[asterisk-bugs] [Asterisk 0015504]: [patch] G726 Codec has choppy audio on Version 1.6.1

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15504 
====================================================================== 
Reported By:                globalnetinc
Assigned To:                tilghman
====================================================================== 
Project:                    Asterisk
Issue ID:                   15504
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     feedback
Target Version:             1.6.1.8
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-07-15 00:44 CDT
Last Modified:              2009-10-11 21:57 CDT
====================================================================== 
Summary:                    [patch] G726 Codec has choppy audio on Version 1.6.1
Description: 
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 and the g726 current name needed a change. Then the audio is
crystal clear. 

====================================================================== 

---------------------------------------------------------------------- 
 (0112194) tilghman (administrator) - 2009-10-11 21:57
 https://issues.asterisk.org/view.php?id=15504#c112194 
---------------------------------------------------------------------- 
One thing that the g726nonstandard option does NOT do is swap an identified
AAL2 codec into the normal mode.  I have attached a patch that does this. 
IF THIS WORKS, it would mean that the AAL2 and G726 codecs are completely
switched somewhere, which would suggest why your change worked.  However, a
test is needed before we can suggest the total fix.

To clarify, with this patch applied, please set g726nonstandard=yes in the
[general] section (never in the individual user section, please!) and test
transcoding. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-11 21:57 tilghman       Note Added: 0112194                          
======================================================================