<html>
<body>
<div style="font-family: Verdana, Arial, Helvetica, Sans-Serif;">
<table bgcolor="#f9f3c9" width="100%" cellpadding="8" style="border: 1px #c9c399 solid;">
<tr>
<td>
This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2827/">https://reviewboard.asterisk.org/r/2827/</a>
</td>
</tr>
</table>
<br />
<table bgcolor="#fefadf" width="100%" cellspacing="0" cellpadding="8" style="background-image: url('https://reviewboard.asterisk.org/static/rb/images/review_request_box_top_bg.png'); background-position: left top; background-repeat: repeat-x; border: 1px black solid;">
<tr>
<td>
<div>Review request for Asterisk Developers, Joshua Colp, Matt Jordan, and Mark Michelson.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated Sept. 9, 2013, 7:57 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Update the patch to check RTP remote address and changes to the to tag.</pre>
</td>
</tr>
</table>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22424">ASTERISK-22424</a>
</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">One of our SIP tests was previously pushing 200 OKs without SDP and Asterisk would accept these calls without question. According to Mark this should not be accepted because there will be no way to know where to send media to or receive media from in these circumstances. The approach this patch takes is to forcibly hang up the call at this point if there is no SDP on the response provided that it's not a response to a reinvite (in which case the behavior is the same as if there were an SDP that couldn't be parsed properly).</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Tested it against SIP_hold before and after
Tested it against a number of testsuite tests against SIP (any of the ones I could run before the patch)
Tested regular SIP phone calls (they didn't hit the modified code path though).</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(398378)</span></li>
<li>/branches/1.8/channels/sip/include/sip.h <span style="color: grey">(398378)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/2827/diff/" style="margin-left: 3em;">View Diff</a></p>
</td>
</tr>
</table>
</div>
</body>
</html>