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<h2><a href="https://wiki.asterisk.org/wiki/display/AST/Media+Overhaul">Media Overhaul</a></h2>
<h4>Page <b>edited</b> by <a href="https://wiki.asterisk.org/wiki/display/~kpfleming">Kevin P. Fleming</a>
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<h4>Changes (1)</h4>
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<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >** Allow re-negotiation during a call <br>** Allow negotiation to include codec attributes <br></td></tr>
<tr><td class="diff-changed-lines" >* Add support for SILK, CELT and other <span class="diff-deleted-words"style="color:#999;background-color:#fdd;text-decoration:line-through;">essentially 'free'</span> codecs <br></td></tr>
<tr><td class="diff-unchanged" >* Add support for variable sampling rates with conferencing <br>* Rework codec translator infrastructure to not be audio specific <br></td></tr>
<tr><td class="diff-snipped" >...<br></td></tr>
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</div> <h4>Full Content</h4>
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<ul>
<li><a href='#MediaOverhaul-ProjectRequirements'>1. Project Requirements</a></li>
<ul>
<li><a href='#MediaOverhaul-HighLevelRequirements'>1.1. High Level Requirements</a></li>
</ul>
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<h1><a name="MediaOverhaul-ProjectRequirements"></a>1. Project Requirements</h1>
<div class='panelMacro'><table class='warningMacro'><colgroup><col width='24'><col></colgroup><tr><td valign='top'><img src="/wiki/images/icons/emoticons/forbidden.gif" width="16" height="16" align="absmiddle" alt="" border="0"></td><td>This section is incomplete.</td></tr></table></div>
<h2><a name="MediaOverhaul-HighLevelRequirements"></a>1.1. High Level Requirements</h2>
<ul>
        <li>Rework codec representation in Asterisk
        <ul>
                <li>Define a real data structure that includes codec attributes instead of the simple current mapping of a bit field into codecs</li>
        </ul>
        </li>
        <li>Improve codec negotiation
        <ul>
                <li>Allow re-negotiation during a call</li>
                <li>Allow negotiation to include codec attributes</li>
        </ul>
        </li>
        <li>Add support for SILK, CELT and other codecs</li>
        <li>Add support for variable sampling rates with conferencing</li>
        <li>Rework codec translator infrastructure to not be audio specific</li>
</ul>
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