[asterisk-bugs] [Asterisk 0015504]: [patch] G726 Codec has choppy audio on Version 1.6.1
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Oct 26 09:00:40 CDT 2009
The following issue has been set as RELATED TO issue 0016118.
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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: 2009-10-26 09:00 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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Relationships ID Summary
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related to 0016118 G726 is choppy on IAX - 1.6
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Issue History
Date Modified Username Field Change
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2009-10-26 09:00 lmadsen Relationship added related to 0016118
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