[asterisk-bugs] [Asterisk 0016512]: rtp.c:2482 ast_rtcp_write_sr: rtcp halted Operation not permitted

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16512 
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Reported By:                jonaskellens
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16512
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.4 - please test a newer version 
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:             2009-12-24 05:11 CST
Last Modified:              2010-01-04 13:15 CST
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Summary:                    rtp.c:2482 ast_rtcp_write_sr: rtcp halted Operation
not permitted
Description: 
While making a call the CLI is flooded with the above message.

It begins already during ringing, and it goes on while the correspondents
are talking.

The code in rtp.c says :
if (!rtp->rtcp->them.sin_addr.s_addr) {  /* This'll stop rtcp for this rtp
session */
       ast_verbose("RTCP SR transmission error, rtcp halted\n");
       AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
       return 0;
}

Why is rtcp-transport aborted ??

I would like to make use of this "quality surveillance" of rtcp.

Is it a setting in Asterisk, my firewall, my SIP client  ??

[Dec 24 11:45:51] ERROR[14035]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR
transmission error to ip_of_ITSP:40483, rtcp halted Operation not
permitted
[Dec 24 11:45:51] ERROR[14035]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR
transmission error to public_ip_of_my_firewall:11003, rtcp halted Operation
not permitted


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---------------------------------------------------------------------- 
 (0115980) lmadsen (administrator) - 2010-01-04 13:15
 https://issues.asterisk.org/view.php?id=16512#c115980 
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Could you provide the following information?

* SIP debug
* Console output
* SIP history

These can be enabled via logger.conf and sip.conf (use 'full' in
logger.conf)

After reloading the modules, then use 'core set verbose 10' and 'core set
debug 10', reproduce the issue, and trim the log file prior to uploading
it.

Also, are you running Asterisk as non-root? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-04 13:15 lmadsen        Note Added: 0115980                          
2010-01-04 13:15 lmadsen        Status                   new => feedback     
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