Atticus [grobe...@gmail.com] wrote: > It isn't just SIP. You will need to set up NAT traversal and make sure RTP > traffic can pass as well. Setting up a STUN server and configuring the > clients to use it should aid in the NAT portion. The RTP traffic should be > fine as long as pf is being stateful. If the phones register over SIP fine, > but have no audio, then the RTP traffic isn't getting where it should. IMO, > it makes more sense to use an on-site PBX so you only have to deal with > traffic to/from the one host, but that doesn't sound like an option for you. >
Using Asterisk as a SIP server and media gateway, on a public IP with phones behind NAT, you can get NAT traversal via RFC 3581+symmetric RTP (sip.conf nat=yes) without STUN and without a firewall SIP ALG. Freeswitch and also Kamailio + rtpproxy can do similar. These gateways are all capable of symmetric RTP, and have been since forever. If you are connecting phones inside of your NAT to an outside SIP provider, or PBX device, make sure the PBX has a public IP (not behind another NAT itself) and has symmetric RTP enabled.
