Avaya IP Telephony Guía de configuración Pagina 43

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Chapter 3 Installing IP telephones 43
IP Telephony Configuration Guide
Choosing a codec
The default codec is used when an IP client has not been configured to use a preferred Codec.
Refer to the next section for individual IP client Codec settings. If the default Codec is set to
AUTO, the Business Communications Manager will choose the appropriate CODEC when an IP
client makes a call. For example, if both endpoints of the call are IP telephones on the same subnet,
the Business Communications Manager chooses G.711 for maximum voice quality. If the
telephones are on different subnets, the Business Communications Manager will choose G.729 to
minimize network bandwidth consumption by voice data packets.
For IP telephones, the Business Communications Manager supports both a-law and mu-law
variants of the G.711 CODEC, as well as the G.729 and G.723 CODECS.
The G.711 CODEC samples the voice stream at a rate of 64Kbps (Kilo bits per second), and is
the CODEC to use for maximum voice quality. Choose the G.711 CODEC with the
companding law (alaw or ulaw) that matches your system requirements.
The G.729 CODEC samples the voice stream at 8Kbps. The voice quality is slightly lower
using a G.729 but it reduces network traffic by approximately 80%.
The G.723 CODEC should be used only with third party devices that do not support G.729 or
G.711.
Codecs with VAD (Voice Activity Detection) make VAD active on the system, which
performs the same function as having silence suppression active.
Choosing a Jitter Buffer
A jitter buffer is used to prevent the jitter associated with arriving (Rx) voice packets at the IP
telephones. The jitter is caused by packets arriving out of order due to having used different
network paths, and varying arrival rates of consecutive voice packets.The greater the size of the
jitter buffer, the better sounding the received voice appears to be. However, voice latency (delay)
also increases. Latency is very problematic for telephone calls, as it increases the time between
when one user speaks and when the user at the other end hears the voice.
Note: If the IP telephones are using VoIP trunks for the call, the codec set for the trunks
overrides the telephone settings.
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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