<div dir="ltr"><div>my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry...<br></div><div>better be safe than sorry....<br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <span dir="ltr">&lt;<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>&gt;</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">
    I mean you can use a private gist, but you will be publishing the
    link in a public email list.  In general I personally dont believe
    revealing ip addresses etc. is any problem - to put my money where
    my mouth is here is a gist link to an unaltered SIP trace on my
    server :)<br>
    <br>
    <a href="https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52" target="_blank">https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52</a><span class="HOEnZb"><font color="#888888"><br>
    <br>
    -Eric</font></span><div><div class="h5"><br>
    <br>
    <div>On 06/23/2016 12:23 PM, John Nash
      wrote:<br>
    </div>
    <blockquote type="cite">
      <div dir="ltr">Ok i am ready with logs. About gist may I use
        private option as traces have our IPs, user</div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Thu, Jun 23, 2016 at 10:32 PM, Eric
          Tamme <span dir="ltr">&lt;<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>&gt;</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"> Hey John,<br>
              <br>
              Please paste a full UNALTERED sip trace into a gist (<a href="http://gist.github.com" target="_blank">gist.github.com</a>) from the proxy
              servers perspective and provide a link so that we can see
              what comes in, and what goes out from both sides.<br>
              <br>
              EG: ngrep -qtd any -W byline port 5060<br>
              <br>
              This will show us the traffic that is leaving the proxy
              destined for the Freeswitch box, and what the freeswitch
              box sends back.<br>
              <br>
              Also - you can look in your browsers console log and
              provide the SIP trace from there in a seperate gist, so
              that we can see what opensips sends back up to your
              browser.<span><font color="#888888"><br>
                  <br>
                  -Eric</font></span>
              <div>
                <div><br>
                  <br>
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div><span style="font-size:12.8px">Am I using
                          correct sip.js example? I copied it to my
                          server and accessing it using https: (used
                          letsencrypt)</span></div>
                    </div>
                    <div class="gmail_extra"><br>
                      <div class="gmail_quote">On Thu, Jun 23, 2016 at
                        7:58 PM, Eric Tamme <span dir="ltr">&lt;<a href="mailto:eric@uphreak.com" target="_blank"></a><a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>&gt;</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"> 1. I
                            would suggest using SIP.js - <a href="https://github.com/onsip/SIP.js" target="_blank"></a><a href="https://github.com/onsip/SIP.js" target="_blank">https://github.com/onsip/SIP.js</a>
                            it is a much more active project that
                            sipml5.<br>
                            <br>
                            2. Im guessing that you are not properly
                            passing flags to RTPEngine.  If you want to
                            have DTLS-SRTP between the browser, and
                            plain RTP/AVP between RTPEngine and
                            freeswitch, you need to &quot;offer&quot; rtp/avp to
                            freeswitch, and &quot;answer&quot; dtls-srtp back up
                            to the browser.<br>
                            <br>
                            the offer to freeswitch would be:  <br>
                            <pre>        $var(rtpengine_flags) = &quot;RTP/AVP replace-session-connection replace-origin ICE=remove&quot;;

</pre>
                            and the answer back up to the browswer would
                            be:<br>
                            <br>
                            <pre>        $var(rtpengine_flags) = &quot;UDP/TLS/RTP/SAVPF ICE=force&quot;;</pre>
                            <br>
                            -Eric
                            <div>
                              <div><br>
                                <br>
                                <br>
                                <div>On 06/23/2016 08:20 AM, John Nash
                                  wrote:<br>
                                </div>
                              </div>
                            </div>
                            <blockquote type="cite">
                              <div>
                                <div>
                                  <div dir="ltr">I am following <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2" target="_blank"></a><a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2" target="_blank">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2</a>
                                    and trying to test a call 
                                    <div><br>
                                    </div>
                                    <div>sipml5 -----------&gt;Opensips
                                      + rtpengine --------&gt; SIP end
                                      point (Freeswitch)<br>
                                      <div><br>
                                      </div>
                                      <div>But I do not have any audio
                                        on both sides. I see this error
                                        at rtpengine log &quot;SRTP output
                                        wanted, but no crypto suite was
                                        negotiated&quot;<br>
                                      </div>
                                    </div>
                                    <div><br>
                                    </div>
                                    <div>Anyone tested this scenario
                                      positive?</div>
                                  </div>
                                  <br>
                                  <fieldset></fieldset>
                                  <br>
                                </div>
                              </div>
                              <pre>_______________________________________________
Users mailing list
<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
                            </blockquote>
                            <br>
                          </div>
                          <br>
_______________________________________________<br>
                          Users mailing list<br>
                          <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br>
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                          <br>
                        </blockquote>
                      </div>
                      <br>
                    </div>
                    <br>
                    <fieldset></fieldset>
                    <br>
                    <pre>_______________________________________________
Users mailing list
<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
                  </blockquote>
                  <br>
                </div>
              </div>
            </div>
            <br>
            _______________________________________________<br>
            Users mailing list<br>
            <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br>
            <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" rel="noreferrer" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
            <br>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
Users mailing list
<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
    </blockquote>
    <br>
  </div></div></div>

<br>_______________________________________________<br>
Users mailing list<br>
<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" rel="noreferrer" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
<br></blockquote></div><br></div>