
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 121, 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
• 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.
This is how the jitter buffer works:
Assume a jitter buffer setting of five frames.
• The IP telephone firmware places the first five arriving frames in the jitter buffer.
• When frame six arrives, the IP telephone firmware places it in the buffer, and sends frame one
to the handset speaker.
• When frame seven arrives, the IP telephone buffers it, and sends frame two to the handset
speaker.
The net effect of using a jitter buffer is that the arriving packets are delayed slightly in order to
ensure a constant rate of arriving frames at the handset speaker.
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