[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
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Patch Set 1:

Can you explain what you mean by "sip_read processing lost packets as zero sized packets"?

That function should only be executed when there is data to be read on an RTP, RTCP, or UDPTL port.

Is data still coming in?

What kind of environment is required to test this?

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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>
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