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G.729 and G.723.1 codecs for Asterisk open source PBX

Primary website / Google group

Asterisk 1.4, 1.6, 1.8, 10.0 - 21.x are supported.

To compile the codecs it is recommended to install Intel IPP libraries for better performance. Alternatively, download and install Bcg729 - a slightly slower implementation written in portable C99. Only G.729 will be available in that case.

The codecs are tested against Bcg729 1.0.2, IPP 5.3 - 8.2. Users of IPP 9.0 and IPP 2017 must also install IPP Legacy libraries. Later IPP versions may not work. AMD processors works with IPP too.

IPP

To install legacy IPP libraries:

tar xf ipp90legacy_lin_9.0.0.008.tar
cd ipp90legacy_lin/
unzip linux.zip
... password: accept
mv linux /opt/intel/ipp/legacy

Place legacy/ in the root of IPP installation directory (under ipp symlink).

Additionally, static libraries from the Intel compiler are required:

cd /opt/intel/ipp/legacy/lib
wget http://asterisk.hosting.lv/bin/icc-static-libs.tar.bz2
tar xjf icc-static-libs.tar.bz2

Codecs

Use ./autogen.sh to generate GNU Autoconf files, then ./configure. Check available options with ./configure --help. Specify --prefix in case Asterisk is installed in non-standard location.

G.723.1 send rate is configured in Asterisk codecs.conf file:

[g723]
; 6.3kbps stream, default
sendrate=63
; 5.3kbps
;sendrate=53

This option is for outgoing voice stream only. It does not affect incoming stream that should be decoded automatically whatever the bit-rate is.

There are also two Asterisk CLI commands g723 debug and g729 debug to print statistics about received frames sizes. This can aid in debugging audio problems. Bump Asterisk debug level to 1 to see the numbers.

astconv is audio format conversion utility similar to Asterisk file convert command. Build it with supplied build-astconv.sh script against Asterisk 16 or later. astconv loads codec_*.so modules directly to perform the conversion. Use codec module that was compiled against same Asterisk version the astconv was built against.

The translation result could be used to: (a) confirm the codec is working properly; (b) prepare voice-mail prompts, for example:

./astconv ./codec_g729.so -e 160 file.slin file.g729
./astconv ./codec_g729.so -d 10  file.g729 file.slin
./astconv ./codec_g723.so -e 480 file.slin file.g723
./astconv ./codec_g723.so -d 24  file.g723 file.slin

file.slin is signed linear 16-bin 8kHz mono audio, you can play it with alsa-utils:

aplay -f S16_LE -r 8000 file.slin

and convert between other formats with SOX:

sox input.wav -e signed-integer -b 16 -c 1 -r 8k -t raw output.slin
sox -t raw -e signed-integer -b 16 -c 1 -r 8k input.slin output.wav

Files:

  • codec_g72x.c, astconv.c, build-astconv.sh - GNU GPL v3;
  • autoconf files initially contributed by Michael.Kromer at computergmbh dot de;
  • g723_slin_ex.h, g729_slin_ex.h, slin_g72x_ex.h - sample speech data;
  • ipp/ files are a copy from IPP samples, IPP license apply.

Before reporting problem with the codecs, please read the website - compiling the codecs is not a trivial task. Asking Asterisk G.729 Google group first is also good idea.