[Freeswitch-users] Convert Sip to Webrtc-like call

[Kamil Nigmatullin]

Maybe it is a silly question, but cannot find anything. I have the opensips
that handles auth and all routing, and freeswitch is responsible for
handling media, prepaid and so on.

Now, I want to implement browser clinent based on sip.js and therefore i
pass all sip signalling through opensips's implementetion for websockets.
For outgoing invites from browser clients this works.

But for calls that are directed to browser clients it doesn't. I beleive
that this is due to simlpe sip call does not have ice-candidates. So, the
question is how to make freeswitch convert simple sip call to web-rtc
oritenteated. I didn't implement mod_verto.

Thanks in advance

-- 
Kamil Nigmatullin
Skype: kamil.nigmatullin
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