Sorry, a mistake: on outgoing webrtc it MUST have RTP/SAVP or RTP/SAVPF
2015-06-13 21:54 GMT+03:00 Alexandru Covalschi <568...@gmail.com>: > 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/ > -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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