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    <tt>As per t</tt><tt>hat capture, I assume that </tt><tt>162.249.6.110
      is your server.  And there is nothing send further from that IP -
      only incoming traffic.</tt><tt><br>
    </tt><tt><br>
    </tt><tt>The next question - is this INVITE reaching your opensips
      script ? to be sure that the OS delivers the UDP packet to the
      opensips application.</tt><tt><br>
    </tt><tt><br>
    </tt><tt>Regards,</tt><br>
    <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
    <div class="moz-cite-prefix">On 06.05.2016 19:28, Nabeel wrote:<br>
    </div>
    <blockquote
cite="mid:CA+vx6K++zkM32rd7D+o5_+Uy9euqYB21N7ZFJyZCKC15oBxyzg@mail.gmail.com"
      type="cite">
      <p dir="ltr">The trace I posted earlier is what I see with tcpdump
        when attempting a call. There is no other INVITE shown in the
        trace: <a moz-do-not-send="true"
          href="http://pastebin.com/raw/C4iymTbh">http://pastebin.com/raw/C4iymTbh</a></p>
      <p dir="ltr">The trace seems to end abruptly in the middle of the
        SDP, so I think it could be due to packet fragmentation. </p>
      <div class="gmail_quote">On 6 May 2016 4:18 pm, "Bogdan-Andrei
        Iancu" &lt;<a moz-do-not-send="true"
          href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>&gt;
        wrote:<br type="attribution">
        <blockquote class="gmail_quote" style="margin:0 0 0
          .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <div bgcolor="#FFFFFF" text="#000000"> <tt>So that meas the
              INVITE never gets to the callee ?? maybe it is not
              properly routed . <br>
              <br>
              Do you see (with ngrep or tcpdump) the INVITE being sent
              out by opensips towards callee ?<br>
              <br>
              Regards,<br>
            </tt>
            <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
            <div>On 06.05.2016 12:56, Nabeel wrote:<br>
            </div>
            <blockquote type="cite">
              <p dir="ltr">Hi,</p>
              <p dir="ltr">Thanks for the idea about packet compression.
                By 'call fails to connect', I meant the call does not
                connect to the callee, ie. the callee's phone does not
                ring after the INVITE (despite using TURN server). </p>
              <p dir="ltr">This was a public WiFi network and that was
                all I could get at the time. I am using OpenSIPS version
                2.1.</p>
              <p dir="ltr">Nabeel</p>
              <div class="gmail_quote">On 6 May 2016 9:16 am,
                "Bogdan-Andrei Iancu" &lt;<a moz-do-not-send="true"
                  href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>&gt;

                wrote:<br type="attribution">
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  <div bgcolor="#FFFFFF" text="#000000"> <tt>Hi,<br>
                      <br>
                      Hard to analyze a call based on the INVITE packet
                      only :). Still the SIP signaling does not show any
                      ALG interference (also not sure if the capture was
                      done before or after the ALG). Also, what you mean
                      by "call fails" ?no reply, negative reply , no
                      audio ?<br>
                      <br>
                      Regards,<br>
                    </tt>
                    <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
                    <div>On 05.05.2016 22:35, Nabeel wrote:<br>
                    </div>
                    <blockquote type="cite">
                      <div dir="ltr">
                        <div class="gmail_extra"><br>
                        </div>
                        <div class="gmail_extra">Please check the
                          following SIP trace taken within a WiFi
                          network. The call fails to connect despite the
                          INVITE request and using a non-standard port.
                          Could this be caused by SIP ALG, or some
                          unopened RTP port on the router?</div>
                        <div class="gmail_extra"><br>
                        </div>
                        <div class="gmail_extra"><a
                            moz-do-not-send="true"
                            href="http://pastebin.com/raw/C4iymTbh"
                            target="_blank"><a class="moz-txt-link-freetext" href="http://pastebin.com/raw/C4iymTbh">http://pastebin.com/raw/C4iymTbh</a></a><br>
                        </div>
                      </div>
                      <br>
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                      <br>
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