<div dir="ltr"><div>Sorry, a mistake: on outgoing
webrtc it MUST have RTP/SAVP or RTP/SAVPF<br></div><br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-13 21:54 GMT+03:00 Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Well, I performed that by creating a media relay consisting of 2 freeswitches and using rtpengine. <br><br></div>You just need to handle WebRTC by kamailio using kamailio websocket module:<br><a href="http://kamailio.org/docs/modules/4.3.x/modules/websocket.html" target="_blank">http://kamailio.org/docs/modules/4.3.x/modules/websocket.html</a><br>caruzdias-es configuration helped me a lot in understanding how websockets work on Kamailio:<br><a href="https://github.com/caruizdiaz/kamailio-ws" target="_blank">https://github.com/caruizdiaz/kamailio-ws</a><br></div><div>But be aware, this configuration is for peer2peer connections, not for dispatching!<br></div><div><br></div><div>Kamailio will send simple SIP packets to the media relay then.<br></div><div><br></div><div>Also I used different NAT-traversal mechanism for sip and ws traffic (different routes based on client's transport protocol).<br></div><div>Also
you'll maybe need to have different rtpengine flags for sip and ws -
remember that WebRTC MUST have SRTP, but I had some issues in
transfering the SRTP handshake in
sipphone<-->kamailio<-->freeswitch scheme, so on webrtc
connection my "incoming" rtpengine had RTP/AVP flag, and on outgoing
webrtc it MUST have RTP/SAVP<br></div><div>For usual SIP calls I also conveted everything to RTP/AVP.<br></div><div><br></div><div>So you'll need to know to which type of user - ws or tcp/udp you're calling to understand which type of RTP to send them.<br></div></div><div class="gmail_extra"><div><div class="h5"><br><div class="gmail_quote">2015-06-13 19:07 GMT+03:00 Murugan Pandian <span dir="ltr"><<a href="mailto:manpower13.cse@gmail.com" target="_blank">manpower13.cse@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">it's posible dispatching websocket request?<div><br></div><div>I am try to connect browser(WebRTC) to sip-phone and vice versa,How i can achieve more concurrent call(more then 1000 call)</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div style="background-color:rgb(255,255,255);line-height:initial"> <div style="width:100%;font-size:initial;font-family:Calibri,'Slate Pro',sans-serif;color:rgb(31,73,125);text-align:initial;background-color:rgb(255,255,255)">That question is difficult to answer without some elaboration on your part as to what you want to achieve.</div> <div style="width:100%;font-size:initial;font-family:Calibri,'Slate Pro',sans-serif;color:rgb(31,73,125);text-align:initial;background-color:rgb(255,255,255)"><br style="display:initial"></div> <div></div> <div style="font-size:initial;font-family:Calibri,'Slate Pro',sans-serif;color:rgb(31,73,125);text-align:initial;background-color:rgb(255,255,255)">--<br>Alex Balashov | Principal | Evariste Systems LLC<br>303 Perimeter Center North, Suite 300<br>Atlanta, GA 30346<br>United States<br><br>Tel: <a href="tel:%2B1-800-250-5920" value="+18002505920" target="_blank">+1-800-250-5920</a> (toll-free) / <a href="tel:%2B1-678-954-0671" value="+16789540671" target="_blank">+1-678-954-0671</a> (direct)<br>Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" target="_blank">http://www.csrpswitch.com/</a><br><br>Sent from my BlackBerry.</div> <table style="background-color:white;border-spacing:0px" width="100%"> <tbody><tr><td colspan="2" style="font-size:initial;text-align:initial;background-color:rgb(255,255,255)"> <div style="border-style:solid none none;border-top-color:rgb(181,196,223);border-top-width:1pt;padding:3pt 0in 0in;font-family:Tahoma,'BB Alpha Sans','Slate Pro';font-size:10pt"> <div><b>From: </b>Murugan Pandian</div><div><b>Sent: </b>Saturday, June 13, 2015 09:47</div><div><b>To: </b><a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a></div><div><b>Reply To: </b>Kamailio (SER) - Users Mailing List</div><div><b>Subject: </b>[SR-Users] SIP-over-Websocket Load Balancing</div></div></td></tr></tbody></table><span><div style="border-style:solid none none;border-top-color:rgb(186,188,209);border-top-width:1pt;font-size:initial;text-align:initial;background-color:rgb(255,255,255)"></div><br><div><div dir="ltr">HI,<div><br></div><div> how to handle sip-over-websocket load balancing (WebRTC)</div></div>
<br></div></span></div>
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<br></blockquote></div><br><br clear="all"><br></div></div><span class="HOEnZb"><font color="#888888">-- <br><div><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</font></span></div>
</blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature"><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: +37367398493<br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
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