Aastra Telecom 41-001343-02 Basic Codecs G.711 u-Law, G.711 a-Law, G.729, Default Codec Settings

Global SIP Settings

For example, if the RTP base port value is 5000, the first voice patch sends RTP on port 5000 and RTCP on port 5001. Addi- tional calls would then use ports 5002, 5003, etc.

You can configure the RTP port on a global-basis only, using the configuration files, the IP Phone UI, or the Aastra Web UI.

Basic Codecs (G.711 u-Law, G.711 a-Law, G.729)

CODEC is an acronym for COmpress-DECompress. It consists of a set of instructions that together implement one or more algorithms. In the case of IP telephony, these algorithms are used to compress the sampled speech data, to decrease the content's file size and bit-rate (the amount of network bandwidth in kilobits per second) required to transfer the audio.

With smaller file sizes and lower bit rates, the network equipment can store and stream digital media content over a net- work more easily.

Aastra IP phones support the International Telecommunications Union (ITU) transmission standards for the following CODECs:

•Waveform CODECs: G.711 pulse code modulation (PCM) with a-Law or u-Law companding

•Parametric CODEC: G.729a conjugate structure - algebraic code excited linear prediction (CS_ACELP)

All codecs have a sampling rate of 8,000 samples per second, and operate in the 300 Hz to 3,700 Hz audio range. The fol- lowing 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	Packetization 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

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 configure a custom set of codecs and attributes instead of using the defaults (see “Customized Codec Preference List” below).

Note:

The basic and custom codec parameters apply to all calls, and are configured on a global-basis only using the config- uration files or the Aastra Web UI.

AMR and AMR-WB (G722.2) Codecs (Licensed Feature for 6735i , 6737i, 6863i, 6865i, and 6867i Only)

Administrators can configure Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR -WB) codecs on the 6735i, 6737i, 6863i, 6865i, and 6867i IP phones. AMR/AMR-WB codecs provide improved speech quality during calls due to wider speech bandwidth, and cover both real-time transfers through Real-time Transport Protocol (RTP) and non-real-time transfers through stored files. AMR supports eight narrowband speech encoding modes (0-7) with bit-rates ranging from

4.75to 12.2 kilobits per second (kbps). AMR-WB supports nine wideband speech encoding modes (0-8), with bit-rates ranging from 6.60 to 23.85 kbps.

Note:

AMR/AMR-WB codecs is a licensed feature on the SIP IP phones. To confirm that the license is active, Administrators can view the license through the phone's Web UI on the Licensing Status page. AMR/AMR-WB should be listed if the fea- ture is available to be used. If Administrators configure AMR/AMR-WB when there is no license, the codec will be ignored and not negotiated.

Administrators can configure AMR/AMR-WB on the IP phones in the Customized Codec Preference List on the Web UI or in the configuration files using the existing “sip customized codec” parameter.

41-001343-02 REV04 – 05.2014 4-75