[test-results] [Bamboo] Asterisk - Trunk > Ubuntu Lucid (10.04) > #1007 has FAILED (1 tests failed). Change made by Russell Bryant.

[Bamboo]

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Asterisk - Trunk > Ubuntu Lucid (10.04) > #1007 failed.
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Code has been updated by Russell Bryant.
1/175 tests failed.

http://bamboo.asterisk.org/browse/ASTTRUNK-LUCID-1007/


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Failing Jobs
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  - amd64 (Default Stage): 1 of 175 tests failed.


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Code Changes
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Russell Bryant (336879):

>Merged revisions 336878 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/10
>
>................
>  r336878 | russell | 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) | 43 lines
>  
>  Merged revisions 336877 via svnmerge from 
>  https://origsvn.digium.com/svn/asterisk/branches/1.8
>  
>  ........
>    r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines
>    
>    Fix crashes in ast_rtcp_write().
>    
>    This patch addresses crashes related to RTCP handling.  The backtraces just
>    show a crash in ast_rtcp_write() where it appears that the RTP instance is no
>    longer valid.  There is a race condition with scheduled RTCP transmissions and
>    the destruction of the RTP instance.  This patch utilizes the fact that
>    ast_rtp_instance is a reference counted object and ensures that it will not get
>    destroyed while a reference is still around due to scheduled RTCP
>    transmissions.
>    
>    RTCP transmissions are scheduled and executed from the chan_sip scheduler
>    context.  This scheduler context is processed in the SIP monitor thread.  The
>    destruction of an RTP instance occurs when the associated sip_pvt gets
>    destroyed (which happens when the sip_pvt reference count reaches 0).  However,
>    the SIP monitor thread is not the only thread that can cause a sip_pvt to get
>    destroyed.  The sip_hangup function, executed from a channel thread, also
>    decrements the reference count on a sip_pvt and could cause it to get
>    destroyed.
>    
>    While this is being changed anyway, the patch also removes calling
>    ast_sched_del() from within the RTCP scheduler callback.  It's not helpful.
>    Simply returning 0 prevents the callback from being rescheduled.
>    
>    (closes issue ASTERISK-18570)
>    
>    Related issues that look like they are the same problem:
>    
>    (issue ASTERISK-17560)
>    (issue ASTERISK-15406)
>    (issue ASTERISK-15257)
>    (issue ASTERISK-13334)
>    (issue ASTERISK-9977)
>    (issue ASTERISK-9716)
>    
>    Review: https://reviewboard.asterisk.org/r/1444/
>  ........
>................
>


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Tests
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New Test Failures (1)
   - AsteriskTestSuite: S/channels/ s i p/sip channel params

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