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9-7

Cisco IP Telephony Network Design Guide

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Chapter 9 Catalyst DSP Provisioning

Catalyst MTP Transcoding Services

Introducing the WAN into an IP telephony implementation forces the issue of

voice compression. In the previous designs shown in this document, all

campus-oriented voice was uncompressed (G.711) to provide the highest quality

while incurring the fewest complications. Once a WAN-enabled network is

implemented, voice compression between sites is the recommended design

choice. This calls into question how WAN users use the conferencing services or

IP-enabled applications, which support only G.711 voice connections. The

solution is to use hardware-based MTP transcoding services to convert the

compressed voice streams into G.711.

MTP Transcoding Design Details

The following points summarize the design capabilities and requirements of the

MTP transcoding:

• Provision MTP transcoding resources appropriately for the number of IP

WAN callers to G.711 endpoints.

• The Catalyst 4000 WS-X4604-GWY module supports 16 transcoding

sessions per module.

• The Catalyst 6000 WS-X6608-T1 or WS-X6608-E1 modules support

31 G.723 or G.711 transcoding sessions per physical port (248 per module)

or 24 G.729 to G.711 transcoding sessions per physical port (192 per

module).

• Transcoding is supported only in low bit rate to high bit rate (G.729a or

G.723.1 to G.711), or vice versa, configurations.

• Each Cisco CallManager must have its own MTP transcoding resources.

• Each transcode has its own jitter buffer of 20-40 ms.

IP-to-IP Packet Transcoding and Voice Compression

Voice compression between IP phones is easily configured through the use of

regions and locations in Cisco CallManager. However, the Catalyst conferencing

services and some applications currently support only G.711, or uncompressed,

connections. For these situations, MTP transcoding or packet-to-packet gateway

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Cisco WS-X4604-GWY - VoIP Gateway Especificaciones Pagina 193