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

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