9. Security Considerations
9.1. SSRC collision and two-time pad
Any fixed keystream output, generated from the same key and index
MUST only be used to encrypt once. Re-using such keystream (jokingly
called a "two-time pad" system by cryptographers), can seriously
compromise security. The NSA's VENONA project [C99] provides a
historical example of such a compromise. It is REQUIRED that
automatic key management be used for establishing and maintaining
SRTP and SRTCP keying material; this requirement is to avoid
keystream reuse, which is more likely to occur with manual key
management. Furthermore, in SRTP, a "two-time pad" is avoided by
requiring the key, or some other parameter of cryptographic
significance, to be unique per RTP/RTCP stream and packet. The pre-
defined SRTP transforms accomplish packet-uniqueness by including the
packet index and stream-uniqueness by inclusion of the SSRC.
The pre-defined transforms (AES-CM and AES-f8) allow master keys to
be shared across streams belonging to the same RTP session by the
inclusion of the SSRC in the IV. A master key MUST NOT be shared
among different RTP sessions.
Thus, the SSRC MUST be unique between all the RTP streams within the
same RTP session that share the same master key. RTP itself provides
an algorithm for detecting SSRC collisions within the same RTP
session. Thus, temporary collisions could lead to temporary two-time
pad, in the unfortunate event that SSRCs collide at a point in time
when the streams also have identical sequence numbers (occurring with
probability roughly 2^(-48)). Therefore, the key management SHOULD
take care of avoiding such SSRC collisions by including the SSRCs to
be used in the session as negotiation parameters, proactively
assuring their uniqueness. This is a strong requirements in
scenarios where for example, there are multiple senders that can
start to transmit simultaneously, before SSRC collision are detected
at the RTP level.
Note also that even with distinct SSRCs, extensive use of the same
key might improve chances of probabilistic collision and time-
memory-tradeoff attacks succeeding.
As described, master keys MAY be shared between streams belonging to
the same RTP session, but it is RECOMMENDED that each SSRC have its
own master key. When master keys are shared among SSRC participants
and SSRCs are managed by a key management module as recommended
above, the RECOMMENDED policy for an SSRC collision error is for the
participant to leave the SRTP session as it is a sign of malfunction.
9.2. Key Usage
The effective key size is determined (upper bounded) by the size of
the master key and, for encryption, the size of the salting key. Any
additive stream cipher is vulnerable to attacks that use statistical
knowledge about the plaintext source to enable key collision and
time-memory tradeoff attacks [MF00] [H80] [BS00]. These attacks take
advantage of commonalities among plaintexts, and provide a way for a
cryptanalyst to amortize the computational effort of decryption over
many keys, or over many bytes of output, thus reducing the effective
key size of the cipher. A detailed analysis of these attacks and
their applicability to the encryption of Internet traffic is provided
in [MF00]. In summary, the effective key size of SRTP when used in a
security system in which m distinct keys are used, is equal to the
key size of the cipher less the logarithm (base two) of m.
Protection against such attacks can be provided simply by increasing
the size of the keys used, which here can be accomplished by the use
of the salting key. Note that the salting key MUST be random but MAY
be public. A salt size of (the suggested) size 112 bits protects
against attacks in scenarios where at most 2^112 keys are in use.
This is sufficient for all practical purposes.
Implementations SHOULD use keys that are as large as possible.
Please note that in many cases increasing the key size of a cipher
does not affect the throughput of that cipher.
The use of the SRTP and SRTCP indices in the pre-defined transforms
fixes the maximum number of packets that can be secured with the same
key. This limit is fixed to 2^48 SRTP packets for an SRTP stream,
and 2^31 SRTCP packets, when SRTP and SRTCP are considered
independently. Due to for example re-keying, reaching this limit may
or may not coincide with wrapping of the indices, and thus the sender
MUST keep packet counts. However, when the session keys for related
SRTP and SRTCP streams are derived from the same master key (the
default behavior, Section 4.3), the upper bound that has to be
considered is in practice the minimum of the two quantities. That
is, when 2^48 SRTP packets or 2^31 SRTCP packets have been secured
with the same key (whichever occurs before), the key management MUST
be called to provide new master key(s) (previously stored and used
keys MUST NOT be used again), or the session MUST be terminated. If
a sender of RTCP discovers that the sender of SRTP (or SRTCP) has not
updated the master or session key prior to sending 2^48 SRTP (or 2^31
SRTCP) packets belonging to the same SRTP (SRTCP) stream, it is up to
the security policy of the RTCP sender how to behave, e.g., whether
an RTCP BYE-packet should be sent and/or if the event should be
Note: in most typical applications (assuming at least one RTCP packet
for every 128,000 RTP packets), it will be the SRTCP index that first
reaches the upper limit, although the time until this occurs is very
long: even at 200 SRTCP packets/sec, the 2^31 index space of SRTCP is
enough to secure approximately 4 months of communication.
Note that if the master key is to be shared between SRTP streams
within the same RTP session (Section 9.1), although the above bounds
are on a per stream (i.e., per SSRC) basis, the sender MUST base re-
key decision on the stream whose sequence number space is the first
to be exhausted.
Key derivation limits the amount of plaintext that is encrypted with
a fixed session key, and made available to an attacker for analysis,
but key derivation does not extend the master key's lifetime. To see
this, simply consider our requirements to avoid two-time pad: two
distinct packets MUST either be processed with distinct IVs, or with
distinct session keys, and both the distinctness of IV and of the
session keys are (for the pre-defined transforms) dependent on the
distinctness of the packet indices.
Note that with the key derivation, the effective key size is at most
that of the master key, even if the derived session key is
considerably longer. With the pre-defined authentication transform,
the session authentication key is 160 bits, but the master key by
default is only 128 bits. This design choice was made to comply with
certain recommendations in [RFC2104] so that an existing HMAC
implementation can be plugged into SRTP without problems. Since the
default tag size is 80 bits, it is, for the applications in mind,
also considered acceptable from security point of view. Users having
concerns about this are RECOMMENDED to instead use a 192 bit master
key in the key derivation. It was, however, chosen not to mandate
192-bit keys since existing AES implementations to be used in the
key-derivation may not always support key-lengths other than 128
bits. Since AES is not defined (or properly analyzed) for use with
160 bit keys it is NOT RECOMMENDED that ad-hoc key-padding schemes
are used to pad shorter keys to 192 or 256 bits.
9.3. Confidentiality of the RTP Payload
SRTP's pre-defined ciphers are "seekable" stream ciphers, i.e.,
ciphers able to efficiently seek to arbitrary locations in their
keystream (so that the encryption or decryption of one packet does
not depend on preceding packets). By using seekable stream ciphers,
SRTP avoids the denial of service attacks that are possible on stream
ciphers that lack this property. It is important to be aware that,
as with any stream cipher, the exact length of the payload is
revealed by the encryption. This means that it may be possible to
deduce certain "formatting bits" of the payload, as the length of the
codec output might vary due to certain parameter settings etc. This,
in turn, implies that the corresponding bit of the keystream can be
deduced. However, if the stream cipher is secure (counter mode and
f8 are provably secure under certain assumptions [BDJR] [KSYH] [IK]),
knowledge of a few bits of the keystream will not aid an attacker in
predicting subsequent keystream bits. Thus, the payload length (and
information deducible from this) will leak, but nothing else.
As some RTP packet could contain highly predictable data, e.g., SID,
it is important to use a cipher designed to resist known plaintext
attacks (which is the current practice).
9.4. Confidentiality of the RTP Header
In SRTP, RTP headers are sent in the clear to allow for header
compression. This means that data such as payload type,
synchronization source identifier, and timestamp are available to an
eavesdropper. Moreover, since RTP allows for future extensions of
headers, we cannot foresee what kind of possibly sensitive
information might also be "leaked".
SRTP is a low-cost method, which allows header compression to reduce
bandwidth. It is up to the endpoints' policies to decide about the
security protocol to employ. If one really needs to protect headers,
and is allowed to do so by the surrounding environment, then one
should also look at alternatives, e.g., IPsec [RFC2401].
9.5. Integrity of the RTP payload and header
SRTP messages are subject to attacks on their integrity and source
identification, and these risks are discussed in Section 9.5.1. To
protect against these attacks, each SRTP stream SHOULD be protected
by HMAC-SHA1 [RFC2104] with an 80-bit output tag and a 160-bit key,
or a message authentication code with equivalent strength. Secure
RTP SHOULD NOT be used without message authentication, except under
the circumstances described in this section. It is important to note
that encryption algorithms, including AES Counter Mode and f8, do not
provide message authentication. SRTCP MUST NOT be used with weak (or
SRTP MAY be used with weak authentication (e.g., a 32-bit
authentication tag), or with no authentication (the NULL
authentication algorithm). These options allow SRTP to be used to
provide confidentiality in situations where
* weak or null authentication is an acceptable security risk, and
* it is impractical to provide strong message authentication.
These conditions are described below and in Section 7.5. Note that
both conditions MUST hold in order for weak or null authentication to
be used. The risks associated with exercising the weak or null
authentication options need to be considered by a security audit
prior to their use for a particular application or environment given
the risks, which are discussed in Section 9.5.1.
Weak authentication is acceptable when the RTP application is such
that the effect of a small fraction of successful forgeries is
negligible. If the application is stateless, then the effect of a
single forged RTP packet is limited to the decoding of that
particular packet. Under this condition, the size of the
authentication tag MUST ensure that only a negligible fraction of the
packets passed to the RTP application by the SRTP receiver can be
forgeries. This fraction is negligible when an adversary, if given
control of the forged packets, is not able to make a significant
impact on the output of the RTP application (see the example of
Weak or null authentication MAY be acceptable when it is unlikely
that an adversary can modify ciphertext so that it decrypts to an
intelligible value. One important case is when it is difficult for
an adversary to acquire the RTP plaintext data, since for many
codecs, an adversary that does not know the input signal cannot
manipulate the output signal in a controlled way. In many cases it
may be difficult for the adversary to determine the actual value of
the plaintext. For example, a hidden snooping device might be
required in order to know a live audio or video signal. The
adversary's signal must have a quality equivalent to or greater than
that of the signal under attack, since otherwise the adversary would
not have enough information to encode that signal with the codec used
by the victim. Plaintext prediction may also be especially difficult
for an interactive application such as a telephone call.
Weak or null authentication MUST NOT be used when the RTP application
makes data forwarding or access control decisions based on the RTP
data. In such a case, an attacker may be able to subvert
confidentiality by causing the receiver to forward data to an
attacker. See Section 3 of [B96] for a real-life example of such
Null authentication MUST NOT be used when a replay attack, in which
an adversary stores packets then replays them later in the session,
could have a non-negligible impact on the receiver. An example of a
successful replay attack is the storing of the output of a
surveillance camera for a period of time, later followed by the
injection of that output to the monitoring station to avoid
surveillance. Encryption does not protect against this attack, and
non-null authentication is REQUIRED in order to defeat it.
If existential message forgery is an issue, i.e., when the accuracy
of the received data is of non-negligible importance, null
authentication MUST NOT be used.
9.5.1. Risks of Weak or Null Message Authentication
During a security audit considering the use of weak or null
authentication, it is important to keep in mind the following attacks
which are possible when no message authentication algorithm is used.
An attacker who cannot predict the plaintext is still always able to
modify the message sent between the sender and the receiver so that
it decrypts to a random plaintext value, or to send a stream of bogus
packets to the receiver that will decrypt to random plaintext values.
This attack is essentially a denial of service attack, though in the
absence of message authentication, the RTP application will have
inputs that are bit-wise correlated with the true value. Some
multimedia codecs and common operating systems will crash when such
data are accepted as valid video data. This denial of service attack
may be a much larger threat than that due to an attacker dropping,
delaying, or re-ordering packets.
An attacker who cannot predict the plaintext can still replay a
previous message with certainty that the receiver will accept it.
Applications with stateless codecs might be robust against this type
of attack, but for other, more complex applications these attacks may
be far more grave.
An attacker who can predict the plaintext can modify the ciphertext
so that it will decrypt to any value of her choosing. With an
additive stream cipher, an attacker will always be able to change
An attacker may be able to subvert confidentiality due to the lack of
authentication when a data forwarding or access control decision is
made on decrypted but unauthenticated plaintext. This is because the
receiver may be fooled into forwarding data to an attacker, leading
to an indirect breach of confidentiality (see Section 3 of [B96]).
This is because data-forwarding decisions are made on the decrypted
plaintext; information in the plaintext will determine to what subnet
(or process) the plaintext is forwarded in ESP [RFC2401] tunnel mode
(respectively, transport mode). When Secure RTP is used without
message authentication, it should be verified that the application
does not make data forwarding or access control decisions based on
the decrypted plaintext.
Some cipher modes of operation that require padding, e.g., standard
cipher block chaining (CBC) are very sensitive to attacks on
confidentiality if certain padding types are used in the absence of
integrity. The attack [V02] shows that this is indeed the case for
the standard RTP padding as discussed in reference to Figure 1, when
used together with CBC mode. Later transform additions to SRTP MUST
therefore carefully consider the risk of using this padding without
proper integrity protection.
9.5.2. Implicit Header Authentication
The IV formation of the f8-mode gives implicit authentication (IHA)
of the RTP header, even when message authentication is not used.
When IHA is used, an attacker that modifies the value of the RTP
header will cause the decryption process at the receiver to produce
random plaintext values. While this protection is not equivalent to
message authentication, it may be useful for some applications.
10. Interaction with Forward Error Correction mechanisms
The default processing when using Forward Error Correction (e.g., RFC
2733) processing with SRTP SHALL be to perform FEC processing prior
to SRTP processing on the sender side and to perform SRTP processing
prior to FEC processing on the receiver side. Any change to this
ordering (reversing it, or, placing FEC between SRTP encryption and
SRTP authentication) SHALL be signaled out of band.
SRTP can be used as security protocol for the RTP/RTCP traffic in
many different scenarios. SRTP has a number of configuration
options, in particular regarding key usage, and can have impact on
the total performance of the application according to the way it is
used. Hence, the use of SRTP is dependent on the kind of scenario
and application it is used with. In the following, we briefly
illustrate some use cases for SRTP, and give some guidelines for
recommended setting of its options.
A typical example would be a voice call or video-on-demand
Consider one bi-directional RTP stream, as one RTP session. It is
possible for the two parties to share the same master key in the two
directions according to the principles of Section 9.1. The first
round of the key derivation splits the master key into any or all of
the following session keys (according to the provided security
SRTP_encr_key, SRTP_auth_key, SRTCP_encr_key, and SRTCP_auth key.
(For simplicity, we omit discussion of the salts, which are also
derived.) In this scenario, it will in most cases suffice to have a
single master key with the default lifetime. This guarantees
sufficiently long lifetime of the keys and a minimum set of keys in
place for most practical purposes. Also, in this case RTCP
protection can be applied smoothly. Under these assumptions, use of
the MKI can be omitted. As the key-derivation in combination with
large difference in the packet rate in the respective directions may
require simultaneous storage of several session keys, if storage is
an issue, we recommended to use low-rate key derivation.
The same considerations can be extended to the unicast scenario with
multiple RTP sessions, where each session would have a distinct
11.2. Multicast (one sender)
Just as with (unprotected) RTP, a scalability issue arises in big
groups due to the possibly very large amount of SRTCP Receiver
Reports that the sender might need to process. In SRTP, the sender
may have to keep state (the cryptographic context) for each receiver,
or more precisely, for the SRTCP used to protect Receiver Reports.
The overhead increases proportionally to the size of the group. In
particular, re-keying requires special concern, see below.
Consider first a small group of receivers. There are a few possible
setups with the distribution of master keys among the receivers.
Given a single RTP session, one possibility is that the receivers
share the same master key as per Section 9.1 to secure all their
respective RTCP traffic. This shared master key could then be the
same one used by the sender to protect its outbound SRTP traffic.
Alternatively, it could be a master key shared only among the
receivers and used solely for their SRTCP traffic. Both alternatives
require the receivers to trust each other.
Considering SRTCP and key storage, it is recommended to use low-rate
(or zero) key_derivation (except the mandatory initial one), so that
the sender does not need to store too many session keys (each SRTCP
stream might otherwise have a different session key at a given point
in time, as the SRTCP sources send at different times). Thus, in
case key derivation is wanted for SRTP, the cryptographic context for
SRTP can be kept separate from the SRTCP crypto context, so that it
is possible to have a key_derivation_rate of 0 for SRTCP and a non-
zero value for SRTP.
Use of the MKI for re-keying is RECOMMENDED for most applications
(see Section 8.1).
If there are more than one SRTP/SRTCP stream (within the same RTP
session) that share the master key, the upper limit of 2^48 SRTP
packets / 2^31 SRTCP packets means that, before one of the streams
reaches its maximum number of packets, re-keying MUST be triggered on
ALL streams sharing the master key. (From strict security point of
view, only the stream reaching the maximum would need to be re-keyed,
but then the streams would no longer be sharing master key, which is
the intention.) A local policy at the sender side should force
rekeying in a way that the maximum packet limit is not reached on any
of the streams. Use of the MKI for re-keying is RECOMMENDED.
In large multicast with one sender, the same considerations as for
the small group multicast hold. The biggest issue in this scenario
is the additional load placed at the sender side, due to the state
(cryptographic contexts) that has to be maintained for each receiver,
sending back RTCP Receiver Reports. At minimum, a replay window
might need to be maintained for each RTCP source.
11.3. Re-keying and access control
Re-keying may occur due to access control (e.g., when a member is
removed during a multicast RTP session), or for pure cryptographic
reasons (e.g., the key is at the end of its lifetime). When using
SRTP default transforms, the master key MUST be replaced before any
of the index spaces are exhausted for any of the streams protected by
one and the same master key.
How key management re-keys SRTP implementations is out of scope, but
it is clear that there are straightforward ways to manage keys for a
multicast group. In one-sender multicast, for example, it is
typically the responsibility of the sender to determine when a new
key is needed. The sender is the one entity that can keep track of
when the maximum number of packets has been sent, as receivers may
join and leave the session at any time, there may be packet loss and
delay etc. In scenarios other than one-sender multicast, other
methods can be used. Here, one must take into consideration that key
exchange can be a costly operation, taking several seconds for a
single exchange. Hence, some time before the master key is
exhausted/expires, out-of-band key management is initiated, resulting
in a new master key that is shared with the receiver(s). In any
event, to maintain synchronization when switching to the new key,
group policy might choose between using the MKI and the <From, To>,
as described in Section 8.1.
For access control purposes, the <From, To> periods are set at the
desired granularity, dependent on the packet rate. High rate re-
keying can be problematic for SRTCP in some large-group scenarios.
As mentioned, there are potential problems in using the SRTP index,
rather than the SRTCP index, for determining the master key. In
particular, for short periods during switching of master keys, it may
be the case that SRTCP packets are not under the current master key
of the correspondent SRTP. Therefore, using the MKI for re-keying in
such scenarios will produce better results.
11.4. Summary of basic scenarios
The description of these scenarios highlights some recommendations on
the use of SRTP, mainly related to re-keying and large scale
- Do not use fast re-keying with the <From, To> feature. It may, in
particular, give problems in retrieving the correct SRTCP key, if
an SRTCP packet arrives close to the re-keying time. The MKI
SHOULD be used in this case.
- If multiple SRTP streams in the same RTP session share the same
master key, also moderate rate re-keying MAY have the same
problems, and the MKI SHOULD be used.
- Though offering increased security, a non-zero key_derivation_rate
is NOT RECOMMENDED when trying to minimize the number of keys in
use with multiple streams.
12. IANA Considerations
The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), to refer to transport methods. This profile
registers the name "RTP/SAVP".
SRTP uses cryptographic transforms which a key management protocol
signals. It is the task of each particular key management protocol
to register the cryptographic transforms or suites of transforms with
IANA. The key management protocol conveys these protocol numbers,
not SRTP, and each key management protocol chooses the numbering
scheme and syntax that it requires.
Specification of a key management protocol for SRTP is out of scope
here. Section 8.2, however, provides guidance on the parameters that
need to be defined for the default and mandatory transforms.
David Oran (Cisco) and Rolf Blom (Ericsson) are co-authors of this
document but their valuable contributions are acknowledged here to
keep the length of the author list down.
The authors would in addition like to thank Magnus Westerlund, Brian
Weis, Ghyslain Pelletier, Morgan Lindqvist, Robert Fairlie-
Cuninghame, Adrian Perrig, the AVT WG and in particular the chairmen
Colin Perkins and Stephen Casner, the Transport and Security Area
Directors, and Eric Rescorla for their reviews and support.
14.1. Normative References
[AES] NIST, "Advanced Encryption Standard (AES)", FIPS PUB 197,
http://www.nist.gov/aes/[RFC2104] Krawczyk, H., Bellare, M. and R. Canetti, "HMAC: Keyed-
Hashing for Message Authentication", RFC 2104, February
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2401] Kent, S. and R. Atkinson, "Security Architecture for
Internet Protocol", RFC 2401, November 1998.
[RFC2828] Shirey, R., "Internet Security Glossary", FYI 36, RFC 2828,
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-time Applications", RFC
3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", RFC 3551, July
14.2. Informative References
[AES-CTR] Lipmaa, H., Rogaway, P. and D. Wagner, "CTR-Mode
Encryption", NIST, http://csrc.nist.gov/encryption/modes/
workshop1/papers/lipmaa-ctr.pdf[B96] Bellovin, S., "Problem Areas for the IP Security
Protocols," in Proceedings of the Sixth Usenix Unix
Security Symposium, pp. 1-16, San Jose, CA, July 1996
[BDJR] Bellare, M., Desai, A., Jokipii, E. and P. Rogaway, "A
Concrete Treatment of Symmetric Encryption: Analysis of DES
Modes of Operation", Proceedings 38th IEEE FOCS, pp. 394-
[BS00] Biryukov, A. and A. Shamir, "Cryptanalytic Time/Memory/Data
Tradeoffs for Stream Ciphers", Proceedings, ASIACRYPT 2000,
LNCS 1976, pp. 1-13, Springer Verlag.
[C99] Crowell, W. P., "Introduction to the VENONA Project",
[CTR] Dworkin, M., NIST Special Publication 800-38A,
"Recommendation for Block Cipher Modes of Operation:
Methods and Techniques", 2001.
[f8-a]3GPP TS 35.201 V4.1.0 (2001-12) Technical Specification 3rd
Generation Partnership Project; Technical Specification
Group Services and System Aspects; 3G Security;
Specification of the 3GPP Confidentiality and Integrity
Algorithms; Document 1: f8 and f9 Specification (Release
[f8-b]3GPP TR 33.908 V4.0.0 (2001-09) Technical Report 3rd
Generation Partnership Project; Technical Specification
Group Services and System Aspects; 3G Security; General
Report on the Design, Specification and Evaluation of 3GPP
Standard Confidentiality and Integrity Algorithms (Release
[GDOI] Baugher, M., Weis, B., Hardjono, T. and H. Harney, "The
Group Domain of Interpretation, RFC 3547, July 2003.
[HAC] Menezes, A., Van Oorschot, P. and S. Vanstone, "Handbook
of Applied Cryptography", CRC Press, 1997, ISBN 0-8493-
[H80] Hellman, M. E., "A cryptanalytic time-memory trade-off",
IEEE Transactions on Information Theory, July 1980, pp.
[IK] T. Iwata and T. Kohno: "New Security Proofs for the 3GPP
Confidentiality and Integrity Algorithms", Proceedings of
[KINK] Thomas, M. and J. Vilhuber, "Kerberized Internet
Negotiation of Keys (KINK)", Work in Progress.
[KEYMGT] Arrko, J., et al., "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming Protocol
(RTSP)", Work in Progress.
[KSYH] Kang, J-S., Shin, S-U., Hong, D. and O. Yi, "Provable
Security of KASUMI and 3GPP Encryption Mode f8",
Proceedings Asiacrypt 2001, Springer Verlag LNCS 2248, pp.
[MIKEY] Arrko, J., et. al., "MIKEY: Multimedia Internet KEYing",
Work in Progress.
[MF00] McGrew, D. and S. Fluhrer, "Attacks on Encryption of
Redundant Plaintext and Implications on Internet Security",
the Proceedings of the Seventh Annual Workshop on Selected
Areas in Cryptography (SAC 2000), Springer-Verlag.
[PCST1] Perrig, A., Canetti, R., Tygar, D. and D. Song, "Efficient
and Secure Source Authentication for Multicast", in Proc.
of Network and Distributed System Security Symposium NDSS
2001, pp. 35-46, 2001.
[PCST2] Perrig, A., Canetti, R., Tygar, D. and D. Song, "Efficient
Authentication and Signing of Multicast Streams over Lossy
Channels", in Proc. of IEEE Security and Privacy Symposium
S&P2000, pp. 56-73, 2000.
[RFC1750] Eastlake, D., Crocker, S. and J. Schiller, "Randomness
Recommendations for Security", RFC 1750, December 1994.
[RFC2675] Borman, D., Deering, S. and R. Hinden, "IPv6 Jumbograms",
RFC 2675, August 1999.
[RFC3095] Bormann, C., Burmeister, C., Degermark, M., Fukuhsima, H.,
Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K.,
Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke,
T., Yoshimura, T. and H. Zheng, "RObust Header Compression:
Framework and Four Profiles: RTP, UDP, ESP, and
uncompressed (ROHC)", RFC 3095, July 2001.
[RFC3242] Jonsson, L-E. and G. Pelletier, "RObust Header Compression
(ROHC): A Link-Layer Assisted Profile for IP/UDP/RTP ", RFC
3242, April 2002.
[SDMS] Andreasen, F., Baugher, M. and D. Wing, "Session
Description Protocol Security Descriptions for Media
Streams", Work in Progress.
[SWO] Svanbro, K., Wiorek, J. and B. Olin, "Voice-over-IP-over-
wireless", Proc. PIMRC 2000, London, Sept. 2000.
[V02] Vaudenay, S., "Security Flaws Induced by CBC Padding -
Application to SSL, IPsec, WTLS...", Advances in
Cryptology, EUROCRYPT'02, LNCS 2332, pp. 534-545.
[WC81] Wegman, M. N., and J.L. Carter, "New Hash Functions and
Their Use in Authentication and Set Equality", JCSS 22,
Appendix A: Pseudocode for Index Determination
The following is an example of pseudo-code for the algorithm to
determine the index i of an SRTP packet with sequence number SEQ. In
the following, signed arithmetic is assumed.
if (s_l < 32,768)
if (SEQ - s_l > 32,768)
set v to (ROC-1) mod 2^32
set v to ROC
if (s_l - 32,768 > SEQ)
set v to (ROC+1) mod 2^32
set v to ROC
return SEQ + v*65,536
Appendix B: Test Vectors
All values are in hexadecimal.
B.1. AES-f8 Test Vectors
SRTP PREFIX LENGTH : 0
RTP packet header : 806e5cba50681de55c621599
RTP packet payload : 70736575646f72616e646f6d6e657373
ROC : d462564a
key : 234829008467be186c3de14aae72d62c
salt key : 32f2870d
key-mask (m) : 32f2870d555555555555555555555555
key XOR key-mask : 11baae0dd132eb4d3968b41ffb278379
IV : 006e5cba50681de55c621599d462564a
IV' : 595b699bbd3bc0df26062093c1ad8f73
B.3. Key Derivation Test Vectors
This section provides test data for the default key derivation
function, which uses AES-128 in Counter Mode. In the following, we
walk through the initial key derivation for the AES-128 Counter Mode
cipher, which requires a 16 octet session encryption key and a 14
octet session salt, and an authentication function which requires a
94-octet session authentication key. These values are called the
cipher key, the cipher salt, and the auth key in the following.
Since this is the initial key derivation and the key derivation rate
is equal to zero, the value of (index DIV key_derivation_rate) is
zero (actually, a six-octet string of zeros). In the following, we
shorten key_derivation_rate to kdr.
The inputs to the key derivation function are the 16 octet master key
and the 14 octet master salt:
master key: E1F97A0D3E018BE0D64FA32C06DE4139
master salt: 0EC675AD498AFEEBB6960B3AABE6
We first show how the cipher key is generated. The input block for
AES-CM is generated by exclusive-oring the master salt with the
concatenation of the encryption key label 0x00 with (index DIV kdr),
then padding on the right with two null octets (which implements the
multiply-by-2^16 operation, see Section 4.3.3). The resulting value
is then AES-CM- encrypted using the master key to get the cipher key.
index DIV kdr: 000000000000
master salt: 0EC675AD498AFEEBB6960B3AABE6
xor: 0EC675AD498AFEEBB6960B3AABE6 (x, PRF input)
x*2^16: 0EC675AD498AFEEBB6960B3AABE60000 (AES-CM input)
cipher key: C61E7A93744F39EE10734AFE3FF7A087 (AES-CM output)
Next, we show how the cipher salt is generated. The input block for
AES-CM is generated by exclusive-oring the master salt with the
concatenation of the encryption salt label. That value is padded and
encrypted as above.
index DIV kdr: 000000000000
master salt: 0EC675AD498AFEEBB6960B3AABE6
xor: 0EC675AD498AFEE9B6960B3AABE6 (x, PRF input)
x*2^16: 0EC675AD498AFEE9B6960B3AABE60000 (AES-CM input)
30CBBC08863D8C85D49DB34A9AE17AC6 (AES-CM ouptut)
cipher salt: 30CBBC08863D8C85D49DB34A9AE1
We now show how the auth key is generated. The input block for AES-
CM is generated as above, but using the authentication key label.
index DIV kdr: 000000000000
master salt: 0EC675AD498AFEEBB6960B3AABE6
xor: 0EC675AD498AFEEAB6960B3AABE6 (x, PRF input)
x*2^16: 0EC675AD498AFEEAB6960B3AABE60000 (AES-CM input)
Below, the auth key is shown on the left, while the corresponding AES
input blocks are shown on the right.
auth key AES input blocks
Questions and comments should be directed to the authors and
Cisco Systems, Inc.
5510 SW Orchid Street
Portland, OR 97219 USA
Phone: +1 408-853-4418
Phone: +46 8 50877040
David A. McGrew
Cisco Systems, Inc.
San Jose, CA 95134-1706
Phone: +1 301-349-5815
Phone: +46 8 58533739
Phone: +46 8 4044502
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