<div dir="ltr"><p dir="ltr">Im afraid thats not how this works, regarding your assumptions in last paragraph. </p><p dir="ltr">The NAT related parameters get saved in location table as OpenSIPS sets flags and upon a call to that user those are used immediately and no timer is used as such. </p><p dir="ltr">Now at the time of signalling rtpproxy has no role to play to work NAT or anything. RTPproxy will just use IP addresses in SDP as instructed in the script to have a media path. </p><p>From the SIP trace you shared it do looks like its sending to the right public IP for callee. Can you do a small test ? Re register your callee phone and within same second make a call to callee. I'm having doubts about a firewall closing the connection between the Callee UA and the OpenSIPS. </p><p><br></p>
<div class="gmail_quote">On Aug 6, 2015 8:00 AM, "Nabeel" <<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">My OpenSIPS runs on a public IP. The callee was connected to Wi-Fi in my first test earlier, but in the second test the callee was connected to a public IP (public mobile network). In both cases, the same '404 timeout' error occurred on call attempt. The SIP trace for the second case is at this link: </p>
<p dir="ltr"><a href="http://pastebin.com/jGxRQ34q" target="_blank">http://pastebin.com/jGxRQ34q</a></p>
<p dir="ltr">Regarding private IP, you said it's impossible to route from public IP to private IP. Although at the IP level this may be true, even if the user is on Wi-Fi, the whole point of NAT traversal is that the user's public IP is discovered and the call can get connected, is that not right? I'm fact, using a TURN server and a different SIP proxy, I was able to connect these same devices under the same networks, so I know this should be possible. I feel something is not configured correctly in OpenSIPS / rtpproxy.</p>
<p dir="ltr">I did "opensipsctl ul show" and the results seem normal; please check it:</p>
<p dir="ltr"><a href="http://pastebin.com/n1BbTuMK" target="_blank">http://pastebin.com/n1BbTuMK</a></p>
<p dir="ltr">Perhaps the NAT processing just needs a bit more time; in thar case what are the config options to increase the request timeout for UDP? I have seen the 'tcp_send_timeout' and 'tcp_connect_timeout' options for TCP, but please let me know if there are similar options for UDP.</p>
<div class="gmail_quote">On 6 Aug 2015 12:08, "Bogdan-Andrei Iancu" <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<tt>Nabeel,<br>
<br>
I suppose you OpenSIPS seats on a public IP, right ? The callee
looks to have a private IP. And, at IP level, it is impossible to
route from a public IP to a private one.<br>
<br>
I see your script has NAT traversal support. My question is - did
the callee properly registered via this script ? can you do an
"opensipsctl ul show" to see the callee's registration ?<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>On 06.08.2015 07:14, Nabeel wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hi,
<div><br>
</div>
<div>Yes, the destination IP is <span style="font-size:12.8000001907349px"> </span><span><a href="http://192.168.0.19:60912/" target="_blank">192.168.0.19:60912</a> and both phones are
registered to OpenSIPS. In this case, the callee is
connected to Wi-Fi (hence 192.xx IP address) and the caller
is connected to a mobile network.</span></div>
<div><span><br>
</span></div>
<div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px">The opensips.cfg I
am using was generated from 'make menuconfig', except with
the addition of "alias=<a href="http://domain.com" target="_blank">domain.com</a>". I have
attached my config file at this link:</span></font></div>
<div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><br>
</span></font></div>
<div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><a href="http://pastebin.com/0QRyC938" target="_blank">http://pastebin.com/0QRyC938</a></span><br>
</font></div>
<div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><br>
</span></font></div>
<div><font color="#000000" face="Consolas, Menlo, Monaco, Lucida
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style="font-size:12px;line-height:21px"><br>
</span></font></div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On 6 August 2015 at 05:00, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi Nabeel,
<div>Quick question; what is this destination ip? <span><a href="http://192.168.0.19:60912" target="_blank">192.168.0.19:60912</a>
?</span><span> - </span>Destination
User Agent Registered on OpenSIPS?</div>
<div>Can you share the opensips.cfg code snippet for this
call ?</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">
<div>
<div>On Wed, Aug 5, 2015 at 11:55 PM,
Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>></span>
wrote:<br>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div>
<div dir="ltr">
<div>Hi,</div>
<div><br>
</div>
<div>I am using the residential script generated
by 'make menuconfig', with UDP and NAT support
enabled. I added "alias=<a href="http://domain.com" target="_blank">domain.com</a>"
to the config because otherwise the UA did not
register with my domain (<a href="mailto:username@domain.com" target="_blank">username@domain.com</a>). When
I attempt to make a call, I see '408 Request
Timeout' in the sip trace and the call does
not connect. Please check the log/trace below
and advise how to fix this.</div>
<div><br>
</div>
<div>SIP trace:</div>
<div><br>
</div>
<a href="http://pastebin.com/u5h9qGNr" target="_blank">http://pastebin.com/u5h9qGNr</a><br>
<div><br>
</div>
<div>OpenSIPS log: </div>
<div><br>
</div>
<div><a href="http://pastebin.com/B8PUCKh0" target="_blank">http://pastebin.com/B8PUCKh0</a><br>
</div>
</div>
<br>
</div>
</div>
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