[asterisk-bugs] [Asterisk 0010332]: SIP - ACK not processed, 200 OK retransmitted

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10332 
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Reported By:                rudinsky
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   10332
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.17 
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:             2007-07-30 05:10 CDT
Last Modified:              2008-08-11 13:27 CDT
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Summary:                    SIP - ACK not processed, 200 OK retransmitted
Description: 
Making a SIP call over long-delay connection causes Asterisk to quit the
session unexpectedly.

Jul 30 10:48:54 WARNING[24077]: chan_sip.c:1228 retrans_pkt: Maximum
retries 
exceeded on transmission NWZiNGViNTlhNGQ5ODFkNjQ0YzU0MmNmZTdkODRkMmM. for
seqno 2 (Critical Response)
Jul 30 10:48:54 WARNING[24077]: chan_sip.c:1245 retrans_pkt: Hanging up
call 
NWZiNGViNTlhNGQ5ODFkNjQ0YzU0MmNmZTdkODRkMmM. - no reply to our critical
packet.
Jul 30 10:48:54 WARNING[24103]: file.c:592 ast_readaudio_callback: Failed
to write frame
  == Spawn extension (default, 334, 1) exited non-zero on
'SIP/honza-0819bd20'

Probably the ACK is not processed and Asterisk resends 200 OK response.
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Relationships       ID      Summary
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duplicate of        0009335 High latency (satellite) hangup on inte...
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 (0091310) pj (reporter) - 2008-08-11 13:27
 http://bugs.digium.com/view.php?id=10332#c91310 
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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?
SVN-trunk-r136787
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.
uploaded: 
sipOK-retransmit.pcap 
sipOK-retransmit.png 

Issue History 
Date Modified    Username       Field                    Change               
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2008-08-11 13:27 pj             Note Added: 0091310                          
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