<div dir="ltr">I double checked my rtpengine offer answer calls and now using <a href="https://github.com/onsip/sipjs-examples/tree/master/demo-phone">https://github.com/onsip/sipjs-examples/tree/master/demo-phone</a> but I face same issue (no audio either side) and error <span style="font-size:12.8px"> &quot;SRTP output wanted, but no crypto suite was negotiated&quot; Rtpengine also I updated to the latest now.</span><div><span style="font-size:12.8px"><br></span></div><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">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">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 class="h5"><br>
    <br>
    <br>
    <div>On 06/23/2016 08:20 AM, John Nash
      wrote:<br>
    </div>
    </div></div><blockquote type="cite"><div><div class="h5">
      <div dir="ltr">I am following <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>_______________________________________________
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</pre>
    </blockquote>
    <br>
  </div>

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<br></blockquote></div><br></div>