Details
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Type:
Security
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Status: Closed
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Severity:
Blocker
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Resolution: Fixed
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Affects Version/s: 13.27.0
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Component/s: Channels/chan_sip/Interoperability
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Labels:
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Environment:asterisk-13.23.0 on Debian/Jessie, with T.38, and sip_preferred_codec_only reINVITE enabled
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Regression:No
Description
Our gateways (asterisk-13 based) experienced occasional segfaults last days, and inspecting with GDB their coredumps, we finally concluded they are caused by a very specific case in process_sdp() of chan_sip.c:
1) Asterisk has been configured with preferred_codec_only for the relevant peer, and e list, possibly restrictive, of codecs
2) the SIP peer starts a valid session through Asterisk (chan_sip) as a B2BUA
3) Asterisk issue a T.38 reINVITE (for example with ReceiveFAX application, even if it was not our case)
4) the SIP UA (UAS in this case) responds with a "broken" SDP with two m-lines, one for an audio codec not included in the SIP peer allowed list, and another with image/t38
Such an SDP is broken because a SIP UA is not allowed to responds with multiple m-lines whenever it received just one m-line.
We reproduced it on a lab with SIPp and the last version 13 released (13.27.0), but I see no change on that part of code also on master.
The reason I choose to tag it as a security issue, is that:
- with specific configurations
- a malevolent, authenticated (it can setup a call through the Asterisk server) user
- can tear down the service
But I'm not sure it is so serious: I'm inviting you to properly change it.
Issue Links
- is a clone of
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