[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. 
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https://issues.asterisk.org/view.php?id=15504 
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Reported By:                globalnetinc
Assigned To:                tilghman
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Project:                    Asterisk
Issue ID:                   15504
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     feedback
Target Version:             1.6.1.7
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-07-15 00:44 CDT
Last Modified:              2009-09-30 17:43 CDT
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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. 

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 (0111683) globalnetinc (reporter) - 2009-09-30 17:43
 https://issues.asterisk.org/view.php?id=15504#c111683 
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Here is an example of what does not work.  Using a Grandstream ATA I call
and invalid number.  A voice plays from the media server in Asterisk
stating that that it is not a valid extension.  The voice is clear and
perfect the codec is g726 or g726aal2.  Both work and dthe device supports
both.  Next I call an outbound number.  The SIP provider is set to only
support g711u.  The sound both ways is now horrible and broken both ways. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-30 17:43 globalnetinc   Note Added: 0111683                          
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