<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">I built something similar - a sip.js-based browser coupled with a webrtc proxy using freeswitch for the media proxy (no verto), with a separate SIP signaling engine (in my case instead of opensips I used my open open source framework, called drachtio <a href="https://github.com/davehorton/drachtio-server" class="">https://github.com/davehorton/drachtio-server</a>).<div class=""><br class=""></div><div class="">TLDR; For the inbound call to the browser, send a 3pcc INVITE to the browser</div><div class=""><br class=""></div><div class="">Meaning:</div><div class=""><br class=""></div><div class="">1. Send INVITE with no SDP to the browser</div><div class="">2. Get 200 OK with SDP and ICE candidates</div><div class="">3. Send INVITE to freeswitch with that SDP</div><div class="">4. Get 200 OK from freeswitch with its SDP</div><div class="">5. Send ACK to browser with SDP from freeswitch</div><div class=""><br class=""></div><div class="">Freeswitch then bridges the call from the webrtc side to the sip side.</div><div class=""><br class=""></div><div class="">FYI, my browser app can be seen at <a href="https://webrtc.drachtio.org/" class="">https://webrtc.drachtio.org/</a>, feel free to try it out, with a couple of notes:</div><div class="">1) only google authentication is supported</div><div class="">2) you need to have sip credentials from a hosted VoIP provider</div><div class="">3) when you first log in, select the ‘Settings’ menu from the upper right and add your sip credentials</div><div class="">4) its meant to be an operator type console where you can also monitor other extensions in your business using sip presence, do call pickup etc. </div><div class="">5) video is currently not supported</div><div class=""><br class=""></div><div class="">Browser app is built using google polymer material design, and code is here: <a href="https://github.com/davehorton/webrtc-operator-console" class="">https://github.com/davehorton/webrtc-operator-console</a></div><div class="">Webrtc proxy app code is here: <a href="https://github.com/davehorton/ws-proxy" class="">https://github.com/davehorton/ws-proxy</a>, and as noted depends on drachtio server and freeswitch (using drachtio-fsmrf module for controlling freeswitch).</div><div class=""><br class=""></div><div class="">For more details or help, feel free to email me directly or open issues on any of those projects.<br class=""><div class=""><br class=""></div><div class=""><br class=""><div><div class="">On Mar 18, 2017, at 6:03 AM, Kamil Nigmatullin <<a href="mailto:kamil.nigmatullin@gmail.com" class="">kamil.nigmatullin@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">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" class=""><div class=""><br class=""></div><div class="">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 class=""><br class="">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 class=""><br class=""></div><div class="">Thanks in advance<br class=""></div><div class=""><br class="">-- <br class=""><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr" class="">Kamil Nigmatullin<br class="">Skype: kamil.nigmatullin</div></div>
</div></div>
_________________________________________________________________________<br class="">Professional FreeSWITCH Consulting Services: <br class=""><a href="mailto:consulting@freeswitch.org" class="">consulting@freeswitch.org</a><br class="">http://www.freeswitchsolutions.com<br class=""><br class="">Official FreeSWITCH Sites<br class="">http://www.freeswitch.org<br class="">http://confluence.freeswitch.org<br class="">http://www.cluecon.com<br class=""><br class="">FreeSWITCH-users mailing list<br class="">FreeSWITCH-users@lists.freeswitch.org<br class="">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br class="">UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br class="">http://www.freeswitch.org</div></div><br class=""></div></div></body></html>