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Global SIP Settings
41-001160-01, Rev 00, Releaes 2.3 4-91
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.
Customized Codec Preference List
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).
CODEC Bit Rate Algorithm
Packetizatio
n Time
Silence
Suppression
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
Note: 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.