<div dir="ltr">I tested and it seems like I do have <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146">https://issues.asterisk.org/jira/browse/ASTERISK-24146</a> but in a different way. If I take more than 7s to answer the call, I don't get audio for a few seconds (about 3), after that it works okay.<div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-08-28 10:43 GMT-03:00 Marek Červenka <span dir="ltr"><<a href="mailto:cervajs@fpf.slu.cz" target="_blank">cervajs@fpf.slu.cz</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
  
    
  
  <div text="#000000" bgcolor="#FFFFFF">
    <div>are you sure you dont have this
      problem?<br>
      <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-24146</a><br>
      <br>
      i'm now fighting with<br>
      <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24602" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-24602</a><br>
      <br>
      Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a):<br>
    </div><div><div class="h5">
    <blockquote type="cite">
      <div dir="ltr">I have it working now!
        <div><br>
        </div>
        <div><b>I had to install Asterisk 13 with PJSIP support.That's
            important, even if you won't use PJSIP.</b> Then I did this
          configuration, which is working fine under NAT:</div>
        <div><br>
        </div>
        <div><b>sip.conf:</b></div>
        <div>
          <div>[6001]</div>
          <div>type=friend</div>
          <div>secret=REDACTED</div>
          <div>host=dynamic</div>
          <div>context=interno</div>
          <div>disallow=all</div>
          <div>;allow=alaw,h263,h264,vp8</div>
          <div>allow=g722</div>
          <div>dtmf=auto</div>
          <div>videosupport=yes</div>
          <div>transport=ws,udp</div>
          <div>avpf=yes</div>
          <div>callerid="WebRTC" <6001></div>
          <div>encryption=yes</div>
          <div>qualify=yes</div>
          <div>directmedia=no</div>
          <div>nat=force_rport,comedia</div>
          <div>icesupport=yes</div>
          <div>dtlsenable=yes ; Tell Asterisk to enable DTLS for this
            peer</div>
          <div>dtlsverify=no ; Tell Asterisk to not verify your DTLS
            certs</div>
          <div>dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell
            Asterisk where your DTLS cert file is</div>
          <div>dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell
            Asterisk where your DTLS private key is</div>
          <div>dtlssetup=actpass ; Tell Asterisk to use actpass SDP
            parameter when setting up DTLS</div>
        </div>
        <div><br>
        </div>
        <div><b>rtp.conf:</b></div>
        <div>
          <div>icesupport=true</div>
          <div>stunaddr=<a href="http://stun.l.google.com:19302" target="_blank">stun.l.google.com:19302</a></div>
        </div>
        <div><br>
        </div>
        <div><b>res_stun_monitor.conf:</b></div>
        <div>stunaddr = <a href="http://stun.l.google.com:19302" target="_blank">stun.l.google.com:19302</a>
             ; Address of the STUN server to query.<b><br>
          </b></div>
        <div>stunrefresh = 30<br>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">2015-08-12 5:23 GMT-03:00 Marek
          Červenka <span dir="ltr"><<a href="mailto:cervajs@fpf.slu.cz" target="_blank">cervajs@fpf.slu.cz</a>></span>:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Dne
            11.8.2015 v 12:18 Joshua Colp napsal(a):<span><br>
              <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                Vinicius Fontes wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  I'm having the same issue! The difference in my case
                  is Asterisk server<br>
                  has a public IPv4 and the browser is behind a single
                  NAT.<br>
                  <br>
                  I'm forwarding my configuration below (which I posted
                  previously on<br>
                  asterisk-users).<br>
                  <br>
                  How can we debug ICE negotiation?<br>
                </blockquote>
                <br>
                You have to do a packet capture, look at the exchange in
                Wireshark, and see how the negotiation flows. It
                requires a basic understanding of ICE.<br>
                <br>
              </blockquote>
              <br>
            </span>
            it looks like we are facing this problem <a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146" rel="noreferrer" target="_blank"></a><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-24146</a>
            too<br>
            if we use "[]" in sipml5 expert config "To disable TURN/STUN
            to speedup ICE candidates gathering you can use an empty
            array. e.g. []."<br>
            it works better<span><br>
              <br>
              <br>
              <br>
              <br>
              -- <br>
              ---------------------------------------<br>
              Marek Cervenka<br>
              =======================================<br>
              <br>
              <br>
            </span>
            <div>
              <div>
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              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
    </blockquote>
    <br>
    <br>
    <pre cols="72">-- 
---------------------------------------
Marek Cervenka
=======================================
</pre>
  </div></div></div>

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