[asterisk-bugs] [Asterisk 0015642]: [patch] Fix for Sonus DTMF issues

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15642 
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Reported By:                jasonshugart
Assigned To:                twilson
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Project:                    Asterisk
Issue ID:                   15642
Category:                   Core/RTP
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Target Version:             1.6.1.18
Asterisk Version:           SVN 
JIRA:                       SWP-406 
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:             2009-08-03 12:28 CDT
Last Modified:              2010-02-23 16:19 CST
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Summary:                    [patch] Fix for Sonus DTMF issues
Description: 
In some cases when Asterisk sends DTMF to Sonus platforms as RFC2833, Sonus
will not recognize the DTMF.  Several articles have identified the problem
as being the gap in the audio prior to the DTMF packets being sent.  This
patch sends a single G.711 ulaw packet prior to the rfc2833 packets.  In
our testing across with two carriers (Level3 and 360 Networks) this patch
fixed our DTMF issues.  The rtp.c file seems very similar for the 1.4
branch, so minor changes could also be applied there.  I added an option to
the rtp.conf file to enable this fix, called rtpfixdtmf.  Attached are both
the rtp.c fix, and the rtp.conf change.
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Relationships       ID      Summary
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has duplicate       0016625 RFC2833 DTMF is not passed correctly wh...
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 (0118432) twilson (administrator) - 2010-02-23 16:19
 https://issues.asterisk.org/view.php?id=15642#c118432 
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I am attaching another patch (for 1.4--1.6.x should apply with one easy
modification--add 'struct ast_silence_generator *gen;' to the end of the
sip_pvt struct definition) that interleaves audio with the dtmf events if
the 'transmit_silence' option in asterisk.conf is enabled (be sure to
uncomment '[options]' as well).  Other things that I have found that fixed
things on various peoples machines were:
1) setting constantssrc=yes in sip.conf
2) setting rfc2833compensate=no in sip.conf

If I could get people to test out 1 and 2 (by themselves) to see if that
fixes the problem, and if they don't then to test out the
fix_sonus_dtmf.patch.txt patch, I would greatly appreciate it. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-23 16:19 twilson        Note Added: 0118432                          
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