<div><div>webrtc_media=true</div></div><div><br><div class="gmail_quote"><div>On Sat, Mar 18, 2017 at 8:50 AM Dave Horton <<a href="mailto:daveh@beachdognet.com">daveh@beachdognet.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word" class="gmail_msg">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="gmail_msg" target="_blank">https://github.com/davehorton/drachtio-server</a>).<div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">TLDR; For the inbound call to the browser, send a 3pcc INVITE to the browser</div><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">Meaning:</div><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">1. Send INVITE with no SDP to the browser</div><div class="gmail_msg">2. Get 200 OK with SDP and ICE candidates</div><div class="gmail_msg">3. Send INVITE to freeswitch with that SDP</div><div class="gmail_msg">4. Get 200 OK from freeswitch with its SDP</div><div class="gmail_msg">5. Send ACK to browser with SDP from freeswitch</div><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">Freeswitch then bridges the call from the webrtc side to the sip side.</div><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">FYI, my browser app can be seen at <a href="https://webrtc.drachtio.org/" class="gmail_msg" target="_blank">https://webrtc.drachtio.org/</a>, feel free to try it out, with a couple of notes:</div><div class="gmail_msg">1) only google authentication is supported</div><div class="gmail_msg">2) you need to have sip credentials from a hosted VoIP provider</div><div class="gmail_msg">3) when you first log in, select the ‘Settings’ menu from the upper right and add your sip credentials</div><div class="gmail_msg">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="gmail_msg">5) video is currently not supported</div><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">Browser app is built using google polymer material design, and code is here: <a href="https://github.com/davehorton/webrtc-operator-console" class="gmail_msg" target="_blank">https://github.com/davehorton/webrtc-operator-console</a></div><div class="gmail_msg">Webrtc proxy app code is here: <a href="https://github.com/davehorton/ws-proxy" class="gmail_msg" target="_blank">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="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">For more details or help, feel free to email me directly or open issues on any of those projects.</div></div><div style="word-wrap:break-word" class="gmail_msg"><div class="gmail_msg"><br class="gmail_msg"><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg"><br class="gmail_msg"><div class="gmail_msg"><div class="gmail_msg">On Mar 18, 2017, at 6:03 AM, Kamil Nigmatullin <<a href="mailto:kamil.nigmatullin@gmail.com" class="gmail_msg" target="_blank">kamil.nigmatullin@gmail.com</a>> wrote:</div><br class="m_-2749592596467677745Apple-interchange-newline gmail_msg"><div class="gmail_msg"><div class="gmail_msg">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="gmail_msg"><div class="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">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="gmail_msg"><br class="gmail_msg">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="gmail_msg"><br class="gmail_msg"></div><div class="gmail_msg">Thanks in advance<br class="gmail_msg"></div><div class="gmail_msg"><br class="gmail_msg">-- <br class="gmail_msg"><div class="m_-2749592596467677745gmail_signature gmail_msg" data-smartmail="gmail_signature"><div class="gmail_msg">Kamil Nigmatullin<br class="gmail_msg">Skype: kamil.nigmatullin</div></div>
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