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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2228/">https://reviewboard.asterisk.org/r/2228/</a>
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<div>Review request for Asterisk Developers, Mark Michelson and Matt Jordan.</div>
<div>By jrose.</div>
<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;">If srtp_create fails, session we create for temp will either not be created or else will be dealloced within srtp_create. At present, when we then run ast_srtp_destroy, attempting to dealloc this can cause Asterisk to crash. This patch addresses that by setting the session pointer to NULL so ast_srtp_destroy doesn't attempt to dealloc it.
As one might expect, this patch doesn't resolve the reporter's problems with actually setting up an srtp enabled SIP call, but it does fix a crash which shouldn't be happening.</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;">The reporter has tested the patch and confirmed that it eliminates the 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-20499">ASTERISK-20499</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_srtp.c <span style="color: grey">(377034)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2228/diff/" style="margin-left: 3em;">View Diff</a></p>
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