Yealink SIP-T2 Series User Manual

Configuring Audio Features
803
The Opus codec supports various audio bandwidths, defined as follows:
Abbreviation
Audio Bandwidth
Sample Rate (Effective)
NB (narrowband)
4 kHz
8 kHz
MB (medium-band)
6 kHz
12 kHz
WB (wideband)
8 kHz
16 kHz
SWB (super-wideband)
12 kHz
24 kHz
FB (fullband)
20 kHz
48 kHz
Note
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 3551
5.3 Kbps
6.3 Kbps
8 Ksps
30ms
iLBC
iLBC
RFC 3952
15.2 Kbps
13.33 Kbps
8 Ksps
20ms
30ms
opus
opus
RFC 6716
16 Kbps
20 Kbps
8 Ksps
16 Ksps
20ms
Note
The network bandwidth necessary to send the encoded audio is typically 5~10% higher than the
bit rate due to packetization overhead. For example, a two-way G.722 audio call at 64 Kbps
consumes about 135 Kbps of network bandwidth.
The SIP-T40P/T40G/T23P/T23G/T21(P) E2/T19(P) E2 IP phones support using WB for opus
encoding; and using NB, MB, WB, SWB and FB for opus decoding.

Yealink SIP-T2 Series Specifications

General

Brand	Yealink
Model	SIP-T2 Series
Category	IP Phone
Language	English