[asterisk-bugs] [Asterisk 0011157]: Asterisk does not send a provisional response at every minute
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jan 22 09:38:26 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=11157
======================================================================
Reported By: rjain
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 11157
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
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:
======================================================================
Date Submitted: 11-04-2007 16:29 CST
Last Modified: 01-22-2008 09:38 CST
======================================================================
Summary: Asterisk does not send a provisional response at
every minute
Description:
The issue here is that Asterisk does not send a non-100 response at every
minute for calls that are in the provisional response state. This causes
most UACs and/or proxies to terminate the call after 3 minutes. There are
many legitimate reasons why a call could remain in an unanswered state for
more than 3 minutes such as early-media (183), call queuing (182), call
forwarding (181) and ringing (180).
Following is quote from section 13.3.1.1 of RFC 3261 which explains what a
UAS should do under such a circumstance:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
This issue was first reported by Alex Balashov on the asterisk-dev mailing
list:
http://lists.digium.com/pipermail/asterisk-dev/2007-November/030341.html.
I've reproduced this problem and collected wireshark and debug traces,
which are attached to this bug report.
======================================================================
----------------------------------------------------------------------
oej - 01-22-08 09:38
----------------------------------------------------------------------
This would add yet another scheduled timer to our poor sip stack... Well,
gotta be fixed. No coders?
Issue History
Date Modified Username Field Change
======================================================================
01-22-08 09:38 oej Note Added: 0081019
======================================================================
More information about the asterisk-bugs
mailing list