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

Thu May 27 06:05:28 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:                     acknowledged
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:              2010-05-27 06:05 CDT
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Summary:                    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
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related to          0017096 C keeps ringing when hanging A and B af...
related to          0013626 [patch] CANCEL before Trying
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 (0122551) davidw (reporter) - 2010-05-27 06:05
 https://issues.asterisk.org/view.php?id=17192#c122551 
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Applying the patch doesn't fix the problem.  Removing it may make it appear
to work better for you, but would make Asterisk violate a MUST clause in
the SIP protocol specification!

What Asterisk should probably be doing is keeping the CANCEL pending, and
probably continuing to try and set up the call, until it gets either a
non-final response, in which case it should send the CANCEL, or a final
one, in which case it should send a BYE.  If the first response from the
phone is Ringing, this means there will be, at least, a short burst of
ringing, if Asterisk is to comply with the RFC.

Those changes are of at least medium complexity, because this bit of the
protocol involves a state machine, rather than an OS thread, controlling
the sequencing of operations.

(I seem to remember, from my first reading, that CANCEL is fire and
forget, which means a lost CANCEL would leave the phone ringing, so it
might be OK to not retransmit the INVITE, if there is no initial response,
and simply time out the dialoge, on the basis that the whole CANCEL process
is not resillient in SIP.) 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-27 06:05 davidw         Note Added: 0122551                          
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