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    <div class="moz-cite-prefix">Dne 15.9.2015 v 13:37 Marek Červenka
      napsal(a):<br>
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    <blockquote cite="mid:55F802FC.3010207@fpf.slu.cz" type="cite">
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      hi,<br>
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        <div><span>i'm fighting with webrtc for 14 days</span></div>
        <div><span>reporting my experience to minimize number of crazy
            asterisk users <br>
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            i have working webrtc with simpl5 + asterisk 13 + pjproject
            2.4.5 + chan_pjsip + secure websockets + secure audio +
            audio in both ways<br>
            <br>
            problems<br>
            first, i needed run chan_sip for old hard phones and wss
            with chan_pjsip only for webrtc. this is possible with patch
            from<br>
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                href="https://issues.asterisk.org/jira/browse/ASTERISK-24106"><a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-24106">https://issues.asterisk.org/jira/browse/ASTERISK-24106</a></a></span><br>
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            chan_sip is not usable for webrtc because of<br>
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        <div><span><a moz-do-not-send="true"
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              href="https://issues.asterisk.org/jira/browse/ASTERISK-24602">https://issues.asterisk.org/jira/browse/ASTERISK-24602</a><br>
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    <br>
    this is the blocking issue
    <a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-24146">https://issues.asterisk.org/jira/browse/ASTERISK-24146</a> <br>
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    <pre class="moz-signature" cols="72">-- 
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Marek Cervenka
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