Aastra Telecom 57I CT, 55I, 53I manual Default Codec Settings, Customized Codec Preference List

IP Phone Administrator Guide Global SIP Settings

Configuring the IP Phones

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.

CODEC	Bit Rate	Algorithm	Packetizatio	Silence
			n Time	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	CS-ACELP	30 ms	enabled
	Kb/s

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.

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.

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).

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