[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16382 
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Reported By:                lftsy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-478 
Regression:                 No 
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:             2009-12-03 10:04 CST
Last Modified:              2010-03-15 09:03 CDT
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Summary:                    SIP OPTIONS qualify message forever
Description: 
Hello, I have a trouble with different Asterisk versions (1.4.26, 1.4.27,
1.4.27.1). When I use the steps below, Asterisk starts to send SIP OPTIONS
to the previous IP/port used by a SIP realtime peer (that has been pruned)
and will keep trying to send SIP OPTIONS pings forever, event if the peer
is connected with a new IP/port address.

I have just checked with my old Asterisk 1.2.27 with the same sip.conf and
I do not have the problem, the SIP OPTIONS stops once register timer has
expired.

During my experience to reproduce the bug, I have been able to have 10
IP/port currently pinged by the Asterisk server for one single peer.
And the only way to stop it is to restart Asterisk...

Thank you for your attention!
Marc Leurent
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Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0016764 Sip Channels Colapse
related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
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 (0119353) lftsy (reporter) - 2010-03-15 09:03
 https://issues.asterisk.org/view.php?id=16382#c119353 
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Good afternoon,
Problem is still there on Asterisk 1.4.30 final release!

If needed, I'm available and I can grant access to our servers to an
Asterisk developers if needed, please simply provide your phone number or
ask for mine on a private channel (email)!


I have been able to reproduce the bug in doing some stress tests, simply
doing 120 calls through the following design:
SIP-Tester -> OpenSIPs Proxy -> Asterisk 1.4.30 -> Asterisk Voicemail
1.4.26

After 2 hours, Asterisk server was sending flood OPTIONS to the OpenSIPs
server, using 20Mb/s of bandwidth and flooding the MySQL server too. I have
the pcap file showing that the server was flooding the IP of the proxy, and
"sip show channels" was showing 15000 SIP OPTIONS transaction ongoing...

Best Regards, 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-15 09:03 lftsy          Note Added: 0119353                          
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