Yealink SIP-T2XP Administrator's Manual page 461

The following table summarizes the supported audio codecs on IP phones:
Codec Algorithm
G722 G.722
PCMA G.711
a-law
PCMU G.711
u-law
G729 G.729
G726-16 G.726
G726-24 G.726
G726-32 G.726
G726-40 G.726
G723_53/
G.723.1
G723_63
iLBC iLBC
Packetization Time
Ptime (Packetization Time) is a measurement of the duration (in milliseconds) of the
audio data in each RTP packet sent to the destination, and defines how much network
bandwidth is used for the RTP stream transfer. Before establishing a conversation, codec
and ptime are negotiated through SIP signaling. The valid values of ptime range from
10 to 60, in increments of 10 milliseconds. The default ptime is 20ms. You can also
disable the ptime negotiation.
Codecs and priorities of these codecs are configurable on a per-line basis. 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 Methods
Configuration Files
G722
Web User Interface
Configuration Files
PCMU
Web User Interface
Configuration Files
PCMA
Web User Interface
G729 Configuration Files
Reference Bit Rate
RFC 3551 64 Kbps
RFC 3551 64 Kbps
RFC 3551 64 Kbps
RFC 3551 8 Kbps
RFC 3551 16 Kbps
RFC 3551 24 Kbps
RFC 3551 32 Kbps
RFC 3551 40 Kbps
5.3kbps
RFC 3951
6.3kbps
13.33 Kbps
RFC 3952
15.2 Kbps
Priority
1
2
3
4
Configuring Audio Features
Sample Packetization
Rate Time
20ms
16 Ksps
8 Ksps 20ms
20ms
8 Ksps
20ms
8 Ksps
20ms
8 Ksps
20ms
8 Ksps
20ms
8 Ksps
20ms
8 Ksps
8 Ksps 30ms
20ms
8 Ksps
30ms
RTPmap
9
0
8
18
443