Hello, I read a couple of thorough articles on SIP, and I'd like to make sure I got things right when it comes to using SIP with NAT routers. I know that, ideally, the IP PBX should be located in the DMZ to void NAT-related issues in SIP, but SOHO routers don't necessarily support this, so I'll assume that the SIP caller "Alice" and the IP PBX (eg. Freeswitch or Asterisk) server are located in a non-routable, private LAN, while the remote callee "Bob" is located on the Internet (either behind their own NAT router, or connected with a public, routable address). The SIP phone of Alice and Bob are both logged on to the Freeswitch server: http://img46.imageshack.us/img46/5120/sipnatrouters.jpg 1. When Alice wants to call Bob, her SIP phone sends an SIP packet to the Freeswitch server with her private IP address and a UDP port that it opened to let incoming RTP packets from Bob 2. Freeswitch rings Bob's phone through the UDP port is used to register with Freeswitch (usually, UDP5060). Bob's phone replies to Freeswitch with his public IP address and the RTP port it chose to receive voice packets from Alice 3. Once Bob picks up the phone, RTP voice packets flow directly between Alice and Bob, while Freeswitch remains in the loop to handle call signaling such as closing the connection when someone hangs up the call. Provided this is how things work... there are three issues when one or all SIP end-points are located in a (different) private LAN: 1. End-points use their private IP and a private UDP port for RTP. A server has to translate this into a routable IP address, and... 2. it must negotiate with the NAT firewall to make sure this RTP port is available, and if not, open some other port, and... 3. the server must rewrite the SDP packet to use this public port I have a couple of questions: 1. Can Freeswitch/Asterisk handle this rewriting/negotiation? 2. Provided the NAT firewall doesn't support UPnP/NAT-PMP, does it mean I must a) enable STUN in Freeswitch, b) set SIP end-points so that they use a fixed port for RTP, and c) configure the NAT firewall to map this UDP port to point to the SIP end-point? 3. Should SIP end-points be configured to use STUN/NAT, or should I let the server handle the IP/port rewriting itself? Thank you for any help. -- View this message in context: http://old.nabble.com/IP-PBX-and-NAT-firewalls-tp27075600p27075600.html Sent from the Freeswitch-users mailing list archive at Nabble.com.