Aastra Telecom 9143i Series User Manual

41-001160-03, Rev 00, Releaes 2.4

4-95

Configuring Network and Session Initiation Protocol (SIP) Features

All Codecs have a sampling rate of 8,000 samples per second, and operate and 
operate in the 300 Hz to 3,700 Hz audio range. The following table lists the 
default settings for bit rate, algorithm, packetization time, and silence suppression 
for each Codec, based on a minimum packet size.

Default Codec Settings.

You can enable the IP phones to use a default "basic codec" set, which consists of 
the set of codecs and packet sizes shown above. 

Or you can instead configure a custom set of codecs and attributes instead of using 
the defaults.

You can also configure the IP phones to use preferred Codecs. To do this, you 
must enter the payload value (payload), the packetization time in milliseconds 
(ptime), and enable or disable silence suppression (silsupp).

Payload is the codec type to be used. This represents the data format carried 
within the RTP packets to the end user at the destination. You can enter payload 
values for G.711 a-law, G.711 u-law, and G.729a.

Ptime (packetization time) is a measurement of the duration of PCM data within 
each RTP packet sent to the destination, and hence defines how much network 
bandwidth is used for transfer of the RTP stream. You enter the ptime values for 
the customized Codec list in milliseconds. (See table below).

G.711 a-law

64 Kb/s

PCM

30 ms

enabled

G.711 u-law

64 Kb/s

PCM

30 ms

enabled

G.729a

8 
Kb/s

CS-ACELP

30 ms

enabled

The basic and custom codec parameters apply to all calls, and are 

configured on a global-basis only using the configuration files or the 
Aastra Web UI.

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