[asterisk-bugs] [Asterisk 0016936]: Qualify frequency has big pauses. Asterisk stops sending SIP OPTIONS to keep NAT alive

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Mar 3 09:43:22 CST 2010


The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16936 
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Reported By:                ib2
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16936
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.4 
JIRA:                       SWP-993 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-03-01 13:54 CST
Last Modified:              2010-03-03 09:43 CST
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Summary:                    Qualify frequency has big pauses. Asterisk stops
sending SIP OPTIONS to keep NAT alive
Description: 
We have several SIP phone peers that that becomes UNREACHABLE since
upgrading to Asterisk 1.6.2.x

[10:08:44] chan_sip.c: Peer '202_117' is now UNREACHABLE!  Last qualify:
100
[10:11:25] chan_sip.c: Peer '202_117' is now Reachable. (86ms / 2000ms)
[11:59:03] chan_sip.c: Peer '202_117' is now UNREACHABLE!  Last qualify:
91
[12:11:27] chan_sip.c: Peer '202_117' is now Reachable. (85ms / 2000ms)
[13:17:21] chan_sip.c: Peer '202_117' is now UNREACHABLE!  Last qualify:
90
[13:41:27] chan_sip.c: Peer '202_117' is now Reachable. (92ms / 2000ms)

The phone is UNREACHABLE until it registers again. The phone does not know
that it is UNREACHABLE.
Asterisk reports the phone as UNREACHABLE after a big pause in sending SIP
OPTIONS to keep NAT alive. Therefore NAT table is lost and asterisk cannot
receive SIP OK reply from the phone.

The typical interval between the occurrence is shown above
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---------------------------------------------------------------------- 
 (0118839) lmadsen (administrator) - 2010-03-03 09:43
 https://issues.asterisk.org/view.php?id=16936#c118839 
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Could you please provide the following information?

 * sip.conf configuration (relevant parts)
 * SIP console trace
 * SIP history (configured per sip.conf)

If you can show the time prior to and after the UNREACHABLE that would be
ideal. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-03 09:43 lmadsen        Note Added: 0118839                          
2010-03-03 09:43 lmadsen        Status                   acknowledged =>
feedback
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