<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"> "SRTP output wanted, but no crypto suite was negotiated" 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"><<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></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 "offer"
rtp/avp to freeswitch, and "answer" dtls-srtp back up to the
browser.<br>
<br>
the offer to freeswitch would be: <br>
<pre> $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> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";</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 ----------->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>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
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</pre>
</blockquote>
<br>
</div>
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