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

Thu Feb 11 06:35:25 CST 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:                
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Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
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:              2010-02-11 06:35 CST
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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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---------------------------------------------------------------------- 
 (0117987) lftsy (reporter) - 2010-02-11 06:35
 https://issues.asterisk.org/view.php?id=16382#c117987 
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To summarize the problem, I would say that the first thing that is not
normal, (from my point of view), since a peer can have only one location,
is the fact (you can see it in the 1st pcap file I have posted
https://issues.asterisk.org/file_download.php?file_id=24694&type=bug) that
a realtime peer can have several Contact location in the same time.

And I think that this is what leads to the flooding...
Ex:
OPTIONS sip:0245667911 at 194.38.160.113:5060 SIP/2.0
OPTIONS sip:0245667911 at 213.162.3.159:5060;transport=udp;user=phone
SIP/2.0

And on my last pcap file today, the same problem appears, one peer has
many Contact entries that are relayed via the proxy IP 212.147.45.91:5060
(but as I said before, the problem can appear with several destination
IP/port in the same time on a direct connection without going through the
proxy)

OPTIONS sip:044444xxyy at 92.104.53.35:65082;line=0ahhazhb SIP/2.0
OPTIONS sip:044444xxyy at 92.104.53.35:59188;rinstance=66d72254a0b7f923
SIP/2.0
OPTIONS sip:044444xxyy at 92.104.53.35:65082;line=0ahhazhb SIP/2.0
OPTIONS sip:044444xxyy at 92.104.53.35:59188;rinstance=66d72254a0b7f923
SIP/2.0
OPTIONS sip:044444xxyy at 92.104.53.35:65246;line=snv7vwjm SIP/2.0
OPTIONS sip:044444xxyy at 92.104.53.35:64749;line=1y8wt5tb SIP/2.0
OPTIONS sip:044444xxyy at 92.104.53.35:55382;line=7tp7l2hd SIP/2.0
...

And of course the peer in not reachable and had a "Expire : -1" even if we
have set a max expiration timer.

pul-lav-vp-ast-03*CLI> sip show peer 044444xxyy

  * Name       : 044444xxyy
  Realtime peer: Yes, cached
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : sipresidential
  Subscr.Cont. : <Not set>
  Language     : de
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      : 044444xxyy at default
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 6
  Dynamic      : Yes
  Callerid     : "044444xxyy" <044444xxyy>
  MaxCallBR    : 384 kbps
  Expire       : -1LI>
  Insecure     : no
  Nat          : Always
  ACL          : No
  T38 pt UDPTL : Yes
  CanReinvite  : NoLI>
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : (Unspecified) Port 0
  Defaddr->IP  : 0.0.0.0 Port 5060
  Reg. exten   :
  Def. Username: 044444xxyy
  SIP Options  : (none)
  Codecs       : 0x102 (gsm|g729)
  Codec Order  : (g729:20,gsm:20)
  Auto-Framing:  No
  Status       : UNKNOWN
  Useragent    :
  Reg. Contact :

And pruning this peer does not stop the flooding OPTIONS...
I hope all these informations will help you pinpoint the problem.

Best Regards, 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-11 06:35 lftsy          Note Added: 0117987                          
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