Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic.
ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario? In a near future i plan to offer disaster recovery, and i want all users binding to C (us...@mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918 address? , thx you PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration _______________________________________________ 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