[asterisk-bugs] [Asterisk 0013409]: [patch] Huge memory leak because memory of channel cdr struct is never returned
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Oct 6 11:30:57 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=13409
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Reported By: tomaso
Assigned To: murf
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Project: Asterisk
Issue ID: 13409
Category: Core/Channels
Reproducibility: always
Severity: block
Priority: normal
Status: closed
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0
SVN Revision (number only!): 137818
Disclaimer on File?: N/A
Request Review:
Resolution: suspended
Fixed in Version:
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Date Submitted: 2008-09-01 09:48 CDT
Last Modified: 2008-10-06 11:30 CDT
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Summary: [patch] Huge memory leak because memory of channel
cdr struct is never returned
Description:
After two days of stress testing by making lots of calls across sip and
dahdi channels the asterisk process memory reached dizzy values: VSZ=3,2GB,
RSS=1,6GB before asterisk stucked completely (even the RAM of our server
(2GB) is finite ;-) ).
Actually this problem is not a question of load, but appears for each
single call.
Using valgrind the reason for that was quickly found: The memory of the
channel cdr struct (chan->cdr) is never returned, not for sip, not for
dahdi channels, when a channel is cleared.
Reproduce:
a.f.a.p. default configuration (modules.conf, etc.)
Make lots of calls and see ps's VSZ and RSS values.
Interested in a patch ? Or is someone revise this anyway?
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Relationships ID Summary
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related to 0013235 Memory leak in Asterisk 1.4 and Trunk
related to 0013444 After update from 140415 to 141991 get ...
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(0093229) tomaso (reporter) - 2008-10-06 11:30
http://bugs.digium.com/view.php?id=13409#c93229
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murf--
Here I am back again with that issue. I checked the current branch 1.6.0
(SVN Revision 146598) where all(?) the patches should have been committed
in now.
It seems that there is a little memory leak left which is not dependent on
dead calls (see attachment - after 20 minutes of load testing, 2x20
calls/sec, no dead calls).
Since my test setup is a little different the sip info you requested won't
help to identify the dead calls. I've got 2x2 PRI loops inbetween:
SIPP client -> Asterisk -> 2xPRI -> loop -> 2xPRI -> Asterisk -> SIPP
server.
If i increase load from 2x20 calls/sec to 2x25 calls/sec dead channels
begin to start.
I guess you want me to remove the PRI loop ... I'll do what I can ...
Issue History
Date Modified Username Field Change
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2008-10-06 11:30 tomaso Note Added: 0093229
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