Re: [SR-Users] SIP-over-Websocket Load Balancing

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 &lt;manpower13.cse(a)gmail.com&gt;om>: -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/