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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/~russell">Russell Bryant</a>
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<h4>Changes (1)</h4>
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<tr><td class="diff-unchanged" >{warning} <br> <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;">h2. Relevant Problems That Exist Today <br> <br>* Codec negotiation (both with Asterisk, and across a bridge) <br>** Support for audio codecs with attributes (SILK) <br>** Support for video codecs with attributes <br>* Limitation on the number of codecs Asterisk can support <br>* Translation paths are audio specific (with no concept of attributes) <br>* There is no way to renegotiate codecs after a call is up <br>* Conferencing is limited to 8 kHz <br>* There is no way to easily get Asterisk to pass through a media type that it does not understand (proprietary data) <br>* Once Asterisk supports codecs with attributes, users will need to be able to specify codecs with attributes <br>* Asterisk is not able to handle a call with more than one audio/video/text stream (only one stream per type). <br>* Asterisk has no RTCP support relevant to audio and video synchronization <br>* Asterisk does not support Gtalk video <br> <br></td></tr>
<tr><td class="diff-unchanged" >h2. High Level Requirements <br> <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-RelevantProblemsThatExistToday'>1.1. Relevant Problems That Exist Today</a></li>
<li><a href='#MediaOverhaul-HighLevelRequirements'>1.2. High Level Requirements</a></li>
</ul>
</ul></div>
<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-RelevantProblemsThatExistToday"></a>1.1. Relevant Problems That Exist Today</h2>
<ul>
<li>Codec negotiation (both with Asterisk, and across a bridge)
<ul>
<li>Support for audio codecs with attributes (SILK)</li>
<li>Support for video codecs with attributes</li>
</ul>
</li>
<li>Limitation on the number of codecs Asterisk can support</li>
<li>Translation paths are audio specific (with no concept of attributes)</li>
<li>There is no way to renegotiate codecs after a call is up</li>
<li>Conferencing is limited to 8 kHz</li>
<li>There is no way to easily get Asterisk to pass through a media type that it does not understand (proprietary data)</li>
<li>Once Asterisk supports codecs with attributes, users will need to be able to specify codecs with attributes</li>
<li>Asterisk is not able to handle a call with more than one audio/video/text stream (only one stream per type).</li>
<li>Asterisk has no RTCP support relevant to audio and video synchronization</li>
<li>Asterisk does not support Gtalk video</li>
</ul>
<h2><a name="MediaOverhaul-HighLevelRequirements"></a>1.2. 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>
</div>
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