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

[Joshua Colp]

Joshua Colp has posted comments on this change.

Change subject: channels/chan_sip: Fix rtptimeout
......................................................................


Patch Set 1:

If you've disconnected a leg then there should be no RTP being received, which is what the option is written for. I'd suggest following what Rusty mentioned on the original issue as well and submitting a Wireshark packet capture.

Personally I'm not comfortable with changing the option in such a fundamental way.

-- 
To view, visit https://gerrit.asterisk.org/981
To unsubscribe, visit https://gerrit.asterisk.org/settings

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: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kelvin <kelchy at gmail.com>
Gerrit-HasComments: No