<div dir="ltr">Hugh,<div><br></div><div>Thank you for the response, however, now neither the callee or the caller are getting a feed. I am not sure if the nat requires symmetric RTP. If there is a way to determine that, please let me know.</div>
<div><br></div><div>Summery, even with Hugh's possible solution, this still does not work. </div><div><br></div><div>Thank you again,</div><div><br></div><div>ben</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Fri, Dec 20, 2013 at 10:26 AM, Hugh Waite <span dir="ltr"><<a href="mailto:hugh.waite@crocodile-rcs.com" target="_blank">hugh.waite@crocodile-rcs.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>Hi Ben,<br>
<br>
Some NATs require symmetric RTP, so that source and destination
ports are the same for incoming and outgoing media streams. This
can be configured using the 'w' flag to rtpproxy_manage in the
NATDETECT route.<br>
<br>
<tt><span style="vertical-align:baseline;font-variant:normal;font-style:normal;font-size:15px;background-color:transparent;text-decoration:none;font-weight:normal">rtpproxy_manage("cow");</span></tt><br>
<br>
Regards,<br>
Hugh<div><div class="h5"><br>
<br>
On 20/12/2013 16:13, Benjamin Trent wrote:<br>
</div></div></div>
<blockquote type="cite"><div><div class="h5">
<div dir="ltr">
<div style="font-family:arial,sans-serif;font-size:13px">Hey
all,</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style="font-family:arial,sans-serif;font-size:13px">**I
apologize if this is a duplicate, I received a bounce back on
my first email.</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br>
</div>
<span style="font-family:arial,sans-serif;font-size:13px">I have
kamailio set up behind a nat(port restricted, with firewall
rules to allow sip transactions and allowing rtpproxy packet
handling if needed) on Amazon EC2. I can register and calls
complete, however, the Caller(the one initiating the
transaction) receives no rtp media feed. I am running with NAT
enabled on kamailio and have rtpproxy installed listening on
the public IP. Kamailio and the rtpproxy are communicating(I
have verified via the kamailio debug logs). If I make a call
between the exact same voip machines directly via local IP on
the same Nat(skipping kamailio), the calls complete and they
both receive feeds.</span><br style="font-family:arial,sans-serif;font-size:13px">
<div style="font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div style="font-family:arial,sans-serif;font-size:13px">Both
the Caller(party making the call) and the Callee(party
receiving the call) are behind a Port Restricted Nat.
<div>
This is a folder containing the debug output for one of
these calls and the kamailio.cfg settings</div>
<div><br>
</div>
<div><a href="https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sharing" target="_blank">https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sharing</a><br>
</div>
<div><br>
</div>
<div>Quick FYI, the Caller Display Name and the Callee SIP
UserName are the same string. However, in my understanding
about sip, the display name means pretty much nothing and is
just a human readable string that does not effect packet
transport. If I am wrong and should test with a different
display name, let me know.</div>
<div><br>
</div>
<div><br>
</div>
<div>Thank you for the assistance,</div>
<div><br>
</div>
<div>ben</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
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</font></span></pre><span class="HOEnZb"><font color="#888888">
</font></span></blockquote><span class="HOEnZb"><font color="#888888">
<br>
<br>
<pre cols="72">--
Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.</pre>
</font></span></div>
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