[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:             1.6.1.8
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:              
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Date Submitted:             2009-07-15 00:44 CDT
Last Modified:              2009-10-11 20:18 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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---------------------------------------------------------------------- 
 (0112192) globalnetinc (reporter) - 2009-10-11 20:18
 https://issues.asterisk.org/view.php?id=15504#c112192 
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if it worked like that then:

[4065511212]
qualify=60000
ignoresdpversion=yes
nat=yes
accountcode=216.166.169.249
sendrpid=yes
callerid=Scott Johnson <4065511212>
usereqphone=yes
context=from-inside
call-limit=2
canreinvite=no
vmexten=4065511212
secret=zluSel
host=dynamic
trustrpid=no
username=4065511212
subscribecontext=local-extensions
dtmfmode=rfc2833
t38pt_udptl=yes,redundancy,maxdatagram=400
g726nonstandard=yes
type=friend
mailbox=4065511212
disallow=all
allow=g729
allow=g726
allow=g726aal2

Should work.  But it does not. If that did not work then 
g726nonstandard=no

Should work.  It does not either. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-11 20:18 globalnetinc   Note Added: 0112192                          
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