[asterisk-dev] [Code Review] 2794: bridge_native_rtp: Fix bugs caused by HOLD/UNHOLD/UPDATE_RTP_PEER
svnbot
reviewboard at asterisk.org
Fri Aug 23 14:05:24 CDT 2013
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2794/
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(Updated Aug. 23, 2013, 2:05 p.m.)
Status
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This change has been marked as submitted.
Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark Michelson, and rmudgett.
Changes
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Committed in revision 397578
Bugs: ASTERISK-22217
https://issues.asterisk.org/jira/browse/ASTERISK-22217
Repository: Asterisk
Description
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Issuing hold/unhold would lead to odd behavior. Between two chan_sip devices, a hold could cause an endless chain of updates while with pjsip a similar chain would begin but then end somewhat randomly. This patch fixes that by no longer tweaking the RTP glue on both sides of the call for every HOLD/UNHOLD/UPDATE_RTP_PEER frame.
Overall this restores holding behavior with chan_sip to how it worked with 11 (which includes some bugs I've noticed). chan_pjsip seems to be working as well, but it doesn't have the same bugs chan_sip exhibited with 11 (chan_sip does), which I guess is a good thing.
This patch doesn't fully address the test failures noted in the bug report. Asterisk 12 will needs a new version of the test with Asterisk 12 specific sipp scenarios to deal with that.
Diffs
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/trunk/bridges/bridge_native_rtp.c 397424
Diff: https://reviewboard.asterisk.org/r/2794/diff/
Testing
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Polycom to Polycom holds and unholds from both sides with directmedia using chan_sip
Polycom to Polycom holds and unholds from both sides with directmedia using chan_pjsip
Polycom to JITSI softphone same scenario (pjsip)
Polycom to Digium phone same scenario (pjsip)
Thanks,
jrose
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