<div dir="ltr">Hi Nabeel,<div><br></div><div>Which test are you referring to ?</div><div>Have you tried adding the following by Bogdan in your .cfg file ?</div><div><br></div><div><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></div><div><tt><br></tt></div>The above worked or you haven't tried them ?<div><tt><br></tt></div><div>Why do you've this at line 294 in your script: <span style="color:rgb(0,0,0);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px"> </span><span style="color:rgb(0,0,0);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px">if ( 0 ) setflag(TCP_PERSISTENT);</span>?? Just above or after this you can write this: <tt>setbflag(SIP_PING_FLAG); </tt>as mentioned by Bogdan.</div><div><tt><br></tt></div>BR,<br>Sammy</div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Aug 7, 2015 at 6:21 AM, 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 Nabeel,<br>
<br>
Using TCP gives an advantage over UDP as being connection
oriented, it force the NAT pinhole to stay open. Nevertheless,
using the SIP pinging with UDP should also have fixed the problem.<br>
<br>
I will update the default cfg to use SIP pinging rather than
simple UDP pinging.<br>
<br>
Thanks and Regards,<br>
</tt><span class="">
<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 07.08.2015 12:32, Nabeel wrote:<br>
</div>
<blockquote type="cite">
<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><tt>Reg</tt><tt>ards,</tt><br>
<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>
<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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