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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4277/">https://reviewboard.asterisk.org/r/4277/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Mark Michelson.</div>
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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;">We were using the pjsip_dialog's secure flag to indicate if the transport in use was secure. However, there is a difference between dialog security and transport security. In RFC 3261 sections 12.1.1 and 12.1.2, it indicates that for a dialog to be secure, the transport in use must be secure AND the target URI must be a SIPS URI. Since we're only interested in if the transport in use is secure, we have to use a different method to determine that.
This patch seeks to fix this by asking PJSIP for information about the dialog's target URI and then checking if the transport in use is a secure transport.</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 has been tested by John Bigelow by placing a call over TLS into the dialplan and seeing the value of ${CHANNEL(pjsip,secure)}. With a TLS transport, this returns 1. When re-run without a TLS transport, this returns 0. It has also been tested that this value functions independently of ${CHANNEL(rtp, secure)} as expected.</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/13/channels/pjsip/dialplan_functions.c <span style="color: grey">(429672)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4277/diff/" style="margin-left: 3em;">View Diff</a></p>
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