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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Aug 18 16:01:39 CDT 2010


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:                jpeeler
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Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Target Version:             1.4.36
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:              2010-08-18 16:01 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
======================================================================
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
related to          0017643 [patch] dialplan reload deadlocks in as...
====================================================================== 

---------------------------------------------------------------------- 
 (0126123) lftsy (reporter) - 2010-08-18 16:01
 https://issues.asterisk.org/view.php?id=16382#c126123 
---------------------------------------------------------------------- 
At least it has worked! We successfully re-created the flooding process on
Asterisk 1.4.26, since it was a version I was able to reproduce the bug
easier!

Going to check with asterisk SVN 1.4 branch ASAP to see if I can also
easily recreate it!


Here is what I have done to "start the flooding", sorry I do not have the
very short procedure,
but for sure, launching these automatic scripts for at least one hour and
you should have some flood on 1.4.26!


What you need is:
* an Asterisk server 1.4.26 (to reproduce it quickly), 1.4.30 is broken
too, 1.4.35 is tomorrow's step
* 2 real phones like 0215667818 and 0245667911 for me
* a twinkle phone also configured with 0215667818

The prune process is a way I have found to accelerate the process of
flooding but it's not compulsory, since peer in production are not using
this command except when changing password or codecs, and it's happening on
peer that doesn't have changed password or codecs


After doing the procedure below,
I have now peer 0215667818 being pinged on (10min after unpluging all
phones):
192.168.10.206:14436
192.168.10.206:10619
192.168.10.206:10749
192.168.10.206:16566
192.168.10.126:2048

And if we stop our scripts and wait for up to 2 days, the flood can grow
to >4Mb/s (max I have seen was 20Mb/s on one single customer, just too bad
for hime!)


############# HERE IS THE COMMAND TO SIMULATE A PEER BEHIND UNSTABLE
CONNECTION ##########
# Here is the command I use to simulate an unstable xDSL line (changing
connection port)
mlr at nb-ge-015:[~]$ while `sleep 1`
do sed -i "s/sip_port=.*/sip_port=$RANDOM/" .twinkle/twinkle.sys
sleep 5s
done


# Here is the command I'm using to kill Twinkle without unregistering
mlr at nb-ge-015:[~/.twinkle]$ while `sleep 1` 
do killall -9 twinkle
sleep 10s
done

# Here is the command I'm using to restart Twinkle sometimes
mlr at nb-ge-015:[~/.twinkle]$ while `sleep 1`
do twinkle
sleep 13s
done

############# AND HERE IS THE ACTION IF WAS ALLOWED TO DO ON ASTERISK
##################
asterisk -rx "sip prune realtime 0215667818"
and call from 0245667911 to 0215667818

###################################################################### 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-18 16:01 lftsy          Note Added: 0126123                          
======================================================================




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