liuzhe02 / doubango

Automatically exported from code.google.com/p/doubango
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webrtc2sip crashes before/on dtls-srtp handshake. #398

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
What steps will reproduce the problem?
1.install webrtc2sip latest revision on ubuntu, install asterisk 11.7 on a 
seperate server
2. register the sipml5 clients via wss.
3. call from chrome or firefox to chrome/firefox, and answer the call

What is the expected output? What do you see instead?
the media stream should be established correctly but on answering the call 
webrtc2sip crashes with segmentation fault (core dumped)

What version of the product are you using? On what operating system?
webrtc2sip latest revision and asterisk 11.7

Please provide any additional information below.
self signed certificates are generated as in 
https://groups.google.com/forum/#!topic/doubango/asAfP5ZCgdI. 
sipml5 clients in chrome and firefox are connected successfully using wss.
the clients are configured to use h263 and pcmu and pcma in asterisk sip.conf

Original issue reported on code.google.com by hamed...@gmail.com on 16 Jul 2014 at 5:55