[asterisk-dev] [Code Review] 3431: Fix channel staging assertion failure.

Matt Jordan reviewboard at asterisk.org
Mon Apr 14 14:32:35 CDT 2014


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Ship it!


Please address the rtp_engine documentation finding before committing.

- Matt Jordan


On April 9, 2014, 2:19 p.m., rmudgett wrote:
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> https://reviewboard.asterisk.org/r/3431/
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> (Updated April 9, 2014, 2:19 p.m.)
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> 
> Review request for Asterisk Developers.
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> Repository: Asterisk
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> Description
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> The failing assertion ensures that the final snapshot gets generated so CDR records can get finalized.  The only place where a channel staging snapshot flag could be left set is in the handle_request_bye().  The function could return before clearing the flag because the channel could dissappear while the function had to have the channel unlocked.
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> * Fixed handle_request_bye() channel snapshot staging coverage area to not have a return in the middle of it and be unable to clear the staging flag.
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> * Pushed the channel snapshot staging coverage area into ast_rtp_instance_set_stats_vars() to ensure that the staging is not interrutped.
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> * Made callers of ast_rtp_instance_set_stats_vars() not call it with channels or channel driver private locks held to eliminate the deadlock potential.  The callers must hold references to the passed in channel and rtp objects.
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> * Eliminated sip_hangup() trying to get the bridge peer.  It is futile at this point because the channel could never be in a bridge.
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> * Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end of the function.  The unref needs to happen after the last use of the pointer.
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> Diffs
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>   /branches/12/main/rtp_engine.c 412047 
>   /branches/12/channels/chan_sip.c 412047 
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> Diff: https://reviewboard.asterisk.org/r/3431/diff/
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> Testing
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> I was unsuccessful in reproducing the testsuite channel staging assertion failure.
> However, SIP calls can still setup and teardown with the patch installed.
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> Thanks,
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> rmudgett
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>

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