<html>
<head>
<base href="https://wiki.asterisk.org/wiki">
<link rel="stylesheet" href="/wiki/s/2030/1/7/_/styles/combined.css?spaceKey=AST&forWysiwyg=true" type="text/css">
</head>
<body style="background: white;" bgcolor="white" class="email-body">
<div id="pageContent">
<div id="notificationFormat">
<div class="wiki-content">
<div class="email">
<h2><a href="https://wiki.asterisk.org/wiki/display/AST/Roadmap">Roadmap</a></h2>
<h4>Page <b>edited</b> by <a href="https://wiki.asterisk.org/wiki/display/~russell">Russell Bryant</a>
</h4>
<div id="versionComment">
<b>Comment:</b>
copy in development wishlist from AstriDevCon<br />
</div>
<br/>
<h4>Changes (5)</h4>
<div id="page-diffs">
<table class="diff" cellpadding="0" cellspacing="0">
<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >h1. Future Development Wishlist <br> <br></td></tr>
<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">* Make ConfBridge() as feature complete as MeetMe() <br>* SIP SIMPLE messaging support <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;">The items in this list primarily came from [AstriDevCon 2010]. <br> <br>h2. (P0) <br> <br>P0 are committed projects that are going to get done. <br> <br>* [AST:T.38 Gateway] (Digium) <br>* Performance of State Change Processing (Stefan Schmidt) <br>** (work is already being done on this front) <br>* SIP path support (Olle) <br>** (first generation of code exists, needs more work, simple patch, going to get it done, needs an extra field in astdb; helps when there are 2 or more load balancing proxies in front of asterisk, when you'd like the call to be able to get back to Asterisk; see https://reviewboard.asterisk.org/r/991/) <br>** [https://issues.asterisk.org/view.php?id=18223] <br>* Group variables (Kobaz) <br>** (on review board, in progress) <br>* Pre-Dial (Kobaz) <br>** (practically done, or something) <br>* Distributed extension state using SIP (Olle) <br>** (resources in place, doing it, 1.4 done before Christmas, project pinana) <br>* Manager event docs (Paul Belanger) <br>* Cross-platform documentation (Ben Klang) <br>** (caveats for using Asterisk on operating system xyz; pull a PDF of the Wiki documentation into the source, don't forget to include basic installation information, and do it all in .txt - Ben) <br>* Fix libs to optionally init OpenSSL (Digium) <br>** (or use existing tools; sort of a bug) <br>* Make ast_channel an opaque type (Digium) <br> <br>h2. (P1) <br> <br>P1 is the highest priority. <br> <br>* [Codecs (SILK, Opal), Media Negotiation|AST:Media Overhaul] (Digium) <br>* RTCP (Olle) <br>** Pinefrog; Work to be done - Ported to trunk, added to CEL <br>* Conferencing that supports a new magic media (Digium) <br>** higher sampling rates <br> <br> <br>h2. (P2) <br> <br>- Async DNS (TCP DNS and use a good resolver) <br>- Named ACLs (deluxepine) <br>- [AST:SIP Security Events] <br>- Light weight means of holding NAT open in SIP (less complex than current qualify, Consider it done) <br>- IPv6 for the restivus (IAX, Jabber/XMPP/Gtalk, Manager, etc.) <br>- ConfBridge feature complete with MeetMe <br>- Support sound file containers (matroska) <br> <br>h2. (P3) <br> <br>* Who hung up? (there's a branch, shouldn't take too much time - Olle) <br>* Unique identifier for filtering log data to a call <br>** (finishing what was already begun w/ Clod's project, CLI filtering; should take a look at what Stephan from Unlimitel.ca's created) <br> <br> <br>h2. (P4, Simon's features) <br> <br>* Multiple SIP Sockets <br>** (Listen on multiple ports or on multiple interfaces, but not all; also set binding for RTP)...alternate idea / solution would be to make Asterisk capable of loading multiple SIP profiles, it might be easier <br>* Multiple DNS results <br>** (need to be able to traverse a list of DNS results, rather than just getting back one result) <br>* ICE-lite <br>** (no code, responding correctly to ICE connectivity checks (STUN multiplexed on the RTP port) and understanding the SDP); it makes NAT traversal work for clients that do ICE; also addressed lightweight NAT refresh) <br> <br>h2. (P5) <br> <br>* AstDB replacement <br>** (realtime, there's code, nearly ready) <br>* SIP identity <br>** (on reviewboard; needs to be forward ported; important for organizations w/ federated identities; a requirement for DTLS SRTP; not widely deployed) <br></td></tr>
<tr><td class="diff-changed-lines" >* RTMP <span class="diff-added-words"style="background-color: #dfd;">client</span> channel driver <br></td></tr>
<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">* XML-ify AMI <br>* Unify AMI and CLI support so that capabilities (options) are the same across both <br>* Zero Restart Configuration Management <br>* Alternate Language (PHP, Perl) dialplan interpreters <br>* XMPP call initialization <br>* Multiple address bindings per transport in chan_sip <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;"> <br> <br>h2. (P6) <br> <br>* Structured identifiers for errors <br>** (tag an error message with a unique string, specific to the error message and where it came from; should be alphanumeric to keep them short) <br>* AMI SetVar, Context limits <br>** (there's code already...Olle has it) <br>* AMI filters on demand <br>* DTLS SRTP <br>** (not likely to be widely deployed in the next 12 months) <br> <br> <br>h2. (P7, not kobaz) <br> <br>* Asterisk register for XMPP account (Leif) <br>* Write a Specification for AMI (not kobaz) <br>* Multiple TLS server certs <br>** (1 socket, requires support by OpenSSL; simpler to implement than multiple SIP profiles; don't know if any clients use it yet; needs more research) <br> <br>h2. (P8, nice to have) <br> <br>- Make resource modules that talk to DBs attempt reconnects <br>- Apple's new file streaming format, derived from .m3u <br>- Make MixMonitor and Monitor feature compatible <br> <br> <br>h2. (P?, Research Required) <br> <br>* New app_queue (as if? no, seriously? talking about this scares Russell) <br>* Identify and fix all bugs in AMI <br>* Broadsoft or Dialog Info shared line appearance (SLA) support <br>** (Tabled for later discussion) <br>* LDAP from within the dialplan <br>** (we may already have it, needs research to see if the realtime driver does what's desired - Leif) <br>* Device state normalization <br>* Anything DB over HTTP(s) with failover handling <br>* Use a channel as a MoH Source <br>* Kill Masquerades <br>* Bridging thread pool <br>* Threadify chan_sip <br>* Export ISDN ROSE information up to Asterisk channels <br></td></tr>
</table>
</div> <h4>Full Content</h4>
<div class="notificationGreySide">
<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 content of this page is a draft and is subject to change at any time.</td></tr></table></div>
<h1><a name="Roadmap-RelatedLinks"></a>Related Links</h1>
<ul>
<li>The <a href="https://issues.asterisk.org/roadmap_page.php" class="external-link" rel="nofollow">Roadmap Page</a> on the <a href="https://issues.asterisk.org" class="external-link" rel="nofollow">issue tracker</a> has information about various issues and patches that have been targeted for a specific Asterisk release.</li>
</ul>
<ul>
<li>The <a href="http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup" class="external-link" rel="nofollow">CHANGES</a> file in Asterisk lists new features that have been added to each version.</li>
</ul>
<h1><a name="Roadmap-Asterisk1.10"></a>Asterisk 1.10</h1>
<p><b>Committed Features</b> - developed by Digium</p>
<ul>
<li><a href="/wiki/display/AST/Media+Overhaul" title="Media Overhaul">Media Overhaul</a></li>
<li><a href="/wiki/display/AST/SIP+Security+Events" title="SIP Security Events">SIP Security Events</a></li>
<li><a href="/wiki/display/AST/T.38+Gateway" title="T.38 Gateway">T.38 Gateway</a></li>
<li><a href="/wiki/display/AST/Documentation+Improvements" title="Documentation Improvements">Documentation Improvements</a></li>
<li>Project Infrastructure Improvements
<ul>
<li>Arrivederci Subversion, Ciao Git</li>
<li>Adios Mantis, Hola JIRA</li>
</ul>
</li>
</ul>
<p><b>Candidate Features</b> - developed by Digium, time permitting</p>
<ul>
<li>SILK Support</li>
</ul>
<h1><a name="Roadmap-FutureDevelopmentWishlist"></a>Future Development Wishlist</h1>
<p>The items in this list primarily came from <a href="/wiki/display/AST/AstriDevCon+2010" title="AstriDevCon 2010">AstriDevCon 2010</a>.</p>
<h2><a name="Roadmap-%28P0%29"></a>(P0)</h2>
<p>P0 are committed projects that are going to get done.</p>
<ul>
<li><a href="/wiki/display/AST/T.38+Gateway" title="T.38 Gateway">T.38 Gateway</a> (Digium)</li>
<li>Performance of State Change Processing (Stefan Schmidt)
<ul>
<li>(work is already being done on this front)</li>
</ul>
</li>
<li>SIP path support (Olle)
<ul>
<li>(first generation of code exists, needs more work, simple patch, going to get it done, needs an extra field in astdb; helps when there are 2 or more load balancing proxies in front of asterisk, when you'd like the call to be able to get back to Asterisk; see <a href="https://reviewboard.asterisk.org/r/991/" class="external-link" rel="nofollow">https://reviewboard.asterisk.org/r/991/</a>)</li>
<li><a href="https://issues.asterisk.org/view.php?id=18223" class="external-link" rel="nofollow">https://issues.asterisk.org/view.php?id=18223</a></li>
</ul>
</li>
<li>Group variables (Kobaz)
<ul>
<li>(on review board, in progress)</li>
</ul>
</li>
<li>Pre-Dial (Kobaz)
<ul>
<li>(practically done, or something)</li>
</ul>
</li>
<li>Distributed extension state using SIP (Olle)
<ul>
<li>(resources in place, doing it, 1.4 done before Christmas, project pinana)</li>
</ul>
</li>
<li>Manager event docs (Paul Belanger)</li>
<li>Cross-platform documentation (Ben Klang)
<ul>
<li>(caveats for using Asterisk on operating system xyz; pull a PDF of the Wiki documentation into the source, don't forget to include basic installation information, and do it all in .txt - Ben)</li>
</ul>
</li>
<li>Fix libs to optionally init OpenSSL (Digium)
<ul>
<li>(or use existing tools; sort of a bug)</li>
</ul>
</li>
<li>Make ast_channel an opaque type (Digium)</li>
</ul>
<h2><a name="Roadmap-%28P1%29"></a>(P1)</h2>
<p>P1 is the highest priority.</p>
<ul>
<li><a href="/wiki/display/AST/Media+Overhaul" title="Media Overhaul">Codecs (SILK, Opal), Media Negotiation</a> (Digium)</li>
<li>RTCP (Olle)
<ul>
<li>Pinefrog; Work to be done - Ported to trunk, added to CEL</li>
</ul>
</li>
<li>Conferencing that supports a new magic media (Digium)
<ul>
<li>higher sampling rates</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P2%29"></a>(P2)</h2>
<ul class="alternate" type="square">
<li>Async DNS (TCP DNS and use a good resolver)</li>
<li>Named ACLs (deluxepine)</li>
<li><a href="/wiki/display/AST/SIP+Security+Events" title="SIP Security Events">SIP Security Events</a></li>
<li>Light weight means of holding NAT open in SIP (less complex than current qualify, Consider it done)</li>
<li>IPv6 for the restivus (IAX, Jabber/XMPP/Gtalk, Manager, etc.)</li>
<li>ConfBridge feature complete with MeetMe</li>
<li>Support sound file containers (matroska)</li>
</ul>
<h2><a name="Roadmap-%28P3%29"></a>(P3)</h2>
<ul>
<li>Who hung up? (there's a branch, shouldn't take too much time - Olle)</li>
<li>Unique identifier for filtering log data to a call
<ul>
<li>(finishing what was already begun w/ Clod's project, CLI filtering; should take a look at what Stephan from Unlimitel.ca's created)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P4%2CSimon%27sfeatures%29"></a>(P4, Simon's features)</h2>
<ul>
<li>Multiple SIP Sockets
<ul>
<li>(Listen on multiple ports or on multiple interfaces, but not all; also set binding for RTP)...alternate idea / solution would be to make Asterisk capable of loading multiple SIP profiles, it might be easier</li>
</ul>
</li>
<li>Multiple DNS results
<ul>
<li>(need to be able to traverse a list of DNS results, rather than just getting back one result)</li>
</ul>
</li>
<li>ICE-lite
<ul>
<li>(no code, responding correctly to ICE connectivity checks (STUN multiplexed on the RTP port) and understanding the SDP); it makes NAT traversal work for clients that do ICE; also addressed lightweight NAT refresh)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P5%29"></a>(P5)</h2>
<ul>
<li>AstDB replacement
<ul>
<li>(realtime, there's code, nearly ready)</li>
</ul>
</li>
<li>SIP identity
<ul>
<li>(on reviewboard; needs to be forward ported; important for organizations w/ federated identities; a requirement for DTLS SRTP; not widely deployed)</li>
</ul>
</li>
<li>RTMP client channel driver</li>
</ul>
<h2><a name="Roadmap-%28P6%29"></a>(P6)</h2>
<ul>
<li>Structured identifiers for errors
<ul>
<li>(tag an error message with a unique string, specific to the error message and where it came from; should be alphanumeric to keep them short)</li>
</ul>
</li>
<li>AMI SetVar, Context limits
<ul>
<li>(there's code already...Olle has it)</li>
</ul>
</li>
<li>AMI filters on demand</li>
<li>DTLS SRTP
<ul>
<li>(not likely to be widely deployed in the next 12 months)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P7%2Cnotkobaz%29"></a>(P7, not kobaz)</h2>
<ul>
<li>Asterisk register for XMPP account (Leif)</li>
<li>Write a Specification for AMI (not kobaz)</li>
<li>Multiple TLS server certs
<ul>
<li>(1 socket, requires support by OpenSSL; simpler to implement than multiple SIP profiles; don't know if any clients use it yet; needs more research)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P8%2Cnicetohave%29"></a>(P8, nice to have)</h2>
<ul class="alternate" type="square">
<li>Make resource modules that talk to DBs attempt reconnects</li>
<li>Apple's new file streaming format, derived from .m3u</li>
<li>Make MixMonitor and Monitor feature compatible</li>
</ul>
<h2><a name="Roadmap-%28P%3F%2CResearchRequired%29"></a>(P?, Research Required)</h2>
<ul>
<li>New app_queue (as if? no, seriously? talking about this scares Russell)</li>
<li>Identify and fix all bugs in AMI</li>
<li>Broadsoft or Dialog Info shared line appearance (SLA) support
<ul>
<li>(Tabled for later discussion)</li>
</ul>
</li>
<li>LDAP from within the dialplan
<ul>
<li>(we may already have it, needs research to see if the realtime driver does what's desired - Leif)</li>
</ul>
</li>
<li>Device state normalization</li>
<li>Anything DB over HTTP(s) with failover handling</li>
<li>Use a channel as a MoH Source</li>
<li>Kill Masquerades</li>
<li>Bridging thread pool</li>
<li>Threadify chan_sip</li>
<li>Export ISDN ROSE information up to Asterisk channels</li>
</ul>
</div>
<div id="commentsSection" class="wiki-content pageSection">
<div style="float: right;">
<a href="https://wiki.asterisk.org/wiki/users/viewnotifications.action" class="grey">Change Notification Preferences</a>
</div>
<a href="https://wiki.asterisk.org/wiki/display/AST/Roadmap">View Online</a>
|
<a href="https://wiki.asterisk.org/wiki/pages/diffpagesbyversion.action?pageId=4259934&revisedVersion=22&originalVersion=21">View Changes</a>
|
<a href="https://wiki.asterisk.org/wiki/display/AST/Roadmap?showComments=true&showCommentArea=true#addcomment">Add Comment</a>
</div>
</div>
</div>
</div>
</div>
</body>
</html>