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Cisco IOS Software Releases 12.4 Special and Early Deployments

G.722-64 and iLBC Codec Support on Cisco Unified Border Elements, DSP Farms, and Voice Gateways

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G.722-64 and iLBC Codec Support on Cisco Unified Border Elements, DSP Farms, and Voice Gateways

Table Of Contents

G.722-64 and iLBC Codec Support on Cisco Unified Border Elements, DSP Farms, and Voice Gateways

Contents

How to Configure G.722-64 and iLBC Codecs for Voice Gateways

Conferencing and Transcoding Session Capacities

How to Configure G.722-64 and iLBC Codecs for Cisco Unified Border Elements

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Command Reference

codec (dial-peer)

codec (DSP Farm profile)

codec preference

Feature Information for G.722-64 and iLBC Codec Support on Cisco UBEs, DSP Farms, and Voice Gateways


G.722-64 and iLBC Codec Support on Cisco Unified Border Elements, DSP Farms, and Voice Gateways


First Published: December 17, 2007
Last Updated: December 27, 2007

The G.722-64 and iLBC codecs are supported for Cisco Unified Border Elements (Cisco UBEs), DSP farms, and voice gateways. Conferencing and universal transcoding are supported on both codecs.

Finding Feature Information in This Module

Your Cisco IOS software release may not support all of the features documented in this module. To reach links to specific feature documentation in this module and to see a list of the releases in which each feature is supported, use the "Feature Information for G.722-64 and iLBC Codec Support on Cisco UBEs, DSP Farms, and Voice Gateways" section.

Finding Support Information for Platforms and Cisco IOS Software Images

Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.

Contents

How to Configure G.722-64 and iLBC Codecs for Voice Gateways

How to Configure G.722-64 and iLBC Codecs for Cisco Unified Border Elements

Additional References

Command Reference

Feature Information for G.722-64 and iLBC Codec Support on Cisco UBEs, DSP Farms, and Voice Gateways

How to Configure G.722-64 and iLBC Codecs for Voice Gateways

The G.722-64 and iLBC codecs can be used to enable conferencing and transcoding on Cisco IOS voice gateways in a Cisco Unified Communications Manager network. Digital signal processor (DSP) farms provide conferencing and transcoding services using DSP resources on high-density digital voice/fax network modules.

To configure conferencing and transcoding for voice gateway routers, see the "Configuring Enhanced Conferencing and Transcoding for Voice Gateway Routers" chapter of the Cisco CallManager and Cisco IOS Interoperability Guide.

For more information on configuring iLBC codecs for H.323 and SIP, see the "Dial Peer Overview" chapter and "Dial Peer Features and Configuration" chapter in Dial Peer Configuration on Voice Gateway Routers

The following changes apply to this chapter:

Codecs

End-user devices must be equipped with one of the following codecs:

Codec
Packetization Periods for Transcoding (ms)

G.711 a-law, G.711 u-law, G.722-64

10, 20, or 30

G.729, G.729A, G.729B, G.729AB

10, 20, 30, 40, 50, or 60

iLBC

20 or 30


Conferencing and Transcoding Session Capacities

Each DSP is individually configurable to support either conferencing or transcoding and standard voice termination. The total number of conferencing, transcoding, and voice termination sessions is limited by the capacity of the entire system, which includes the DSPs, hardware platform, physical voice interface, and Cisco Unified Communications Manager.

Table 1 and Table 2 list the maximum number of conference calls and transcoding sessions that DSPs can handle, in theory. Actual capacity may be less based on the total system design.

Table 1 DSP Theoretical Session Capacities 

Application
NM-HD-1V/2V
(1 DSP)
NM-HD-2VE
(3 DSPs)
NM-HDV2
(16 DSPs)
2801/2811
(2 PVDM2-64)
2821/2851
(3 PVDM2-64)
3825, 3845
(4 PVDM2-64)
Conferencing

G.711

8 sessions
(64 conferees)

24 sessions
(192 conferees)

50 sessions
(400 conferees)

50 sessions
(400 conferees)

50 sessions
(400 conferees)

50 sessions
(400 conferees)

G.722-64

2 sessions
(16 conferees)

6 sessions
(48 conferees)

32 sessions
(256 conferees)

16 sessions
(128 conferees)

24 sessions
(192 conferees)

32 sessions
(256 conferees)

G.729

2 sessions
(16 conferees)

6 sessions
(48 conferees)

32 sessions
(256 conferees)

16 sessions
(128 conferees)

24 sessions
(192 conferees)

32 sessions
(256 conferees)

iLBC

1 session
(8 conferees)

3 sessions
(24 conferees)

16 sessions
(128 conferees)

8 sessions
(64 conferees)

12 sessions
(96 conferees)

16 sessions
(128 conferees)

Transcoding

G.711 a-law/u-law <-> G.729a/G.729ab/ GSM FR

8 sessions

24 sessions

128 sessions

64 sessions

96 sessions

128 sessions

G.711 a-law/u-law <-> G.729/G.729b/ GSM EFR

6 sessions

18 sessions

96 sessions

48 sessions

72 sessions

96 sessions

G.722-64 <-> G.711

8 sessions

24 sessions

128 sessions

64 sessions

96 sessions

128 sessions

G.722-64<-> any

4 sessions

12 sessions

64 sessions

32 sessions

48 sessopms

64 sessions

iLBC <-> G.711

6 sessions

18 sessions

96 sessions

48 sessions

72 sessions

96 sessions

iLBC <-> any

3 sessions

9 sessions

48 sessions

24 sessions

36 sessions

48 sessions

Voice Termination

G.711 a-law/u-law

16 sessions

48 sessions

256 sessions

128 sessions

192 sessions

256 sessions

G.722-64, G.726, G.729a, G.729ab, iLBC

8 sessions

24 sessions

128 sessions

64 sessions

96 sessions

128 sessions

G.729, G.729b, G.723.1, G.728

6 sessions

18 sessions

96 sessions

48 sessions

72 sessions

96 sessions


Table 2 Theoretical System Capacities for One DSP 

Application
G.711 a-law/u-law
G.722-64
G729 a/ab
G.729, G.729b
iLBC
Conferencing

8 sessions
(8 x 8 = 64 conferees)

2 sessions
(8 x 2 = 16 conferees)

2 sessions
(8 x 2 = 16 conferees)

2 sessions
(8 x 2 = 16 conferees)

1 session
(1 x 8 = 8 conferees)

Conferencing on PVDM2-8

4 sessions
(4 x 8 = 32 conferees)

1 session
(1 x 8 = 8 conferees)

1 session
(1 x 8 = 8 conferees)

1 session
(1 x 8 = 8 conferees)

1 session
(1 x 8 = 8 conferees)

Hardware MTP

16 sessions

Transcoding

8 sessions

8 sessions

8 sessions

6 sessions

8 sessions


How to Configure G.722-64 and iLBC Codecs for Cisco Unified Border Elements

The G.722-64 and iLBC codecs can be used to set up transcoding on Cisco Unified Border Elements (Cisco UBEs). To configure these codecs on a Cisco UBE, see the "Fundamental Cisco Multiservice IP-to-IP Gateway Configuration" chapter of the Cisco Multiservice IP-to-IP Gateway document.

Additional References

The following sections provide references related to G.722-64 and iLBC Codec Support on Cisco UBEs, DSP farms, and voice gateways.

Related Documents

Related Topic
Document Title

Conferencing and transcoding for voice gateways

Cisco Communications Manager and Cisco IOS Interoperability Guide

Transcoding for Cisco Unified Communications Manager Express

Cisco Unified Communications Manager Express System Administrator Guide

Transcoding for Cisco Unified Border Elements

Cisco Multiservice IP-to-IP Gateway

Dial-peer configuration

Dial Peer Configuration on Voice Gateway Routers


Standards

Standard
Title

H245, Annex S

Control protocol for multimedia communication


MIBs

MIB
MIBs Link

CISCO-VOICE-COMMON-DIAL-CONTROL-MIB.my

CISCO-VOICE-DIAL-CONTROL-MIB.my

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

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


RFCs

RFC
Title

RFC3951

Internet Low Bit Rate Codec (iLBC)

RFC3952

Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech


Technical Assistance

Description
Link

The Cisco Technical Support & Documentation website contains thousands of pages of searchable technical content, including links to products, technologies, solutions, technical tips, and tools. Registered Cisco.com users can log in from this page to access even more content.

http://www.cisco.com/techsupport


Command Reference

This section documents new and modified commands:

codec (dial-peer)

codec (DSP Farm profile)

codec preference

codec (dial-peer)

To specify the voice coder rate of speech for a dial peer, use the codec command in dial-peer configuration mode. To reset the default value, use the no form of this command.

Cisco 1750 and Cisco 1751 Modular Access Routers, Cisco AS5300 and AS5800 Universal Access Servers, and Cisco MC3810 Multiservice Concentrators

codec codec [bytes payload_size]

no codec codec [bytes payload_size]

Cisco 2600 and 3600 Series Routers and Cisco 7200 and 7500 Series Routers

codec {codec [bytes payload_size] | transparent}

no codec {codec [bytes payload_size] | transparent}

Syntax Description

codec

Codec options available for the various platforms are described in Table 3, below.

bytes

(Optional) Specifies the number of bytes in the voice payload of each frame.

payload-size

(Optional) Number of bytes in the voice payload of each frame. See Table 4 for valid entries and default values.

transparent

Enables codec capabilities to be passed transparently between endpoints in a Cisco Unified Border Element.

Note The transparent keyword is only available on the Cisco 2600 and 3600 Series Router and Cisco 7200 and 7500 Series Router platforms.


Table 3 Codec support by platform  

Codec
Cisco 1750 and Cisco 1751 Modular Access Routers
Cisco 2600 and 3600 Series Routers and Cisco 7200 and 7500 Series Routers
Cisco AS5300 and AS5800 Universal Access Servers
Cisco MC3810 Multiservice Concentrators

clear-channelClear channel at 64,000 bits per second (bps)

Yes

Yes

Yes

g711alawG.711 A-Law at 64,000 bps

Yes

Yes

Yes

Yes

g711ulawG.711 u-Law at 64,000 bps

Yes

Yes

Yes

Yes

g722-64G.722-64 at 64,000 bps

Yes

Yes

Yes

g723ar53G.723.1 Annex A at 5300 bps

Yes

Yes

Yes

g723ar63G.723.1 Annex A at 6300 bps

Yes

Yes

Yes

g723r53G.723.1 at 5300 bps

Yes

Yes

Yes

g723r63G.723.1 at 6300 bps

Yes

Yes

Yes

g726r16G.726 at 16,000 bps

Yes

Yes

Yes

Yes

g726r24G.726 at 24,000 bps

Yes

Yes

Yes

Yes

g726r32G.726 at 32,000 bps

Yes

Yes

Yes

Yes

g726r53G.726 at 53,000 bps

Yes

Yes

Yes

g726r63G.726 at 63,000 bps

Yes

Yes

Yes

g728G.728 at 16,000 bps

Yes

Yes

Yes

g729abr8G.729 Annex A and B at 8000 bps

Yes

Yes

Yes

Yes

g729ar8G729 Annex A at 8000 bps

Yes

Yes

Yes

Yes

g729br8G.729 Annex B at 8000 bps

Yes

Yes

Yes

Yes

g729r8G.729 at 8000 bps. This is the default codec

Yes

Yes

Yes

Yes


 

Defaults

g729r8, 30-byte payload for VoFR and VoATM
g729r8, 20-byte payload for VoIP
See Table 4 for valid entries and default values.

Command Modes

dial-peer configuration

Command History

Release
Modification

11.3(1)T

This command was introduced on the Cisco 3600 series.

11.3(3)T

This command was implemented on the Cisco 2600 series.

12.0(3)T

This command was implemented on the Cisco AS5300. This release does not support the clear-channel keyword.

12.0(4)T

This command was implemented on the Cisco 3600 series, Cisco 7200 series and the Cisco MC3810. This release modified the command for VoFR dial peers.

12.0(5)XE

Additional codec choices and other options were implemented.

12.0(5)XK

The g729br8 and pre-ietf codec choices were added for the Cisco 2600 and Cisco 3600 series.

12.0(7)T

This command was integrated into Cisco IOS Release 12.0.(7)T and implemented on the Cisco AS5800. Additional voice coder rates of speech were added. This release does not support the clear-channel keyword on this platform.

12.0(7)XK

The g729abr8 and g729ar8 codec choices were for the Cisco MC3810, and the keyword pre-ietf was deleted.

12.1(1)T

This command was integrated in Cisco IOS Release 12.1(1)T.

12.1(5)T

The gsmefr and gsmfr codec keywords were added.

12.2(8)T

The command was implemented on Cisco 1750 and Cisco 1751.

12.2(13)T3

The transparent keyword was added. This keyword is available only in js2 images.

12.4(4)T

The gsmefr and gsmfr codec keywords were removed.

12.4(15)XY

The g722-64 keyword was added.


Usage Guidelines

Use this command to define a specific voice coder rate of speech and payload size for a VoIP or VoFR dial peer. This command is also used for VoATM.

A specific codec type can be configured on the dial peer as long as it is supported by the setting used with the codec complexity voice-card configuration command. The codec complexity command is voice-card specific and platform specific. The codec complexity voice-card configuration command is set to either high or medium.

If the codec complexity command is set to high, the following keywords are available: g711alaw, g711ulaw, g722-64, g723ar53, g723ar63, g723r53, g723r63, g726r16, g726r24, g726r32, g728, g729r8, and g729br8.

If the codec complexity command is set to medium, the following keywords are available: g711alaw, g711ulaw, g726r16, g726r24, g726r32, g729r8, and g729br8.

The codec dial-peer configuration command is particularly useful when you must change to a small-bandwidth codec. Large-bandwidth codecs, such as G.711, do not fit in a small-bandwidth link. However, the g711alaw and g711ulaw codecs provide higher quality voice transmission than other codecs. The g729r8 codec provides near-toll quality with considerable bandwidth savings.

If codec values for the dial peers of a connection do not match, the call fails.

You can change the payload of each VoIP frame by using the bytes keyword; you can change the payload of each VoFR frame by using the bytes keyword with the payload-size argument. However, increasing the payload size can add processing delay for each voice packet.

Table 4 describes the voice payload options and default values for the codecs and packet voice protocols.

Table 4 Voice Payload-per-Frame Options and Defaults  

Codec
Protocol
Voice Payload Options (in Bytes)
Default Voice Payload (in Bytes)

g711alaw
g711ulaw

VoIP
VoFR
VoATM

80, 160
40 to 240 in multiples of 40
40 to 240 in multiples of 40

160
240
240

g722-64

VoIP

80, 160. 240

160

g723ar53
g723r53

VoIP
VoFR
VoATM

20 to 220 in multiples of 20
20 to 240 in multiples of 20
20 to 240 in multiples of 20

20
20
20

g723ar63
g723r63

VoIP
VoFR
VoATM

24 to 216 in multiples of 24
24 to 240 in multiples of 24
24 to 240 in multiples of 24

24
24
24

g726r16

VoIP
VoFR
VoATM

20 to 220 in multiples of 20
10 to 240 in multiples of 10
10 to 240 in multiples of 10

40
60
60

g726r24

VoIP
VoFR
VoATM

30 to 210 in multiples of 30
15 to 240 in multiples of 15
30 to 240 in multiples of 15

60
90
90

g726r32

VoIP
VoFR
VoATM

40 to 200 in multiples of 40
20 to 240 in multiples of 20
40 to 240 in multiples of 20

80
120
120

g728

VoIP
VoFR
VoATM

10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10

40
60
60

g729abr8
g729ar8
g729br8
g729r8

VoIP
VoFR
VoATM

10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10

20
30
30


For toll quality, use the g711alaw or g711ulaw keyword. These values provide high-quality voice transmission but use a significant amount of bandwidth. For nearly toll quality (and a significant savings in bandwidth), use the g729r8 keyword.


Note The clear-channel keyword is not supported on Cisco AS5300.



Note The G.723 and G.728 codecs are not supported on the 1700 platform for Cisco Hoot and Holler applications.



Note The transparent keyword affects only H.323 to H.323 connections.



Note The G.722-64 codec is only supported for H.323 and SIP.


Examples

The following example shows how to configure a voice coder rate that provides toll quality voice with a payload of 120 bytes per voice frame on a router that is acting as a terminating node. The sample configuration begins in global configuration mode and is for VoFR dial peer 200.

dial-peer voice 200 vofr
 codec g711ulaw bytes 240
 
   

The following example configures a voice coder rate for VoIP dial peer 10 that provides toll quality but uses a relatively high amount of bandwidth:

dial-peer voice 10 voip
 codec g711alaw
 
   

The following example configures the transparent codec used by the Cisco Unified Border Element:

dial-peer voice 1 voip
 incoming called-number .T
 destination-pattern .T
 session target ras
 codec transparent

Related Commands

Command
Description

codec (DSP interface dsp farm)

Specifies call density and codec complexity.

codec (voice port)

Specifies voice compression.

codec complexity

Specifies call density and codec complexity based on the codec used.

show dial peer voice

Displays the codec setting for dial peers.


codec (DSP Farm profile)

To specify the codecs supported by a digital signal processor (DSP) farm profile, use the codec command in DSP farm profile configuration mode. To remove the codec, use the no form of this command.

codec {codec-type | pass-through}

no codec {codec-type | pass-through}

Syntax Description

codec-type

Specifies the codec preferred.

g711alaw—G.711 a-law 64,000 bps.

g711ulaw—G.711 u-law 64,000 bps.

g722r-64G.722-64 at 64,000 bps

g729abr8—G.729 ANNEX A and B 8000 bps.

g729br8—G.729 ANNEX B 8,000 bps.

g729ar8—G.729 ANNEX A and R 8000 bps.

g729r8—G.729 8000 bps.

pass-through

Enables codec pass-through. Supported for transcoding and MTP profiles.


Command Default

Transcoding

g711alaw

g711ulaw

g729abr8

g729ar8

Conferencing

g711alaw

g711ulaw

g729abr8

g729ar8

g729br8

g729r8

MTP

g711ulaw

Command Modes

DSP farm profile configuration

Command History

Release
Modification

12.3(8)T

This command was introduced.

12.4(4)T

The pass-through keyword was added.

12.4(15)XY

The g722-64 keyword was added.


Usage Guidelines

Only one codec is supported for each Media Termination Point (MTP) profile. To support multiple codecs, you must define a separate MTP profile for each codec.

Hardware MTPs support only G.711 a-law and G.711 u-law. If you configure a profile as a hardware MTP, and you want to change the codec to other than G.711, you must first remove the hardware MTP by using the no maximum sessions hardware command.

The pass-through keyword is supported for transcoding and MTP profiles only; it is not supported for conferencing profiles. To support the RSVP agent on a SCCP device, you must use the codec pass-through command. In pass-through mode, the SCCP device processes the media stream using a pure software MTP regardless of the nature of the stream. This enables video and data streams to be processed in addition to audio. When pass-through mode is set in a transcoding profile, no transcoding is done for the session; the transcoding device performs a pure software MTP function. Pass-through mode can be used for Secure RTP sessions.

Examples

The following example shows the call density and codec complexity being set to GSMEFR:

Router(config)# dspfarm profile 123 transcode
Router(config-dspfarm-profile)# codec gsmefr

Related Commands

Command
Description

associate application

Associates the SCCP protocol to the DSP farm profile.

dspfarm profile

Enters DSP farm profile configuration mode and defines a profile for DSP farm services.

maximum sessions (DSP Farm profile)

Specifies the maximum number of sessions that are supported by the profile.

rsvp

Enables RSVP support on a transcoding or MTP device.

shutdown (DSP Farm profile)

Disables a DSP farm profile.


codec preference

To specify a list of preferred codecs to use on a dial peer, use the codec preference command in voice-class configuration mode. To disable this functionality, use the no form of this command.

codec preference value codec-type [mode frame_size][bytes payload-size] [packetization-period 20] [encap rfc3267] [frame-format {bandwidth-efficient | octet-aligned [crc | no-crc]}] [modes modes-value]

no codec preference value codec-type

Syntax Description

value

The order of preference, with 1 being the most preferred and 14 being the least preferred.

codec-type

The codec preferred. Values are as follows:

clear-channel—Clear Channel 64,000 bps

g711alaw—G.711 a law 64,000 bps

g711ulaw—G.711 mu-law 64,000 bps

g722r-64G.722-64 at 64,000 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 16,000 bps

g726r24—G.726 24,000 bps

g726r32—G.726 32,000 bps

g728—G.728 16,000 bps

g729abr8—G.729 ANNEX-A and B 8000 bps

g729br8—G.729 ANNEX-B 8000 bps

g729r8—G.729 8000 bps

gsmamr-nb—Enables GSMAMR codec capability

ilbcinternet Low Bitrate Codec (iLBC) at 13,330 bps or 15,200 bps.

transparent—Enables codec capabilities to be passed transparently between endpoints

Note The transparent keyword not supported when the call-start command is configured.

mode

(Optional) For iLBC codecs only. Specifies the iLBC operating frame mode that is encapsulated in each packet.

frame_size

(Optional) For iLBC codecs only. iLBC operating frame in milliseconds (ms). Valid entries are:

20—20ms frames for 15.2kbps bit rate

30—30ms frames for 13.33 kbps bit rate

Default is 20.

bytes

(Optional) Specifies that the size of the voice frame is in bytes.

payload-size

(Optional) Number of bytes you specify as the voice payload of each frame. Values depend on the codec type and the packet voice protocol.

packetization-period 20

(Optional) Sets the packetization period at 20 ms. Applicable only to GSMAMR-NB codec support.

encap rfc3267

(Optional) Sets the encapsulation value to comply with RFC 3267. Applicable only to GSMAMR-NB codec support.

frame-format

(Optional) Specifies a frame format. Supported values are octet-aligned and bandwidth-efficient. The default is octet-aligned. Applicable only to GSMAMR-NB codec support.

crc | no-crc

(Optional) CRC is applicable only for octet-aligned frame format. If you enter bandwidth-efficient frame format, the crc | no-crc options will not be available because they are inapplicable. Applicable only to GSMAMR-NB codec support.

modes modes-values

(Optional) Valid values are from 0 to 7. You can specify modes as a range (for example, 0-2), or individual modes separated by commas (for example, 2,4,6), or a combination of the two (for example, 0-2,4,6-7). Applicable only to GSMAMR-NB codec support.


Command Default

If the gsmamr-nb keyword is entered, the default values are as follows:

Packetization period is 20 ms.
Encap is rfc3267.
Frame format is octet-aligned.
CRC is no-crc.
Modes value is 0-7.

Command Modes

Voice-class configuration

Command History

Release
Modification

12.0(2)XH

This command was introduced on the Cisco AS5300.

12.0(7)T

This command was implemented on the Cisco 2600 series and Cisco 3600 series.

12.0(7)XK

This command was implemented on the Cisco MC3810.

12.1(2)T

This command was integrated into Cisco Release IOS Release 12.1(2)T.

12.1(5)T

The codecs gsmefr and gsmfr were added.

12.2(13)T3

The transparent keyword was added.

12.4(4)T

The codecs gsmefr and gsmfr were removed.

12.4(4)XC

This command was extended to include GSMAMR-NB codec parameters on the Cisco AS5350XM and Cisco AS5400XM platforms.

12.4(9)T

This command was integrated into Cisco IOS Release 12.4(9)T.

12.4(11)T

The ilbc codec and mode keyword were added.

12.4(15)XY

The g722r-64 keyword was added.


Usage Guidelines

The routers at opposite ends of the WAN may have to negotiate the codec selection for the network dial peers. The codec preference command specifies the order of preference for selecting a negotiated codec for the connection. Table 5 describes the voice payload options and default values for the codecs and packet voice protocols.


Note The transparent keyword not supported when the call start command is configured.


Table 5 Voice Payload-per-Frame Options and Defaults

Codec
Protocol
Voice Payload Options (in Bytes)
Default Voice Payload (in Bytes)
g711alaw
g711ulaw

VoIP
VoFR
VoATM

80, 160
40 to 240 in multiples of 40
40 to 240 in multiples of 40

160
240
240

g722r-64

VoIP

80, 160, 240

160

g723ar53
g723r53

VoIP
VoFR
VoATM

20 to 220 in multiples of 20
20 to 240 in multiples of 20
20 to 240 in multiples of 20

20
20
20

g723ar63
g723r63

VoIP
VoFR
VoATM

24 to 216 in multiples of 24
24 to 240 in multiples of 24
24 to 240 in multiples of 24

24
24
24

g726r16

VoIP
VoFR
VoATM

20 to 220 in multiples of 20
10 to 240 in multiples of 10
10 to 240 in multiples of 10

40
60
60

g726r24

VoIP
VoFR
VoATM

30 to 210 in multiples of 30
15 to 240 in multiples of 15
30 to 240 in multiples of 15

60
90
90

g726r32

VoIP
VoFR
VoATM

40 to 200 in multiples of 40
20 to 240 in multiples of 20
40 to 240 in multiples of 20

80
120
120

g728

VoIP
VoFR
VoATM

10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10

40
60
60

g729abr8
g729ar8
g729br8
g729r8

VoIP
VoFR
VoATM

10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10

20
30
30

ilbc

VoIP

For mode 20, 38, 76, 114, 152, 190, 228.

For mode 30, 50, 100, 150, 200

38

50


Examples

The following example sets the codec preference to the GSMAMR-NB codec and specifies parameters:

Router(config-voice-class)# codec preference 1 gsmamr-nb packetization-period 20 encap 
rfc3267 frame-format octet-aligned crc

The following example creates codec preference list 99 and applies it to dial peer 1919:

voice class codec 99
codec preference 1 g711alaw 
codec preference 2 g711ulaw bytes 80 
codec preference 3 g723ar53 
codec preference 4 g723ar63 bytes 144 
codec preference 5 g723r53 
codec preference 6 g723r63 bytes 120 
codec preference 7 g726r16 
codec preference 8 g726r24 
codec preference 9 g726r32 bytes 80 
codec preference 10 g729br8 
codec preference 11 g729r8 bytes 50
codec preference 12 gsmefr
end 
dial-peer voice 1919 voip 
 voice-class codec 99
 
   

The following example configures the transparent codec used by the Cisco Unified Border Element:

voice class codec 99
codec preference 1 transparent 
 
   
codec preference 1 transparent
 
   

Note You can only assign a preference value of 1 to the transparent codec. Additional codecs assigned to other preference values are ignored if the transparent codec is used.


The following example shows how to configure the iLBC codec used by the Cisco Unified Border Element:

voice class codec 99
codec preference 1 ilbc 30 200
 
   

Related Commands

Command
Description

call-start

Forces an H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer.

voice class codec

Enters voice-class configuration mode and assigns an identification tag number to a codec voice class.

voice-class codec (dial peer)

Assigns a previously configured codec selection preference list to a dial peer.


Feature Information for G.722-64 and iLBC Codec Support on Cisco UBEs, DSP Farms, and Voice Gateways

Table 6 lists the release history for this feature.

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

Cisco IOS software images are specific to a Cisco IOS software release, a feature set, and a platform. Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.


Note Table 6 lists only the Cisco IOS software release that introduced support for a given feature in a given Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS software release train also support that feature.


Table 6 Feature Information for G.722-64 and iLBC Codec Support on Cisco UBEs, DSP Farms, and Voice Gateways

Feature Name
Releases
Feature Information

iLBC Codec Support

12.4(11)T

The internet Low Bitrate Codec (iLBC) is a standard, high-complexity speech codec that is suitable for robust voice communication over IP. iLBC has built-in error correction functionality that helps the codec perform in networks with a high-packet loss.

G.722-64 and iLBC Codec Support on Cisco UBEs, DSP Farms, and Voice Gateways

12.4(15)XY

The G.722-64 and iLBC codecs are supported for Cisco UBEs, DSP farms, and voice gateways. Conferencing and universal transcoding are supported on both codecs.