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The problem occurs when I make or receive a call from Firefox or Chrome, via
webrtc2sip and asterisk to a softphone (Ekiga). I've attached the sipml5 log
file.
I think I'm getting audio in both directions (the mic is faulty on one
machine), but video only in one direction. Video is sent from the web browser
to the softphone, but doesn't get back to the web browser. There are messages
appearing in the webrtc2sip debug:
*[DOUBANGO INFO]: Incoming SIP INFO(picture_fast_update)
*[DOUBANGO INFO]: ***IDR request tooo close(2 ms)...ignoring****
and
**[DOUBANGO WARN]: function: "tdav_codec_h263_decode()"
file: "src/codecs/h263/tdav_codec_h263.c"
line: "440"
MSG: Buffer overflow
Also, before video starts I get thousands of
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
appearing in the webrtc2sip log.
The asterisk config allows g722 and h263 only.
It's probably something that isn't configured correctly, or a mistake in my
sipml5 code, but I'd be grateful for some help solving this problem.
thanks,
Charles.
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
What version of the product are you using? On what operating system?
Please provide any additional information below.
Original issue reported on code.google.com by charles....@gmail.com on 15 Jun 2015 at 3:53
oops. I didn't fill out all the requested information.
webrtc2sip v2.6.0
sipml5 - trunk, downloaded 11/6/15
operating system - sipml5 on Windows 7, Firefox 38.0.5. Webrtc2sip is on Ubuntu
Trusty
asterisk is v13.3.2
Original comment by charles....@gmail.com on 16 Jun 2015 at 8:53
Original issue reported on code.google.com by
charles....@gmail.com
on 15 Jun 2015 at 3:53Attachments:
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