[asterisk-bugs] [Asterisk 0016774]: 1.6.1.13 and 1.6.1.14 UDP ports not freed =>
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Feb 18 03:46:18 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16774
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Reported By: kowalma
Assigned To:
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Project: Asterisk
Issue ID: 16774
Category: Core/RTP
Reproducibility: always
Severity: block
Priority: normal
Status: feedback
Target Version: Potential Blocker
Asterisk Version: 1.6.1.14
JIRA:
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: 2010-02-04 12:46 CST
Last Modified: 2010-02-18 03:46 CST
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Summary: 1.6.1.13 and 1.6.1.14 UDP ports not freed =>
Description:
Since 1.6.1.13 looks there is problem with freeing UDP ports. 2-3 days ago
I upgraded one box from 1.6.0.5 and second from 1.6.1.1 to 1.6.1.13 and
every few thusants calls I'm running out of UDP ports.
mg0:~# rasterisk -x " sip show channels"
Peer User/ANR Call ID Format Hold
Last Message Expiry
192.168.9.2 mg0 28f9a3d00edd874 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 77d13e113c8409e 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 7297a36515d38d1 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 07b22e6a461f5b3 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 3076b8603492c24 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 30180433452ebf7 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 28c1ad56233d208 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 0182d70a3ec9d14 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 76a5f0c46328c99 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 56be88b8268d261 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 626ef5624a83e0c 0x0 (nothing) No
Rx: ACK
192.168.9.2 mg0 1af2adc275f0b10 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 645b2a390a66a11 0x0 (nothing) No
Rx: ACK
192.168.9.2 mg0 502a709e10ff8dc 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 58ab1eac5e50a92 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 6967aeee16f8e71 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 618b46651909df3 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 4ddae9dc5f91236 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 250e1d32111b2c5 0x0 (nothing) No
Rx: ACK
192.168.9.2 mg0 72cfef2c5fc67eb 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 6d70624c7eb9942 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 5e2c546e27a0af2 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 53fa535541908ce 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 0f8cb2514fd7e4f 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 5cdeead033ebb17 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 4083b3aa6c3a4c5 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 4bd036370f2ac5d 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 50467ac42828e9d 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 17faaccd5554f00 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 5bfa754b209122e 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 78d3f4fa2920ac2 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 2fcbbb020e2cf0a 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 2cf255430ced865 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 385ac7317e336fc 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 633b90226eaac79 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 36fe9f8b78ce7b2 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 61b9d37358c31de 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 30a1736605b2c28 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 6fe7902c4ae25f0 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 7a3168e81aabb71 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 4bf8f6e05a1efc9 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 6665165a33c355e 0x8 (alaw) No
Rx: INVITE
192.168.9.2 mg0 3446ee9b07593d4 0x8 (alaw) No
Rx: ACK
192.168.9.2 mg0 145209e01d8b3d4 0x8 (alaw) No
Tx: ACK
192.168.9.2 mg0 45a79c7c44628cf 0x8 (alaw) No
Rx: INVITE
45 active SIP dialogs
mg0:~#
On this box I have 4xE1 so max I can handle 120 calls.
Before upgrade this MG was rock-stable and had asterisk uptime counted in
weeks.
I have one box running 1.6.1.12 and there seems there is no RTP-UDP ports
problem.
mg0:~# while true
> do
> lsof | grep UDP -c ; sleep 10
> done
682
688
691
682
703
730
745
757
781
^C
mg0:~#
Every few hours I need to restart asterisk as it runs out of free ports
/etc/asterisk/rtp.conf
[general]
rtpstart=5000
rtpend=51000
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Relationships ID Summary
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related to 0016721 after a few minutes it takes down the s...
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----------------------------------------------------------------------
(0118210) armin (reporter) - 2010-02-18 03:46
https://issues.asterisk.org/view.php?id=16774#c118210
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With version 1.4.28 it is easy to reproduce:
Start SIP client to register at asterisk, then disconnect that SIP phone
from network. In the time the registration is still active in asterisk,
call this phone. Asterisk will send INVITEs (of course with no answer),
then hangup after about 30 seconds. The asterisk channels are released, but
the sip channel for that "Init: INVITE" is not released.
If this is not related to this PR, we need to open a new one.
Issue History
Date Modified Username Field Change
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2010-02-18 03:46 armin Note Added: 0118210
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