Yealink SIP-T27P [668/841] Packetization time
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Yealink SIP-T27P [668/841] Packetization time

Yealink SIP-T27P [668/841] Packetization time
Administrator’s Guide for SIP-T2 Series/T19(P) E2/T4 Series IP Phones
652
SIP-T27P/T23P/T23G/
T21(P) E2
G722, PCMA, PCMU, G729,
G726-16, G726-24, G726-32,
G726-40, iLBC
G722, PCMA, PCMU,
G729
The following table summarizes the supported audio codecs on IP phones:
Codec
Algorithm
Reference
Bit Rate
Sample
Rate
G722
G.722
RFC 3551
64 Kbps
16 Ksps
PCMA
G.711 a-law
RFC 3551
64 Kbps
8 Ksps
PCMU
G.711 u-law
RFC 3551
64 Kbps
8 Ksps
G729
G.729
RFC 3551
8 Kbps
8 Ksps
G726-16
G.726
RFC 3551
16 Kbps
8 Ksps
G726-24
G.726
RFC 3551
24 Kbps
8 Ksps
G726-32
G.726
RFC 3551
32 Kbps
8 Ksps
G726-40
G.726
RFC 3551
40 Kbps
8 Ksps
G723_53/
G723_63
G.723.1
RFC 3951
5.3kbps
6.3kbps
8 Ksps
iLBC
iLBC
RFC 3952
13.33 Kbps
15.2 Kbps
8 Ksps
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

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