 00 0F 00 03 56 34 
32 01 01 03 02 02 04 00 03 01 20
*Jan 11 05:42:08.847:ModemRelay pkt[0:D:13]. sqn 4 tm 10236 IN V42, pb=2, payload: 03 7F

Related Commands

Command
Description

debug hpi all

To view gateway DSP modem relay termination codes.

debug modem relay errors

To view modem relay network errors.


mgcp modem relay voip mode

To enable modem relay mode support in a gateway for Media Gateway Control Protocol (MGCP) Voice over IP (VoIP) calls, use the mgcp modem relay voip mode command in global configuration mode. To disable this function, use the no form of this command.

mgcp modem relay voip mode nse [codec {g711alaw | g711ulaw } [redundancy] | redundancy]

no mgcp modem relay voip mode

Syntax Description

nse

This keyword instructs the gateway to use nse (named signaling event) mode for upspeeding.

codec

This keyword instructs the gateway which codec to use for upspeeding:

g711alaw—Codec G.711 a-law 64000 bits per second for E1 environments.

g711ulaw—Codec G.711 u-law 64000 bits per second for T1 environments.

For example, if the codec is set to g711ulaw and the initial codec for the call is G.729, while upspeeding the gateway will use the G.711 codec.

redundancy

(Optional) Packet redundancy for modem traffic during modem passthrough phase. But default redundancy is disabled. "Redundancy" causes the DSP to generate duplicate (redundant) data packets for fax/modem passthrough calls as per RFC 2198. Redundant packets transmission is needed to take care of excessive loss of packets in VoIP networks to make fax/modem passthrough calls more reliable.


Defaults

Disabled

Command Modes

Global configuration

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

This command enables the modem relay mode for MGCP VoIP calls. By default, modem relay mode is enabled. The remaining part of the CLI is for upspeeding. The keywords in the remaining part of the CLI are included because modem passthrough is an intermediate step while switching from voice calls to modem relay calls.

The mgcp modem relay voip mode command enables MGCP modem relay.

If this command is not used, all modem calls go through as "passthrough" calls, which are less reliable and use higher bandwidth.

If modem relay is configured but "codec" is not configured, G.711 u-law codec is used for upspeeding.

If modem relay is configured but "redundancy" is not configured, redundancy is disabled and no duplicate data packets are sent while in modem/fax passthrough mode.

Even if one of the gateways is configured with "redundancy," the calls will go through. DSPs can handle asymmetric (one way) redundancy.

Examples

The following example enables MGCP modem relay and specifies NSE mode for upspeeding, codec g711ulaw, and packet redundancy for modem traffic during modem passthrough phase:

mgcp modem relay voip mode nse codec g711ulaw redundancy

Related Commands

Command
Description

mgcp modem relay voip mode

Sets in-band negotiation of compression parameters between two VoIP gateways.

mgcp tse payload

Enables telephony signaling events (TSEs) for communications between gateways, required for modem relay over VoIP using MGCP.

mgcp modem relay voip gateway-xid

Optimizes the modem relay transport protocol and the estimated one-way delay across the IP network.

mgcp modem relay voip sprt retries

Sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.


mgcp tse payload

To enable telephony signaling events (TSEs) for communications between gateways as required for modem relay over VoIP using Media Gateway Control Protocol (MGCP), use the mgcp tse payload command in global configuration mode. To disable TSEs, use the no form of this command.

mgcp tse payload payload-value

no mgcp tse payload

Syntax Description

payload-value

Valid range is from 98 through 119. If the value is not configured, the default of 100 is used.


Defaults

Default payload value is 100.

Command Modes

Global configuration

Command History

Release
Modification

12.0(7)XK

This command was introduced for Simple Gateway Control Protocol (SGCP) on the Cisco MC3810 and on the Cisco 3600 series router (except the Cisco 3620).

12.1(5)XM

This command was modified to support Media Gateway Control Protocol (MGCP).

12.2(2)T

This command was implemented on the Cisco 7200 series router and integrated into Cisco IOS Release 12.2(2)T.

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

This command must be enabled for modem-relay over VoIP using MGCP.

Telephony service events (TSE) are also known as named service events (NSEs). TSE is used to communicate telephony events between gateways . They are RTP packets with different payload values that indicate if the packet contains a TSE. Both gateways should have the same payload value field.

Examples

The following example sets the payload value at 98 for the MGCP telephony signaling events:

mgcp tse payload 98

Related Commands

Command
Description

mgcp modem relay voip mode

Sets in-band negotiation of compression parameters between two VoIP gateways.

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp modem relay voip sprt retries

Sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.

mgcp modem relay voip gateway-xid

Optimizes the modem relay transport protocol and the estimated one-way delay across the IP network.


mgcp modem relay voip gateway-xid

To set in-band negotiation of compression parameters between two VoIP gateways using Media Gateway Control Protocol (MGCP), use the mgcp modem relay voip gateway-xid command in global configuration mode. To disable this function, use the no form of this command.

mgcp modem relay voip gateway-xid [compress {backward | both | forward | no} [dictionary value [string-length value] | string-length value] | dictionary value [string-length value] | string-length value ]

no mgcp modem relay voip gateway-xid

Syntax Description

voip

Indicate that this CLI affects only VoIP calls and not VoAAL2 calls. This is necessary because MGCP also supports VoAAL2 calls (voice and fax/modem), but modem relay calls for VoAAL2 are not supported.

gateway-xid

If enabled on both gateways, in-band negotiation of compression parameters occurs between the two VoIP gateways. The gateways negotiate whether they are going to have an in-band negotiation between the two DSPs on either side of the network. This is not for negotiation of any parameters, but instead it is to determine whether there is going to be a negotiation process itself or not. The actual parameters are negotiated by the in-band negotiation process and not by the host at a later stage (assuming the two hosts agreed to this negotiation by having gateway-xid enabled).

The remaining parameters specify the negotiation posture of this gateway in the subsequent in-band negotiation step (assuming in-band negotiation was agreed on by the two gateways).

compress

This keyword specifies the direction in which the data flow is compressed. For normal dial-up, compression should be enabled in both directions.

You may want to disable compression in one or more directions. This is normally done during testing, and perhaps for gaming applications, but not for normal dial-up, when compression is enabled on both directions.

backward—Enables compression only in the backward direction.

forward—Enables compression only in the forward direction.

both (default)—For normal dial-up, this is the preferred setting.

no—Disables compression in both directions.

dictionary value

V.42bis parameter that specifies characterstics of the compression algorithm. Range: 512 to 2048. Default: 1024.

string-length value

V.42bis parameter that specifies characterstics of the compression algorithm. Range: 16 to 32. Default: 32.


Defaults

Enabled

Command Modes

Global configuration

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

XID negotition is a DSP-specific feature used in modem relay. The mgcp modem relay voip gateway-xid command enables this negotiation feature. By default it is enabled.

The parameters compress and dictionary and string-length are DSP-specific and related to xid negotiation. If the mgcp modem relay voip gateway-xid command is disabled, they are all irrelevant. The application (MGCP or H.323) just passes these configured values to the DSPs, and it is the DSP that requires them.

Examples

The following example enables in-band negotiation of compression parameters on the VoIP gateway, with compression in both directions, and dictionary size of 1024 and string length of 32 for the compression algorithm:

mgcp modem relay voip gateway-xid compress both dictionary 1024 string-length 32

Related Commands

Command
Description

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp tse payload

Enables telephony signaling events (TSEs) for communications between gateways, required for modem relay over VoIP using MGCP.

mgcp modem relay voip gateway-xid

Optimizes the modem relay transport protocol and the estimated one-way delay across the IP network.

mgcp modem relay voip sprt retries

Sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.


mgcp modem relay voip latency

To optimize the modem relay transport protocol and the estimated one-way delay across the IP network using Media Gateway Control Protocol (MGCP), use the mgcp modem relay voip latency command in global configuration mode. To disable this function, use the no form of this command.

mgcp modem relay voip latency value

no mgcp modem relay voip latency

Syntax Description

value

Estimated one-way delay across the IP network, used to optimize the Modem Relay Transport Protocol. Valid range is from 100 through 1000 ms. The default value of 200 does not need to be changed for most networks.


Defaults

Default value is 200.

Command Modes

Global configuration

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this command to adjust the retransmission timer of the SPRT protocol, if required, by setting this value to the estimated one-way delay (in milliseconds) across the IP network. Changing this value may affect the throughput or delay characteristics of the modem relay call. The default value of 200 does not need to be changed for most networks.

Examples

The following example sets the estimated one-way delay across the IP network to 100 ms.

mgcp modem relay voip latency 100

Related Commands

Command
Description

mgcp modem relay voip mode

Sets in-band negotiation of compression parameters between two VoIP gateways.

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp tse payload

Enables telephony signaling events (TSEs) for communications between gateways, required for modem relay over VoIP using MGCP.

mgcp modem relay voip sprt retries

Sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.


mgcp modem relay voip sprt retries

To set the maximum number of times that the SPRT protocol tries to send a packet before disconnecting, use the mgcp modem relay voip sprt retries command in global configuration mode. To disable this function, use the no form of this command.

mgcp modem relay voip sprt retries value

no mgcp modem relay voip sprt retries

Syntax Description

retries value

The maximum number of times that the SPRT protocol tries to send a packet before disconnecting. Range: 6 to 30. Default: 12.


Defaults

Default value is 12.

Command Modes

Global configuration

Command History

Release
Modification

12.2(11)T

This command was introduced on the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this command to set the maximum number of times that the Simple Packet Relay Transport (SPRT) protocol tries to send a packet before disconnecting.

Examples

The following example sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting to 15:

mgcp modem relay voip sprt retries 15

Related Commands

Command
Description

mgcp modem relay voip mode

Sets in-band negotiation of compression parameters between two VoIP gateways.

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp tse payload

Enables telephony signaling events (TSEs) for communications between gateways, required for modem relay over VoIP using MGCP.

mgcp modem relay voip gateway-xid

Optimizes the Modem Relay Transport Protocol and the estimated one-way delay across the IP network.


modem relay (dial-peer)

To configure modem relay over Voice over IP (VoIP) for a specific dial peer, use the modem relay command in dial-peer configuration mode. To disable modem relay for a specific dial peer, use the no form of this command.

modem relay {nse {[payload-type number] codec {g711alaw | g711ulaw} [redundancy]} | system}

no modem relay { nse | system}

Syntax Description

nse

Named signaling event.

payload-type

(Optional) NSE payload-type.

number

(Optional) The value of the payload type (98 to 119).

codec

Sets the upspeed voice compression selection for speech or audio signals. The upspeed method is used to dynamically change the codec type and speed to meet network conditions. A faster codec speed may be required to support both voice and data calls and a slower one for only voice traffic.

g711ulaw

Codec G.711 u-law 64000 bits per second for T1.

g711alaw

Codec G.711 a-law 64000 bits per second for E1.

redundancy

(Optional) Packet redundancy (RFC 2198) for modem traffic.

system

This default setting uses the global configuration parameters set by using the modem-relay command in voice-service configuration mode for VoIP.


Defaults

Disabled

Command Modes

Dial-peer configuration

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

This command applies to VoIP dial peers.

Use this command to configure modem relay over VoIP for a specific dial peer.

Use the same codec type for the originating and terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.

The redundancy keyword is an optional parameter for sending redundant packets for modem traffic during the passthrough phase.

When the system keyword is enabled, the following parameters are not available: nse, payload-type, codec, and redundancy. The system keyword overrides the configuration for the dial peer and the values from the modem-relay command in voice-service configuration mode for VoIP are used.

Examples

The following example shows modem relay over VoIP configured for a specific dial peer in dial-peer configuration mode.

modem relay nse codec g711ulaw redundancy

Related Commands

Command
Description

modem relay (voice-service)

Configures modem relay globally for all VoIP dial peers.


modem relay (voice-service)

To configure modem relay over VoIP for all connections for the Cisco AS5300, use the modem relay command in voice-service configuration mode. To disable modem relay for all connections, use the no form of this command.

modem relay nse [payload-type number] codec {g711ulaw | g711alaw}
[
redundancy] [maximum-sessions value]

no modem relay nse

Syntax Description

nse

Named signaling event.

payload-type

(Optional) NSE payload type.

number

(Optional) The value of the payload type, in a range from 98 through 120. The default is 100.

codec

Sets the upspeed voice compression selection for speech or audio signals. The upspeed method is used to dynamically change the codec type and speed to meet network conditions. A faster codec speed may be required to support both voice and data calls and a slower one for only voice traffic.

g711ulaw

Codec G.711 u-law 64000 bits per second for T1.

g711alaw

Codec G.711 a-law 64000 bits per second for E1.

redundancy

(Optional) Packet redundancy (RFC2198) for modem traffic.

maximum-sessions value

Maximum number of redundant, simultaneous modem passthrough sessions. Valid range is 1 through 10000.


Defaults

Disabled

Command Modes

Voice-service configuration

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this command to configure modem relay over VoIP for the Cisco AS5300. The default behavior for this command is no modem relay. Configuration of modem relay for VoIP dial peers via the modem relay dial-peer configuration command overrides this voice-service command for the specific VoIP dial peer where the dial-peer command is configured.

The payload type is an optional parameter for the nse keyword. Use the same payload-type number for both the originating gateway and the terminating gateway. The payload-type number can be set from 96 to 119. If you do not specify the payload-type number, the number defaults to 100.

Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec is required for T1, and g711alaw codec is required for E1.

The redundancy keyword is an optional parameter for sending redundant packets for modem traffic during the passthrough phase.

The maximum-sessions keyword is an optional parameter for the modem relay command. This parameter determines the maximum number of redundant, simultaneous modem relay sessions. The recommended value for the maximum-sessions is 16. The value can be set from 1 through 10000. The maximum-sessions keyword applies only if the redundancy keyword is used.

When using the voice service voip and modem relay nse commands on a terminating gateway to globally set up fax or modem relay with NSEs, you must also ensure that each incoming call will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate calls with dial peers by using the incoming called-number command to specify a sequence of digits that incoming calls can match. You can ensure that all calls will match at least one dial peer by using the following commands:

Router(config)# dial-peer voice tag voip
Router(config-dial-peer)# incoming called-number .

Examples

The following example shows modem relay configuration in voice-service configuration mode for NSE payload type 101 using codec G.711:

modem relay nse payload-type 101 codec g711ulaw redundancy maximum-session 1

Related Commands

Command
Description

incoming called-number

Defines an incoming called number to match a specific dial peer.

modem relay (dial-peer)

Configures modem relay on a specific VoIP dial peer.


modem relay gateway-xid

To set in-band negotiation of compression parameters between two VoIP gateways, use the modem relay gateway-xid command in dial-peer or voice-service configuration mode. To disable this function, use the no form of this command.

modem relay gateway-xid {[compress [backward | forward | both | no] ] [ dictionary value] [ string-length value ]}

no modem relay gateway-xid

Syntax Description

compress

This keyword specifies the direction the data flow is compressed. For normal dial-up, compression should be enabled on both directions.

You may want to disable compression in one or more directions (this is normally done during testing, and perhaps for gaming applications; but not for normal dial-up, when compression is enabled on both directions).

backward—enables compression only in the backward direction.

forward—enables compression only in the forward direction.

both (default)—For normal dial-up, this will be the preferred setting.

no—disables compression in both directions.

dictionary value

V.42 bis parameter that specifies characteristics of the compression algorithm. Default value is 1024.

Note Your modem may not support higher values.

string-length value

V.42 bis parameter that specifies characteristics of the compression algorithm. Default value is 32.

Note Your modem may not support higher values.


Defaults

Enabled

Command Modes

Dial-peer configuration
Voice-service configuration

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

If enabled on both gateways, in-band negotiation of compression parameters occurs between the two VoIP gateways. Inband negotiation occurs between the two DSPs on either side of the network. This is not for negotiation of any parameters, but to decide if there will be a negotiation process. The actual parameters are negotiated by the in-band negotiation process and not by the host at a later stage, assuming the two hosts have agreed to do this negotiation, by having their gateway-xid enabled.

The remaining parameters specify the negotiation posture of this gateway in the subsequent inband negotiation step (assuming inband negotiation was agreed on by the two gateways).

xid negotiation is a DSP-specific feature used in modem relay. The modem relay gateway-xid command enables this negotiation feature. By default it is disabled. The parameters compress and dictionary and string-length are DSP-specific and related to xid negotiation. If gateway-xid is disabled, they are all irrelevant. The application passes these configured values to the DSP that needs them.

Examples

The following example enables in-band negotiation of compression parameters on the VoIP gateway, with compression in both directions, and dictionary size of 1024 and string length of 32 for the compression algorithm:

modem relay gateway-xid compress both dictionary 1024 string-length 32

Related Commands

Command
Description

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp tse payload

Enables TSEs for communications between gateways, required for modem relay over VoIP using MGCP.

mgcp modem relay voip gateway-xid

Optimizes the modem relay transport protocol and the estimated one-way delay across the IP network.

mgcp modem relay voip sprt retries

Sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.


modem relay latency

To optimize the modem relay transport protocol and the estimated one-way delay across the IP network, use the modem relay latency command in dial-peer or voice-service configuration mode. To disable this function, use the no form of this command.

modem relay latency value

no modem relay latency

Syntax Description

value

Estimated one-way delay across the IP network, used to optimize the modem relay transport protocol. Valid range is from 100ms to 1000 ms. The default value is 200 ms, which does not need to be changed for most networks.


Defaults

Default value is 200.

Command Modes

Dial-peer configuration
Voice-service configuration

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this command to adjust the retransmission timer of the SPRT protocol, if required, by setting this value to the estimated one-way delay (in milliseconds) across the IP network. Changing this value may affect the throughput or delay characteristics of the modem relay call. The default value of 200 does not need to be changed for most networks.

Examples

The following example sets the estimated one-way delay across the IP network to 100 msec.

Router(config-dial-peer)# modem relay latency 100

Related Commands

Command
Description

mgcp modem relay voip mode

Sets in-band negotiation of compression parameters between two VoIP gateways.

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp tse payload

Enables telephony signaling events (TSEs) for communications between gateways, required for modem relay over VoIP using MGCP.

mgcp modem relay voip sprt retries

Sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.


modem relay sprt retries

To set the maximum number of times that the Simple Packet Relay Transport (SPRT) protocol tries to send a packet before disconnecting, use the modem relay sprt retries command in dial-peer or voice-service configuration mode. To disable this function, use the no form of this command.

modem relay sprt retries value

no modem relay sprt retries

Syntax Description

sprt

Simple packet relay transport. It is the protocol that carries modem relay traffic between two gateways.

retries value

The maximum number of times that the SPRT protocol tries to send a packet before disconnecting. The default value is 12.


Defaults

Default value is 12.

Command Modes

Dial-peer configuration
Voice-service configuration

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this dial-peer or voice-service configuration command to set the maximum number of times that the SPRT protocol tries to send a packet before disconnecting.

Examples

The following example sets the maximum number of times that the SPRT protocol tries to send a packet before disconnecting at 15.

modem relay sprt retries 15

Related Commands

Command
Description

modem relay gateway-xid

Sets in-band negotiation of compression parameters between two VoIP gateways.

mgcp modem relay voip mode

Enables modem relay mode support in a gateway for MGCP VoIP calls.

mgcp tse payload

Enables TSEs for communications between gateways, required for modem relay over VoIP using MGCP.

modem relay latency

Optimizes the modem relay transport protocol and the estimated one-way delay across the IP network.


show call active voice

To show current call information for a call in progress, use the show call active command in privileged EXEC mode.

show call active voice [brief [id value] | compact [duration {less time | more time}] | id value]

Syntax Description

voice

Specifies that the active call table displays voice call information.

brief

(Optional) Displays a truncated version.

compact

(Optional) Displays a compact version.

id value

Call identifier as shown in compact or brief format. Range: 1 to FFFF.

duration

(Optional) Displays active calls of a specified duration.

less time

Displays shorter calls. Range: 1 to 2147483647.

more time

Displays longer calls. Range: 1 to 2147483647.


Defaults

No default behavior or values.

Command Modes

Privileged EXEC

Command History

Release
Modification

11.3(1)T

This command was introduced.

12.0(4)XJ

This command was modified for store and forward fax.

12.1(3)T

This command was modified for modem passthrough over VoIP on the Cisco AS5300.

12.(2)11T

This command was modified for modem relay over VoIP.


Usage Guidelines

Usethis command to display the contents of the active call table. If you use the voice keyword, the active call table displays information about all the voice calls currently connected through the router or access server.

Examples

The following is a brief sample output from the show call active voice brief command:

router# show call active voice brief

<ID>:<start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
  dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
 IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
  delay:<last>/<min>/<max>ms <codec>
  MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
   last <buf event time>s dur:<Min>/<Max>s
 FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
  sig:<on/off> <codec> (payload size)
 ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
  sig:<on/off> <codec> (payload size)
 Tele <int>:tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
  MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>
 Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt:<type>/<manf>
 bw:<req>/<act> codec:<audio>/<video>
  tx:<audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
 rx:<audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Total call-legs:2
11E1 :1347833hs.1 +80 pid:50281 Answer 4085260287 active
 dur 00:00:38 tx:347/14028 rx:336/13640
 Tele 0:D:5:tx:5070/2340/0ms modem-relay noise:0 acom:0  i/0:0/0 dBm
  MODEMRELAY info:10/10/0 xid:0/0 total:76/10/0

11E1 :1347834hs.1 +59 pid:25 Originate 50281 active
 dur 00:00:39 tx:336/13640 rx:347/12312
 IP 1.8.84.15:18986 rtt:0ms pl:2380/40ms lost:13/1/0 delay:57/57/70ms g729r8

Shown below is a sample output for the show call active voice command:

router# show call active  voice
Total call-legs:2
GENERIC:
SetupTime=1347833 ms
Index=1
PeerAddress=40812345677
PeerSubAddress=
PeerId=50181
PeerIfIndex=59
LogicalIfIndex=30
ConnectTime=1347913
CallDuration=00:01:20
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=2
TransmitPackets=521
TransmitBytes=15072
!
!
SPRTTotalInfoBytesReceived=76
SPRTTotalInfoBytesSent=10
SPRTPacketDrops=0

 GENERIC:
SetupTime=1347834 ms
Index=1
PeerAddress=50181
PeerSubAddress=
PeerId=25
PeerIfIndex=58
LogicalIfIndex=0
ConnectTime=1347893
CallDuration=00:01:21
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=515
TransmitBytes=14714
ReceivePackets=525
ReceiveBytes=11244
VOIP:
ConnectionId[0x99001380 0xBF3411D3 0x800E9165 0x694C2CE8]
IncomingConnectionId[0x99001380 0xBF3411D3 0x800E9165 0x694C2CE8]
RemoteIPAddress=1.8.84.15
RemoteUDPPort=18986
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2
SessionTarget=1.8.84.15
OnTimeRvPlayout=2380
GapFillWithSilence=10 ms
GapFillWithPrediction=30 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=70 ms
LoWaterPlayoutDelay=57 ms
ReceiveDelay=57 ms
LostPackets=13
EarlyPackets=1
LatePackets=0
VAD = enabled
CoderTypeRate=g729r8
CodecBytes=20
SignalingType=ext-signal
CallerName=
CallerIDBlocked=False

Table 2 provides an alphabetical listing of the show call active command fields and a description of each field.

Table 2 show call active Field Descriptions 

Field
Description

ACOM Level

Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.

Buffer Drain Events

Total number of jitter buffer drain events.

Buffer Fill Events

Total number of jitter buffer fill events.

CallDuration

Length of the call in hours, minutes and seconds, hh:mm:ss.

CallOrigin

Call origin: answer or originate.

CallState

Current state of the call.

ChargedUnits

Total number of charging units applying to this peer since system startup. The unit of measure for this field is in hundredths of seconds.

CodecBytes

Payload size in bytes for the codec used.

CoderTypeRate

Negotiated coder rate. This value specifies the send rate of voice/fax compression to its associated call leg for this call.

ConnectionId

Global call identifier for this gateway call.

ConnectTime

Time when the call was connected.

Consecutive-packets-lost Events

Total number of consecutive (two-or-more) packet loss events.

Corrected packet-loss Events

Total number of packet loss events that were corrected using the RFC 2198 method.

Dial-Peer

Tag of the dial peer sending this call.

ERLLevel

Current Echo Return Loss (ERL) level for this call.

FaxTxDuration

Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

GapFillWithInterpolation

Duration of the voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

GapFillWithRedundancy

Duration of the voice signal played out with a signal synthesized from redundancy parameters available because voice data was lost or not received in time from the voice gateway for this call.

GapFillWithPrediction

Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms.

GapFillWithSilence

Duration of voice signal replaced with silence because voice data was lost or not received in time for this call.

HiWaterPlayoutDelay

High-water mark Voice Playout FIFO Delay during this call.

Index

Dial peer identification number.

InfoActivity

Active information transfer activity state for this call.

InfoType

Information type for this call, for example, voice or fax.

InSignalLevel

Active input signal level from the telephony interface used by this call.

Last Buffer Drain/Fill Event

Time since the last jitter buffer drain/fill event, in seconds.

LogicalIfIndex

Index number of the logical interface for this call.

LoWaterPlayoutDelay

Low water mark Voice Playout FIFO Delay during this call.

Modem passthrough signaling method in use

Indicates that this is a modem passthrough call and that named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signaling codec upspeed. The upspeed method is used to dynamically change the codec type and speed to meet network conditions. A faster codec may be needed for both voice and data calls, and then a slower one for voice-only traffic.

Modem relay signaling method in use

Indicates that this is a modem relay call and named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signaling codec upspeed. The upspeed method is used to dynamically change the codec type and speed to meet network conditions. A faster codec may be needed for both voice and data calls, and then a slower one for voice-only traffic.

NoiseLevel

Active noise level for this call.

OnTimeRvPlayout

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

OutSignalLevel

Active output signal level to telephony interface used by this call.

PeerAddress

Destination pattern or number associated with this peer.

PeerId

ID value of the peer table entry to which this call was made.

PeerIfIndex

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

PeerSubAddress

Subaddress when this call is connected.

Percent Packet Loss

Total percent packet loss.

ReceiveBytes

Number of bytes received by the peer during this call.

ReceiveDelay

Average Playout FIFO Delay plus the Decoder Delay during this voice call.

ReceivePackets

Number of packets received by this peer during this call.

RemoteIPAddress

Remote system IP address for the VoIP call.

RemoteUDPPort

Remote system UDP listener port to which voice packets are sent.

RoundTripDelay

Voice packet round trip delay between the local and remote system on the IP backbone for this call.

SelectedQoS

Selected RSVP quality of service (QoS) for this call.

SessionProtocol

Session protocol used for an Internet call between the local and remote router through the IP backbone.

SessionTarget

Session target of the peer used for this call.

SetupTime

Value of the system UpTime when the call associated with this entry was started.

SignalingType

Signaling type for this call, for example, channel-associated signaling (CAS) or common-channel signaling (CCS).

Time between Buffer Drain/Fills

Minimum and maximum durations between jitter buffer drain/fill events, in seconds.

TransmitBytes

Number of bytes sent by this peer during this call.

TransmitPackets

Number of packets sent by this peer during this call.

TxDuration

Duration of transmit path open from this peer to the voice gateway for this call.

VAD

Whether voice activation detection (VAD) was enabled for this call.

VoiceTxDuration

Duration of voice transmission from this peer to the voice gateway for this call. Derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.


Related Commands

Command
Description

show call history voice

Displays the VoIP call history table.

show dial-peer voice

Displays configuration information for dial peers.

show num-exp

Displays how the number expansions are configured in VoIP.

show voice port

Displays configuration information about a specific voice port.


show call history voice

To display the call history table, use the show call history command in privileged EXEC mode.

show call history voice [brief [id value] | compact [duration {less time | more time}] | [id value] | last number]

Syntax Description

voice

Specifies that the call history table shows voice call information.

brief

(Optional) Displays a truncated version of the call history table.

compact

(Optional) Displays a compact version.

id value

Call identifier as shown in compact or brief format. Range: 1 to FFFF.

duration

(Optional) Displays active calls of a specified duration.

less time

Displays shorter calls. Range: 1 to 2147483647.

more time

Displays longer calls. Range: 1 to 2147483647.

last number

(Optional) Displays the last calls connected, where the number of calls that appear is defined by the number argument. Range: 1 to 100.


Defaults

No default behavior or values

Command Modes

Privileged EXEC

Command History

Release
Modification

11.3(1)T

This command was introduced.

12.0(4)XJ

This command was modified for store and forward fax.

12.1(3)T

This command was modified for modem passthrough on VoIP.

12.2(11)T

This command was modified for modem relay on VoIP.


Usage Guidelines

Use this command to display the voice call history table. The call history table contains a listing of all calls connected through this router in descending time order since VoIP was enabled. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the last keyword, and define the number of calls you want to see with the number argument. To display a truncated version of the call history table, use the brief keyword.

Examples

The following is sample output from the show call history command updated to show modem relay over Voice over IP information:

Router# show call history voice
Total call-legs:2

GENERIC:
SetupTime=43003 ms
Index=1
PeerAddress=50110
PeerSubAddress=
PeerId=102
PeerIfIndex=55
LogicalIfIndex=0
DisconnectCause=0
DisconnectText=
ConnectTime=43018
DisconnectTime=58646
CallDuration=00:02:36
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=0
VOIP:
ConnectionId[0x29448C7E 0x4F140003 0x0 0x2147B]
IncomingConnectionId[0x29448C7E 0x4F140003 0x0 0x2147B]
RemoteIPAddress=1.14.82.60
RemoteUDPPort=16530
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=TRUE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=cisco
SessionTarget=
OnTimeRvPlayout=0
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=0 ms
LoWaterPlayoutDelay=0 ms
ReceiveDelay=0 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = enabled
CoderTypeRate=g711ulaw
CodecBytes=0
cvVoIPCallHistoryIcpif=0
SignalingType=cas
CallerName=50110,Voice/ENA
CallerIDBlocked=False

GENERIC:
SetupTime=43009 ms
Index=2
PeerAddress=55250
PeerSubAddress=
PeerId=101
PeerIfIndex=56
LogicalIfIndex=54
DisconnectCause=10
DisconnectText=normal call clearing.
ConnectTime=43018
DisconnectTime=58647
CallDuration=00:02:36
CallOrigin=1
ChargedUnits=0
InfoType=speech
TransmitPackets=922
TransmitBytes=44779
ReceivePackets=1021
ReceiveBytes=59857
TELE:
ConnectionId=[0x29448C7E 0x4F140003 0x0 0x2147B]
IncomingConnectionId=[0x29448C7E 0x4F140003 0x0 0x2147B]
TxDuration=4790 ms
VoiceTxDuration=2660 ms
FaxTxDuration=0 ms
CoderTypeRate=g711ulaw
NoiseLevel=0
ACOMLevel=0
SessionTarget=
ImgPages=0
CallerName=50110,Voice/ENA
CallerIDBlocked=False
SPRTInfoFramesReceived=10
SPRTInfoTFramesSent=9
SPRTInfoTFramesResent=0
SPRTXidFramesReceived=0
SPRTXidFramesSent=0
SPRTTotalInfoBytesReceived=10
SPRTTotalInfoBytesSent=76
SPRTPacketDrops=678

Table 3 provides an alphabetical listing of the show call history command fields and a description of each field.

Table 3 show call history Field Descriptions 

Field
Description

ACOMLevel

Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.

Buffer Drain Events

Total number of jitter buffer drain events.

Buffer Fill Events

Total number of jitter buffer fill events.

CallDuration

Length of the call in hours, minutes, and seconds; hh:mm:ss.

CallOrigin

Call origin: answer or originate.

ChargedUnits

Total number of charging units applying to this peer since system startup. The unit of measure for this field is in hundredths of seconds.

CodecBytes

Payload size in bytes for the codec used.

CoderTypeRate

Negotiated coder rate. This value specifies the send rate of voice/fax compression to its associated call leg for this call.

ConnectionID

Global call identifier for the gateway call.

ConnectTime

Time when this call was connected.

Consecutive-packets-lost Events

Total number of consecutive (two-or-more) packet loss events.

Corrected packet-loss Events

Total number of packet loss events that were corrected using the RFC 2198 method.

DisconnectCause

Description explaining why this call was disconnected.

DisconnectText

Descriptive text explaining the disconnect reason.

DisconnectTime

Time when this call was disconnected.

FaxTxDuration

Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

GapFillWithInterpolation

Duration of the voice signal played out with a signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.

GapFillWithRedundancy

Duration of the voice signal played out with a signal synthesized from redundancy parameters available because voice data was lost or not received in time from the voice gateway for this call.

GapFillWithSilence

Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.

GapFillWithPrediction

Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call.

HiWaterPlayoutDelay

High water mark Voice Playout FIFO Delay during this voice call.

Index

Dial peer identification number.

InfoType

Information type for this call, for example, voice or fax.

Last Buffer Drain/Fill Event

Time since the last jitter buffer drain/fill event, in seconds.

LogicalIfIndex

Index number of the logical voice port for this call.

LoWaterPlayoutDelay

Low-water mark Voice Playout FIFO Delay during this voice call.

Modem passthrough signaling method is nse

Indicates that this is a modem passthrough call and named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signaling codec upspeed. The upspeed method is used to dynamically change the codec type and speed to meet network conditions. A faster codec may be needed for both voice and data calls, and then a slower one for voice-only traffic.

Modem relay signaling method is nse

Indicates that this is a modem relay call and named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signaling codec upspeed. The upspeed method is used to dynamically change the codec type and speed to meet network conditions. A faster codec may be needed for both voice and data calls, and then a slower one for voice-only traffic.

NoiseLevel

Average noise level for this call.

OnTimeRvPlayout

Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

Percent Packet Loss

Total percent packet loss.

PeerAddress

Destination pattern or number associated with this peer.

PeerId

ID value of the peer entry table to which this call was made.

PeerIfIndex

Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.

PeerSubAddress

Subaddress where this call is connected.

ReceiveBytes

Number of bytes received by the peer during this call.

ReceiveDelay

Average Playout FIFO Delay plus the Decoder Delay during this voice call.

ReceivePackets

Number of packets received by this peer during this call.

RemoteIPAddress

Remote system IP address for this call.

RemoteUDPPort

Remote system UDP listener port to which voice packets are sent.

RoundTripDelay

Voice packet round trip delay between the local and remote system on the IP backbone for this call.

SelectedQoS

Selected RSVP QoS for this call.

Session Protocol

Session protocol used for an Internet call between the local and remote router through the IP backbone.

Session Target

Session target of the peer used for this call.

SetUpTime

Value of the system UpTime when the call associated with this entry was started.

SignalingType

Signaling type for this call, for example, channel-associated signaling (CAS) or common-channel signaling (CCS).

Time between Buffer Drain/Fills

Minimum and maximum durations between jitter buffer drain/fill events, in seconds.

TransmitBytes

Number of bytes sent by this peer during this call.

TransmitPackets

Number of packets sent by this peer during this call.

TxDuration

Duration of the transmit path open from this peer to the voice gateway for this call.

VAD

Whether voice activation detection (VAD) was enabled for this call.

VoiceTxDuration

Duration of voice transmission from this peer to the voice gateway for this call. Derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.


show modem relay statistics

To display various statistics for modem relay, use the show modem relay statistics command in privileged EXEC mode.

show modem relay statistics {all | phy | pkt | queue | sprt | timer | v42} [call-identifier call-setup-time call-index]

Syntax Description

all

All statistics associated with the modem relay feature.

phy

Modem relay physical layer statistics.

pkt

Modem relay Packetizer statistics.

queue

Modem relay queue statistics.

sprt

Modem relay SPRT layer statistics.

timer

Mmodem relay timer statistics.

v42

Mmodem relay V.42 statistics.

call-identifier

Identifies a particular call.

call-setup-time

Value of the system UpTime when the call associated with this entry was started. Range: 0 to 4294967295.

call-index

Dial peer identification number used to distinguish between calls with the same setup time. Range: 0 to 4294967295.


Defaults

No statistics are displayed.

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(11)T

This command was introduced for the Cisco 2600, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 7200, and Cisco AS5300.


Usage Guidelines

Use this command to display various modem relay call statistics, including counts of different types of packets, errors and events, for all modem relay calls.

Display statistics for a specific modem relay call by using the call-identifier keyword and specifying the call setup time and call index of the desired call. Obtain the call setup time and call index values from the SetupTime and Index fields at the start of each call record in the show call active command output.

Examples

Router# show modem relay statistics all call-identifier 43009 1

ID:3

SPRT Layer Statistics
        sprt_info_frames_rcvd=10 sprt_xid_frames_rcvd=0
        sprt_tc0_explicit_acks_rcvd=6 sprt_tc1_explicit_acks_rcvd=122
        sprt_tc2_explicit_acks_rcvd=126 sprt_destructive_brks_rcvd=0
        sprt_expedited_brks_rcvd=0
        sprt_non_expedited_brks_rcvd=0
        sprt_info_tframes_sent=9 sprt_info_tframes_resent=0
        sprt_xid_frames_sent=0 sprt_tc0_explicit_acks_sent=8
        sprt_tc1_explicit_acks_sent=129 sprt_tc2_explicit_acks_sent=132
        sprt_destructive_brks_sent=0
        sprt_expedited_brks_sent=0
        sprt_non_expedited_brks_sent=0
        sprt_info_tframes_asked_to_consumed=10
        sprt_info_tframes_consumed=10
        sprt_info_tframes_failed_to_consume=0
        sprt_info_bytes_rcvd=10 sprt_info_bytes_sent=76
        sprt_pkts_dropped_intf_busy=289 sprt_min_rexmit_timeout=500
        sprt_max_rexmit_timeout=500

Queue Statistics
        sprt_tc1_rcv_qdrops=0 sprt_tc1_xmit_qdrops=0
        sprt_tc2_rcv_qdrops=0 sprt_tc2_xmit_qdrops=0
        pktizer_out_qdrops=4 pktizer_in_qdrops=0 v42_xmit_qdrops=0

V42 Layer Statistics
        vs_chng_dueto_timeouts=0 vs_chng_dueto_rej=0
        vs_chng_dueto_rnr_resp_f1_set=0 nr_seq_exception=0
        good_rcvd_lapm_pkts=1385 discarded_rcvd_lapm_pkts=0
        rejected_rcvd_lapm_pkts=0 v42_rcvd_iframe=9
        v42_rcvd_rr=1374 v42_rcvd_rnr=0 v42_rcvd_rej=0
        v42_rcvd_srej=0 v42_rcvd_sabme=0 v42_rcvd_dm=0
        v42_rcvd_ui=0 v42_rcvd_disc=0 v42_rcvd_ua=1
        v42_rcvd_frmr=0 v42_rcvd_xid=1 v42_rcvd_test=0
        v42_rcvd_destructive_brk=0 v42_rcvd_expedited_brk=0
        v42_rcvd_non_expedited_brk=0 v42_rcvd_brkack=0
        v42_sent_iframe=10 v42_sent_rr=1464 v42_sent_rnr=0
        v42_sent_rej=0 v42_sent_srej=0 v42_sent_sabme=1
        v42_sent_dm=0 v42_sent_ui=0 v42_sent_disc=0
        v42_sent_ua=0 v42_sent_frmr=0 v42_sent_xid=1
        v42_sent_test=0 v42_sent_destructive_brk=0
        v42_sent_expedited_brk=0
        v42_sent_non_expedited_brk=0
        v42_sent_brkack=0

Physical Layer Statistics
        num_local_retrain=0 num_remote_retrain=0
        num_local_speed_shift=0 num_remote_speed_shift=0
        num_sync_loss=0

Packetizer Statistics
        frames_inprogress=5 good_crc_frames=1385
        bad_crc_frames=31 frame_aborts=124
        hdlc_sync_detects=1 hdlc_sync_loss_detects=0
        bad_frames=0

Timer Statistics
        xid_timer_cnt=0 sabme_timer_cnt=0 ack_timer_cnt=0
        chkpnt_timer_cnt=1333

Router# show modem relay stat all

ID:3

SPRT Layer Statistics
        sprt_info_frames_rcvd=10 sprt_xid_frames_rcvd=0
        sprt_tc0_explicit_acks_rcvd=6 sprt_tc1_explicit_acks_rcvd=155
        sprt_tc2_explicit_acks_rcvd=158 sprt_destructive_brks_rcvd=0
        sprt_expedited_brks_rcvd=0
        sprt_non_expedited_brks_rcvd=0
        sprt_info_tframes_sent=9 sprt_info_tframes_resent=0
        sprt_xid_frames_sent=0 sprt_tc0_explicit_acks_sent=8
        sprt_tc1_explicit_acks_sent=161 sprt_tc2_explicit_acks_sent=165
        sprt_destructive_brks_sent=0
        sprt_expedited_brks_sent=0
        sprt_non_expedited_brks_sent=0
        sprt_info_tframes_asked_to_consumed=10
        sprt_info_tframes_consumed=10
        sprt_info_tframes_failed_to_consume=0
        sprt_info_bytes_rcvd=10 sprt_info_bytes_sent=76
        sprt_pkts_dropped_intf_busy=357 sprt_min_rexmit_timeout=500
        sprt_max_rexmit_timeout=500

Queue Statistics
        sprt_tc1_rcv_qdrops=0 sprt_tc1_xmit_qdrops=0
        sprt_tc2_rcv_qdrops=0 sprt_tc2_xmit_qdrops=0
        pktizer_out_qdrops=4 pktizer_in_qdrops=0 v42_xmit_qdrops=0

V42 Layer Statistics
        vs_chng_dueto_timeouts=0 vs_chng_dueto_rej=0
        vs_chng_dueto_rnr_resp_f1_set=0 nr_seq_exception=0
        good_rcvd_lapm_pkts=1910 discarded_rcvd_lapm_pkts=0
        rejected_rcvd_lapm_pkts=0 v42_rcvd_iframe=9
        v42_rcvd_rr=1899 v42_rcvd_rnr=0 v42_rcvd_rej=0
        v42_rcvd_srej=0 v42_rcvd_sabme=0 v42_rcvd_dm=0
        v42_rcvd_ui=0 v42_rcvd_disc=0 v42_rcvd_ua=1
        v42_rcvd_frmr=0 v42_rcvd_xid=1 v42_rcvd_test=0
        v42_rcvd_destructive_brk=0 v42_rcvd_expedited_brk=0
        v42_rcvd_non_expedited_brk=0 v42_rcvd_brkack=0
        v42_sent_iframe=10 v42_sent_rr=1988 v42_sent_rnr=0
        v42_sent_rej=0 v42_sent_srej=0 v42_sent_sabme=1
        v42_sent_dm=0 v42_sent_ui=0 v42_sent_disc=0
        v42_sent_ua=0 v42_sent_frmr=0 v42_sent_xid=1
        v42_sent_test=0 v42_sent_destructive_brk=0
        v42_sent_expedited_brk=0
        v42_sent_non_expedited_brk=0
        v42_sent_brkack=0

Physical Layer Statistics
        num_local_retrain=0 num_remote_retrain=0
        num_local_speed_shift=0 num_remote_speed_shift=0
        num_sync_loss=0

Packetizer Statistics
        frames_inprogress=5 good_crc_frames=1910
        bad_crc_frames=31 frame_aborts=124
        hdlc_sync_detects=1 hdlc_sync_loss_detects=0
        bad_frames=0

Timer Statistics
        xid_timer_cnt=0 sabme_timer_cnt=0 ack_timer_cnt=0
        chkpnt_timer_cnt=1809

        Total Modem Relay Call Legs = 1

Router# show modem relay stat sprt

ID:3

SPRT Layer Statistics
        sprt_info_frames_rcvd=10 sprt_xid_frames_rcvd=0
        sprt_tc0_explicit_acks_rcvd=6 sprt_tc1_explicit_acks_rcvd=177
        sprt_tc2_explicit_acks_rcvd=180 sprt_destructive_brks_rcvd=0
        sprt_expedited_brks_rcvd=0
        sprt_non_expedited_brks_rcvd=0
        sprt_info_tframes_sent=9 sprt_info_tframes_resent=0
        sprt_xid_frames_sent=0 sprt_tc0_explicit_acks_sent=8
        sprt_tc1_explicit_acks_sent=183 sprt_tc2_explicit_acks_sent=187
        sprt_destructive_brks_sent=0
        sprt_expedited_brks_sent=0
        sprt_non_expedited_brks_sent=0
        sprt_info_tframes_asked_to_consumed=10
        sprt_info_tframes_consumed=10
        sprt_info_tframes_failed_to_consume=0
        sprt_info_bytes_rcvd=10 sprt_info_bytes_sent=76
        sprt_pkts_dropped_intf_busy=403 sprt_min_rexmit_timeout=500
        sprt_max_rexmit_timeout=500

        Total Modem Relay Call Legs = 1

Router# show modem relay stat queue

ID:3

Queue Statistics
        sprt_tc1_rcv_qdrops=0 sprt_tc1_xmit_qdrops=0
        sprt_tc2_rcv_qdrops=0 sprt_tc2_xmit_qdrops=0
        pktizer_out_qdrops=4 pktizer_in_qdrops=0 v42_xmit_qdrops=0

        Total Modem Relay Call Legs = 1

Router# show modem relay stat v42

ID:3

V42 Layer Statistics
        vs_chng_dueto_timeouts=0 vs_chng_dueto_rej=0
        vs_chng_dueto_rnr_resp_f1_set=0 nr_seq_exception=0
        good_rcvd_lapm_pkts=2442 discarded_rcvd_lapm_pkts=0
        rejected_rcvd_lapm_pkts=0 v42_rcvd_iframe=9
        v42_rcvd_rr=2431 v42_rcvd_rnr=0 v42_rcvd_rej=0
        v42_rcvd_srej=0 v42_rcvd_sabme=0 v42_rcvd_dm=0
        v42_rcvd_ui=0 v42_rcvd_disc=0 v42_rcvd_ua=1
        v42_rcvd_frmr=0 v42_rcvd_xid=1 v42_rcvd_test=0
        v42_rcvd_destructive_brk=0 v42_rcvd_expedited_brk=0
        v42_rcvd_non_expedited_brk=0 v42_rcvd_brkack=0
        v42_sent_iframe=10 v42_sent_rr=2539 v42_sent_rnr=0
        v42_sent_rej=0 v42_sent_srej=0 v42_sent_sabme=1
        v42_sent_dm=0 v42_sent_ui=0 v42_sent_disc=0
        v42_sent_ua=0 v42_sent_frmr=0 v42_sent_xid=1
        v42_sent_test=0 v42_sent_destructive_brk=0
        v42_sent_expedited_brk=0
        v42_sent_non_expedited_brk=0
        v42_sent_brkack=0

        Total Modem Relay Call Legs = 1

Router# show modem relay stat phy

ID:3

Physical Layer Statistics
        num_local_retrain=0 num_remote_retrain=0
        num_local_speed_shift=0 num_remote_speed_shift=0
        num_sync_loss=0

        Total Modem Relay Call Legs = 1

Router# show modem relay stat pkt

ID:3

Packetizer Statistics
        frames_inprogress=5 good_crc_frames=2573
        bad_crc_frames=61 frame_aborts=150
        hdlc_sync_detects=1 hdlc_sync_loss_detects=0
        bad_frames=0

        Total Modem Relay Call Legs = 1

Router# show modem relay stat timer

ID:3

Timer Statistics
        xid_timer_cnt=0 sabme_timer_cnt=0 ack_timer_cnt=0
        chkpnt_timer_cnt=2750

        Total Modem Relay Call Legs = 1

Related Commands

Command
Description

show call active voice

Displays current call information for a call in progress.

show modems

Displays all modem configurations.


voice service voip

To enter voice-service configuration mode and specify the voice encapsulation type, use the voice service command in global configuration mode.

voice service voip

Syntax Description

voip

Specifies VoIP encapsulation.


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

12.1(1)XA

This command was introduced for Voice over ATM on the Cisco MC3810 series.

12.1(2)T

This command was implemented for VoIP in Cisco IOS Release 12.1(2)T on the Cisco MC3810.

12.1(3)T

This command was implemented in Cisco IOS Release 12.1(3)T for VoIP on the Cisco AS5300.

12.2(11)T

This command was implemented to include modem relay for VoIP in addition to modem passthrough.


Usage Guidelines

Use this command to switch to voice-service configuration mode from global configuration mode and to specify a voice encapsulation type. Use the exit command to exit voice-service configuration mode and return to global configuration mode.

Examples

The following example shows how to access voice-service configuration mode and specify VoIP voice encapsulation beginning in global configuration mode:

Router(config)# voice service voip
Router(conf-voi-serv)#

Related Commands

Command
Description

modem passthrough

Configures modem passthrough over VoIP.

modem relay

Configures modem relay over VoIP.


Glossary

Backhaul—A scheme where telephony signaling is passed from a gateway to a separate control for processing. With such a scheme, the gateway does not need to interpret the signaling information. In the MGCP specification, this is called signal tunneling.

CA—Call agent. Also called the Media Gateway Controller (MGC).

CSR—Call success rate

EC—Echo cancellation

FEC—Forward error correction

DS0—64 kbps channel in a T1/E1 line

Jitter—A VOIP impairment describing the amount of variation from the normal packet rate measured at the receiving gateway.

H.323—A standard that specifies the components, protocols and procedures that provide multimedia communication services--real-time audio, video, and data communications--over packet networks, including internet protocol (IP)-based networks. H.323 is part of a family of ITU-T recommendations called H.32X that provides multimedia communication services over a variety of networks.

LAPM—Link Access Procedure for Modems. This is a link layer and error correcting protocol that detects errors through the use of a Cyclic Redundancy Check (CRC) and corrects them through the use of automatic retransmission of data.

Media Gateway—A PSTN/PBX-VoIP/NAS gateway that is controlled by a call agent using MGCP.

MGC—Media Gateway Controller. Also called the call agent (CA).

MGCP—Media Gateway Control Protocol

MNP—Microcom Networking Protocol. This is a series of protocols for modem communication. MNP4 is an error correction protocol and MNP5 is a compression protocol. They are superseded by LAPM and V.42bis, meaning that they should only be used as a fallback if both sides do not support LAPM and V.42bis

Modem passthrough—Transmits modem tones as a pulse-code modulation (PCM) stream, inhibiting voice processing functions such as compression and echo cancellation.

Modem relay—Decomposes modem tones to digital form and reconstitutes them on the egress gateway.

NSE—Named Signaling Event. Also called a Telephony signaling event (TSE) in RFC 2833. Used to signal codec upspeed.

PCM—Pulse Code Modulation. Transmission of analog information in digital form through sampling and encoding the samples with a fixed number of bits.

SIP— Session Initiation Protocol

SPRT—Simple Packet Relay Transport

SCTP—Simple Control Transmission Protocol

TDM—Time Division Multiplexing. Technique in which information from multiple channels is allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.

TSE—Telephony Signaling Event. Also called a Named signaling event (NSE) in RFC 2833. Used to signal codec upspeed.

Upspeeding—The method used to dynamically change the codec type and speed to meet network conditions. This means that the modem moves to a faster codec when both voice and data calls are connected, and then slows down when there's only voice traffic. Named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signaling codec upspeed.

V.34—ITU-T Standard that specifies a serial line protocol. V.34 offers improvements to the V.32 standard, including higher transmission rate (28.8 kbps) and enhanced data compression.

V.42—ITU-T Recommendation for error correction and link layer protocol between modems. This protocol provides a procedure for establishing an error-corrected session and describes LAPM. This is the most common protocol used for high speed modems.

V.42bis—ITU-T Recommendation for data compression using error correction procedures.

VAD—Voice activity detection.

VoIP—Voice over Internet Protocol.


Note For a list of other internetworking terms, see the Internetworking Terms and Acronyms document available on the Documentation CD-ROM and Cisco.com at the following URL: http://www.cisco.com/univercd/cc/td/doc/cisintwk/ita/index.htm.