[asterisk-bugs] [Asterisk 0012013]: SIP with canreinvite=yes through multiple Asterisk instances fails

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12013 
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Reported By:                alx
Assigned To:                blitzrage
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Project:                    Asterisk
Issue ID:                   12013
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-02-17 20:55 CST
Last Modified:              2008-12-06 16:45 CST
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Summary:                    SIP with canreinvite=yes through multiple Asterisk
instances fails
Description: 
This is an update to issue http://bugs.digium.com/view.php?id=10481 which was
closed [apologies if there's
another way to report this against the original bug]

The problem relating to 'reinvite glare' as originally reported in 10481
seems to still be present with asterisk 1.4.18. The only difference is that
instead of the call being torn down, the 491 results in 'reinvites' being
lost. The result is that the call stays up, but the media path is not
modified to go directly between the two endpoints involved in the call.

The scenario we are testing is as follows:
SIP phone A ---> Asterisk A ---> Asterisk B ---> SIP phone B

"canreinvite=yes" is set on SIP phone A and SIP phone B. The connection
between Asterisk A and Asterisk B is a 'Custom' trunk. Both Asterisk boxes
are running 1.4.18.

The call proceeds as follows:
SIP Phone A     Asterisk A     Asterisk B      SIP Phone B
     -----INVITE--->     
     <--100 Trying--      ------INVITE-->  
                          <--100 Trying--  -----INVITE-->
                                           <--100 Trying-
                                           <--180 Ringing-
                          <--180 Ringing--
     <--180 Ringing
                                           <--200 OK
                          <--200 OK        --ACK-->
     <--200 OK
     --ACK-->                  
               --INVITE----> 
               <--INVITE----
               <--491 ------
               --491------->
               ---ACK------>
               <--ACK-------

At this point, the two Asterisk boxes just sit idle. No further reinvites
take place until the call is released. The media path remains pinned
between Asterisk A and Asterisk B, instead of going between SIP Phone A and
SIP Phone B.
 
[[Will attach SIP debug output once I figure out how to do so in this
reporting form]]
                
                                     


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---------------------------------------------------------------------- 
 (0095932) diegoviola (reporter) - 2008-12-06 16:45
 http://bugs.digium.com/view.php?id=12013#c95932 
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Just use FreeSWITCH, it's a lot better.

Asterisk is worthless and broken software. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-06 16:45 diegoviola     Note Added: 0095932                          
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