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

[Asterisk Bug Tracker]

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 15:32 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.
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 (0107597) mmichelson (administrator) - 2009-07-10 15:32
 https://issues.asterisk.org/view.php?id=15442#c107597 
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I think I see what's going on. I had a discussion about this sort of thing
earlier today with some people.

So the 183 is being used as a "watchdog" in a way. It's not actually
trying to convey the presence of early media. It just is a way of saying
"I'm still alive, don't cancel the call." 

Just so I can be sure of its usage, would you mind uploading a sip debug
from Asterisk during one of these failed calls? Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-10 15:32 mmichelson     Note Added: 0107597                          
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