Information on Audio Codecs
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.
Each service, program, phone, gateway, etc typically supports several different codecs, and when talking to each other, negotiate which codec they will use.
X-lite has various audio codecs implemented. The implemented audio codecs of the X-lite 3.0 version are DVI4, G.711 uLaw/aLaw, GSM, iLBC as (narrow band audio codecs) and BV-32, BV FEC, DVI4, L16 PCM (wide band audio codecs). Eyebeam version 1.5 has a wider range of codecs available.
G.711A & G.711U
G.711 is a codec primarily used in telephony, released in 1972. It samples the analogue voice input signal using pulse-code modulation (PCM) at a sampling rate of 8000 samples/second.
The G.711 standard has two algorithms: the A-law (mostly used in Europe) and the U-law algorithm (used in North America and Japan).
The G.711 codec encodes the pulse-code modulation samples (sampling rate of 8000 samples/second) to logarithmic 8-bit samples. This will lead to a (8000 times 8-bit=) 64 kbit/s bitstream. Note that this is the datastream without information send in the headers of the packages, so the actual datastream will be higher. Because the sampling rate of 8000 samples/second and the encoding to 8-bit samples are intrinsic properties of the codec, the bitstream can not exceed 64 kbit/s. If one wants to increase the quality of the voice by increasing the used bandwidth (> 64 kbit/s) one has to use another codec.
Various codecs compared
The following sites were used to build the following comparison:
|Codec||Compression method||Transfer rate||NEB||Frame size|
|G.711||Pulse Code Modulation (PCM)||64 Kbps||87.2 Kbps||20 ms|
|G.726||Adaptive Differential Pulse Code Modulation (ADPCM)||16-40 Kbps||55.2 Kbps|
|G.728||Low Delay Code Excited Linear Prediction (LD-CELP)||16 Kbps||31.5 Kbps||0,625ms|
|G.729/G.729A||Conjugate Structure Algebraic Code Excited Linear Prediction (CD-ACELP)||8 Kbps||31.2 Kbps||10 ms|
|G.723.1||Mulptiple Maximum Likehood Quantization (MPMLQ)||6.3 Kbit/s||21.9 Kbps||37.5 ms|
|G.723||ACELP||5,3 Kbps||20.8 Kbps|
|GSM||Global System for Mobile communications||13 Kbps||20ms|
|Speex Narrowband (8kHz)||Patent-free audio compression||2.15-24.6 Kbps||20ms|
|Speex Wideband (16kHz)||Patent-free audio compression||4-44.2 Kbps||20ms|
|iLBC||Internet Low Bandwidth Codec (free)||13,3 Kbps||30ms|
|iLBC||Internet Low Bandwidth Codec (free)||15,2 Kbps||27.7 Kbps||20ms|
|L16 PCM||PCM||128 Kbps|
|EVCR||Enhanced Variable Rate Codec||8 Kbps|
NEB = Nominal Ethernet Bandwidth (one direction)
- G711u/a - Requires 64+Kbps (inbound and outbound) continuous data stream for a quality call, plus the SIP signal. A 128Kbps connection has shown to be the minimal connection needed in our testing.
- G723 - Requires 6.4+Kbps (I and O) continuous data stream and also needs room for the SIP signal, 28.8Kbps connection has shown to be the minimal connection.
- From the maximum voice quality point of view, when enough bandwidth is available, the best quality can be obtained by using the L16 PCM codec. This L16 PCM codec is the same as the G.711 codec, however the L16 PCM codec samples to 16 bits samples instead of the 8 bits samples in the case of the G.711 codec.
- Asterisk will resample all codecs internally to 8000 samples/sec. This means using for example the Speex Wide-band (16 kHz sample-rate) or even Ultra-wide-band (32 kHz sample-rate) with Asterisk is useless.
- Codecs with a frame-size other than 20 ms, appear to work bad with Asterisk. This may have to do with the IAX-protocol.
Audio codec translations
When Asterisk can choose between different codecs (prerecorded samples) and also when conversions between codecs have to be made (e.g. bandwidth=low), Asterisk will choose the best codec out of the conversion-matrix.
In the Asterisk CLI this machine-specific matrix can be shown: (N.B. matrix is taken from a P3 machine)
asterisk1*CLI> show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 3 3 7 3 2 8 - 86 30 ulaw - 7 - 1 6 2 1 7 - 85 29 alaw - 7 1 - 6 2 1 7 - 85 29 g726 - 10 5 5 - 5 4 10 - 88 32 adpcm - 7 2 2 6 - 1 7 - 85 29 slin - 6 1 1 5 1 - 6 - 84 28 lpc10 - 12 7 7 11 7 6 - - 90 34 g729 - - - - - - - - - - - speex - 12 7 7 11 7 6 12 - - 34 ilbc - 14 9 9 13 9 8 14 - 92 -
The list of available codecs can be made visible like this:
asterisk1*CLI> show audio codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESC -------------------------------------------------------------------------------- 1 (1 << 0) (0x1) audio g723 (G.723.1) 2 (1 << 1) (0x2) audio gsm (GSM) 4 (1 << 2) (0x4) audio ulaw (G.711 u-law) 8 (1 << 3) (0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726 (G.726) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC)