[asterisk-bugs] [Asterisk 0016674]: [patch] channels stuck in ringing state forever

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16674 
====================================================================== 
Reported By:                under
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16674
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           Older 1.4 - please test a newer version 
JIRA:                       SWP-781 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-01-22 11:13 CST
Last Modified:              2010-01-26 04:51 CST
====================================================================== 
Summary:                    [patch] channels stuck in ringing state forever
Description: 
Scenario:
CALLER -> INVITE            ->  ASTERISK -> INVITE            -> CALLEE
CALLER <- SESSION PROGRESS  <-  ASTERISK <- SESSION PROGRESS  -> CALLEE
CALLER <- RINGING           <-  ASTERISK <- RINGING           -> CALLEE

After this short network out of order causes further packets lost and
channels are stuck in "show channels" forever.

Caller and callee gateways seem to hangup channel by retransmission
timeout, so they are clean and tidy.

You can reproduce this with attached sipp scenario being run on callee
side.
It simply doesn't response anything after Ringing - you'll see channels
stuck.
====================================================================== 

---------------------------------------------------------------------- 
 (0117178) under (reporter) - 2010-01-26 04:51
 https://issues.asterisk.org/view.php?id=16674#c117178 
---------------------------------------------------------------------- 
Issue can be workaround'ed using Dial's timeout option (it controls the
amount of time connection procedure can take). 
But anyway, some mechanism is needed to control timeouts of SIP messages,
that is independent from Dial.
And my patch seems to be only a workaround - some more thorough
implementation is needed. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-26 04:51 under          Note Added: 0117178                          
======================================================================