Ip Audio Settings Window - Avaya T3 IP Service Manual

2 Service operation
2.10.5

IP audio settings window

In this window you can change settings for the digital voice packets.
At the receiving end this is the jitter buffer. The buffer size determines how long voice pack-
ets are collected for before being processed for voice output. The size of the individual
transmitted voice packets and the type of encoding is specified by the Codec and Delay
settings. You can specify which Codec should be used and with which priority.
Note:
With the WebTerminal, only RTP voice packages can be prioritised with the TOS / DiffServ
Bits setting. This does not affect signalling data. The settings for prioritisation of voice pak-
kages can be managed under TOS byte settings in the IP phone's audit menu (→ S. 135) or
the IP admin tool.
Jitter buffer size
Codec
4.999.099.262 - 07/07
short Short delay, little compensation for packet losses
normal Default
long Long delay, better compensation for packet losses
auto Connection–dependent automatic setting
Do not change
G711A 64 kbit voice resolution by A–law encoding
G729A 6.4 kbit voice resolution
none None No codec prioritized.
The combination fields of the lower priorities are barred.
2.10 The web terminal
Page 99