[asterisk-dev] [Code Review] 2421: One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX

[Michael Young]

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(Updated April 12, 2013, 1:16 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Bugs: ASTERISK-21374
    https://issues.asterisk.org/jira/browse/ASTERISK-21374


Repository: Asterisk


Description
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I found another case where the force_rport and comedia flags are not being set automatically when using the new auto_* settings. This time it involves calls initiated by the PBX.

When we reload asterisk the default flags turned on and off by auto_force_rport (force_rport) and auto_comedia (comedia) go back to the default setting of off. These flags are turned on, as needed, when a peer re-registers or initiates a call. This would apply to even just having the default global setting "nat=auto_force_rport".

Everything is good except in the following scenario:
We reload Asterisk and the peer's registration has not expired. We load in the default settings for the peer which turns force_rport and comedia back to off. Since the peer has not re-registered or placed a call yet, they remain off. We then initiate a call to the peer from the PBX. The force_rport and comedia flags stay off. If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not.

This patch should be applied after the patch for ASTERISK-21225 is committed. (Review - https://reviewboard.asterisk.org/r/2385/)


Diffs
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  /branches/11/channels/chan_sip.c 385376 

Diff: https://reviewboard.asterisk.org/r/2421/diff/


Testing
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Tested on machine in production where this problem occurred.


Thanks,

Michael Young

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