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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3275/">https://reviewboard.asterisk.org/r/3275/</a>
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<p>Ship it!</p>
<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Ship It!</pre>
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<p>- Joshua Colp</p>
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<p>On March 3rd, 2014, 6:02 p.m. UTC, Jonathan Rose wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp, Kevin Harwell, and Matt Jordan.</div>
<div>By Jonathan Rose.</div>
<p style="color: grey;"><i>Updated March 3, 2014, 6:02 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22911">ASTERISK-22911</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<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;">This patch provides a fix for the hold problem by doing the following:
Once an ICE session is marked as started, we start adding any new remote candidates into a separate list until we get another attempt to start the ICE session.
Once a call to start the ice session is made, instead of immediately quitting if the session is already started, we check for a difference in the two candidates lists. If the lists are identical, we wipe out the new list and keep the old one and just quit then going on with the current ICE session. If the lists are changed, we toss the old list and adopt the new one and restart the ICE session.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<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;">SIPML client to Asterisk to Desk Phone
SIPML calls desk phone
audio test, got two way audio
SIPML holds call
SIPML resumes call
audio test, got two way audio (previously this would cause one way audio from the SIPML client to the desk phone)</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/11/res/res_rtp_asterisk.c <span style="color: grey">(409155)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3275/diff/" style="margin-left: 3em;">View Diff</a></p>
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