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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1444/">https://reviewboard.asterisk.org/r/1444/</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;">A 5 second delayed destruction isn't that big of a deal. I'm fine with this and completely agree that introducing any sort of synchronization here would just make this more complicated than is necessary. Great Work!</pre>
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<p>- David</p>
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<p>On September 18th, 2011, 9:20 p.m., Russell Bryant wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Russell Bryant.</div>
<p style="color: grey;"><i>Updated Sept. 18, 2011, 9:20 p.m.</i></p>
<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;">I opened ASTERISK-18570 for this issue. The rest of the issues are ones that I have found so far that appear to be the same problem.
This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions.
RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed.
While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled.</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;">This patch has been applied to a test environment with a couple of servers running Asterisk 1.8.7.0-rc1. The servers have processed over 1 million calls without hitting this crash.</pre>
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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-13334">ASTERISK-13334</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-15257">ASTERISK-15257</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-15406">ASTERISK-15406</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17560">ASTERISK-17560</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18570">ASTERISK-18570</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-9716">ASTERISK-9716</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-9977">ASTERISK-9977</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/res/res_rtp_asterisk.c <span style="color: grey">(335789)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1444/diff/" style="margin-left: 3em;">View Diff</a></p>
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