Audio Codecs On - Yealink SIP-T2 SERIES Administrator's Manual

Hide thumbs Also See for SIP-T2 SERIES:
Table of Contents

Advertisement

Administrator's Guide for SIP-T2 Series/T19(P) E2/T4 Series/CP860 IP Phones
The following table summarizes the supported audio codecs on IP phones:
Codec
G.722.1c
G.722.1c
G.722.1c
G.722.1
G722
PCMA
PCMU
G729
G726-16
G726-24
G726-32
G726-40
G723/
G723_53/
G723_63
iLBC
Opus
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.
842
Algorithm
Reference
RFC 5577
G.722.1
RFC 5577
RFC 5577
G.722.1
RFC 5577
G.722
RFC 3551
G.711
RFC 3551
a-law
G.711
RFC 3551
u-law
G.729
RFC 3551
G.726
RFC 3551
G.726
RFC 3551
G.726
RFC 3551
G.726
RFC 3551
G.723.1
RFC 3551
iLBC
RFC 3952
Opus
RFC 6716
Sample
Bit Rate
Rate
48 Kbps
32 Ksps
32 Kbps
32 Ksps
24 Kbps
32 Ksps
24 Kbps
16 Ksps
64 Kbps
16 Ksps
64 Kbps
8 Ksps
64 Kbps
8 Ksps
8 Kbps
8 Ksps
16 Kbps
8 Ksps
24 Kbps
8 Ksps
32 Kbps
8 Ksps
40 Kbps
8 Ksps
5.3 Kbps
8 Ksps
6.3 Kbps
15.2 Kbps
8 Ksps
13.33 Kbps
16 Kbps
16 Ksps
20 Kbps
Packetization
Time
20ms
20ms
20ms
20ms
20ms
20ms
20ms
20ms
20ms
20ms
20ms
20ms
30ms
20ms
30ms
20ms

Advertisement

Table of Contents
loading

This manual is also suitable for:

Sip-t19p e2/t4 seriesSip-t19 e2/t4 seriesCp860 series

Table of Contents