Yealink SIP-T38G

Configuring Audio Features
173
This chapter provides information for making configuration changes for the following
audio features:
 Audio Codecs
 Acoustic Clarity Technology
CODEC is an abbreviation of COmpress-DECompress. It is capable of coding or
decoding a digital data stream or signal by implementing an algorithm. The object of
the algorithm is to represent the high-fidelity audio signal with minimum number of bits
while retaining the quality. This can effectively reduce the frame size and the bandwidth
required for transmission of the audio.
The default codecs used on the IP phone are summarized in the following table:
Codec
Algorithm
Bit Rate
Sample Rate
Packetization Time
PCMA
G.711 a-law
64 Kbps
8 Ksps
20ms
PCMU
G.711 u-law
64 Kbps
8 Ksps
20ms
G729
G.729
8 Kbps
8 Ksps
20ms
G722
G.722
64 Kbps
16 Ksps
20ms
In addition to the codecs introduced above, the IP phone also supports the codecs:
G723_53, G723_63, G726_16, G726_24, G726_32, G726_40, iLBC_13_3 and iLBC_15_2.
You can configure the preferred codecs to use on a per-account basis instead of using
the default codecs. You can also configure the priorities for the enabled codecs. The
attribute ―rtpmap‖ is used to define a mapping from RTP payload codes to a codec,
clock rate and other encoding parameters.
The corresponding attributes of the codec are listed as follows:
Codec
Configuration Method
Priority
RTPmap
PCMU
Configuration Files
Web User Interface
1
0
PCMA
Configuration Files
Web User Interface
2
8
G729
Configuration Files
Web User Interface
3
18