[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #1484 has FAILED (40 tests failed, 2 failures were new). Change made by Matthew Jordan.

[Bamboo]

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Asterisk Testing > Asterisk Trunk > #1484 failed (rerun once).
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Code has been updated by Matthew Jordan.
40/740 tests failed, 2 failures were new.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-1484/


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Failing Jobs
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  - Asterisk CentOS 6 32-Bit (CentOS 6): 40 of 615 tests failed.



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Code Changes
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Matthew Jordan (393740):

>Refactor RTCP events over to Stasis; associate with channels
>
>This patch does the following:
>
>* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
>  information in the RTCP events. Because Stasis provides a cache, Jaco's
>  patch was modified to pass the channel uniqueid to the RTP layer as
>  opposed to a pointer to the channel. This has the following benefits:
>  (1) It keeps the RTP engine 'clean' of references back to channels
>  (2) It prevents circular dependencies and other potential ref counting issues
>* The RTP engine now allows any RTP implementation to raise RTCP messages.
>  Potentially, other implementations (such as res_rtp_multicast) could also
>  raise RTCP information. The engine provides structs to represent RTCP headers
>  and RTCP SR/RR reports.
>* Some general refactoring in res_rtp_asterisk was done to try and tame the
>  RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
>  but it does feel marginally better.
>* A few random bugs were fixed in the RTCP statistics. (Example: performing an
>  assignment of a = a is probably not correct)
>* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
>  raise an event when we sent a RR report.
>
>Note that this work will be of use to others who want to monitor call quality
>or build modules that report call quality statistics. Since the events are now
>moving across the Stasis message bus, this is far easier to accomplish. It is
>also a first step (though by no means the last step) towards getting Olle's
>pinefrog work incorporated.
>
>Again: note that the patch by Jaco Kroon was modified slightly for this work;
>however, he did all of the hard work in finding the right places to set the
>channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
>for his hard work here.
>
>Review: https://reviewboard.asterisk.org/r/2603/
>
>(closes issue ASTERISK-20574)
>Reported by: Jaco Kroon
>patches:
>  asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
>
>(closes issue ASTERISK-21471)
>Reported by: Matt Jordan
>
>



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Tests
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New Test Failures (2)
   - AsteriskTestSuite: S/fastagi/control-stream-file
   - AsteriskTestSuite: S/apps/mixmonitor audiohook inherit
Existing Test Failures (38)
   - AsteriskTestSuite: S/apps/queues/position priority maxlen
   - AsteriskTestSuite: S/channels/iax2/basic-call
   - AsteriskTestSuite: S/channels/ s i p/sip one legged transfer
   - AsteriskTestSuite: S/apps/queues/queue baseline
   - AsteriskTestSuite: S/apps/queues/set penalty
   - AsteriskTestSuite: S/bridge/automixmon
   - AsteriskTestSuite: S/bridge/blindxfer setup
   - AsteriskTestSuite: S/channels/ s i p/sip hold
   - AsteriskTestSuite: S/channels/ s i p/sip blind transfer/callee with reinvite
   - AsteriskTestSuite: S/bridge/transfer failure
   - AsteriskTestSuite: S/bridge/parkcall
   - AsteriskTestSuite: S/manager/bridge actions
   - AsteriskTestSuite: S/channels/ s i p/sip blind transfer/caller with reinvite
   - AsteriskTestSuite: S/bridge/atxfer setup
   - AsteriskTestSuite: S/channels/gulp/incoming calls without auth
   - AsteriskTestSuite: S/channels/ s i p/sip blind transfer/caller refer only
   - AsteriskTestSuite: S/channels/ s i p/acl call
   - AsteriskTestSuite: S/fax/local channel t38 queryoption
   - AsteriskTestSuite: S/bridge/disconnect
   - AsteriskTestSuite: S/feature attended transfer
   - AsteriskTestSuite: S/masquerade
   - AsteriskTestSuite: S/bridge/dial l s options
   - AsteriskTestSuite: S/bridge/automon

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