Rtp port, Basic codecs (g.711 u-law, g.711 a-law, g.729) – AASTRA 6700i series, 9143, 9480i, 9480i CT SIP Administrator Guide EN User Manual

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41-001343-01 Rev 03, Release 3.2.2

4-91

RTP Port

RTP is described in RFC1889. The UDP port used for RTP streams is traditionally an

even-numbered port, and the RTCP control is on the next port up. A phone call therefore uses one

pair of ports for each media stream.

The RTP port is assigned to the first line on the phone, and is then incremented for each

subsequent line available within the phone to provided each line a unique RTP port for its own use.

On the IP phone, the initial port used as the starting point for RTP/RTCP port allocation can be

configured using "RTP Port Base". The default RTP base port on the IP phones is 3000.

For example, if the RTP base port value is 5000, the first voice patch sends RTP on port 5000 and

RTCP on port 5001. Additional 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

network 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 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

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

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