search

mikesimb / sipml5

Automatically exported from code.google.com/p/sipml5
BSD 3-Clause "New" or "Revised" License
0 stars 0 forks source link

WebRTC Not working with Google Chrome 35. #180

Closed GoogleCodeExporter closed 8 years ago

GoogleCodeExporter commented 8 years ago 
What steps will reproduce the problem?
1. Update to Google Chrome 35.0.1916.114 
2. Attempt to use Web RTC
3. Confirmed issue on OSX and Windows 7. 

Google Chrome 34.0+ was working fine. 
Attempted to update 30+ people to Chrome 35 - All of them were unable to use 
the webRTC service. 
Chrome 34 still working fine.

Original issue reported on code.google.com by taylo...@gmail.com on 22 May 2014 at 6:54

GoogleCodeExporter commented 8 years ago 
Same here, we encountered a problem with DTLS. SDP is missing a=crypto lines.

Original comment by dca...@gmail.com on 22 May 2014 at 8:01

GoogleCodeExporter commented 8 years ago 
DTLS must *not* use a=crypto. You should attach full logs if you want help.

Original comment by boss...@yahoo.fr on 23 May 2014 at 1:15

GoogleCodeExporter commented 8 years ago 
Issue 179 has been merged into this issue.

Original comment by boss...@yahoo.fr on 23 May 2014 at 1:15

GoogleCodeExporter commented 8 years ago 
https://groups.google.com/forum/#!topic/doubango/iQ_8rgpbm0k

Original comment by boss...@yahoo.fr on 23 May 2014 at 1:26

GoogleCodeExporter commented 8 years ago 
Same problem, when update to Google Chrome 35, webrtc stop working. With 
version 34 still works fine.

-- Registered SIP '595983222446' at 64.33.235.148:53688
== Using SIP RTP CoS mark 5
[May 23 01:24:50] WARNING[5557][C-00000018]: chan_sip.c:10512 process_sdp: 
Rejecting secure audio stream without encryption details: audio 57120 
UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126

Original comment by lrdl...@gmail.com on 23 May 2014 at 1:26

GoogleCodeExporter commented 8 years ago 
@lrdluis
Please read the link I have provided. In short: all is correct in SIPML5 and 
you have to enable DTLS in Asterisk. Don't think DTLS implementation in 
Asterisk is ready for the prime and recommend using webrtc2sip breaker.

Original comment by boss...@yahoo.fr on 23 May 2014 at 1:30

GoogleCodeExporter commented 8 years ago 
I already use webrtc breaker, but the behavior is the same.

Original comment by lrdl...@gmail.com on 23 May 2014 at 1:51

GoogleCodeExporter commented 8 years ago 
If it's the case this means you're not using it correctly. As already said, you 
must provide logs for both sipml5 and webrtc2sip if you want help.

Original comment by boss...@yahoo.fr on 23 May 2014 at 2:17

GoogleCodeExporter commented 8 years ago 
Tengo el mismo problema, alguna manera para solucionar esto, uso sipml5 !!!

Original comment by angelarb...@gmail.com on 18 Jun 2014 at 7:31