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

[jrose]

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/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/1924/#comment11539>

    sip_get_rtp_peer and sip_get_vrtp_peer are actually no longer necessary since bridging will always use the _multi variants of them instead.  sip_get_trtp_peer was actually never useful. I haven't removed any of these since rtp_engine seems to expect them to be there, and the functions themselves are still valid.  They have been stripped of their acl application code though since it was implemented incorrectly.  The get_udptl_peer function has similarly been stripped of its acl application code.


- jrose


On May 16, 2012, 4:41 p.m., jrose wrote:
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> (Updated May 16, 2012, 4:41 p.m.)
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> 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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> Diffs
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>   /trunk/channels/chan_sip.c 366591 
>   /trunk/include/asterisk/rtp_engine.h 366591 
>   /trunk/main/rtp_engine.c 366591 
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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,
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> jrose
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