[Asterisk-code-review] channels/chan sip: Fix rtptimeout (asterisk[master])

[Kelvin]

Kelvin has posted comments on this change.

Change subject: channels/chan_sip: Fix rtptimeout
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Patch Set 1:

i put a log of packet size on the line before setting lastrtprx which also includes channel details so i would know what leg is receiving the rtp. started a call, then disconnect network on one leg. i can still see 2 sets or rtp received, 1 set has all packet sizes zero.

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Gerrit-MessageType: comment
Gerrit-Change-Id: I2650224ee99df87c102c58fcb54ffae383d0c8c3
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Kelvin <kelchy at gmail.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kelvin <kelchy at gmail.com>
Gerrit-HasComments: No