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1. I would suggest using SIP.js - <a class="moz-txt-link-freetext" href="https://github.com/onsip/SIP.js">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 "offer"
rtp/avp to freeswitch, and "answer" dtls-srtp back up to the
browser.<br>
<br>
the offer to freeswitch would be: <br>
<pre class="escaped"> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
</pre>
and the answer back up to the browswer would be:<br>
<br>
<pre class="escaped"> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";</pre>
<br>
-Eric<br>
<br>
<br>
<div class="moz-cite-prefix">On 06/23/2016 08:20 AM, John Nash
wrote:<br>
</div>
<blockquote
cite="mid:CAA4rGPwTD=g0RO2ifcnzE7+UL-j4YYxye++A_bgJFW8LLcrEWA@mail.gmail.com"
type="cite">
<div dir="ltr">I am following <a moz-do-not-send="true"
href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2</a>
and trying to test a call
<div><br>
</div>
<div>sipml5 ----------->Opensips + rtpengine --------> 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 "SRTP output wanted, but no crypto
suite was negotiated"<br>
</div>
</div>
<div><br>
</div>
<div>Anyone tested this scenario positive?</div>
</div>
<br>
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</pre>
</blockquote>
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