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<h2><a href="https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Developer+Call+-+04212011+-+1700+EST">Asterisk SCF Developer Call - 04212011 - 1700 EST</a></h2>
<h4>Page <b>edited</b> by <a href="https://wiki.asterisk.org/wiki/display/~bmj">Bryan M. Johns</a>
</h4>
<br/>
<h4>Changes (11)</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" >Specific areas tested: <br> <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> Session setup/teardown <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> UDP/TCP/TLS calls on IPv4 and IPv6 <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> SIP dynamic configuration (add/remove endpoints) <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> RTP state replication <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> Removing components in mid-operation (routing, bridging, media) <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> Attempting calls with components not running (routing, bridging, media) <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> SIP PRACK <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> SIP attended transfer (see bug list) <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-added-words"style="background-color: #dfd;">*</span> Unhandled SIP methods <br></td></tr>
<tr><td class="diff-unchanged" > <br>Issues encountered: <br></td></tr>
<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >\* Pushed endpoints do not update routing \-> Fixed <br>\* pjsip_tls_transport_start being used while not present \-> Fixed <br></td></tr>
<tr><td class="diff-changed-lines" >\* Crash in routing service when updating <span class="diff-changed-words">desti<span class="diff-deleted-chars"style="color:#999;background-color:#fdd;text-decoration:line-through;">o</span>nation</span> ids \-> Fixed <br></td></tr>
<tr><td class="diff-unchanged" >\* Some endpoint config items not being processed \-> Fixed <br> <br></td></tr>
<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" > <br>Notes: <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;"> <br>Mark Michaelson gives an overview of SIPit 28 findings specific to Asterisk SCF <br> <br>Kevin Fleming gives general development update <br> <br>Developer APIs contingent upon near-beta status <br></td></tr>
</table>
</div> <h4>Full Content</h4>
<div class="notificationGreySide">
<p><b>Digium Developer Conference Bridge:</b></p>
<p>Bridge: 35039</p>
<p>PIN: 35806</p>
<p><b>Proposed Agenda:</b></p>
<p>Welcome & Announcements:</p>
<ul>
<li>Introduction of Digium participants</li>
</ul>
<p>Old Business: </p>
<ul>
<li>None</li>
</ul>
<p>New Business:</p>
<ul>
<li>Discussion of current SCF development status</li>
<li>Discussion of progress and anticipated dates for developer APIs</li>
<li>Discussion of results from SIPit 28:</li>
</ul>
<p>Specific areas tested:</p>
<ul>
<li>Session setup/teardown</li>
<li>UDP/TCP/TLS calls on IPv4 and IPv6</li>
<li>SIP dynamic configuration (add/remove endpoints)</li>
<li>RTP state replication</li>
<li>Removing components in mid-operation (routing, bridging, media)</li>
<li>Attempting calls with components not running (routing, bridging, media)</li>
<li>SIP PRACK</li>
<li>SIP attended transfer (see bug list)</li>
<li>Unhandled SIP methods</li>
</ul>
<p>Issues encountered:</p>
<p>April 11th<br/>
* Configurator should output help message if nothing is specified on command line -> Tweaked message<br/>
* Pushed transports do not work -> Fixed<br/>
* Pushed endpoints do not update routing -> Fixed<br/>
* pjsip_tls_transport_start being used while not present -> Fixed<br/>
* Crash in routing service when updating destination ids -> Fixed<br/>
* Some endpoint config items not being processed -> Fixed</p>
<p>April 12th<br/>
* Dialing something that does not exist causes deadlock -> Fixed, configuration issue (thread pool size was not large enough, need to document)<br/>
* Trying to call an endpoint with no configured target host and port causes assertion -> Fixed<br/>
* Trying to end a call for an endpoint that is not answering fails -> Fixed<br/>
* TLS in pjsip and linking against OpenSSL is a slight mess, we need to link against it<br/>
* We do not use sips URIs for TLS, this causes the pjsip transport selector to select wrong transport<br/>
* Attempting to contact an IPv6 UA without the needed transport crashes SIP component -> Fixed</p>
<p>April 13th<br/>
* PRACK support incomplete -> Fixed<br/>
* Found issue with AMI migration of transfer related calls to routing, could cause a crash (405)<br/>
* We silently drop SIP packets for methods we do not support/have implemented -> Fixed<br/>
* Attempting to place a call when RTP component goes down causes failure -> Fixed<br/>
* Attempting to hangup a call when RTP component goes down causes failure -> Fixed<br/>
* RTP state replication fails to replicate state items -> Fixed</p>
<p>April 14th</p>
<p>* Found potential race condition in RTP state replication -> Fixed<br/>
* Regression found in fix for silently dropping SIP packets -> Fixed<br/>
* Added support for SIP session timers</p>
<p>April 15th</p>
<p>Areas of improvement:</p>
<p>* Configuration mechanism should have some manner of notifying that specific configuration could not be applied<br/>
* Configuration mechanism should have some manner of notifying that a requirement for configuration was not met<br/>
* SIP configurator does not have a log message when invalid section is read<br/>
* Create separate module for replying to SIP packets that contain methods we do not support/have implemented<br/>
* Configuration mechanism should have some manner of notifying that something non-configuration related went wrong<br/>
* Make waits in RTP unit test timed in case state replicator fails to push out items<br/>
* Remove requirement for predefined global adapter port numbers for replicated components<br/>
* Replicate configuration as well as state</p>
<p>Open Floor</p>
<ul>
<li>What are you doing with Asterisk SCF today?</li>
</ul>
<p>Notes:</p>
<p>Mark Michaelson gives an overview of SIPit 28 findings specific to Asterisk SCF</p>
<p>Kevin Fleming gives general development update</p>
<p>Developer APIs contingent upon near-beta status</p>
</div>
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