<div dir="ltr"><p dir="ltr">Hi Hamid, </p>
<p dir="ltr">I am not using the location table, I am using memory only to store the contacts. </p>
<p dir="ltr">If you use TCP within the same network, where a TURN server is not required (TURN server not set in Linphone settings), I found that the calls then work depending on what the network allows. Within the same Wi-Fi network, in my case audio calls work but not video calls. But using 3G/4G only, with or without a TURN server set in Linphone settings, calls do not work at all. This is why I think the behaviour of TURN server has something to do with this. To those people stating that TCP works with Linphone, please test over 3G/4G where a TURN server might be required. </p><p dir="ltr"><br></p>
<div class="gmail_quote">On 2 Feb 2016 7:07 am, "Hamid Hashmi" <<a href="mailto:hamid2kviii@hotmail.com" target="_blank">hamid2kviii@hotmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr"><font color="#2672EC">Nabeel I have been using Linphone 3.6.1 with opensips for a long time. And its working fine on both UDP and TCP. I have gone through your logs, there is log line </font><span style="color:rgb(51,51,51);font-family:Consolas,'Liberation Mono',Menlo,Courier,monospace;font-size:11.9px;line-height:19.04px;background-color:rgba(0,0,0,0.0392157)">DBG:tm:matching_3261: RFC3261 transaction matching failed</span><font color="#2672EC"> Please check contact of your "to number" in location table. <br></font><br><br><b><font color="#5133ab" face="Times New Roman" size="3"><i>Hamid R. Hashmi</i></font></b><div><font size="2">Software Engineer - VoIP</font></div><div><font color="#008a17" size="2" style="font-size:10pt">Vopium A/S</font></div><br><br><div><hr>Date: Mon, 1 Feb 2016 19:35:53 +0000<br>From: <a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a><br>To: <a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>; <a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a><br>Subject: Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP<br><br><div dir="ltr">Hi Bogdan,<div><br></div><div>Below is the requested log for TCP call attempt. User 1111 is trying to call user 2222 via OpenSIPS server <span style="color:rgb(51,51,51);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:24px;background-color:rgb(248,248,248)">162.248.6.120</span> :</div><div><br></div><div><a href="http://pastebin.com/UQ9mEemd" target="_blank">http://pastebin.com/UQ9mEemd</a><br></div></div><div><br><div>On 1 February 2016 at 09:41, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><blockquote style="border-left:1px #ccc solid;padding-left:1ex">
<div>
<tt>Hi,<br>
<br>
I strongly suggest to look into the opensips logs and see what
opensips try to do with the call. Based on your saying (that you
see a timeout), I suspect your OpenSIPS tries to deliver the call
over TCP to a destination which does not listen on TCP.<br>
If you do not know hoe to interpret the logs, run opensips in
debug=4 mode, upload the logs corresponding the INVITE execution
and provide the link.<br>
<br>
Regards,<br>
</tt>
<pre>Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre><div><div>
<div>On 31.01.2016 16:28, Nabeel wrote:<br>
</div>
</div></div><blockquote><div><div>
<div dir="ltr">Without using alias=<a href="http://domain.com" target="_blank">domain.com</a>, TCP still does not
work. My initial request for someone to test this using
Linphone remains. Please test and let me know if you can call
using TCP with OpenSIPS listening on an IP address.</div>
<div><br>
<div>On 31 January 2016 at 09:28, Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>></span>
wrote:<br>
<blockquote style="border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">On further testing, using the IP address
instead of the domain name in the URI setting of Linphone
works with TCP, so I think this might be to do with
SRV/NAPTR records associated with the domain.</div>
<div>
<div>
<div><br>
<div>On 31 January 2016 at 08:29,
Nabeel <span dir="ltr"><<a href="mailto:nabeelshikder@gmail.com" target="_blank"></a><a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>></span>
wrote:<br>
<blockquote style="border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><span style="font-size:12.8px">Hello,</span>
<div style="font-size:12.8px"><br>
</div>
<div style="font-size:12.8px">There seems to be
a problem with calls over TCP using Linphone,
and since Linphone is a popular open source
application, I would like someone to please
verify this problem. Calls work fine with
Linphone over UDP, but after registering with
TCP using the same credentials, calls do not
connect at all and lead to a request timeout.
A request timeout does not say much about the
cause, but in this case I suspect there is
something wrong with TCP on the server side. I
would like someone to please install Linphone
on your phone and connect to your OpenSIPS
server using UDP and TCP. Please report here
if the calls work over both transports. </div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
<br>
<fieldset></fieldset>
<br>
</div></div><span><pre>_______________________________________________
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</blockquote></div><br></div>
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