Hi, > a) Can kamailio be used as sip-proxy while using WebRTC based UA > calling to plain UAC/WebRTC based UAC ?
Yes, kamailio can do SIP over websocket, so all you need is a javascript SIP stack (e.g. JsSIP, jain-sip JS, ...) on your WebRTC enabled client. > b) What to use for media proxying (this really baffles me..) rtpproxy > or rtpengine (?) or mediaproxy or rtpproxy-ng ? Is there any relation > between them anywhere? you will need to be able to translate WebRTC RTP (RTP/SAVPF) to other RTP profiles like RTP/AVP. Only rtpengine can do this (note that mediaproxy-ng is the old name for rtpengine). > c) I am not behind NAT and do not want secure web-sockets, so any > sample config I can refer to ? If you familiar with kamailio cfg scripting you can try to start something from scratch (building a simple proxy is quite straightforward). Otherwise i don't know any example file that does all you need. See examples/websocket.cfg for websocket handling. You can disable the registrar and the NAT stuff in it if you don't need them. Cheers, -- Camille _______________________________________________ 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
