Dear sr-users, 

I am trying to set up a simple Kamailio server with rtpproxy on the same host computer, behind a NAT router. “Run your own Skype-like service” is my set-up guide, but I’m using Kamailio version 5.1 so I modified the config files that came with the download. My problem is that SIP clients can register over the internet via TLS encryption with the server and establish a call session connection, but the ZRTP media connection is not being made. Once the call session connection is made, both clients send out UDP packets to each other, but neither receives them—the media traffic is one way and being dropped somewhere. 

I’ve tried different rtpproxy start settings. If I start the rtpproxy service with “sudo /etc/init.d/rtpproxy start” then it uses the default “unix:/var/run/rtpproxy/rtpproxy.sock”. I’ve also tried starting with “sudo rtpproxy –l my_public_ip –s udp:localhost:7722” which then asks for either –F (superuser) or –u (user). I’ve tried it both with options “–F” and “-u kamailio kamailio”. All the variations tried so far have the same result as noted above. 

The server behind the router is in a DMZ zone that bypasses the router firewall.

Is there anything in the user-list archives that addresses this issue? Or some likely place to start troubleshooting the problem?

Any suggestions would be greatly appreciated. Thanks in advance.

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