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 07:56:30.000-0600 | Date Closed: | 2009-01-22 12:55:46.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Core/RTP |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
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 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 ~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. ****** ADDITIONAL INFORMATION ****** 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. ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=161013 |