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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:                     acknowledged
Asterisk Version:           Older 1.4 - please test a newer version 
JIRA:                       SWP-631 
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:              2010-04-14 09:51 CDT
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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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---------------------------------------------------------------------- 
 (0120395) lmadsen (administrator) - 2010-04-14 09:51
 https://issues.asterisk.org/view.php?id=16512#c120395 
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This branch by oej may or may not help: 
http://www.voip-forum.com/opensource/2010-01/test-rtcp-test-branch-based-asterisk-14/


Issue History 
Date Modified    Username       Field                    Change               
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2010-04-14 09:51 lmadsen        Note Added: 0120395                          
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