28 Nov
2012
28 Nov
'12
9:58 p.m.
Jeremy, it is doesn't work at all. I've made a lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during a
call.
Another issue is - sipml5 is sending a malformed Contact field, and
asterisk is trying to contact to invalid destination and finally closing a
call.
2012/11/28 Jeremy Jongepier <jeremy(a)autostatic.com>
On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
I think at the moment only a patched Asterisk might be able to do this:
http://code.google.com/p/**sipml5/wiki/Asterisk<http://code.google.com/p…
But I haven't got this to work yet because I get the feeling the patch
Doubango provides is incomplete.
Or maybe webrtc2sip by the same makers: http://code.google.com/p/**
webrtc2sip/ <http://code.google.com/p/webrtc2sip/>
I've tried setting up a webrtc2sip server today but it crashes after a few
minutes, is horrible to set up (the web GUI is lightyears behind Siremis)
and for the moment I can't get it to work because I can't find any proper
documentation on how to set it up.
If anyone could build webrtc support into rtpproxy or any other media
proxy that can work together with Kamailio I'd be more than happy to test
it.
Regards,
Jeremy
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