[asterisk-bugs] [Asterisk 0018898]: Large number of active sip dialogs PUBLISH in the output "sip show channels".

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18898 
====================================================================== 
Reported By:                Obi Van
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18898
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.2.4 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-02-28 07:55 CST
Last Modified:              2011-02-28 13:55 CST
====================================================================== 
Summary:                    Large number of active sip dialogs PUBLISH in the
output "sip show channels".
Description: 
On Debian 5.0 and Asterisk 1.8.2.4 (also 1.8.2.3) in the output "sip show
channels" I see the following (IP addresses is fake):
123.45.678.910  (None)           OTMxYmI3YWRjOGN  0x0 (nothing)    No     
 Rx: PUBLISH                <guest>   
123.45.678.910  (None)           ZmNmNmRhMTUyNDQ  0x0 (nothing)    No     
 Rx: PUBLISH                <guest>   
123.45.678.910    (None)           NTZiYjhlNGRlNzY  0x0 (nothing)    No   
   Rx: PUBLISH                <guest>
To these addresses are registered softphones clients. Execute a command
"sip show channel" on any of the PUBLISH dialogues gives the following
results (123.45.678.900 - is address Asterisk):
*CLI>sip show channel NTZiYjhlNGRlNzY
* SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                NTZiYjhlNGRlNzY1NGIxMTBhMzFiMTgxNTlkNGNjNmU.
  Owner channel ID:       <none>
  Our Codec Capability:   0x10d (g723|ulaw|alaw|g729)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   0x0 (nothing)
  Joint Codec Capability:   0x0 (nothing)
  Format:                 0x0 (nothing)
  T.38 support            No
  Video support           No
  MaxCallBR:              0 kbps
  Theoretical Address:    123.45.678.910:5060
  Received Address:       123.45.678.910:5060
  SIP Transfer mode:      open
  Force rport:            Yes
  Audio IP:               123.45.678.900 (local)
  Our Tag:                as2b4a5359
  Their Tag:              27384736
  SIP User agent:         Zoiper rev.6313
  Need Destroy:           No
  Last Message:           Rx: PUBLISH
  Promiscuous Redir:      No
  Route:                  N/A
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive
Number of such dialogues can reach up to 100 or more! CLI command "sip
reload" does not help. Only helps "core stop now". I noticed that
 Session-Timer:      Inactive
With the execution of commands for any active dialogue, for example ACK, I
get the following:

 Session-Timer:          Active
  S-Timer Interval:       600
  S-Timer Refresher:      uas
  S-Timer Expirys:        0
  S-Timer Sched Id:       162202
  S-Timer Peer Sts:       Inactive
  S-Timer Cached Min-SE:  0
  S-Timer Cached SE:      600
  S-Timer Cached Ref:     auto
  S-Timer Cached Mode:    Originate
While the output is consistent with the settings in the file sip.conf. It
is seen that:
Session-Timer:          Active
It seems to me that the dialogue PIBLISH does not work Session-Timer.

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

---------------------------------------------------------------------- 
 (0132447) lmadsen (administrator) - 2011-02-28 13:55
 https://issues.asterisk.org/view.php?id=18898#c132447 
---------------------------------------------------------------------- 
Here is a request for additional information after discussing this with
another developer. Find his responses below:

So the issue is apparently that incoming PUBLISH messages create a sip_pvt
structure that is not going away. Well, I assume the sip_pvt does not go
away based on how the issue description is worded.

Now, in past Asterisk versions, Asterisk had no support at all for SIP
PUBLISH, so those packets would be dropped immediately and such a problem
couldn't occur.

In 1.8, PUBLISH support was added because call completion required it. So
it's likely a problem that only affects 1.8.

What would be helpful to find out is 1) Why are these PUBLISHes being sent
to Asterisk? 2) Do the PUBLISH dialogs go away eventually or do they stick
around in the system forever? I'm assuming they're sticking around, but
it's not 100% clear in that regard.

But the other thing you can note for the reporter is that if the PUBLISHes
do not relate to call completion (which I am 99.99% sure they are not
related), find the source of the PUBLISHes and stop sending them.

Because if the PUBLISHes are for presence or something like that, Asterisk
is just going to completely ignore them anyway. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-28 13:55 lmadsen        Note Added: 0132447                          
======================================================================