[asterisk-dev] [Code Review] chan_sip: Fix directmedia's use of ACL to limit remotebridging to certain host addresses for trunk

Mark Michelson reviewboard at asterisk.org
Thu May 24 10:46:36 CDT 2012


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Ship it!


Excellent! This will work great. I have one nit-picky optimization-esque suggestion that you are free to ignore if you want.


/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/1924/#comment11778>

    You don't have to do this, but you can skip over these lines entirely if res is 0.


- Mark


On May 21, 2012, 9:29 a.m., jrose wrote:
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> (Updated May 21, 2012, 9:29 a.m.)
> 
> 
> Review request for Asterisk Developers, Mark Michelson and Matt Jordan.
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> Summary
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> Related to https://reviewboard.asterisk.org/r/1899/ which has some locking changes that need to be made still...
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> This patch takes Mark's suggested approach for adding callbacks for the rtp_bridge function to be able to supply two channels for determining rtp glue stuff in cases where a channel driver needs data about both peers in order to determine what types of bridging are permissible.
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> This addresses bug AST-876.
>     https://issues.asterisk.org/jira/browse/AST-876
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> 
> Diffs
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>   /trunk/channels/chan_sip.c 366775 
>   /trunk/include/asterisk/rtp_engine.h 366775 
>   /trunk/main/rtp_engine.c 366775 
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> Diff: https://reviewboard.asterisk.org/r/1924/diff
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> 
> Testing
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> Similar to the testing done for the review 1899 version.  Calls were tested with and without directmediapermit/deny on both sides of a call with calls being started from both directions. Specific things tested include whether or not the host address lists were copied properly (because that was a rather substantial problem earlier on) and the results of ast_apply_ha in each case.
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> Thanks,
> 
> jrose
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>

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