G726 codec choppy audio

Asterisk Asterisk Support
1

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.

Just did not know if you had experienced this.

2

Just a stab in the dark, but are you sure that it is not bridging the two phones then using g711 or something with better call quality once they are bridged?

3

After several hours I found that the source file for 1.6.1 main/frame.c
had to be editied. 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 */

4

What kind of phone were you using ? I had this issue with 1.4.X using SNOM phones. Polycom phones had no issue. Also you may want to post this on bugs.digium.com.

EDIT: Sorry I had audio issues with G722 and not G726.