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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>
</h4>
<br/>
<h4>Changes (5)</h4>
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<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >* Asterisk does not support Gtalk video <br> <br></td></tr>
<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">h2. High Level Requirements <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;">h2. Phase 1 Requirements <br></td></tr>
<tr><td class="diff-unchanged" > <br></td></tr>
<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">* Rework codec representation in Asterisk <br>** Define a real data structure that includes codec attributes instead of the simple current mapping of a bit field into codecs <br>* Improve codec negotiation <br>** Allow re-negotiation during a call <br>** Allow negotiation to include codec attributes <br>* Add support for SILK, CELT and other codecs <br>* 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-added-lines" style="background-color: #dfd;">Rework media representation completely across all of Asterisk, while maintaining existing functionality. Only add functionality that is required to exercise what has been done. <br></td></tr>
<tr><td class="diff-unchanged" > <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;">* Design a new way to represent codecs <br>** translation interface <br>** capabilities <br>** ast_frame handling <br>** Initial call setup codec negotiation <br>** everything that touches media ... <br>* Exercise what we have done so far <br>** Add support for SILK (and its attributes) <br>** Add support for H.264 attributes <br>* Custom format definitions with attributes (for setting preferences) <br> <br>h2. Phase Later Requirements <br> <br>* Codec re-negotiation <br>* Improved conferencing (dynamic sample rate support) <br>* Video transcoding (an implementation that proves it works) <br>* A&V sync in RTCP (research required) <br>* GTalk Video Support <br>* Support for unknown media types for pass-through (research required) <br>* Support for more than one stream of the same type (audio/video/text) <br> <br></td></tr>
<tr><td class="diff-unchanged" >{numberedheadings} <br></td></tr>
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</div> <h4>Full Content</h4>
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<div class="error"><span class="error">Error formatting macro: toc: java.lang.NoClassDefFoundError: org/w3c/tidy/EntityTable</span> </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-Phase1Requirements"></a>1.2. Phase 1 Requirements</h2>
<p>Rework media representation completely across all of Asterisk, while maintaining existing functionality. Only add functionality that is required to exercise what has been done.</p>
<ul>
<li>Design a new way to represent codecs
<ul>
<li>translation interface</li>
<li>capabilities</li>
<li>ast_frame handling</li>
<li>Initial call setup codec negotiation</li>
<li>everything that touches media ...</li>
</ul>
</li>
<li>Exercise what we have done so far
<ul>
<li>Add support for SILK (and its attributes)</li>
<li>Add support for H.264 attributes</li>
</ul>
</li>
<li>Custom format definitions with attributes (for setting preferences)</li>
</ul>
<h2><a name="MediaOverhaul-PhaseLaterRequirements"></a>1.3. Phase Later Requirements</h2>
<ul>
<li>Codec re-negotiation</li>
<li>Improved conferencing (dynamic sample rate support)</li>
<li>Video transcoding (an implementation that proves it works)</li>
<li>A&V sync in RTCP (research required)</li>
<li>GTalk Video Support</li>
<li>Support for unknown media types for pass-through (research required)</li>
<li>Support for more than one stream of the same type (audio/video/text)</li>
</ul>
</div>
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