[asterisk-bugs] [Asterisk 0015504]: [patch] G726 Codec has choppy audio on Version 1.6.1
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Nov 10 14:15:43 CST 2009
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: file
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Project: Asterisk
Issue ID: 15504
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: block
Priority: normal
Status: closed
Target Version: 1.6.1.11
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:
Resolution: fixed
Fixed in Version:
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Date Submitted: 2009-07-15 00:44 CDT
Last Modified: 2009-11-10 14:15 CST
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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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Relationships ID Summary
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related to 0016118 G726 is choppy on IAX - 1.6
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(0113609) svnbot (reporter) - 2009-11-10 14:15
https://issues.asterisk.org/view.php?id=15504#c113609
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Repository: asterisk
Revision: 229285
_U branches/1.6.2/
U branches/1.6.2/codecs/codec_g726.c
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r229285 | file | 2009-11-10 14:15:42 -0600 (Tue, 10 Nov 2009) | 22 lines
Merged revisions 229282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | 15 lines
Merged revisions 229281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
lines
Remove broken support for direct transcoding between G.726 RFC3551 and
G.726 AAL2.
On some systems the translation core would actually consider g726aal2
-> g726 -> signed linear
to be a quicker path then g726aal2 -> signed linear which exposed this
problem.
(closes issue https://issues.asterisk.org/view.php?id=15504)
Reported by: globalnetinc
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http://svn.digium.com/view/asterisk?view=rev&revision=229285
Issue History
Date Modified Username Field Change
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2009-11-10 14:15 svnbot Checkin
2009-11-10 14:15 svnbot Note Added: 0113609
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