[asterisk-bugs] [Asterisk 0015442]: Asterisk cannot handle SIP 183 "Session Progress" if no SDP is contained in it

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Jul 10 15:08:41 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15442 
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Reported By:                ffloimair
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15442
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.25.1 
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:             2009-07-02 02:19 CDT
Last Modified:              2009-07-10 15:08 CDT
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Summary:                    Asterisk cannot handle SIP 183 "Session Progress" if
no SDP is contained in it
Description: 
We have a trunk to an Alcatel system running and they send "183 Session
Progress" SIP messages without SDP. However Asterisk cannot handle these
messages if they do not contain an SDP payload.

Alcatel refers to the SIP RFC3261 standard which states:
"The 183 (Session Progress) response is used to convey information
 about the progress of the call that is not otherwise classified.  The
 Reason-Phrase, header fields, or message body MAY be used to convey
 more details about the call progress."

The emphasis lies on MAY in the last sentence, so to be compliant with the
SIP standard Asterisk should be able to handle 183 messages without any SDP
in the message body.
====================================================================== 

---------------------------------------------------------------------- 
 (0107595) tkarl (reporter) - 2009-07-10 15:08
 https://issues.asterisk.org/view.php?id=15442#c107595 
---------------------------------------------------------------------- 
Thank you very much for this first feedback.
In my project the Alcatel PBX provides a link to one public PSTN
interface. When we call a mobile phone through this interface the want to
indicate that the call should not be canceled within the standard 4 seconds
(T2 timer? I'm really not a SIP expert…). 
The asterisk does not crash in my case, but the call is canceled every
time so it is not possible to reach any mobile phone.
I hope I explained it correctly. If you need more information I can try to
get them from my Alcatel partner. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-10 15:08 tkarl          Note Added: 0107595                          
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