[Asterisk-code-review] pjsip: Add rtp timeout and rtp timeout hold endpoint options. (asterisk[13])

[Matt Jordan]

Matt Jordan has posted comments on this change.

Change subject: pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
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Patch Set 1:

We're going to need tests for the two options.

An easy way of doing this would be to just use SIPp with a very low rtp_timeout/rtp_timeout_hold. Since Asterisk won't receive any RTP, it should terminate the call after n seconds.

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Gerrit-MessageType: comment
Gerrit-Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-HasComments: No