<div dir="ltr"><div>SamyGo,</div><div><br></div><div>I tried the test you suggested but it did not work.</div><div><br></div><div>Bogdan,</div><div><br></div><div>After switching to TCP and adding your suggestions along with the following parameters, the timeout errors seem to be resolved:</div><div><br></div><div><b>tcp_async=1</b><br></div><div><b>tcp_connect_timeout=99999</b></div><div><b>tcp_send_timeout=99999</b></div><div><b><div>modparam("nathelper", "nortpproxy_str", "a=nortpproxy:yes\r\n")</div><div><br></div></b></div><div>I'm not sure in exactly what combination works best, but perhaps these should be included in the default residential script?</div><div><b><br></b></div><div>Thanks for the help... I'll be back with more questions.</div><div><br></div><div><br></div><div class="gmail_extra"><div class="gmail_quote">On 6 August 2015 at 15:41, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</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">
<tt>Hi Nab</tt><tt>eel,</tt><tt><br>
</tt><tt><br>
</tt><tt>This time the SIP trace looks ok - the message is sent to a
public IP and not a private one (as shown in the prev capture).
Also the usrloc data looks ok, telling that the NAT traversal work</tt><tt>s
ok.</tt><tt><br>
</tt><tt>As you have Mobile Data, some operators do filter SIP, so
not sure what say.</tt><tt><br>
</tt><tt><br>
</tt><tt>Do you see the sip messages going back and forward during
the </tt><tt>registration process ? </tt><tt><br>
</tt><tt><br>
</tt><tt>An issue may be the fact the default script does non-SIP
pinging </tt><tt>(which is unidirectional), so the NAT may close.
Add the followings:</tt><tt><br>
</tt><tt>1) </tt><tt>modparam("nathelper", "sipping_bflag",
"SIP_PING_FLAG")</tt><tt><br>
</tt><tt> </tt><tt>modparam("nathelper", "sipping_from",
<a href="mailto:sip:pinger@xx.xx.xx.xx" target="_blank">"sip:pinger@xx.xx.xx.xx"</a>)</tt><tt><br>
<br>
</tt><tt>2) setbflag(</tt><tt>SIP_PING_FLAG); before doing save()</tt><tt><br>
</tt><tt><br>
</tt><tt>You should see OpenSIPS doing </tt><tt>keep alive with
OPTIONS requests.<br>
</tt><span class=""><tt>Reg</tt><tt>ards,</tt><br>
<tt></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>
</span><div><div class="h5"><div>On 06.08.2015 14:59, Nabeel wrote:<br>
</div>
<blockquote type="cite">
<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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