Well, I performed that by creating a media relay consisting of 2 freeswitches and using rtpengine.
You just need to handle WebRTC by kamailio using kamailio websocket module: http://kamailio.org/docs/modules/4.3.x/modules/websocket.html caruzdias-es configuration helped me a lot in understanding how websockets work on Kamailio: https://github.com/caruizdiaz/kamailio-ws But be aware, this configuration is for peer2peer connections, not for dispatching! Kamailio will send simple SIP packets to the media relay then. Also I used different NAT-traversal mechanism for sip and ws traffic (different routes based on client's transport protocol). 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 For usual SIP calls I also conveted everything to RTP/AVP. 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. 2015-06-13 19:07 GMT+03:00 Murugan Pandian <manpower13....@gmail.com>: > it's posible dispatching websocket request? > > I am try to connect browser(WebRTC) to sip-phone and vice versa,How i can > achieve more concurrent call(more then 1000 call) > > On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov <abalas...@evaristesys.com> > wrote: > >> That question is difficult to answer without some elaboration on your >> part as to what you want to achieve. >> >> -- >> Alex Balashov | Principal | Evariste Systems LLC >> 303 Perimeter Center North, Suite 300 >> Atlanta, GA 30346 >> United States >> >> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ >> >> Sent from my BlackBerry. >> *From: *Murugan Pandian >> *Sent: *Saturday, June 13, 2015 09:47 >> *To: *sr-users@lists.sip-router.org >> *Reply To: *Kamailio (SER) - Users Mailing List >> *Subject: *[SR-Users] SIP-over-Websocket Load Balancing >> >> HI, >> >> how to handle sip-over-websocket load balancing (WebRTC) >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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