What steps will reproduce the problem?
1. Environment setup using Asterisk, WebRTC2SIP and Browser (Chrome and/or FF)
2. Login by SIP client on (http://sipml5.org/call.htm?svn=222#)
2. Initiate call from legacy phone (linphone/microsip)
3. Receive call at call.html
4. Press Hold button
What is the expected output? What do you see instead?
Expected : Call should be move in Hold state and Resume button should be
displayed.
Actual : Behaviour is different with different browsers. Below are the
observations:
1. First I have verified with FF 36 version. One error is found on browser side
as below:
removeStream() not implemented
I have concluded that hold is not supported on Firefox after checking the
release notes at https://code.google.com/p/sipml5/wiki/ReleaseNotes#1.5.222
2. Then I have verified with Chrome latest version and found that webrtc2sip is
crashed during this operation. I have collected webrtc2sip logs and core dump.
Attached with ticket.
What version of the product are you using? On what operating system?
Operating System : CentOS 6.6
Asterisk : 11.6 cert-9
webrtc2sip : latest from global repository
Firefox : 36
Chrome : 40
sipml5 : 1.5.222
Please provide server logs with DEBUG level equal to INFO
Attached (webrtc2sip.log and core.dump)
Please provide browser logs
Attached (browser.log)
Thanks and Regards
Vinod Pandey
Original issue reported on code.google.com by pandey.g...@gmail.com on 5 Mar 2015 at 6:50
Original issue reported on code.google.com by
pandey.g...@gmail.comon 5 Mar 2015 at 6:50Attachments: