cgy987512 / webrtc2sip

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webrtc2sip crash on call hold from chrome browser #177

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
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

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