Hi guys: I am a R&D engineer trying to learn kamailio. After following some tutorials and reading the thread in this mailing list I was able to setup a voip backend with this configuration XLITE/LINPHONE ---> KAMAILIO ----> FREESWITCH I am using Freeswitch as a media server. After configuring RTP Proxy and kamailio to use bridged mode. I was able to successfully setup a voip backend like the one above. I encountered a problem when the UAC I am using is a webclient like sipml5. I noticed that when SIP INVITES from KAMAILIO to FREESWITCH are being passed when a INVITE transaction is initiated from a sipml5 client FREESWITCH is trying to use the public ip of webrtc server of the sipml5 backend. Unfortunately, I am using private ip/LAN IP between kamailio and freeswitch. As a result calls are established but there is no audio that is happening. I am attaching a snapshot of the ngrep that on kamailio and freeswitch server for your reference. I would like to know if there is a setting in kamailio that would allow me to modify the IP in the "o" and "c" sdp parameter when forwarding an invite to Freeswitch. I did another test. XLITE ---> KAMAILIO ---> FREESWITCH ----> KAMAILIO ----> sipml5 And the call works. It has audio. I think it must have something to do with the SDP header that is being generated by sip5ml UAC that is conflicting with my setup. Any help or advice will be greatly appreciated. Thanks. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users