Summary:ASTERISK-13012: "RTCP SR transmission error, rtcp halted" logged when SIP call put on hold
Reporter:matt_b (matt_b)Labels:
Date Opened:2008-11-04 07:56:30.000-0600Date Closed:2009-01-22 12:55:46.000-0600
Versions:Frequency of
Environment:Attachments:( 0) bug13835.patch
( 1) bug13835-trunk.patch
( 2) console.log
( 3) debug.log
( 4) extensions.conf
( 5) sip.conf
Description:I'm running 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 ~5 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.


I have posted a question about this on the forums: http://forums.digium.com/viewtopic.php?t=65210 which directed me to file a bug report
Comments:By: David Woolley (davidw) 2008-11-04 08:11:56.000-0600

I suggested it was a bug, because the code that outputs this message, does:

AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);

which I assume is an attempt to de-schedule the 5 second RTCP transmissions.

By: Jeff Peeler (jpeeler) 2008-12-03 12:46:14.000-0600

I believe the added missing check will fix the problem.

By: Digium Subversion (svnbot) 2008-12-04 12:30:37.000-0600

Repository: asterisk
Revision: 161013

U   branches/1.4/main/rtp.c

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

(closes issue ASTERISK-13012)
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.