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    <p>What codecs are supported by your grandstream? Isn't the g711 in
      the group?</p>
    <p>Cheers,<br>
      Daniel<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 19/05/16 01:51, Moacir Ferreira
      wrote:<br>
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      <div dir="ltr">I did not dig into the problem but on my tests I
        saw that my (old) Grandstream phone was refusing the call for
        not having a compatible codec to talk with the offered ones by
        the browser (Firefox). Being this the case, I guess I must
        include a translator, and all routing logic, in between the
        callers. It points to Asterisk that I would like to avoid for
        now. But I guess this is not a problem that only affects me.
        Someone else must have faced this before. So the question still
        open: What solution would be recommended for such case?<br>
        <br>
        Cheers,<br>
        Moacir<br>
        <br>
        <div>> To: <a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
          > From: <a class="moz-txt-link-abbreviated" href="mailto:rfuchs@sipwise.com">rfuchs@sipwise.com</a><br>
          > Date: Wed, 18 May 2016 19:03:10 -0400<br>
          > Subject: Re: [SR-Users] Browser WebRTC transcoder<br>
          > <br>
          > On 18/05/16 04:57 PM, Moacir Ferreira wrote:<br>
          > > Hey Daniel,<br>
          > ><br>
          > > If you say so, you probably right... I did not try
          it because on the<br>
          > > sipwise GitHub
          (<a class="moz-txt-link-freetext" href="https://github.com/sipwise/rtpengine">https://github.com/sipwise/rtpengine</a>) they mention:<br>
          > ><br>
          > > /"Rtpengine does not (yet) support:/<br>
          > > //<br>
          > ><br>
          > > * /Repacketization or transcoding/<br>
          > <br>
          > This refers to translating one audio codec into another
          (e.g. opus to <br>
          > PCM). Translating between RTP and SRTP (i.e. encrypting
          and decrypting) <br>
          > is supported.<br>
          > <br>
          > Cheers<br>
          > <br>
          > _______________________________________________<br>
          > SIP Express Router (SER) and Kamailio (OpenSER) -
          sr-users mailing list<br>
          > <a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
          >
          <a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
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      <pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
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    <br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
Kamailio World Conference, Berlin, May 18-20, 2016 - <a class="moz-txt-link-freetext" href="http://www.kamailioworld.com">http://www.kamailioworld.com</a></pre>
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