rfc6562
Internet Engineering Task Force (IETF) C. Perkins
Request for Comments: 6562 University of Glasgow
Category: Standards Track JM. Valin
ISSN: 2070-1721 Mozilla Corporation
March 2012
Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP
Abstract
This memo discusses potential security issues that arise when using
variable bit rate (VBR) audio with the secure RTP profile.
Guidelines to mitigate these issues are suggested.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc6562.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Perkins & Valin Standards Track [Page 1]
RFC 6562 VBR Audio with SRTP March 2012
Table of Contents
1. Introduction ...................................................2
2. Scenario-Dependent Risk ........................................2
3. Guidelines for Use of VBR Audio with SRTP ......................3
4. Guidelines for Use of Voice Activity Detection with SRTP .......3
5. Padding the Output of VBR Codecs ...............................4
6. Security Considerations ........................................5
7. Acknowledgements ...............................................5
8. References .....................................................5
8.1. Normative References ......................................5
8.2. Informative References ....................................6
1. Introduction
The Secure RTP (SRTP) framework [RFC3711] is a widely used framework
for securing RTP sessions [RFC3550]. SRTP provides the ability to
encrypt the payload of an RTP packet, and optionally add an
authentication tag, while leaving the RTP header and any header
extension in the clear. A range of encryption transforms can be used
with SRTP, but none of the predefined encryption transforms use any
padding; the RTP and SRTP payload sizes match exactly.
When using SRTP with voice streams compressed using variable bit rate
(VBR) codecs, the length of the compressed packets will depend on the
characteristics of the speech signal. This variation in packet size
will leak a small amount of information about the contents of the
speech signal. This is potentially a security risk for some
applications. For example, [spot-me] shows that known phrases in an
encrypted call using the Speex codec in VBR mode can be recognized
with high accuracy in certain circumstances, and [fon-iks] shows that
approximate transcripts of encrypted VBR calls can be derived for
some codecs without breaking the encryption. How significant these
results are, and how they generalize to other codecs, is still an
open question. This memo discusses ways in which such traffic
analysis risks may be mitigated.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
2. Scenario-Dependent Risk
Whether the information leaks and attacks discussed in [spot-me],
[fon-iks], and similar works are significant is highly dependent on
the application and use scenario. In the worst case, using the rate
information to recognize a prerecorded message knowing the set of all
possible messages would lead to near-perfect accuracy. Even when the
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audio is not prerecorded, there is a real possibility of being able
to recognize contents from encrypted audio when the dialog is highly
structured (e.g., when the eavesdropper knows that only a handful of
possible sentences are possible), and thus contain only little
information. Recognizing unconstrained conversational speech from
the rate information alone is unreliable and computationally
expensive at present, but does appear possible in some circumstances.
These attacks are only likely to improve over time.
In practical SRTP scenarios, how significant the information leak is
when compared to other SRTP-related information must be considered,
such as the fact that the source and destination IP addresses are
available.
3. Guidelines for Use of VBR Audio with SRTP
It is the responsibility of the application designer to determine the
appropriate trade-off between security and bandwidth overhead. As a
general rule, VBR codecs should be considered safe in the context of
low-value encrypted unstructured calls. However, applications that
make use of prerecorded messages where the contents of such
prerecorded messages may be of any value to an eavesdropper (i.e.,
messages beyond standard greeting messages) SHOULD NOT use codecs in
VBR mode. Interactive voice response (IVR) applications would be
particularly vulnerable since an eavesdropper could easily use the
rate information to recognize the prompts being played out.
Applications conveying highly sensitive unstructured information
SHOULD NOT use codecs in VBR mode.
It is safe to use variable rate coding to adapt the output of a voice
codec to match characteristics of a network channel, provided this
adaptation is done in a way that does not expose any information on
the speech signal. For example, VBR audio can be used for congestion
control purposes, where the variation is driven by the available
network bandwidth, not by the input speech (i.e., the packet sizes
and spacing are constant unless the network conditions change). VBR
speech codecs can safely be used in this fashion with SRTP while
avoiding leaking information on the contents of the speech signal
that might be useful for traffic analysis.
4. Guidelines for Use of Voice Activity Detection with SRTP
Many speech codecs employ some form of voice activity detection (VAD)
to either suppress output frames, or generate some form of lower-rate
comfort noise frames, during periods when the speaker is not active.
If VAD is used on an encrypted speech signal, then some information
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about the characteristics of that speech signal can be determined by
watching the patterns of voice activity. This information leakage is
less than with VBR coding since there are only two rates possible.
The information leakage due to VAD in SRTP audio sessions can be much
reduced if the sender adds an unpredictable "overhang" period to the
end of active speech intervals, obscuring their actual length. An
RTP sender using VAD with encrypted SRTP audio SHOULD insert such an
overhang period at the end of each talkspurt, delaying the start of
the silence/comfort noise by a random interval. The length of the
overhang applied to each talkspurt must be randomly chosen in such a
way that it is computationally infeasible for an attacker to reliably
estimate the length of that talkspurt. This may be more important
for short talkspurts, since it seems easier to distinguish between
different single word responses based on the exact word length, than
to glean meaning from the duration of a longer phrase. The audio
data comprising the overhang period must be packetized and
transmitted in RTP packets in a manner that is indistinguishable from
the other data in the talkspurt.
The overhang period SHOULD have an exponentially decreasing
probability distribution function. This ensures a long tail, while
being easy to compute. It is RECOMMENDED to use an overhang with a
"half life" of a few hundred milliseconds (this should be sufficient
to obscure the presence of interword pauses and the lengths of single
words spoken in isolation, for example, the digits of a credit card
number clearly enunciated for an automated system, but not so long as
to significantly reduce the effectiveness of VAD for detecting
listening pauses). Despite the overhang (and no matter what the
duration is), there is still a small amount of information leaked
about the start time of the talkspurt due to the fact that we cannot
apply an overhang to the start of a talkspurt without unacceptably
affecting intelligibility. For that reason, VAD SHOULD NOT be used
in encrypted IVR applications where the content of prerecorded
messages may be of any value to an eavesdropper.
The application of a random overhang period to each talkspurt will
reduce the effectiveness of VAD in SRTP sessions when compared to
non-SRTP sessions. However, it is still expected that the use of VAD
will provide significant bandwidth savings for many encrypted
sessions.
5. Padding the Output of VBR Codecs
For scenarios where VBR is considered unsafe, a constant bit rate
(CBR) codec SHOULD be negotiated and used instead, or the VBR codec
SHOULD be operated in a CBR mode. However, if the codec does not
support CBR, RTP padding SHOULD be used to reduce the information
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leak to an insignificant level. Packets may be padded to a constant
size or to a small range of sizes ([spot-me] achieves good results by
padding to the next multiple of 16 octets, but the amount of padding
needed to hide the variation in packet size will depend on the codec
and the sophistication of the attacker) or may be padded to a size
that varies with time. The most secure and RECOMMENDED option is to
pad all packets throughout the call to the same size.
In the case where the size of the padded packets varies in time, the
same concerns as for VAD apply. That is, the padding SHOULD NOT be
reduced without waiting for a certain (random) time. The RECOMMENDED
"hold time" is the same as the one for VAD.
Note that SRTP encrypts the count of the number of octets of padding
added to a packet, but not the bit in the RTP header that indicates
that the packet has been padded. For this reason, it is RECOMMENDED
to add at least one octet of padding to all packets in a media
stream, so an attacker cannot tell which packets needed padding.
6. Security Considerations
This entire memo is about security. The security considerations of
[RFC3711] also apply.
7. Acknowledgements
ZRTP [RFC6189] contains similar recommendations; the purpose of this
memo is to highlight these issues to a wider audience, since they are
not specific to ZRTP. Thanks are due to Phil Zimmermann, Stefan
Doehla, Mats Naslund, Gregory Maxwell, David McGrew, Mark Baugher,
Koen Vos, Ingemar Johansson, and Stephen Farrell for their comments
and feedback on this memo.
8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
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8.2. Informative References
[RFC6189] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
Path Key Agreement for Unicast Secure RTP", RFC 6189,
April 2011.
[fon-iks] White, A., Matthews, A., Snow, K., and F. Monrose,
"Phonotactic Reconstruction of Encrypted VoIP
Conversations: Hookt on fon-iks", Proceedings of the IEEE
Symposium on Security and Privacy 2011, May 2011.
[spot-me] Wright, C., Ballard, L., Coull, S., Monrose, F., and G.
Masson, "Spot me if you can: Uncovering spoken phrases in
encrypted VoIP conversation", Proceedings of the IEEE
Symposium on Security and Privacy 2008, May 2008.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
UK
EMail: csp@csperkins.org
Jean-Marc Valin
Mozilla Corporation
650 Castro Street
Mountain View, CA 94041
USA
Phone: +1 650 903-0800
EMail: jmvalin@jmvalin.ca
Perkins & Valin Standards Track [Page 6]
ERRATA