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
AbstractThis 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 Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
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 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]. 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
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.
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.
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. RFC3711] also apply. 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. [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.
[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.