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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3603/">https://reviewboard.asterisk.org/r/3603/</a>
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<p style="margin-top: 0;">On November 5th, 2014, 7:55 a.m. EST, <b>Joshua Colp</b> wrote:</p>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Was the behavior for this ever brought up on the -dev list? I don't remember it so I don't think so. Before accepting the behavior you've done I'd really like us to talk about it there amongst everyone. I really think it's the reverse of what a user would expect to happen when a masquerade happens.</pre>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">http://lists.digium.com/pipermail/asterisk-dev/2014-October/071154.html
The only response I received was on IRC from Richard, who said it was a well written email.</pre>
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<p>- Corey</p>
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<p>On October 30th, 2014, 8:06 p.m. EDT, Corey Farrell wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp and Matt Jordan.</div>
<div>By Corey Farrell.</div>
<p style="color: grey;"><i>Updated Oct. 30, 2014, 8:06 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-22409">ASTERISK-22409</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;">During masquerade it is possible for the AST_JITTERBUFFER_FD to be cleared (set to -1). This change adds a check when copying channel fd's to prevent clearing an FD with -1. This seems to resolve the bad audio quality experienced after the masquerade. When AST_JITTERBUFFER_FD was set to -1, this prevented the channel from polling that timer. This caused RTP packets to be received late, and discarded.</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;">Verified the scenario outlined in ASTERISK-22409 no longer experiences audio quality loss.</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/main/channel.c <span style="color: grey">(426593)</span></li>
<li>/branches/11/funcs/func_jitterbuffer.c <span style="color: grey">(426593)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3603/diff/" style="margin-left: 3em;">View Diff</a></p>
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