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    <p>Hello,</p>
    <p>I see this was discussed further on rtpengine issue tracker. Did
      using a newer version of rtpengine made it work?</p>
    <p>The typical hint I have is to look at javascript console in the
      browser, there should be logs printed when some dtls negotiation
      fails.<br>
    </p>
    <p>Cheers,<br>
      Daniel<br>
    </p>
    <div class="moz-cite-prefix">On 05.05.20 02:21, Chirag Desai wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CAFbhCMGBxJz17a5=jSu8ONyZq6LdFcga_STg1mU57uaxcPdXrg@mail.gmail.com">
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              <div><br>
                Hi all,</div>
            </div>
            <div style="font-family:sans-serif" dir="auto"><br>
            </div>
            <div style="font-family:sans-serif" dir="auto">I have
              configured Kamailio for WebSockets following this guide as
              an example: <a
href="https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg"
                moz-do-not-send="true">https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg</a></div>
            <div style="font-family:sans-serif" dir="auto"><br>
            </div>
            <div style="font-family:sans-serif" dir="auto">With sip.js
              and jssip I'm able to initiate a call from WebRTC to SIP
              and establish a call successfully.</div>
            <div style="font-family:sans-serif" dir="auto"><br>
            </div>
            <div style="font-family:sans-serif" dir="auto">The issue
              arises when I try to receive a call from a SIP device. In
              this case the call establishes but there is no audio in
              either direction.</div>
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            </div>
            <div style="font-family:sans-serif" dir="auto">I *think* the
              issue is with RTP Engine and I've raised a bug there, but
              I'm not sure why it is misbehaving <a
                href="https://github.com/sipwise/rtpengine/issues/983"
                moz-do-not-send="true">https://github.com/sipwise/rtpengine/issues/983</a>.
              There are some logs from RTP engine posted here.</div>
            <div style="font-family:sans-serif" dir="auto"><br>
            </div>
            <div style="font-family:sans-serif" dir="auto">The sip
              device communicates with Kamailio over UDP / RTP, nothing
              is encrypted.</div>
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            </div>
            <div style="font-family:sans-serif" dir="auto">I would
              appreciate any guidance.</div>
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            </div>
            <div style="font-family:sans-serif" dir="auto">Thanks in
              advance,</div>
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            </div>
            <div style="font-family:sans-serif" dir="auto">C</div>
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            .8ex;border-left:1px #ccc solid;padding-left:1ex">
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      <pre class="moz-quote-pre" wrap="">_______________________________________________
Kamailio (SER) - Users Mailing List
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>
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</pre>
    </blockquote>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda">www.linkedin.com/in/miconda</a>
Funding: <a class="moz-txt-link-freetext" href="https://www.paypal.me/dcmierla">https://www.paypal.me/dcmierla</a></pre>
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