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

[Kevin Harwell]

Kevin Harwell 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: Code-Review-1

(1 comment)

These options, and their descriptions, should also be added to the pjsip.conf.sample file as well.

https://gerrit.asterisk.org/#/c/941/1/res/res_pjsip.c
File res/res_pjsip.c:

Line 798: 				<configOption name="rtp_timeout" default="0">
        : 					<synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
        : 					<description><para>
        : 						This option configures the number of seconds without RTP (while off hold) before
        : 						considering a channel as dead. When the number of seconds is reached the underlying
        : 						channel is hung up.
        : 					</para></description>
        : 				</configOption>
        : 				<configOption name="rtp_timeout_hold" default="0">
Is the default of 0 considered "off" or not checked?


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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: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Scott Griepentrog <sgriepentrog at digium.com>
Gerrit-HasComments: Yes