[asterisk-bugs] [Asterisk 0011536]: With "pedantic=yes" Asterisk shouldn't match To tag if the dialog is not established
noreply at bugs.digium.com
noreply at bugs.digium.com
Sat Dec 15 19:26:02 CST 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11536
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Reported By: ibc
Assigned To:
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Project: Asterisk
Issue ID: 11536
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.15
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: 12-13-2007 06:58 CST
Last Modified: 12-15-2007 19:26 CST
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Summary: With "pedantic=yes" Asterisk shouldn't match To tag
if the dialog is not established
Description:
Asterisk in SIP pedantic mode.
- Asterisk calls to an external SIP user in OpenSer proxy.
- After "Ringing" Asterisk sends CANCEL.
- OpenSer replies with "200 canceling" but Asterisk doesn't recognize it
so resends the CANCEL.
I've a theory of why it fails:
The "180 Ringing" contains a "To" tag:
To: <sip:ibc at aliax.net>;tag=veuje
When Asterisk sends CANCEL OpenSer replies with:
SIP/2.0 200 canceling
To: <sip:ibc at aliax.net>;tag=373b8885156114ecb2c4bd665f9faf0b-f9aa
So Asterisk asumes wrong To tag (because it's different as the received
one) and discards it.
Of course, OpenSer proxy behaviour is correct: there could be many "180
Ringing" each one with different To tag in case of forking scenario, so the
To tag in the "200 canceling" from OpenSer *shouldn't* match any previous
"To" tag received in any provisional response.
About it I opened a thread in Sip-implementators mailist in which this
issue is explained:
https://lists.cs.columbia.edu/pipermail/sip-implementors/2007-October/017668.html
IMHO the solution would be:
Asterisk shouldn't try to match a To tag if the dialog is not yet
established. And Asterisk should assume that it can receive multiples To
tag in provisional responses in case of forking scenarios (there are more
SIP in the world than just single phones registered in Asterisk XD).
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ibc - 12-15-07 19:26
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Sure, I'll be patient ;)
And thanks a lot for all your work improving SIP stack in Asterisk.
Issue History
Date Modified Username Field Change
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12-15-07 19:26 ibc Note Added: 0075474
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