[asterisk-dev] [Code Review] 3987: Bridging: Fix bouncing native bridge

[opticron]

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https://reviewboard.asterisk.org/r/3987/
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(Updated Sept. 12, 2014, 1:17 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
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Committed in revision 423006


Bugs: ASTERISK-24211
    https://issues.asterisk.org/jira/browse/ASTERISK-24211


Repository: Asterisk


Description
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This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge could cause a bouncing native bridge. In the case of the dial_LS_options test, this was a remote RTP bridge which caused the audio path to continually cycle between Asterisk and the remote endpoints generating a large number of SIP messages and delaying the test long enough to cause it to fail (checking timing was part of the test). The root cause was that the code to decide whether to use native bridging was expecting a time-remaining value of 0 to be the default instead of the actual default value of -1. A value of 0 or negative numbers could also be generated by preceding code in some circumstances. Both issues are addressed in this patch.


Diffs
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  branches/1.8/main/channel.c 422898 

Diff: https://reviewboard.asterisk.org/r/3987/diff/


Testing
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Verified that the test (11-only) operated correctly with this patch.


Thanks,

opticron

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