G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames. An algorithmic look-ahead of 7.5 ms duration means that total algorithmic delay is 37.5 ms. Its official name is Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s. It is sometimes associated with a Truespeech trademark in coprocessors produced by DSP Group.
There are two bit rates at which G.723.1 can operate:
6.3 kbit/s (using 24 byte frames) using a MPC-MLQ algorithm (MOS 3.9)
5.3 kbit/s (using 20 byte frames) using an ACELP algorithm (MOS 3.62)
G.723.1 is mostly used in Voice over IP (VoIP) applications due to its low bandwidth requirement. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus some other method such as G.711 or out-of-band methods should be used to transport these signals.
To install the G.723.1 codec for FreePBX you will need to download and save the module first by below command.
After download completed we need to rename and move the module into the Asterisk module folder by below command.
cp codec_g723-ast130-gcc4-glibc2.2-x86_64-core2-sse4.so /usr/lib64/asterisk/modules/codec_g723.so
Nex we will restart the sterisk in order to load the new module into Asterisk.
service asterisk restart
To confirm the installation you may check the Asterisk transcoding support table by executing below command.
asterisk -rx ‘core show translation’
If you see g723 above then both installed succesfully. Otherwise check the Asterisk logs.