[asterisk-bugs] [Asterisk 0013287]: 200 OK retransmitted even when first SIP OK is ACKed

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13287 
====================================================================== 
Reported By:                pj
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13287
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 136787 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-08-11 15:20 CDT
Last Modified:              2008-09-08 20:38 CDT
====================================================================== 
Summary:                    200 OK retransmitted even when first SIP OK is ACKed
Description: 
I looked at sip callflow in asterisk and seems, that asterisk retransmits
SIP/OK, even in case that first SIP/OK was correctly ACKed by other party.
If you look at attached call graph analysis, you can see it,
also, asterisk delayed initial INVITE about 200ms, before it forwards to
another party, why?

I must also notice, that my asterik server was 100% idle, only one call
was processed, when this dump was taken. Call was over LAN connection.

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0010332 SIP - ACK not processed, 200 OK retrans...
related to          0013115 could NOT get the channel lock
====================================================================== 

---------------------------------------------------------------------- 
 (0092206) svnbot (reporter) - 2008-09-08 20:38
 http://bugs.digium.com/view.php?id=13287#c92206 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 141949

U   trunk/main/channel.c

------------------------------------------------------------------------
r141949 | russell | 2008-09-08 20:38:31 -0500 (Mon, 08 Sep 2008) | 9 lines

Modify ast_answer() to not hold the channel lock while calling
ast_safe_sleep()
or when calling ast_waitfor().  These are inappropriate times to hold the
channel
lock.  This is what has caused "could not get the channel lock" messages
from
chan_sip and has likely caused a negative impact on performance results of
SIP
in Asterisk 1.6.  Thanks to file for pointing out this section of code.

(closes issue http://bugs.digium.com/view.php?id=13287)
(closes issue http://bugs.digium.com/view.php?id=13115)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=141949 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-08 20:38 svnbot         Checkin                                      
2008-09-08 20:38 svnbot         Note Added: 0092206                          
======================================================================