- It works like SILK but, see item 2.
- Its an old codec thus supported by lots of phones as opposed to SILK
Environments (of course, improvise and modify for your respective OSes, it can run too far away from these)
- Debian 6 32bit (somehow I messed up my 64bit or it doesn’t work well with 64 bit)
- Asterisk 11
So the steps are simple as outlined here;
- # apt-get install speex speex libspeexdsp-dev libspeex-dev
- Go to your Asterisk source directory
- Do a #make clean && ./configure && make && make install
- NOTE: It should automatically notice Speex libraries are already installed and it will auto select, so don’t have to menu select whatnot.
- Once done make, restart Asterisk (#amportal kill && amportal start)
- Then you should be autoloading the codec, so # asterisk –rx “core show translation” should show you speex
speex 15000 15000 15000 15000 15000 9000 15000 - 23000 15000 15000 17250 17000 15000 23000 17000 17000 17000 17000 17000 17000 17000
speex16 23500 23500 23500 23500 23500 17500 23500 23500 - 23500 23500 15000 9000 23500 23000 17500 17000 17000 17000 17000 17000 17000
- Otherwise, manually load #asterisk –rx “module load codec_speex.so”
- If you hit errors, lookout for the full log…
- Now, if speex is loaded properly, go to the IAX/SIP Setting pages in FreePBX and enable speex codec respectively
- That should now allow you to use speex in the extensions/devices you’ve configured
And you guessed, it, here’s the cliché, Enjoy Speex-ing…
PS> What Codec we really want to consider in future and see it released to support Asterisk?.
OPUS. I believe this one codec will rule them all. Till then, we use bits like SILK and SPEEX and sometimes g729 to get our very cranky networks to play nice with VoIP audio.