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Configuring Audio Features
477
The following table summarizes the supported audio codecs on IP phones:
Codec Algorithm Reference Bit Rate Sample
Rate
Packetization
Time
G722 G.722 RFC 3551 64 Kbps 16 Ksps
20ms
PCMA G.711
a-law
RFC 3551 64 Kbps 8 Ksps 20ms
PCMU G.711
u-law
RFC 3551 64 Kbps 8 Ksps 20ms
G729 G.729 RFC 3551 8 Kbps 8 Ksps 20ms
G726-16 G.726 RFC 3551 16 Kbps 8 Ksps
20ms
G726-24 G.726 RFC 3551 24 Kbps 8 Ksps
20ms
G726-32 G.726 RFC 3551 32 Kbps 8 Ksps
20ms
G726-40 G.726 RFC 3551 40 Kbps 8 Ksps
20ms
G723_53/
G723_63
G.723.1 RFC 3951
5.3kbps
6.3kbps
8 Ksps 30ms
iLBC iLBC RFC 3952
13.33 Kbps
15.2 Kbps
8 Ksps
20ms
30ms
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 Priority RTPmap
G722
Configuration Files
Web User Interface
1 9
PCMU
Configuration Files
Web User Interface
2 0
PCMA
Configuration Files
Web User Interface
3 8
G729 Configuration Files 4 18