Re: [SR-Users] Kamailio behind NAT

On Jan 21, 2014, at 6:18 AM, "John Smith" <jsmith.15(a)mail.com> wrote: Hello, I am currently deploying one Kamailio behind NAT with one Asterisk as explained in the Asipto KB (Kamailio 4.0.x and Asterisk 11.3.0 using Asterisk Database). The structure is deployed as described in that document, with the only addition of one NAT between Kamailio and Internet: Phone ———————> Nat ———————> Kamailio ——————> Asterisk I have declared the private IP with the advertise option in order to support the NAT, enabled WITH_NAT and I have installed rtpproxy using standard Debian package configured as rtpproxy -l public_ip_ -s udp:localhost:7722 After setting up two phones which register correctly at Asterisk, I have no audio at all. By placing tcpdumps between nodes I see at Kamailio node both audio from public IP to internal Kamailio IP and from the latter to the Asterisk IP. In Asterisk I see audio coming from the Kamailio private IP and then back to the public IP of the phone. My guess is that audio should flow back into Kamailio and then to the phone, not directly from Asterisk as it is right now. Can anyone hint at where I am wrong? Thank you _______________________________________________ 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