Nortel BCM50 2.0 Installation Manual page 47

The information under the following headings describes some of the components that determine
voice quality for IP telephones and trunks:
• "Codecs" on page 47
• "Jitter buffer" on page 47
• "QoS routing" on page 48
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 (kb/s) while G.729 samples at a far lower rate of 8 kb/s.
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.
Note: You can only change the codec on a configured IP telephone if it is online
to the BCM50 2.0, or if Keep DN Alive is enabled for an offline telephone.
The BCM50 2.0 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 jitter buffer on a configured IP telephone if it is
online to the BCM50 2.0, or if Keep DN Alive is enabled for an offline telephone.
Chapter 6 IP telephone overview
Telephony Device Installation Guide
47