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

[jrose]

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Ship it!


- jrose


On April 11, 2013, 9:15 p.m., Michael Young wrote:
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> https://reviewboard.asterisk.org/r/2421/
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> (Updated April 11, 2013, 9:15 p.m.)
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> Review request for Asterisk Developers.
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> Bugs: ASTERISK-21374
>     https://issues.asterisk.org/jira/browse/ASTERISK-21374
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> Repository: Asterisk
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> 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.
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> 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".
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> 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.
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> This patch should be applied after the patch for ASTERISK-21225 is committed. (Review - https://reviewboard.asterisk.org/r/2385/)
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> Diffs
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>   /branches/11/channels/chan_sip.c 385376 
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> Diff: https://reviewboard.asterisk.org/r/2421/diff/
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> Testing
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> Tested on machine in production where this problem occurred.
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> Thanks,
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> Michael Young
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