[asterisk-bugs] [Asterisk 0013835]: "RTCP SR transmission error, rtcp halted" logged when SIP call put on hold

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13835 
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Reported By:                matt_b
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13835
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-11-04 07:56 CST
Last Modified:              2008-12-04 12:30 CST
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Summary:                    "RTCP SR transmission error, rtcp halted" logged
when SIP call put on hold
Description: 
I'm running 1.6.0.1 on Ubuntu 6.06 Server (2.6.15-52-server) with SNOM 370
handsets. Whenever I put a call on hold the message "RTCP SR transmission
error, rtcp halted" is logged on the console approx. every
http://bugs.digium.com/view.php?id=8#c5 seconds until
I take the call off hold. From a functional perspective the caller hears
the hold music correctly, so I think this is a cosmetic issue only, but it
obviously worries anyone reviewing the log files unnecessarily.

I have attached a console log without debugging enabled and one with full
logging enabled, and my sip.conf and extensions.conf.
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 (0095778) svnbot (reporter) - 2008-12-04 12:30
 http://bugs.digium.com/view.php?id=13835#c95778 
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Repository: asterisk
Revision: 161013

U   branches/1.4/main/rtp.c

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r161013 | jpeeler | 2008-12-04 12:30:36 -0600 (Thu, 04 Dec 2008) | 9 lines

(closes issue http://bugs.digium.com/view.php?id=13835)
Reported by: matt_b
Tested by: jpeeler

This mirrors a check that was present in ast_rtp_read to also be in
ast_rtp_raw_write to not schedule sending the receiver report if the remote
RTCP endpoint address isn't present in the RTCP structure.

Closes AST-142.


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http://svn.digium.com/view/asterisk?view=rev&revision=161013 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-04 12:30 svnbot         Checkin                                      
2008-12-04 12:30 svnbot         Note Added: 0095778                          
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