[asterisk-bugs] [Asterisk 0017192]: [patch] Phone keeps ringing when hangup between 'NOTIFY' and 'Status: 180 Ringing'

Fri Aug 20 04:58:20 CDT 2010

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17192 
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Reported By:                ronaldderksen
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17192
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.6.0.26 
JIRA:                       SWP-1350 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 257592 
Request Review:              
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Date Submitted:             2010-04-16 04:49 CDT
Last Modified:              2010-08-20 04:58 CDT
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Summary:                    [patch] Phone keeps ringing when hangup between
'NOTIFY' and 'Status: 180 Ringing'
Description: 
When I call my aastra SIP phone and hangup before the aastra phone sents
back the 'Status: 180 Ringing' the phone keeps ringing 'forever'
This is what I see with tshark when I hangup right after I dialed the
phone.

  0.000000 192.168.12.4 -> 192.168.12.100 SIP Request: NOTIFY
sip:400 at 192.168.12.100:5060;transport=udp
  0.044154 192.168.12.100 -> 192.168.12.4 SIP Status: 200 OK
  0.351948 192.168.12.4 -> 192.168.12.100 SIP/SDP Request: INVITE
sip:400 at 192.168.12.100:5060;transport=udp, with session description
  0.355715 192.168.12.4 -> 192.168.12.100 SIP Request: NOTIFY
sip:400 at 192.168.12.100:5060;transport=udp
  0.525349 192.168.12.100 -> 192.168.12.4 SIP Status: 180 Ringing
  0.529074 192.168.12.100 -> 192.168.12.4 SIP Status: 200 OK

The aastra phone sents the 'Status: 180 Ringing' message after 529ms. That
is a a big window. When I first play a message with 'Playback' application
and the caller decides to hangup, the window of 529ms is too much. There is
a big change people hangup in that 529ms.

I can also reproduce this with a polycom phone. But with polucom it is
harder because it sents the 'Status: 180 Ringing' quicker (186 ms) I cannot
reproduce this problem with a linksys phone. Linksys phone sents the
'Status: 180 Ringing' in 20ms.

A normal Dial -> ringing -> hangup looks like this (with aastra phone):

  0.000000 192.168.12.4 -> 192.168.12.100 SIP Request: NOTIFY
sip:400 at 192.168.12.100:5060;transport=udp
  0.047770 192.168.12.100 -> 192.168.12.4 SIP Status: 200 OK
  0.331322 192.168.12.4 -> 192.168.12.100 SIP/SDP Request: INVITE
sip:400 at 192.168.12.100:5060;transport=udp, with session description
  0.464797 192.168.12.100 -> 192.168.12.4 SIP Status: 180 Ringing
  2.822520 192.168.12.4 -> 192.168.12.100 SIP Request: NOTIFY
sip:400 at 192.168.12.100:5060;transport=udp
  2.823327 192.168.12.4 -> 192.168.12.100 SIP Request: CANCEL
sip:400 at 192.168.12.100:5060;transport=udp
  2.983367 192.168.12.100 -> 192.168.12.4 SIP Status: 200 OK
  2.986478 192.168.12.100 -> 192.168.12.4 SIP Status: 487 Request
Terminated

So When I hangup after the phone ringed ones, I see that asterisk sents a
'CANCEL' message. I think this 'CANCEL' message also need to be sent when
asterisk did not yet receive the 'Status: 180 Ringing' message.

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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0017096 C keeps ringing when hanging A and B af...
related to          0013626 [patch] CANCEL before Trying
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 (0126185) davidw (reporter) - 2010-08-20 04:58
 https://issues.asterisk.org/view.php?id=17192#c126185 
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pabelanger:  Could I suggest that this is reverted to "feedback", as the
candidate fix is reported not to work, but no full SIP debug information
has been provided for the failure of the patched version.

thsgmbh:  Have you tried the proposed patch?  If so, are you able to
provide SIP debug information for it?

Note though, that the basic problem is already understood - the remaining
problem is how to fix it properly.  The problem is that Asterisk is not
sending a deferred CANCEL when it is not allowed to send an immediate one. 
Sending a deferred CANCEL would stop the phone ringing, but the nature of
SIP means it may ring for a short time, as the cancel cannot be issued
until the phone has made some response.

I personally have no say in the prioritisation of applying fixes, but to
some extent it depends on when Asterisk users with the ability to make code
changes find a problem worthwhile to fix.  There is no contractual
obligation to fix anything in the open source version of Asterisk, unless
you take out an explicit contract with someone for that purpose. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-08-20 04:58 davidw         Note Added: 0126185                          
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