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Asterisk Multiple RTP Buffer Overflows
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SUMMARY
<http://www.asterisk.org/> Asterisk is "an open source telephony engine
and toolkit. Asterisk implements the Session Initiation Protocol (SIP)".
The Mu Security Research team has found two security issues in the SDP
parser in Asterisk 1.4.18. One is an invalid write to an
attacker-controllable, almost arbitrary memory location and the other is a
stack buffer overflow with limited attacker-controllable values.
DETAILS
Vulnerable Systems:
* Asterisk version 1.4.18.0 and prior
Immune Systems:
* Asterisk version 1.4.18.1
1) Sending an invalid RTP payload type number in the SDP payload of an
INVITE message can cause a write to an invalid memory location. An
attacker would have some control over the memory location.
The invalid memory write is in ast_rtp_unset_m_type() (main/rtp.c, line
1655) called by process_line() (channels/chan_sip.c, line 5275).
ast_rtp_unset_mt_type() does not validate pt, while it is validated in
ast_rtp_set_mt_type() (line 1642). The attacker controls pt and could
write a 0 to a wide range of memory locations.
Example invalid SDP payload (invalid RTP payload type is 780903144):
v=0
o=- 817933771 817933775 IN IP4 10.10.1.101
s=session-name
c=IN IP4 10.10.1.101
t=0 0
m=audio 5000 RTP/AVP 0
a=rtpmap:780903144 PCMU/8000
a=rtpmap:4 G723/8000/1
a=rtpmap:97 telephone-event/8000
2) Sending more than 32 RTP payload type number attributes in the SDP
payload of a SIP INVITE will overflow a buffer on the stack. An attacker
would have some control over the values written.
In process_sdp() (channels/chan_sip.c, line 4980), rtpmap codecs are
stored in found_rtpmap_codecs, an array of 32 ints. The number of codecs
in the map is stored in last_rtpmap_codec. Codecs are appeneded to the
array without checking the size of the array (line 5258). Up to 64
(SIP_MAX_LINES). An attacker would have some control over the values
written - the codec must be between 0 and 256 (MAX_RTP_PT).
Example SDP payload:
v=0
o=- 817933771 817933775 IN IP4 10.10.1.101
s=session-name
c=IN IP4 10.10.1.101
t=0 0
m=audio 5000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
[... repeat this line ...]
a=rtpmap:4 G723/8000/1
a=rtpmap:97 telephone-event/8000
Vendor Response / Solution:
Fixed in Asterisk 1.4.18.1 and other branches. Available from
http://www.asterisk.org
History:
March 11, 2008 - First contact with vendor
March 18, 2008 - Vendor releases fix and advisory
CVE Information:
<http://www.cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2008-1289>
CVE-2008-1289
ADDITIONAL INFORMATION
The information has been provided by Mu Security.
The original article can be found at:
<http://labs.musecurity.com/advisories/MU-200803-01.txt>
http://labs.musecurity.com/advisories/MU-200803-01.txt
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