[asterisk-dev] [Code Review] 4277: Ensure that CHANNEL(pjsip, secure) gives expected value.

[Mark Michelson]

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(Updated Dec. 18, 2014, 8:44 a.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
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Committed in revision 429739


Repository: Asterisk


Description
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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.


Diffs
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  /branches/13/channels/pjsip/dialplan_functions.c 429672 

Diff: https://reviewboard.asterisk.org/r/4277/diff/


Testing
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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.


Thanks,

Mark Michelson

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