<html>
<head>
    <base href="https://wiki.asterisk.org/wiki">
            <link rel="stylesheet" href="/wiki/s/2042/1/7/_/styles/combined.css?spaceKey=TOP&amp;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/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 \-&gt; Fixed <br>\* pjsip_tls_transport_start being used while not present \-&gt; 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 \-&gt; Fixed <br></td></tr>
            <tr><td class="diff-unchanged" >\* Some endpoint config items not being processed \-&gt; 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:&nbsp; 35039</p>

<p>PIN:&nbsp; 35806</p>


<p><b>Proposed Agenda:</b></p>

<p>Welcome &amp; Announcements:</p>


<ul>
        <li>Introduction of Digium participants</li>
</ul>


<p>Old Business:&nbsp;</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/>
&#42; Configurator should output help message if nothing is specified on command line &#45;&gt; Tweaked message<br/>
&#42; Pushed transports do not work &#45;&gt; Fixed<br/>
&#42; Pushed endpoints do not update routing &#45;&gt; Fixed<br/>
&#42; pjsip_tls_transport_start being used while not present &#45;&gt; Fixed<br/>
&#42; Crash in routing service when updating destination ids &#45;&gt; Fixed<br/>
&#42; Some endpoint config items not being processed &#45;&gt; Fixed</p>

<p>April 12th<br/>
&#42;  Dialing something that does not exist causes deadlock &#45;&gt; Fixed,  configuration issue (thread pool size was not large enough, need to  document)<br/>
&#42; Trying to call an endpoint with no configured target host and port causes assertion &#45;&gt; Fixed<br/>
&#42; Trying to end a call for an endpoint that is not answering fails &#45;&gt; Fixed<br/>
&#42; TLS in pjsip and linking against OpenSSL is a slight mess, we need to link against it<br/>
&#42; We do not use sips URIs for TLS, this causes the pjsip transport selector to select wrong transport<br/>
&#42; Attempting to contact an IPv6 UA without the needed transport crashes SIP component &#45;&gt; Fixed</p>

<p>April 13th<br/>
&#42; PRACK support incomplete &#45;&gt; Fixed<br/>
&#42; Found issue with AMI migration of transfer related calls to routing, could cause a crash (405)<br/>
&#42; We silently drop SIP packets for methods we do not support/have implemented &#45;&gt; Fixed<br/>
&#42; Attempting to place a call when RTP component goes down causes failure &#45;&gt; Fixed<br/>
&#42; Attempting to hangup a call when RTP component goes down causes failure &#45;&gt; Fixed<br/>
&#42; RTP state replication fails to replicate state items &#45;&gt; Fixed</p>

<p>April 14th</p>

<p>&#42; Found potential race condition in RTP state replication &#45;&gt; Fixed<br/>
&#42; Regression found in fix for silently dropping SIP packets &#45;&gt; Fixed<br/>
&#42; Added support for SIP session timers</p>

<p>April 15th</p>

<p>Areas of improvement:</p>

<p>&#42; Configuration mechanism should have some manner of notifying that specific configuration could not be applied<br/>
&#42; Configuration mechanism should have some manner of notifying that a requirement for configuration was not met<br/>
&#42; SIP configurator does not have a log message when invalid section is read<br/>
&#42; Create separate module for replying to SIP packets that contain methods we do not support/have implemented<br/>
&#42; Configuration mechanism should have some manner of notifying that something non-configuration related went wrong<br/>
&#42; Make waits in RTP unit test timed in case state replicator fails to push out items<br/>
&#42; Remove requirement for predefined global adapter port numbers for replicated components<br/>
&#42; 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>
        <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/TOP/Asterisk+SCF+Developer+Call+-+04212011+-+1700+EST">View Online</a>
        |
        <a href="https://wiki.asterisk.org/wiki/pages/diffpagesbyversion.action?pageId=13076358&revisedVersion=3&originalVersion=2">View Changes</a>
                |
        <a href="https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Developer+Call+-+04212011+-+1700+EST?showComments=true&amp;showCommentArea=true#addcomment">Add Comment</a>
            </div>
</div>
</div>
</div>
</div>
</body>
</html>