CODEC is an abbreviation of Compress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with a minimum number of bits while retaining the quality. This can effectively reduce the frame size and the bandwidth required for audio transmission.

The audio codec that the phone uses to establish a call should be supported by the SIP server. When placing a call, the phone will offer the enabled audio codec list to the server and then use the audio codec negotiated with the called party according to the priority.

The following table lists the parameters you can use to configure the audio codecs.

Codec Bit Rate Sample Rate Packetization Time
G.711A 64 Kbps 8 Ksps 20ms
G.711U 64 Kbps 8 Ksps 20ms
G.722 64 Kbps 16 Ksps 20ms
G.729 8 Kbps 8 Kbps 20ms
G.723 5.3 Kbps
6.3 Kbps
8 Kbps 30ms
G726-32 32 Kbps 8 Kbps 20ms
OPUS 8-12 Kbps
28-40 Kbps
8 Ksps
16 Ksps
20ms
iLBC 15.2 Kbps 8 Ksps 20ms
iLBC 13.33 Kbps 8 Ksps 30ms
Parameter DBID_SIP_AUDIO_TYPEY
Description It configures the priority of the enabled audio codec.
Permitted Values 0-G.711A
1-G.711U
2-G.722
3-G.729
4-G.723
5-G726-32
6-OPUS
7-iLBC
8-NONE
Default Y=1-7
DBID_SIP_AUDIO_TYPE1 = 1
DBID_SIP_AUDIO_TYPE2 = 0
DBID_SIP_AUDIO_TYPE3 = 3
DBID_SIP_AUDIO_TYPE4 = 2
DBID_SIP_AUDIO_TYPE5 = 6
DBID_SIP_AUDIO_TYPE6 = 7
DBID_SIP_AUDIO_TYPE7 = 4
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Audio Codec Type Y
Parameter DBID_G726_PAYLOAD
Description
Permitted Values Integer
Default 2
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > G726 Payload
Parameter DBID_G723_BITRATE
Description The G.723 operates at both 6.3kbps and 5.3kbps. The high rate algorithm (6.3kbps) has higher reconstructed speech quality and the low rate algorithm (5.3kbps) has lower computational complexity.
Permitted Values 0-5.3kbps
1-6.3kbps
Default 0
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > G.723 Coding Speed

PTime is a measurement of the duration (in milliseconds) that how long the audio data in each RTP packet is sent to the destination, and defines how much the 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.

Parameter DBID_PACKET_CYCLE
Description It configures the ptime (in milliseconds) for the codec.
Permitted Values 0-10
1-20
2-30
3-40
4-50
5-60
Default 1
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Packet Cycle (ms)
Parameter DBID_ECHO_CANCEL_MGT
Description If enable it, it can effectively reduce the problem of echo during the call.
Permitted Values 0-Disable
1-Enable
Default 1
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Echo Cancel
Parameter DBID_AUTO_GAIN_CONTROL
Description If it is enabled, the phone automatically adjusts the sound volume during calls based on surrounding sounds.
Permitted Values 0-Disable
1-Enable
Default 0
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Auto Gain Control
Parameter DBID_SIP_ONLY_RSP_ONE_CODEC
Description It enable or disable to negotiate with the first speech code in the returned 200OK.
Permitted Values 0-Disable
1-Enable
Default 0
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Use First Matching Vocoder in 200OK SDP
Parameter DBID_SIP_CODEC_PRIORITY
Description It configures the negotiation way of audio codec .
Note: If it is set to 0, the IP phone will negotiate the remote voice codec. If it is set to 1, the IP phone will negotiate the voice codec of the IP phone.
Permitted Values 0-Remote
1-Local
Default 0
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Codec Priority
Parameter DBID_PTIME_FOLLOW_REMOTE_SDP
Description It enable or disable the remote cycle as the packaging cycle of the speech code.
Permitted Values 0-Disable
1-Enable
Default 0
Web UI SIP Account > Line X [1] > Audio Configuration > Codec Setup > Packet Cycle Follows Remote SDP

[1] X is the account ID. X = 1-4

Author:admin  Create time:2023-08-30 17:58
Last editor:admin  Update time:2024-07-29 16:00