[asterisk-bugs] [Asterisk 0018189]: RFC2833 DTMF generation broken due to SSRC change on bridges channels

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18189 
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Reported By:                marcbou
Assigned To:                jpeeler
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Project:                    Asterisk
Issue ID:                   18189
Category:                   Core/RTP
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     assigned
Target Version:             1.8.1
Asterisk Version:           1.8.0 
JIRA:                       SWP-2473 
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:             2010-10-22 12:13 CDT
Last Modified:              2010-12-04 19:25 CST
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Summary:                    RFC2833 DTMF generation broken due to SSRC change on
bridges channels
Description: 
Since upgrading to the latest 1.8.0-rc, DTMF digits sent over SIP/RTP to
our service provider were no longer being detected.

We analyzed packet dumps, comparing old and new RTP packets being
generated by asterisk.

The difference was tracked down to asterisk 1.8.0-rc now changing the SSRC
value for RFC2833 DTMF digit packets.

If in main/channel.c:ast_channel_bridge() I comment out 

    ast_indicate(c0, AST_CONTROL_SRCCHANGE);
    ast_indicate(c1, AST_CONTROL_SRCCHANGE);

the SSRC no longer changes for DTMF digits and the provider can detect
them again.

However I am not sure if the change doesn't adversely affect other things.
Please advise.

Kind regards,

Marc Boucher



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Relationships       ID      Summary
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related to          0018352 SSRC is changing when DTMF sent
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 (0129309) cmbaker82 (reporter) - 2010-12-04 19:25
 https://issues.asterisk.org/view.php?id=18189#c129309 
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We are able to reproduce the delay only sporadically.  We are using sip
unboxed accounts from bandwidth.com and sip accounts from vitelity.com. 
The problem happens more frequently with the bandwidth.com acccounts.  

On bad days the delay occurs on about 10% of calls; however, on most days
it is usually far less.  One time we made 60 consecutive calls without
experiencing the delay.

Our method of testing is for one person to call in from their cell phone,
the person receiving the call picks up the call and immediately begins
counting (1,2,3,4).  When the delay occurs the calling person will hear
silence until 3.

It still happens occasionally so if there are any settings you want to
know or logs you need that will help just let me know. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-04 19:25 cmbaker82      Note Added: 0129309                          
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