[asterisk-bugs] [Asterisk 0015627]: Asterisk runs out of sockets

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15627 
====================================================================== 
Reported By:                falves11
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15627
Category:                   Core/Netsock
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 209626 
Request Review:              
====================================================================== 
Date Submitted:             2009-07-31 17:12 CDT
Last Modified:              2009-08-01 10:03 CDT
====================================================================== 
Summary:                    Asterisk runs out of sockets
Description: 
The Parallels engineers have found a bug that takes down asterisk because
the server runs out of sockets, and also it degrades the performance
because over time it takes more and more time for the processor to find an
empty socket. The load on the processor grows over time,

====================================================================== 

---------------------------------------------------------------------- 
 (0108508) falves11 (reporter) - 2009-08-01 10:03
 https://issues.asterisk.org/view.php?id=15627#c108508 
---------------------------------------------------------------------- 
Actually now that I think about it, SIP timers has nothing to do with this.
If I disable full-media-proxy by including this in my sip.conf:
directrtpsetup=yes
canreinvite=nonat

the issue is non-existent, but SIP Timers are active. The issue of the UDP
sockets happens only when I change my sip.conf to:
directrtpsetup=no
canreinvite=no

However, if I try one single call the amount of UDP sockets goes up and
down correctly. I imagine there is special kind of call that never finishes
"naturally" and thus the SIP Timers or my RTP timeout close it, but then
the sockets never get released. 
Any ideas where to go from here? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-08-01 10:03 falves11       Note Added: 0108508                          
======================================================================