search

kirikouz / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip
0 stars 0 forks source link

Contact header different between REGISTER and INVITE in rtcweb-breaker=yes case. #51

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago 
What steps will reproduce the problem?
1. In Expert Settings Enable RTCWeb Breaker
2. Login with sipML5
3. Check the Sip REGISTER sent by the webrtc2sip it is missing the contact 
header parameter rtcweb-breaker=yes.
4. Perfom an invite and check the Sip INVITE the parameter is now part of the 
contact header.

What is the expected output? What do you see instead?

The sipML5 client REGISTER with Contact header parameter rtcweb-breaker=yes.
But that parameter is not sent to our CSCF by the webrtc2sip.
When I try to call someone the parameter "rtcweb-breaker=yes" is in the contact 
header of INVITE request.

Our CSCF replys SIP/2.0 403 Calling User Not Registered because of the Contact 
header differences.

If the parameter is rtcweb-breaker=no it is part of both REGISTER and INVITE 
and it's working.

What version of the product are you using? On what operating system?
Doubango 2.0 latest code from trunk.
svn co http://webrtc2sip.googlecode.com/svn/trunk/ webrtc2sip
February 6 2013 
SUSE Linux Enterprise Server 11 (x86_64)

Please provide any additional information below.

Original issue reported on code.google.com by martin.n...@gmail.com on 11 Feb 2013 at 8:08

GoogleCodeExporter commented 8 years ago 
Sorry, clarification, should read rtcweb-breaker is in SIP REGISTER but not in 
the SIP INVITE. The parameter is in websocket for both requests.
Test perform with Firefox Nightly 21.0a1 (2013-01-29).

Original comment by martin.n...@gmail.com on 11 Feb 2013 at 10:45

GoogleCodeExporter commented 8 years ago

Original comment by boss...@yahoo.fr on 12 Feb 2013 at 1:48

GoogleCodeExporter commented 8 years ago 
We have the same problem, but it only happens when using a WSS url for 
websocket_proxy_url. For unsecure websockets, it works. 
Is it correct that the full contact header is needed so that requests from the 
callee (i.e. BYE) are forwarded correctly by the webrtc2sip proxy?
Because when the contact parameters are missing the webrtc2sip does not 
recognize the BYE and sends it somewhere else.

Original comment by hoffmann...@gmail.com on 1 Aug 2013 at 9:37