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

jrose reviewboard at asterisk.org
Wed May 16 16:41:57 CDT 2012


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1924/
-----------------------------------------------------------

Review request for Asterisk Developers, Mark Michelson and Matt Jordan.


Summary
-------

Related to https://reviewboard.asterisk.org/r/1899/ which has some locking changes that need to be made still...

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.


Diffs
-----

  /trunk/channels/chan_sip.c 366591 
  /trunk/include/asterisk/rtp_engine.h 366591 
  /trunk/main/rtp_engine.c 366591 

Diff: https://reviewboard.asterisk.org/r/1924/diff


Testing
-------

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.


Thanks,

jrose

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120516/85013c6c/attachment.htm>


More information about the asterisk-dev mailing list