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26 Chapter 1 Introduction

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Codecs

The algorithm used to compress and decompress voice is embedded in a software entity called a

codec (COde-DECode).

Two popular Codecs are G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per

second (kbps) while G.729 samples at a far lower rate of 8 kbps. For actual bandwidth

requirements, refer to “Determining the bandwidth requirements” on page 145, where you will

note that the actual kbps requirements are slightly higher than label suggests.

Voice quality is better when using a G.711 CODEC, but more network bandwidth is used to

exchange the voice frames between the telephones.

If you experience poor voice quality, and suspect it is due to heavy network traffic, you can get

better voice quality by configuring the IP telephone to use a G.729 CODEC.

The Business Communications Manager supports these codecs:

• G.729

• G.723

• G.729 with VAD (Voice Activity Detection)

• G.723 with VAD

• G.711-uLaw

• G.711-aLaw

Jitter Buffer

Voice frames are transmitted at a fixed rate, because the time interval between frames is constant.

If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many

cases, however, some frames can arrive slightly faster or slower than the other frames. This is

called jitter, and degrades the perceived voice quality. To minimize this problem, configure the IP

telephone with a jitter buffer for arriving frames.

Note: You can only change the codec on a configured IP telephone if it is online to the

Business Communications Manager, or if Keep DN Alive is enabled for an offline

telephone.

Note: You can only change the jitter buffer on a configured IP telephone if it is online to

the Business Communications Manager, or if Keep DN Alive is enabled for an offline

telephone.

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Avaya IP Telephony Guía de configuración Pagina 26