[asterisk-bugs] [Asterisk 0013865]: SIP/TLS enabled - just one call possible - 481 Call/Transaction Does Not Exist

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13865 
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Reported By:                st
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   13865
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1-beta1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-11-09 10:03 CST
Last Modified:              2009-05-08 16:15 CDT
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Summary:                    SIP/TLS enabled - just one call possible - 481
Call/Transaction Does Not Exist
Description: 
Settup consits of Asterisk 1.6.1-beta1 and Snom320 firmware 7.3.7
TLS is enabled using standard port 5061 for the first example and port
5062 for the second example.

Dialplan is very simple: Answer; Wait(1); Playback(...); Hangup;

The first call of the first example does work but it is not possible to
call the same number again: 481 Call/Transaction Does Not Exist

The first call of the second example has no "BYE" and has to be cancelled
at the phone.


(IMHO a new category chan_sip/TLS should be created)
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Relationships       ID      Summary
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has duplicate       0015009 Asterisk's not handling BYE sip-tls mes...
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 (0104501) jmacz (reporter) - 2009-05-08 16:15
 http://bugs.digium.com/view.php?id=13865#c104501 
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Hi Vrban, after upgrading the Phone from firmware version 2.2.0.0047 to
3.1.3.0439 (which didn't work either) and doing some more tests with the
Eyebeam softphone (v1.5.7), I found that there's a subtle difference
between the INVITE and BYE messages from/to the IP phone and from/to the
softphone. Calls between Eyebeam softphones always work. Calls from an
Eyebeam Softphone to a Polycom behaves correctly if the Polycom hangs up.
But calls between Polycoms or from a Polycom to an Eyebeam were the
softphone hangs up, get stucked (not the case if the A party -the Polycom
320- , also terminates the call, case in which the BYE message seems to be
well handled).

The only thing different I've noticed so far is the URI of the first
example vs the second one.

The one that works (Eyebeam --> Polycom --> Polycom Answers --> Polycom
Hangs UP --> Eyebeam receives a BYE and hangs up OK):

INVITE sip:<NUMBER>@<UA ADDRESS>:<UA PORT>;transport=tls SIP/2.0
Via: SIP/2.0/TLS <ASTERISK ADDRESS>:5061;branch=z9hG4bK3a21876a;rport

BYE sip:<NUMBER>@<ASTERISK ADDRESS>;transport=tls SIP/2.0
Via: SIP/2.0/TLS <UA ADDRESS>:<UA PORT>;branch=z9hG4bKb62ee8a34F14EC08

Versus the one that doesn't (Polycom --> Eyebeam --> Eyebeam Answers -->
Eyebeam Hangs UP --> Polycom doesn't hangs UP and sends 481 error):

INVITE sip:<NUMBER>@<ASTERISK ADDRESS>;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS <UA ADDRESS>:<UA PORT>;branch=z9hG4bKa1885f262AB2844F

BYE sip:<NUMBER>@<UA ADDRESS>:<UA PORT>;transport=tls SIP/2.0
Via: SIP/2.0/TLS <ASTERISK ADDRESS>:5061;branch=z9hG4bK0cd0e4b4;rport

I've tried many configurations on the phone without luck (specifying the
TLS port as 5061, setting an outbound proxy with the Asterisk's IP, etc).

I'm attaching two calls between the Polycom IP SP 320 (ext 27) with the
new firmware, and the Eyebeam (ext 33):

Call without issues: 20090508_sipdebug_tls-eyebeam
Call still with the issue: 20090508_sipdebug_tls-polycom

I'll continue testing and will try to reach some help from Polycom to see
If I can figure out what's going with this. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-08 16:15 jmacz          Note Added: 0104501                          
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