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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3997/
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Review request for Asterisk Developers.
Bugs: ASTERISK-24237
https://issues.asterisk.org/jira/browse/ASTERISK-24237
Repository: Asterisk
Description
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When a native RTP bridge that is remotely bridging its participants switches to
a softmix bridge, it may not properly re-INVITE the media for one or both
participants back to Asterisk. This is due to two factors:
(1) The current bridge_native_rtp code only re-INVITEs if it believes the
channel will survive the bridge operation. Currently, that code is failing, as
it expects the channels to have a soft hangup flag set on it indicating that a
redirect has occurred or that the channel is going to leave the bridge. (The
code did not take into account a smart bridge operation).
(2) When the bridge layer performs a smart bridge, it passes a dummy bridge
down into the old mixing technology when it is stopped. That breaks the native
RTP bridge, as it looks to bridge->channels to know which channels to re-INVITE
back. That list has no entries, as the dummy bridge does not populate that
value.
This patch modifies bridge_native_rtp such that it keeps track of the channels
itself. Given how tricky this code is - both smart bridging and native RTP
bridging - this keeps the mixing technology insulated from changes in the core,
which is probably a good thing.
Diffs
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/branches/12/bridges/bridge_native_rtp.c 423232
Diff: https://reviewboard.asterisk.org/r/3997/diff/
Testing
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The tests that extercised this code the most are the PJSIP blind transfer
tests, as they change the bridge mixing technology from native_rtp to simple
and back in various tests. Shocking the callee_with_hold/caller_with_hold tests
worked right off the bat. The direct media tests still fail, but this is not
surprising as the messages from Asterisk arrive interleaved, which is not
something SIPp handles well (at all).
Thanks,
Matt Jordan
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