[asterisk-bugs] [Asterisk 0016158]: High CPU usage, choppy sound

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Nov 10 10:01:52 CST 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16158 
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Reported By:                mustardman
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16158
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.1.9 
JIRA:                       SWP-382 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-10-30 19:45 CDT
Last Modified:              2009-11-10 10:01 CST
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Summary:                    High CPU usage, choppy sound
Description: 
After upgrading from Asterisk 1.6.1.6 to 1.6.1.8 I seem to have a random
problem with high CPU usage.  When making a SIP to SIP call or SIP to
System call Asterisk process CPU usage goes much higher than normal and the
sound get's choppy.  This is somewhat random but occurs fairly regularly,
maybe 50% of the time.

If it happens it happens as soon as a call is answered.  CPU usage
sometimes stays high for minutes and will suddenly go low even though I
didn't do anything.  More often than not it only stays high for a short
period of time and then goes back to normal.

I am running 64bit CentOS 5.4.  Log files and Asterisk CLI -vvvvvvvvvvr
look clean.  

Not sure if this is useful but I will attached a gdb.txt file taken while
it is happening
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---------------------------------------------------------------------- 
 (0113566) lmadsen (administrator) - 2009-11-10 10:01
 https://issues.asterisk.org/view.php?id=16158#c113566 
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Issue is Acknowledge. Per the issue guidelines though, as this is part of
chan_sip, please provide the following information (as quoted from the
issue guidelines):

Include debug output! Please include output from "sip debug" if you have a
SIP problem. This seems obvious, but apparently is not. Set debug to 4,
verbose to 4, turn on sip history and dumphistory in sip.conf and capture
all output. A packet trace from ethereal will not tell us what is happening
inside your Asterisk server, so that is not a replacement.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-10 10:01 lmadsen        Note Added: 0113566                          
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