[asterisk-bugs] [Asterisk 0011058]: Asterisk doesn't recognize a "408 Request Timeout"

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11058 
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Reported By:                ibc
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   11058
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.13  
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:             10-23-2007 04:57 CDT
Last Modified:              11-12-2007 02:31 CST
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Summary:                    Asterisk doesn't recognize a "408 Request Timeout"
Description: 
If Asterisk calls a SIP user and receives from him a "480 User not
responding" Asterisk hangups the call OK.

But in my case Asterisk calls a SIP user registered in a OpenSer, and this
OpenSer cancels not responding calls after 60 seconds and replies with a
"408 Request Timeout", but Asterisk ignores this reply and doesn't send the
ACK. So the channel remains open.

About "408 Request Timeout" RFC 3261 says:

    21.4.9 408 Request Timeout

     The server could not produce a response within a suitable amount of
     time, for example, if it could not determine the location of the
user
     in time.  The client MAY repeat the request without modifications at
     any later time.

I think Asterik should accept a "408" and terminate the call.
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---------------------------------------------------------------------- 
 ibc - 11-12-07 02:31  
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IMHO this is not a bug in OpenSer. I can put another example:

- pedantic=yes
- Asterisk INVITE's a OpenSer user.
- Asterisk CANCEL's the INVITE.
- The phone stops ringing and return a "487: Request Terminated".
- Asterisk doesn't accept the "487" and resends the CANCEL again and
again.
- In fact CLI says during long time (after a CANCEL timeout):
  CLI> sip show channels
  Peer       User/ANR  Call ID      Seq (Tx/Rx)  Form  Hold  Last Message
  82.92.0.11 ibc       534f49530ce  00103/00000  unkn  No    Tx: CANCEL


I attach now the above example as "pedantic_mode_fails_CANCEL.txt".


PD: I asked this question in "SIP implementators" and confirmed that this
is not an OpenSer's bug, but Asterisk's one:
 
https://lists.cs.columbia.edu/pipermail/sip-implementors/2007-October/017668.html


Issue History 
Date Modified   Username       Field                    Change               
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11-12-07 02:31  ibc            Note Added: 0073504                          
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