[asterisk-bugs] [Asterisk 0016097]: [patch] asterisk crashes when there are no RTP port left
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Oct 22 01:03:17 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16097
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Reported By: steinwej
Assigned To:
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Project: Asterisk
Issue ID: 16097
Category: Channels/chan_sip/General
Reproducibility: always
Severity: crash
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0
SVN Revision (number only!): 224449
Request Review:
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Date Submitted: 2009-10-19 10:51 CDT
Last Modified: 2009-10-22 01:03 CDT
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Summary: [patch] asterisk crashes when there are no RTP port
left
Description:
To reproduce in the lab I changed sip.conf to consume a lot of UDP ports:
videosupport=yes
t38pt_udptl=yes
rtp.conf:
rtpstart=8000
rtpend=8006
I also configured a sip connection to a different system.
Asterisk 1.4:
The first call (SIP ATA - calls SIP trunk) goes through.
The second call is aborted. Asterisk behaves good.
[Oct 19 17:26:24] ERROR[1976]: rtp.c:1965 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Oct 19 17:26:24] WARNING[1976]: chan_sip.c:4497 sip_alloc: Unable to
create RTP audio and video session: Address already in use
[Oct 19 17:26:24] ERROR[1976]: chan_sip.c:16076 sip_request_call: Unable
to build sip pvt data for 'sip-test-t38/01229922888w' (Out of memory or
socket error)
[Oct 19 17:26:24] WARNING[1976]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'SIP' (cause 42 - Switching equipment congestion)
But asterisk 1.6.0:
First call goes through.
But the second call crashes the system.
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(0112599) steinwej (reporter) - 2009-10-22 01:03
https://issues.asterisk.org/view.php?id=16097#c112599
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I would also consider this as a security issue. One of our customer caused
this kind of crash.
Issue History
Date Modified Username Field Change
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2009-10-22 01:03 steinwej Note Added: 0112599
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