What steps will reproduce the problem?
1. Using Chrome with sipml5 to place calls through webrtc2sip to a sip endpoint.
If the call is terminated at Chrome, by killing the chrome window or doing a
'F5' to reload the page, then the webrtc2sip sever will frequently, but not
always, incur an exception and must be restarted.
Presumably just hanging up on the call normally from the Chrome page would
cause the same issue, but that has not been verified since our application does
not have a 'hangup' button for the caller.
2.
I enabled coredumps, and it appears to trap in the same place each time. I will
attach the backtrace from gdb, here is a snip:
at src/audio/tdav_session_audio.c:269
269
trtp_manager_send_rtp(base->rtp_manager, audio->encoder.buffer, out_size,
TMEDIA_CODEC_F
RAME_DURATION_AUDIO_ENCODING(audio->encoder.codec), tsk_false/*Marker*/,
tsk_true/*lastPacket*/);
3.
What is the expected output? What do you see instead?
Expected is that the server would disconnect from the calls without incurring
an exception. It does that most of time, but occasionally does have an
exception.
What version of the product are you using? On what operating system?
webrtc2sip 2.60
sipml5 1.4.217
chrome 37.0.2062
o/s centos 6.5 x86_64
Please provide server logs with DEBUG level equal to INFO
Will attach.
Also attaching gdb backtrace log.
Please provide browser logs
Original issue reported on code.google.com by SLHank...@verizon.net on 29 Sep 2014 at 6:41
Original issue reported on code.google.com by
SLHank...@verizon.net
on 29 Sep 2014 at 6:41Attachments: