[asterisk-bugs] [Asterisk 0016046]: [patch] crash in ast_rtp_instance_early_bridge_make_compatible() when directmedia=yes
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jul 28 16:35:53 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16046
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Reported By: ebroad
Assigned To:
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Project: Asterisk
Issue ID: 16046
Category: Channels/chan_skinny
Reproducibility: always
Severity: crash
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA: SWP-1958
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-10-08 16:50 CDT
Last Modified: 2010-07-28 16:35 CDT
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Summary: [patch] crash in
ast_rtp_instance_early_bridge_make_compatible() when directmedia=yes
Description:
Looks like glue1->get_codec() is null in
ast_rtp_instance_early_bridge_make_compatible.
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(0125232) wedhorn (developer) - 2010-07-28 16:35
https://issues.asterisk.org/view.php?id=16046#c125232
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Not sure about mgcp, but unlike skinny, SIP is a point to point protocol.
Basically, SIP tech_pvt is extra data for chan. On skinny, tech_pvt is
extra data for chan AND data for the device.
As such, generally for SIP, if there's a chan, there is also a
chan->tech_pvt. However, in skinny we can't rely on that. During dialing,
there is a sub with no associated channel. At the end of a call, we break
chan and sub apart for the cleanup.
If there is a codec request during the hangup we could get a segfault with
your code. However, on second thought it would likely be due to some other
bug where asterisk is not aware the channel is going down. So maybe leave
it as is. I'll ponder it some more.
Issue History
Date Modified Username Field Change
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2010-07-28 16:35 wedhorn Note Added: 0125232
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