I'm pretty sure I'm just missing something very simple here like a port forward. Let me know if you have any ideas! This setup completely works when both csipsiimple clients are inside my LAN and kamailio is configured with alias=192.168.1.LAN. The problem begins when I connect both clients from the WAN and configure kamailio with alias=66.41.221.WAN. The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out. Summary:: The callee and caller both establish 2-way audio and zrtp keys. TLS is working. Both SIP clients are using the OSTN wizard in csipsimple. Please review my Debug logs: http://pastebin.com/9Rw7zSQ0 _______________________________________________ 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