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

Asterisk Bug Tracker noreply at bugs.digium.com
Tue May 4 17:53:21 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16936 
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Reported By:                ib2
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   16936
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     assigned
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:              2010-05-04 17:53 CDT
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Summary:                    [patch] 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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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0017277 [patch] The heap data structure can't c...
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---------------------------------------------------------------------- 
 (0121385) crjw (reporter) - 2010-05-04 17:53
 https://issues.asterisk.org/view.php?id=16936#c121385 
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I had a similar problem:
Asterisk is supposed to re-register to my SIP provider every 75 seconds.
Logging always shows:
 "Expiry for xxx.xxx.xxx is 90 sec (Scheduling reregistration in 75 s)".

With the latest trunk, most re-registrations took place after 75s, but
once every hour or two, there was an unexplained delays of up to 15
minutes.
It appeared the the delays were often preceded by several of my Polycom
phones updating their subscription information for "hints".

I installed the heap-fix.diff patch from issue
https://issues.asterisk.org/view.php?id=17277 about 12 hours ago.
I have not seen any of the delays since installing the patch.  Every thing
looks perfect now! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-05-04 17:53 crjw           Note Added: 0121385                          
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