[asterisk-bugs] G726 Codec has choppy audio on Version 1.6.1

[Johnson, Scott]

I am using G726 to reduce the rtp steam. It all works great for calls.
Quality is good but when VM or a prompt is played the sound is horrible.
It seems the translation is not working correctly. 

If the call is G726 (caller) => Asterisk => G726 (callee) the voice is
great. Sounds as good as G711. 

If: 
G711 (caller) => Asterisk = > G726 (callee) voice is horrible. You
cannot understand most words. Or 
Asterisk (VM or prompt playback) => G726 it is also bad.

 

The hardware is a Linksys spa2102 on the client side and the SIP trunk
provider is using Cicso software.  They work perfectly together and if
Asterisk is not in the middle the call quality is what you would expect.

 

We added the option g726nonstandard = yes in the sip.conf file

 

This made the call to VM or any time Asterisk was involved different but
equally bad.

 

After several hours I found that the source file for 1.6.1 main/frame.c 
had to be edited. The G726_AAL2 had to have the name g726 instead of
g726aal2. Then the audio is crystal clear. 


{ AST_FORMAT_G726_AAL2, "g726", 8000, "G.726 AAL2", 40, 10, 300, 10, 30
}, /*!< codec_g726.c */ 

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