<div dir="ltr">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.<br clear="all"><div><br></div><div>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. <br><br>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. <br><br></div><div>Thanks in advance<br></div><div><br>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr">Kamil Nigmatullin<br>Skype: kamil.nigmatullin</div></div>
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