Re: [SR-Users] NAT Traversal issue

Hi All, Problem solved. It was a CODEC issue. Best Regards, Roy. On Tue, Dec 4, 2012 at 3:58 AM, Raj Roy Ghandhi &lt;roy.gandhi(a)gmail.com <mailto:roy.gandhi@gmail.com>> wrote: Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client->LAN client web page call 2. LAN webrtc client -> 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy. _______________________________________________ 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