Codecs

Description

Codecs are used to convert an analog voice signal to digitally encoded version. Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc

List of codecs supported by MOR can be found here.

Per Call Bandwidth requirements for various codecs

Codec & Bit Rate (Kbps)	Codec Sample Size (Bytes)	Codec Sample Interval (ms)	Voice Payload Size (Bytes)	Voice Payload Size (ms)	Bandwidth Ethernet (Kbps)
G.711 (64 Kbps)	80 Bytes	10ms	160 bytes	20ms	87.2Kbps
G.729 (8 Kbps)	10 Bytes	10ms	20 bytes	20ms	31.2Kbps
G.723.1 (6.3 Kbps)	24 Bytes	30ms	24 bytes	30ms	21.9Kbps
G.723.1 (5.3 Kbps)	20 Bytes	30ms	20 bytes	30ms	20.8Kbps
G.726 (32 Kbps)	20 Bytes	5ms	80 bytes	20ms	55.2Kbps
G.726 (24 Kbps)	15 Bytes	5ms	60 bytes	20ms	47.2Kbps
G.728 (16 Kbps)	10 Bytes	5ms	60 bytes	30ms	31.5Kbps
G722_64k(64 Kbps)	80 Bytes	10ms	160 bytes	20ms	87.2Kbps
ilbc_mode_20(15.2Kbps)	38 Bytes	20ms	38 bytes	20ms	38.4Kbps
ilbc_mode_30(13.33Kbps)	50 Bytes	30ms	50 bytes	30ms	28.8Kbps
speex(24.6 Kbps)	66 Bytes	20ms	56 bytes	25ms	42.4 Kbps
speex(18.2 Kbps)	46 Bytes	20ms	56 bytes	20ms	42.4 Kbps

Transcoding

Transcoding is conversion from one codec to another. It is performed by Asterisk when LegA and LegB (user's and provider's) codecs do not match. Transcoding uses a lot of system resources, so it should be avoided to get better server performance.

For more information please visit

• A more detailed guide about bandwidth calculations ant related metrics
• Also here
• A simple bandwidth calculator can be found here
• Speex codec install

See also

• G729r8
• Which codecs should I choose for devices