[asterisk-bugs] [Asterisk 0014000]: Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Dec 1 07:36:42 CST 2008
The following issue has been SUBMITTED.
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http://bugs.digium.com/view.php?id=14000
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Reported By: folke
Assigned To:
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Project: Asterisk
Issue ID: 14000
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 159896
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 2008-12-01 07:36 CST
Last Modified: 2008-12-01 07:36 CST
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Summary: Wrong usage of sscanf with use of uninitialized
variable caused accidental parsing of RTP/SAVP
Description:
Hi folks!
We are Nortel VoIP-PBX Supporter and have a customer with as SIP-Trunk
between Nortel CS1000 and Asterisk. Worked fine. After an upgrade on Nortel
PBX to a new MediaGatewayCard (MC32s) no calls could be established. The
Nortel offers after the upgrade next to RTP also SRTP.
But asterisk accidentaly parses also the m=audio line with RTP/SAVP in the
incoming SDP-Message causing to refused the call ('488 Not acceptable').
It was because a sscanf is used in chan_sip.c / process_sdp() to compare
and examine the 'm=audio' in a wrong way. As soon as the line starts with
'audio ' followed by a number, sscanf returns '1' regardless of the rest of
the line. The result of '%n' variable of sscanf was then used unitialized.
With my fix now chan_sip ignores now the 'm=audio ... RTP/SAVP' lines and
everything is fine.
see details
patch available
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Issue History
Date Modified Username Field Change
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2008-12-01 07:36 folke New Issue
2008-12-01 07:36 folke Asterisk Version => SVN
2008-12-01 07:36 folke SVN Branch (only for SVN checkouts, not tarball
releases) => trunk
2008-12-01 07:36 folke SVN Revision (number only!) => 159896
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