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

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