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<p class="MsoNormal">Dear sr-users,
</p>
<p class="MsoNormal">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.
</p>
<p class="MsoNormal">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 <i style="mso-bidi-font-style:normal">my_public_ip</i>
–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. <br>
</p>
<p class="MsoNormal">The server behind the router is in a DMZ zone
that bypasses
the router firewall.</p>
<p class="MsoNormal">Is there anything in the user-list archives
that addresses
this issue? Or some likely place to start troubleshooting the
problem?<br>
</p>
<p class="MsoNormal">Any suggestions would be greatly appreciated.
Thanks in advance.<br>
</p>
<span style="font-size:12.0pt;font-family:"Times New
Roman";mso-fareast-font-family:
"Times New
Roman";mso-ansi-language:EN-US;mso-fareast-language:EN-US;
mso-bidi-language:AR-SA">Steve </span></p>
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