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

[Asterisk Bug Tracker]

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:                
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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 14:22 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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---------------------------------------------------------------------- 
 (0118900) ib2 (reporter) - 2010-03-03 14:22
 https://issues.asterisk.org/view.php?id=16936#c118900 
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sip.conf:

[general]
language=aa
port=5060
bindaddr=0.0.0.0
srvlookup=no
defaultexpirey=120
disallow=all
allow=alaw
allow=ulaw
allow=g726
allow=ilbc
allow=gsm
callevents=yes
musicclass=default
relaxdtmf=yes
rtpiholdtimeout=300
rtptimeout=60
tos=184
useragent=PBXIreland
nat=yes
dtmfmode=auto
registerattempts=0
registertimeout=40
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0
externip=217.118.x.x
limitonpeer = yes

[202]
type=friend
host=dynamic
canreinvite=no
qualify=2000
call-limit=4
context=phones
callerid=202 <202>
defaultuser=202
fromdomain=217.118.x.x
secret=verysecret
nat=yes 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-03 14:22 ib2            Note Added: 0118900                          
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