[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

Versions: (draft-mcgrew-avt-srtp) 00 01 02 03 04 05 06 07 08 09 RFC 3711

Internet Engineering Task Force                      Rolf Blom, Ericsson
AVT Working Group                           Elisabetta Carrara, Ericsson
INTERNET-DRAFT                                    David A. McGrew, Cisco
Expires: July 2001                                Mats Naslund, Ericsson
                                                  Karl Norrman, Ericsson
                                                       David Oran, Cisco

                                                           February 2001

                The Secure Real Time Transport Protocol

Status of this memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that other
   groups may also distribute working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time. It is inappropriate to use Internet-Drafts as reference
   material or cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at


   This document describes the Secure Real Time Transport Protocol
   (SRTP), a profile of the Real Time Transport Protocol (RTP) which can
   provide privacy, message authentication, replay protection, and
   implicit header authentication.

   SRTP can achieve high throughput and low packet expansion by using an
   additive stream cipher for encryption, a universal hashing based

Blom et al.                                                     [Page 1]

INTERNET-DRAFT                    SRTP                     February 2001

   function for message authentication, and an 'implicit' index for
   sequencing based on the RTP sequence number.

   In addition, SRTP proves to be a suitable protection for heterogenous
   environments, i.e. environments including both wired and wireless


   1. Notational Conventions.........................................2
   2. Goals..........................................................3
   3. SRTP Overview..................................................4
   3.1 SRTP Cryptographic Contexts...................................5
   3.2 Mapping SRTP Packets to Cryptographic Contexts................5
   3.3 SRTP Packet Processing........................................6
   3.4 Cryptographic Algorithms......................................7
   4. Synchronization................................................8
   4.1. IV Formation for Implicit Header Authentication .............9
   5. Replay Protection.............................................10
   6. Encryption....................................................10
   6.1 Defined Ciphers..............................................11
   6.1.1. Counter Mode AES..........................................11
   6.1.2. AES in f8-Mode............................................12
   6.1.3. NULL Cipher...............................................13
   7. Message Authentication........................................13
   7.1 Default MAC: UMAC............................................14
   8. SRTP Parameters...............................................14
   9. Secure RTCP...................................................15
   10. Rationale....................................................17
   10.1 Synchronization.............................................18
   10.2 Replay Protection...........................................18
   10.3 Source Origin Authentication................................18
   10.4. Choice of Encryption Transform.............................19
   11. Security Considerations......................................20
   11.1. SSRC collision.............................................21
   11.2. Confidentiality of the RTP Payload.........................21
   11.3. Confidentiality of the RTP Header..........................22
   11.4. Integrity of RTP headers...................................22
   12. Multicast and Multi-unicast..................................22
   13. Acknowledgements.............................................23
   14. Author's Addresses...........................................23
   15. References...................................................23
   APPENDIX A: Test Vectors.........................................25

1. Notational Conventions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC-2119 [B97].

Blom et al.                                                     [Page 2]

INTERNET-DRAFT                    SRTP                     February 2001

   By convention, the most left bit (byte) is the most significant one.
   By XOR we mean bitwise addition modulo 2 of binary strings, and ||
   denotes concatenation. E.g. if C = A || B, then the most significant
   bits of C are the same as those of A, and the least significant bits
   of C equals those of B.

2. Goals

   The security goals for SRTP are to ensure:

   * the privacy of the RTP payload,

   * the authentication of the entire RTP packet, including protection
   against replayed RTP packets, and

   * implicit authentication of the header.

   Each of the security services described above is optional. Any
   combination of options can be provided, except the single option of
   implicit header authentication.

   Source origin authentication (e.g., digitally signed packets) may be
   desirable in some situations, but this goal is deferred from
   consideration in this document. See Section 10.3 for a discussion on
   this point.

   Other goals for the protocol are:

   * a low computational cost,

   * a low footprint (i.e., small code size and data memory for key
     schedules and replay lists),

   * limited packet expansion,

   * no error propagation (e.g., changing a single bit of an SRTP packet
   should change no more than one bit of the corresponding RTP packet),

   * the preservation of RTP header compression efficiency,

   * to allow cryptographic keys to be used by multiple RTP sessions

   * independence from the underlying transport used by RTP.

   These properties ensures that SRTP is a suitable protection scheme
   for both wired and wireless scenarios.

Blom et al.                                                     [Page 3]

INTERNET-DRAFT                    SRTP                     February 2001

3. SRTP Overview

   RTP is the Real Time Protocol [SCFJ96].  We define SRTP as a profile
   of RTP, in an analogous way to RFC1890 which defines the audio/video
   profile for RTP. Conceptually, we consider a 'bump in the stack'
   implementation which resides between the RTP application and the
   transport layer, which intercepts RTP packets and then forwards an
   equivalent SRTP packet on the sending side, and which intercepts SRTP
   packets and passes an equivalent RTP packet up the stack on the
   receiving side.

          0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   |   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
   |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   |                           timestamp                           |
   |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   |           synchronization source (SSRC) identifier            |
   |   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |   |            contributing source (CSRC) identifiers             |
   |   |                               ....                            |
   |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   |                   RTP extension (optional)                    |
   | +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | | |                                                               |
   | | |                           payload                             |
   | | |                             ....                              |
   | | |                     authentication tag (optional)             |
   | | |
   | | |                             ....                              |
   | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | |
   | +- Encrypted Portion
   +---- Authenticated Portion

                    Figure 1.  The format of an SRTP packet.

   The format of an SRTP packet is illustrated in Figure 1. The optional
   authentication tag is the only field defined by SRTP that is not in
   RTP. It provides data origin authentication of the header and
   payload, and it indirectly provides replay protection by

Blom et al.                                                     [Page 4]

INTERNET-DRAFT                    SRTP                     February 2001

   authenticating the sequence number. The Encrypted Portion of an
   SRTP packet consists of the RTP payload of the equivalent RTP packet.
   The Authenticated Portion of an SRTP packet consists of the entire
   equivalent RTP packet.

3.1 SRTP Cryptographic Contexts

   Each SRTP session requires the sender and receiver to maintain
   cryptographic state information. This information is called the
   cryptographic context, and it consists of:

   * an encryption key k_e, and a optionally "salting key" k_s. These
   keys must be randomly and independently chosen.

   * a 32-bit rollover counter r (which records how many times the
   16-bit RTP sequence number has been reset to zero after passing
   through 65,535),

   * an 8-bit FLAG used to signal additional information,

   * the mode of operation for the encryption scheme, and

   * the cipher.

   In addition, when authentication and replay protection are provided:

   * a message authentication key k_a,

   * a sequence number s_l (which is the last received and authenticated
     sequence number for the receiver, and is the last sequence number
     sent for the sender), and

   * a replay list L (maintained by the receiver only).

3.2 Mapping SRTP Packets to Cryptographic Contexts

   In this section we define the mapping of RTP and SRTP packets to the
   cryptographic contexts used to protect them.

   The RTP synchronization source (SSRC) identifier is used, along with
   the RTP transport address (e.g., the Destination IP Address and Port
   Number) by a receiver to identify the proper cryptographic context
   for each packet.

   Recall that an RTP session is defined [SCFJ96] by a pair of
   destination Transport Addresses (one network address plus a port pair
   for RTP and RTCP), and that a multimedia session is defined as a
   collection of RTP sessions. For example, a particular multimedia

Blom et al.                                                     [Page 5]

INTERNET-DRAFT                    SRTP                     February 2001

   session could include an audio RTP session, a video RTP session, and
   a text RTP session.

   An SSRC identifier is unique inside an RTP session, and all packets
   with the same SSRC form part of the same timing and sequence number
   space. Thus, the SSRC field and transport address information can be
   used by an SRTP receiver (or by a bump in the stack implementation on
   the sender's side) to identify the proper cryptographic context
   within that session. Note though that, for instance in a multicast
   scenario, the RTP anti-collision mechanism for SSRCs may force these
   identifiers to change over time, see discussion in Section 12.

   SRTP may allow the different RTP sessions to use identical
   cryptographic keys. This is possible if the design of the
   synchronization mechanism (i.e., the IV in the case of the F8 and
   Counter Modes) avoids keystream re-use (the two-time pad, Section 11)
   and with uniqueness requirements on SSRC beyond that dictated by the
   RTP standard, see Section 12. However, different multimedia sessions
   SHOULD use different keys.

   The authentication and encryption keys of each context MUST remain
   fixed for the duration of that context. This ensures that incorrect
   keys will not be used by the receiver due to a synchronization error.

3.3 SRTP Packet Processing

   When Generic Forward Error Correction is performed as specified in
   RFC 2733, then the security processing takes place before FEC on the
   sender's side, and after FEC on the receiver's side.

   To construct a proper SRTP packet, given an RTP packet, the sender
   does the following:

   1. Determine which cryptographic context to use by checking the
   SSRC field of the RTP packet, and the Transport Address information
   of that packet (e.g., the Destination IP Address and Port Number).

   2. Determine the index of the SRTP packet as described in Section 4,
   using the rollover counter in the cryptographic context and the
   sequence number in the RTP packet. Form the current initialization
   vector (IV). If Implicit Header Authentication is provided, this can
   be done as described in Section 4.1.

   3. Encrypt the Encrypted Portion of the packet, as described in
   Section 6, using the IV determined in Step 2 and the encryption key
   and salting key in the context found in Step 1.

   4. If authentication is provided, compute the authentication tag for
   the Authenticated Portion of the packet, as described in Section 7,
   using the index determined in Step 2 and the authentication key in

Blom et al.                                                     [Page 6]

INTERNET-DRAFT                    SRTP                     February 2001

   the context found in Step 1. Note that the Encrypted Portion is
   encrypted before the authentication tag is computed.

   To authenticate and decrypt a SRTP packet, the receiver does the

   1. Determine which cryptographic context to use by checking the
   SSRC field of the RTP packet and the transport address information of
   the underlying transport header (e.g., the Destination IP Address and
   Port Number).

   2. Determine the index of the SRTP packet from the rollover counter
   in the cryptographic context and the sequence number in the RTP
   packet, as described in Section 4. Form the current IV in the same
   way as done in Step 2 in the encryption process.

   3. If authentication is provided, check the Replay List to ensure
   that no packet with that index has been received and authenticated
   before, as described in Section 5. If that index is in the list, then
   the packet has been replayed and is invalid. It MUST be discarded,
   and the event SHOULD be logged.

   Compute the authentication tag for the Authenticated Portion of the
   packet, as described in Section 7, using the index determined in Step
   2 and the authentication key in the context found in Step 1. Note
   that the Encrypted Portion is not decrypted before the authentication
   tag is computed.

   If the authentication tag that is computed matches that in the SRTP
   packet, then the packet is accepted and the index is added to the
   Replay List. Otherwise, the packet is invalid: it MUST be discarded,
   and the event SHOULD be logged.

   4. Decrypt the Encrypted Portion of the packet, as described in
   Section 6, using the IV determined in Step 2 and the encryption key
   and salting key in the context found in Step 1.

   The processing occurring when replay protection is activated has been
   chosen to maximize resistance to denial of service attacks (i.e., to
   minimize the receiver's effort in processing spurious packets).

3.4 Cryptographic Algorithms

   Default encryption and authentication algorithms are specified in
   Sections 6.1 and 7.1. While there are numerous encryption and message
   authentication algorithms that can be used in SRTP, we define default
   algorithms in order to avoid the complexity of specifying the
   encodings for the signaling of algorithm and parameter identifiers.

Blom et al.                                                     [Page 7]

INTERNET-DRAFT                    SRTP                     February 2001

4. Synchronization

   SRTP implementations use an 'implicit' packet index for sequencing.
   Receiver-side implementations use the RTP sequence number to
   reconstruct the correct index (that is, location in the sequence of
   all RTP packets). The index is defined as s + r * 65,536, where the
   sequence number is s and the rollover counter is r.

   A robust approach for the proper use of a rollover counter requires
   that its handling and use be well defined. In particular, out-of-
   order RTP packets with sequence numbers close to 65,536 or zero must
   be properly dealt with.

   A receiver reconstructs the index i of a packet with sequence number
   s using the estimate

   i = 65,536 * t + s,

   where t is chosen from the set { r-1, r, r+1 } such that i is closest
   to the value 65,536 * r + s_l. If the value r+1 is used, then the
   rollover counter r in the cryptographic context is incremented by

   The pseudocode for the algorithm to process a packet with sequence
   number s follows:

      if (s_l < 32,768)
         if (s - s_l > 32,768)
            set i to s + 65,536 * (r-1)
            set i to s + 65,536 * r
         if (s_l - 32,768 > s)
            set r to r + 1
         set i to s + r * 65,536
      set s_l to s

   The index i is used in replay protection (Section 5) when
   authentication is provided, in encryption (Section 6), and in message
   authentication (Section 7).

   This algorithm should be extended by using the information in the
   authenticated RTCP reports.

   When RTP authentication is not present, robust synchronization is not
   possible. In this case, transmission errors or an active attacker may
   force the receiver to erroneously update his rollover counter and
   thus to become completely out of synch. It is not possible to protect

Blom et al.                                                     [Page 8]

INTERNET-DRAFT                    SRTP                     February 2001

   against active attackers in such case, but it is possible to have an
   update policy for the rollover counter which, except in rare cases,
   is robust with respect to random bit errors.

   As the rollover counter is 32 bits long, the maximum number of
   packets in any given SRTP session is 2^48 = 281,474,976,710,656.
   After that number of SRTP packets have been sent, the sender MUST
   not send any more packets with that cryptographic context. This
   limitation enforces a security benefit by providing an upper bound on
   the amount of traffic that can pass before cryptographic keys are

   Other approaches to sequencing were considered and rejected; please
   see Section 10.1 for our rationale.

4.1. IV Formation for Implicit Header Authentication

   There may be several alternatives for the Initialization Vector (IV)
   formation. To guarantee synchronization and avoid keystream re-use,
   we only require the SSRC, rollover counter and sequence number, or
   some function thereof (possibly combined with re-keying mechanisms),
   to be part of the IV. Below, we give a concrete proposal which also
   provides 'implicit' header authentication, and works with every
   cipher having at least 128-bit block size. This particular solution
   also gives a high degree of agreement between bit ordering in the RTP
   packet header and the IV, simplifying data copying.

   When implicit header authentication is provided, data from each RTP
   packet to be encrypted and transmitted, must be included in the(IV).
   This IV shall be computed and supplied as input to the ciphering
   algorithm. This shall be done by taking information of said RTP
   packet, the FLAG, and the rollover counter value, and computing the
   128-bit IV:

    IV = ROC || FLAG || M || PT  || SEQ || TS || SSRC

   where TS (Timestamp, 32 bits), SEQ (Sequence Number, 16 bits), M
   (Marker Bit, 1 bit), PT (Payload Type, 7 bits), and SSRC
   (Synchronization Source, 32 bits) are taken from the current RTP
   header. ROC is the 32-bit rollover counter from the identified
   context. FLAG is a 8-bit value which is used to signal additional
   information. Currently, the only value defined (for RTP) is FLAG =
   00..0. The value 00..01 is reserved for RTCP and MUST not be used
   with RTP.

   With this IV formation, the number of SRTP packets encrypted with any
   fixed encryption key MUST therefore be no more than 2^48. Otherwise,
   the size of the ROC ..||..SEQ .. field will not be large enough to
   avoid keystream reuse.

Blom et al.                                                     [Page 9]

INTERNET-DRAFT                    SRTP                     February 2001

5. Replay Protection

   A packet is 'replayed' when it is stored by an adversary, and then
   re-injected onto the network. SRTP provides protection against such
   attacks whenever authentication is provided, through the storage of
   the indices of the most recently received and authenticated packets.

   Each SRTP receiver maintains a Replay List, which conceptually
   contains the indices of all of the packets which have been received
   and authenticated. In practice, the list can use a 'sliding window'
   approach, so that a fixed amount of storage suffices for replay
   protection. SRTP packet indices which are less than s_l * 65,536 -
   SRTP-WINDOW-SIZE MAY be assumed to have been received, where SRTP-
   WINDOW_SIZE is a parameter that MUST be at least 64, and which  MAY
   be set to a higher value.

   The Replay List can be efficiently implemented by using a bitmap to
   represent which packets have been received, as described in the
   Security Architecture for IP [KA98a].

6. Encryption

   Encryption uses a 'seekable' additive stream cipher, following the
   Stream Cipher ESP [sc-esp]. The stream ciphers that can be used must
   be able to efficiently seek to arbitrary locations in their
   keystream. Ciphers that can do this include SEAL [RC94, RC98],
   LEVIATHAN [MF00b], and any block cipher run in suitable mode. In
   particular, AES in counter mode will provide good security,
   reasonable performance, and conform to emerging U.S. Federal
   standards. Another mode which fulfils the requirements is f8 mode
   [ES3D], used together with AES.

   SRTP encryption consists of generating a keystream segment
   corresponding to the index of the packet, and then bitwise exclusive-
   oring that keystream segment into the RTP packet, starting at the
   first bit of the RTP payload. Decryption is then done the same way,
   but swapping the roles of the plaintext and ciphertext. The
   definition of how the keystream is generated, given the index,
   depends on the cipher and its mode of operation.

   Such a cipher shows features which are desired in a general scenario,
   e.g. low computational cost, and speed. It also shows properties
   which fulfil additional requirements posed by the cellular
   environment [BCNN00], i.e. preservation of RTP header compression
   efficiency, and absence of error propagation and message expansion.

   Hence, we conclude that the proposed profile can be applied to the
   most general heterogenous environment.

Blom et al.                                                    [Page 10]

INTERNET-DRAFT                    SRTP                     February 2001

6.1 Defined Ciphers

   The default cipher is the Advanced Encryption Standard (AES), and we
   define two modes of running AES, Counter Mode AES and AES in f8-Mode.
   Both of these modes provide implicit header authentication through
   the use of the IV formation described in Section 4.1. The NULL cipher
   is also defined, to be used when encryption is not required.

6.1.1. Counter Mode AES

   The default cipher SHALL be AES used in the Segmented Integer Counter
   Mode (SICM) [M00], with a 128-bit key size and a 128-bit block size.

   Conceptually, counter mode consists of encrypting successive
   integers. The actual definition is somewhat more complicated, in
   order to avoid 128 bit integer arithmetic and to randomize the
   starting point of the integer sequence. Each packet is encrypted with
   a distinct keystream segment, which is computed as follows.

   The 128-bit block is divided into three parts: a 64-bit segment
   prefix, a 32-bit block index, which is incremented to generate a
   keystream segment, and a 32-bit segment suffix. The segment
   prefix/suffix pair is unique for each keystream segment.

   A keystream segment is the concatenation of the output blocks of the
   cipher in encrypt mode, in which the block indicies are in increasing
   order. Symbolically, each keystream segment looks like

   E(A || B || C) || E(A || B + 1 mod 2^32 || C) || E(A || B + 2 mod
   2^32 || C) ..

   where A, B, and C are segment prefix, block index, and segment
   suffix, respectively, determined as given below.

   The offsets are computed from the salting key k_s and the IV (from
   Section 4.1) by exclusive-oring k_s and the IV, and setting A to the
   first 64 bits of the result, B as the following 32 and C to the
   remaining 32 bits of the result. Symbolically,

   A || B || C = IV XOR k_s.

   If k_s is less than 128 bits long, then k_s is concatenated with
   itself as many times as needed in order to form the salt which is
   added to the IV. If no salting key is used, this is interpreted as
   k_s = 0.

   Note that the segment prefix/suffix pair is distinct for each packet
   which is encrypted, thus ensuring that keystream segments are
   distinct and non-overlapping.

Blom et al.                                                    [Page 11]

INTERNET-DRAFT                    SRTP                     February 2001

   The restriction on the maximunm number of RTP packets above ensures
   the security of the encryption method by limiting the effectiveness
   of probabilistic attacks [BR98].

   The AES has a block size of 128 bits, so 2^32 output blocks are
   sufficient to generate the 2^7 * 2^32 = 549755813888 bits of
   keystream needed to encrypt the largest possible RTP packet.

6.1.2. AES in f8-Mode

   To encrypt UMTS (Universal Mobile Telecommunications System, as 3G
   networks) data, a solution (see [ES3D]) known as the f8-algorithm has
   been developed. On a high level, the proposed scheme is a variant of
   Output Feedback Mode (OFB) [HAC], with a more elaborate
   initialization and feedback function. As in normal OFB, the core
   consists of a block cipher. We define the use of AES as default block
   cipher to be used in f8-Mode for RTP encryption, with 128-bit key and
   block size.

   Figure 2 shows the structure of an arbitrary b-bit block size cipher,
   E, running in what we shall call "f8-mode of operation".

                |      |
            --->|  E   |
           |    |      |
           |    +------+
           |        |
     m --> *        |---------------------------  ...     -------|
   _____   |    IV' |           |             |                  |
           |        |  ct=1 --> *    ct=2 --> *   ... ct=L-1 --> *
           |        |           |             |                  |
           |        |       --> *         --> *   ...        --> *
           |       \|/     |   \|/       |   \|/            |   \|/
           |    +------+   | +------+    | +------+         | +------+
           |    |      |   | |      |    | |      |         | |      |
     k -------->|  E   |   | |  E   |    | |  E   |         | |  E   |
                |      |   | |      |    | |      |         | |      |
                +------+   | +------+    | +------+         | +------+
                    |      |    |        |    |             |    |
                    |------     |--------     |    ...  ----     |
                    |           |             |                  |
                   \|/         \|/           \|/                \|/

                   S(0)        S(1)          S(2)  . . .       S(L-1)

Blom et al.                                                    [Page 12]

INTERNET-DRAFT                    SRTP                     February 2001

   Figure 2. f8-mode of operation (asterisk, *, denotes bitwise XOR).

   Let E(k,B) be the 128-bit output of E in encrypt mode when applied to
   the 128-bit key k and 128-bit plaintext block B. Let ct, IV, IV',
   S(j), and m denote 128-bit blocks, determined below.

   The S() keystream for an n-bit message is defined by setting IV' =
   E(k XOR m, IV), and ct = S(-1) = 00..0. For j = 0,1,.., L-1 where L =
   n/128 (rounded up to nearest integer) compute

         S(j) = E(k,IV' XOR ct XOR S(j-1)),            (Eq. 1)
         ct   = ct + 1 mod 2^128                       (Eq. 2)

   Notice that the IV (as defined in Section 4.1) is not used directly.
   Instead it is fed through E under another key to produce an internal,
   "salted" value (denoted IV') to prevent an attacker from gaining
   known input/ouput pairs, and the roll of the internal counter is to
   prevent short keystream cycles. The value of the key mask m is
   defined to be

     m = k_s || 0x555..5,

   i.e. the salting key, padded with the the binary pattern 0101.. to
   fill the 128-bit key size. (If no salting key is used, m = 0x55..5.)

   The maxium allowable packet size can be determined as follows.
   The AES has a block size of 128 bits. Assuming that AES behaves like
   a random function, it is (heuristically) secure to generate about
   2^64 output blocks, which is sufficient to generate the 2^71 bits of
   keystream. In practise though, the counter ct above will often be
   sufficient if implemented as a 16- or 32-bit counter. In fact, for
   some security margin, other methods SHOULD be used if packets of size
   exceeding 2^32 * 128 = 549755813888 bits are to be encrypted.

6.1.3. NULL Cipher

   The NULL cipher is used when no confidentiality is requested. It
   simply copies the plaintext input into the ciphertext output.

7. Message Authentication

   Message integrity and authentication (hereafter referred to as just
   "authentication") are optional functions provided by SRTP.
   Authentication can be provided by any message authentication code,
   though the default value is UMAC [KBHHKR00].

Blom et al.                                                    [Page 13]

INTERNET-DRAFT                    SRTP                     February 2001

   The authentication tag is computed by applying the UMAC function to
   the Authenticated Portion of the SRTP packet.

   The authentication tag is appended to the RTP packet. This expansion
   of the RTP packet may cause the packet size to exceed the Maximum
   transmission Unit (MTU) of a network interface on its path,
   especially in circumstances when the application is attempting to
   'optimize' the size of packets. MTU path discovery SHOULD be used to
   avoid this problem.

   Authentication SHOULD be provided by SRTP. The fact that
   authentication is optional is motivated by the fact that, while the
   function is typically highly desired, there are certain cases
   (notably in the cellular environment) where it has an impact in terms
   of cost, as motivated in [BCNN00]. In those cases, it is up to the
   user security profile to request authentication.

7.1 Default MAC: UMAC

   The default message authentication code is UMAC [KBHHKR00], which
   has proven security properties and is quite fast. Furthermore, it
   can be used with short (e.g., two or four byte) authentication tags,
   as well as larger tags.

   UMAC is a parameterized algorithm (see Section 2.1 of [KBHHKR00]).
   The default selection of UMAC parameters for SRTP are:

      WORD-LEN              2
      UMAC-OUTPUT-LEN       4
      L1-KEY-LEN            128
      UMAC-KEY-LEN          16

   This choice of parameters is intended to work well on low-power
   processors, to minimize packet expansion, and to minimize the size of
   the cryptographic context. The WORD-LEN of two will work well on 16
   bit and higher processors. The packet expansion is determined by the
   UMAC-OUTPUT-LEN to be only four bytes. The storage requirement, per
   cryptographic context, is 144 bytes. These parameters ensure a
   forgery probability of no greater than 1/2^30 for each individual
   packet. Please see the security considerations section in [KBHHKR00]
   and the references therein for a more detailed discussion.

8. SRTP Parameters

   The SRTP-WINDOW-SIZE is defined to be at least 64 (Section 5).

Blom et al.                                                    [Page 14]

INTERNET-DRAFT                    SRTP                     February 2001

   The current defined modes are Counter Mode (default), f8 Mode
   (Section 6), and the NULL Cipher. The default cipher is AES (Section
   6), used with a block- and encryption key size of 128 bits.

9. Secure RTCP

   Secure RTCP follows the definition of Secure RTP, but defines the
   index and IV differently. In order to differentiate these quantities,
   we refer to it as the SRTCP index and IV.

   SRTCP is defined as a profile of RTCP, and it adds two new fields
   to the RTCP packet definition, the SRTCP index and the authentication
   tag. Those fields are appended to an RTCP packet in order to form an
   equivalent SRTCP packet, so that they follow any other profile-
   specific extensions. An SRTCP packet is illustrated in Figure 3.

        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   |   |V=2|P|    RC   |   PT=SR=200   |             length            |
   |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   |                         SSRC of sender                        |
   | +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   | | |                              ...                              |
   | | |                          sender info                          |
   | | |                              ...                              |
   | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | | |                              ...                              |
   | | |                         report block 1                        |
   | | |                              ...                              |
   | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | | |                              ...                              |
   | | |                         report block 2                        |
   | | |                              ...                              |
   | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | | |                                                               |
   | | |                              ...                              |
   | | |                                                               |
   | | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   | | |                              ...                              |
   | | |                  profile-specific extensions                  |
   | | |                              ...                              |
   | +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   | | |                           SRTCP index                         |
   | | |                              ...                              |
   | | |                       authentication tag                      |
   | | |                              ...                              |
   | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | |

Blom et al.                                                    [Page 15]

INTERNET-DRAFT                    SRTP                     February 2001

   | +-- Encrypted Portion
   +---- Authenticated Portion

   Figure 3.  The format of a Secure RTCP packet, after Section 6.3.1 of
   [SCFJ96]. In this case, the underlying RTCP packet is a sender report
   packet; the SRTP format is identical for other RTCP packet types.

   The SRTCP index is a 32-bit value. As we allow both encrypted and
   non-encrypted packets belonging to the same flow (see discussion
   below), indices with their most significant bit set to '1' are
   reserved for encrypted packets, and indices with most significant bit
   set to '0' are used for non-encrypted packets. With this restriction,
   the rest of the bits are set to zero before the first SRTCP packet is
   sent, and is incremented by one after each SRTCP is sent. Except for
   differences in the most significant bit, SRTCP indices form a
   strictly increasing sequence. The index is explicitly included in
   each packet, in contrast to the 'implicit' index approach used for

   SRTCP packet processing is identical to that of SRTP packet
   processing, with the following changes:

   * SRTCP replay protection is as defined in Section 5, but using the
   the SRTCP index as the index i.

   * SRTCP encryption is as defined in Section 6, but using the
   definition of the SRTCP Encrypted Portion as defined in this
   section, using the SRTCP index as the index i, and the IV as defined
   in this section.

   * The SRTCP authentication tag is defined as in Section 7, but
   applying the UMAC function to the Authenticated Portion of the SRTCP
   packet as defined in this section, and using the SRTCP index as the
   index i.

   * SRCTP decryption is performed as in Section 6, but only if the
   SRTCP index has its most significant bit equal to 1. If so, the
   encrypted portion is decrypted, using the SRTCP index as the index i,
   and the IV as defined in this section. In case the most significant
   bit of the index is 0, the payload is simply copied.

   The IV for ciphers using 128-bit block size is formed in the
   following way:

   IV = SRTCP index || FLAG || PT || 0..0 || SSRC

   where PT (Payload Type, 8 bit), and SSRC (Synchronization Source, 32
   bits) are taken from the first header in the RTCP compound packet.
   SRTCP index is the added 32-bit index to the packet. A pad of 48
   zeros is inserted between the PT and the SSRC.

Blom et al.                                                    [Page 16]

INTERNET-DRAFT                    SRTP                     February 2001

   FLAG is a 8-bit value which is used to signal additional information.
   Currently, the only value defined (for RTCP) is FLAG = 00..01. The
   value 0..0 is reserved for RTP and MUST not be used for RTCP. This
   allows to use the same key for related RTP and RTCP flows (being the
   IV unique).

   Then this IV is treated in the same way as defined in Section 6,
   according to the chosen encryption mode.

   The encryption prefix (Section 6.1 of [SCFJ96]), which is a random
   32-bit quantity intended to improve privacy, SHOULD NOT be used. This
   is because SRTP encryption uses an additive stream cipher, and thus
   the prefix offers no benefit.

   The maximum number of SRTCP packets is limited to 2^31 =
   2,147,483,648. The last RTCP packet MUST contain an RTCP BYE. SRTCP
   senders MUST send an RTCP BYE in the final packet, if the maximum
   number of SRTCP packets is reached. Similarly, SRTCP receivers MUST
   act as though the last RTCP packet included a BYE, even if no BYE was
   included in the packet, if the maximum number of SRTCP packets is

   Authentication MUST be required for RTCP, being it the control
   protocol (e.g., it has a BYE packet). Moreover, the cost for RTCP
   authentication is not of the same order of RTP authentication, being
   the session bandwidth allocated to RTCP recommended at 5%. However,
   when adding authentication to RTCP, the overhead in bandwidth SHOULD
   be considered (it will be more than 5%).

   It is allowed to split a compound RTCP packet into two lower-layer
   packets, one to be encrypted and one to be sent in the clear, as
   described in Section 9.1 of [SCFJ96].

   Encryption/non-encryption is signaled by the most significant bit of
   the SRTCP index as described above.

10. Rationale

   SRTP achieves high throughput and low packet expansion by using fast
   stream ciphers for encryption, an implicit index for synchronization,
   and universal hash functions for message authentication. SRTP shows
   to be a suitable choice for the most general scenario, and to fit
   also the most demanding one, conversation multimedia over wireless,
   having it the necessary robustness properties.

   Only a single header extension may be appended to the RTP data
   header, so the use of a header extension for SRTP was avoided. SRTP
   and SRTCP are defined as profiles of RTP and RTCP, respectively.

Blom et al.                                                    [Page 17]

INTERNET-DRAFT                    SRTP                     February 2001

10.1 Synchronization

   RTP runs over unreliable transport. Thus, maintaining synchronization
   of the cryptographic context between the sender and receiver is a
   conspicuous challenge. Because of the requirement to minimize packet
   expansion, no explicit sequencing information should be added. RTP
   packets contain two fields for synchronization purposes, the
   timestamp and the sequence number. The timestamp field could be used
   for cryptographic synchronization in some circumstances. However,
   this field is not appropriate for such use. From [SCFJ96]:

   Several consecutive RTP packets may have equal timestamps if they are
   (logically) generated at once, e.g., belong to the same video frame.
   Consecutive RTP packets may contain timestamps that are not monotonic
   if the data is not transmitted in the order it was sampled, as in the
   case of MPEG interpolated video frames.

   The RTP sequence number might be directly used as a unique identifier
   for SRTP packets. However, it has only sixteen bits, which would
   limit the duration of an SRTP security association to only 64,536
   packets, asking therefore for periodically rekeying.

   The 'implicit index' approach works as long as the reorder and loss
   of the packets is not too great. In particular, 32,768 packets would
   need to be lost, or a packet would need to be 32,768 packets out of
   sequence in order for synchronization to be lost. Such drastic loss
   or reorder is likely to disrupt the RTP application itself.

   When a participant joins an SRTP session while that session is in
   progress, the entire cryptographic context except for the replay
   list is sent to that participant. This step is essential for
   security. See also Section 12.

10.2 Replay Protection

   Replay protection is undoubtedly important for multimedia data, and
   SHOULD be provided. Otherwise, it would be possible for an adversary
   to perform simple manipulations on data that subverted security. For
   example, in a voice application, the phrase "yes" could be
   substituted for "no" if replay protection were not present. However,
   there are certain scenarios, e.g. conversation multimedia, where it
   may be difficult to perform such a kind of attacks. Moreover, to be
   useful, replay protection needs to be based on an authentication
   mechanism (i.e., authentication of the sequence number of the RTP
   header), and this has a cost when cellular links are involved on the

10.3 Source Origin Authentication

Blom et al.                                                    [Page 18]

INTERNET-DRAFT                    SRTP                     February 2001

   'Source origin authentication' was listed as an option in the
   security goals, not because it is not an appropriate goal, but
   because it may not be achievable. This goal may be desirable in some
   circumstances, such as multicast environments in which the sender
   is more trusted than the receivers, or when translators or mixers
   (Section 2.3 of [SCFJ96]) are used. However, it is not clear that
   this capability can always be provided, as mixers and translators can
   change the payload. Furthermore, this security service essentially
   requires digital signatures (at least if collusion resistance is
   required [BF00]).

   Two examples of the multicast scenario mentioned above are a
   military commander addressing his troops over RTP, and financial
   market data sent over RTP. In these situations, a 'stream signing'
   method can provide digital signatures on the entire RTP packets. An
   extensive literature on such methods is developing, and it is
   reasonable to expect that one of these methods can be reduced to
   practice and specified for RTP. This suggests that it should be left
   as an option in the current specification. A future effort can define
   a stream signing method as an authentication type for RTP, which
   could be used as a replacement for a message integrity transform.

   Examples of the mixer and translator scenarios include a translator
   re-encoding data at a lower rate or in a different encoding, and a
   mixer combining the audio streams of multiple speakers in a
   teleconference. In these cases, it is not clear that meaningful
   source origin authentication is possible, as the data that is
   received is not the same as the data that is signed. If the
   translator is trusted by the receivers, then it could sign or re-sign
   the data streams, but this scenario may not be prevalent. It may be
   possible to devise a signing scheme that authenticates the source but
   not the content (enabling the receivers to know that "John is one of
   the people talking", but not providing authentication on who said
   what) by signing the concatenation of the Contributing source (CSRC)
   field and some sequencing information (e.g., a timestamp or sequence
   number), but such schemes require synchronization between the
   senders. This synchronization is not required by the RTP protocol
   itself, and may be difficult or impossible to arrange.

10.4 Choice of Encryption Transform

   When adopting a block cipher mode to produce keystreams, the central
   ingredient is the block cipher which is its core. As far as modern
   cryptology knows, the security basically stands (and falls) with the
   security of the block cipher. This means that if a weakness is found,
   replacing the block cipher with a new one will most likely remedy the
   security problems. We define AES (Rijndael) [AES] as default block
   cipher, as it is widely believed to be secure.

Blom et al.                                                    [Page 19]

INTERNET-DRAFT                    SRTP                     February 2001

11. Security Considerations

   The security of UMAC is well understood, and is described in

   Additive ciphers do not provide any security service other than
   privacy. In particular, they do not provide message authentication
   (see [RK99] or [S96] for a discussion of this security service).
   However, SRTP uses a message authentication code to provide that
   security service.

   By using 'seekable' stream ciphers, SRTP avoids the denial of service
   attacks that are possible on stream ciphers that lack this property
   (these attacks are described in Section 3.4 of [B96]).

   No bit of keystream in an additive stream cipher should ever be used
   to encrypt multiple distinct plaintext bits. Such keystream reuse
   (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. In SRTP, a 'two-
   time pad' is avoided by requiring the key or the IV to be unique.

   An SSRC is mapped to a unique crypto context. Multiple crypto
   contexts may contain identical keys; in this case, each context
   together with data from the RTP header MUST produce a unique IV
   (which is typically assured by plugging the unique SSRC in the IV).

   If manual keying is used, two different cryptographic contexts might
   accidentally use the same encryption key with non-negligible
   probability, through manual error or procedural inadequacies. Thus,
   manual keying SHOULD NOT be used for SRTP (or SRTCP).

   An 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,Bi96]. 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, 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".

   In order to provide an effective key size of n bits in a deployment
   in which 2^m SRTP/SRTCP cryptographic contexts will be created, the
   true key size will need to be n+m bits. The value of m SHOULD be 32
   bits for networks with 50,000 connections (fully meshed networks

Blom et al.                                                    [Page 20]

INTERNET-DRAFT                    SRTP                     February 2001

   with up to 200 devices), and SHOULD be 64 bits for networks with
   49e+12 connections (fully meshed networks with up to 7,000,000
   devices). These choices of m ensures that key collision attacks
   amortized over a ten year period offer no advantage over exhaustive
   search, when new SRTP keys are established for every connection
   every hour (note that such an attack requires the storage of all
   network traffic over the ten year period). These choices will suffice
   for many networks, though SRTP deployments with more stringent
   security requirements will need to make a detailed assessment of
   those requirements with respect to the attacks described in [MF00].

   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.

   It is an important point that the m bits of 'extra' key provided to
   thwart these attacks need not be private. In jurisdictions with
   mandated limits on the length of a secret key, the additional key
   bits could be made public. This is because those bits are
   functionally equivalent to the 'salt' that is used to protect
   passwords from dictionary attacks. The fact that the 'extra' key bits
   are distinct for many different keys defeats the key collision and
   time-memory tradeoff attacks by reducing the number of keys over
   which cryptanalytic computation can be amortized.

   Note that other security protocols which use additive ciphers for the
   encryption of Internet traffic (e.g., SSL, TLS, SSH, IPSEC) are also
   vulnerable to the attacks described in [MF00]. Those attacks are
   generic to additive encryption of redundant plaintext, and are not
   particular to SRTP.

11.1 SSRC collision

   Assume that two or more communication parties use the same key.
   Though RTP implements an SSRC collision detection mechanism, it is
   impossible to guarantee that two parties do not accidently choose the
   same SSRC and send a few packets before the collision is detected. In
   a very unfortunate case, the IV formation in Section 4.1 could in
   fact make the keystreams collide and we have a 'two-time pad'. This
   is probably a bigger problem in the case of group communication when
   a single group key is desired. See also some administrative issues
   with SSRC collisions in Section 12.

11.2. Confidentiality of the RTP Payload

   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

Blom et al.                                                    [Page 21]

INTERNET-DRAFT                    SRTP                     February 2001

   parameter settings etc. This, in turn, implies that the corresponding
   bit of the keystream can be deduced. However, if the stream cipher is
   secure, knowledge of a few bits of the keystream will not aid an
   attacker in predicting the following keystream bits. Thus, the
   payload length (and information deducible from this) will leak, but
   nothing else.

11.3. Confidentiality of the RTP Header

   With our proposal, 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".

   Our proposal is a low-cost method, which allows header compression to
   reduce bandwidth. It is up to the endpoints policies to decide about
   the security scheme to employ. If the header compression is omitted,
   other solutions might be applicable, e.g. [sc-esp]. In other words,
   we provide a solution that works in the most general scenario, even
   in the most demanding one (like conversational multimedia over low-
   bandwidth, unreliable media. Of course the solution will then also
   work in less restricted environments, but we suggest that if one
   really needs to protect headers, and is allowed to do so by the
   surrounding environment, then he should also look at alternatives. In
   addition, we strongly recommend the use of profiles to select the
   right trade-off for the required level of security.

11.4 Integrity of RTP headers

   The IV formation in Section 4.1, which depends on the RTP header,
   provides an 'implicit' authentication of that header, which is useful
   when the authentication option is not present. This is because any
   attacks which modify the header of such a packet will cause the SRTP
   receiver to use an incorrect IV in the decryption step, with the
   result that the decrypted RTP payload will be essentially random.

12. Multicast and Multi-unicast

   The scheme described here can be used in case a single, unique key (a
   single pair, encryption group key and authentication group key) is to
   be used inside a multimedia session, for a low complexity key
   management. However, it then becomes necessary to have a way to
   assure that each SSRC is unique inside that multimedia session. This
   is a light and feasible solution in several scenarios, e.g. one
   sender only, streaming, and unicast.

Blom et al.                                                    [Page 22]

INTERNET-DRAFT                    SRTP                     February 2001

   In multicast and multi-unicast, to use the same group key for the
   multimedia session, there should be a way to guarantee uniqueness of
   the SSRC before starting sending. Otherwise, the triggering of the
   anti-collision mechanism will ask for a change in the SSRCs of the
   parties that happened to have the same SSRC, hence giving trouble in
   pointing to the right context.

   The problem remains how to address the context database after the
   anti-collision algorithm has changed the SSRCs. Section 3.3 defines
   the use of SSRC and Transport Address of that packet as selectors to
   the database. In case of UDP, the unchanged transport address can be
   a good indicator that a collision, followed by anti-collision
   triggering, has happened. So, simply try decryptions until a RTCP
   message confirms the change in the SSRC on that transport address and
   then update the database selector triplet.

   If the requirement of unique SSRC inside that multimedia session
   cannot be guaranteed (e.g., for large groups), then a unique key per
   sender should be used. The additional requirement is to have SSRC
   unique per sender, which appears to be feasible enough. However, the
   same consideration on the anti-collision algorithm triggerring

13. Acknowledgements

   The authors would like to thank Brian Weis and Magnus Westerlund for
   their reviews and comments.

14. Author's Addresses

    Questions and comments about this memo can be directed to:

      David A. McGrew
      David Oran
      Cisco Systems, Inc.
      San Jose, CA 95134-1706 USA
      mcgrew@cisco.com, oran@cisco.com

      Rolf Blom
      Elisabetta Carrara
      Mats Naslund
      Karl Norrman
      Ericsson Research
      {rolf.blom, elisabetta.carrara, mats.naslund,

15. References

Blom et al.                                                    [Page 23]

INTERNET-DRAFT                    SRTP                     February 2001

   [AES] NIST, "Advanced Encryption Standard (AES)",

   [B97]   Bradner, S., "Key words for use in RFCs to Indicate
   Requirement Levels", RFC 2119, March 1997.

   [BCNN00]  Blom, R., Carrara, E., Naslund, M., and Norrman, K.,
   "Conversational Multimedia Security in 3G Networks", Internet Draft,
   November 2000, <draft-blom-cmsec-3g-00.txt>.

   [BF00] Boneh, D., and Franklin, M., "Message Authentication in a
   Multicast Environment", the Proceedings of the Seventh Annual
   Workshop on Selected Areas in Cryptography (SAC 2000), Springer-

   [C99]   Crowell, W. P., "Introduction to the VENONA Project",

   [ES3D] ETSI SAGE 3GPP Standard Algorithms Task Force, "Security
   Algorithms Group of Experts (SAGE); General Report on the Design,
   Specification and Evaluation of 3GPP Standard Confidentiality and
   Integrity Algorithms", Public report, Draft Version 1.0, Dec 1999.

   [ES3E] ETSI SAGE 3GPP Standard Algorithms Task Force, "Security
   Algorithms Group of Experts (SAGE) Report on the Evaluation of 3GPP
   Standard Confidentiality and Integrity Algorithms", Public report,
   Draft Version 1.0, Dec 1999.

   [HAC]  Menezes, A., Van Oorschot, P., and Vanstone, S., "Handbook of
   Applied Cryptography", CRC Press, 1997, ISBN 0-8493-8523-7.

   [H80]   Hellman, M. E., "A cryptanalytic time-memory trade-off", IEEE
   Transactions on Information Theory, July 1980, pp. 401-406.

   [KA98a] Kent, S., and R. Atkinson, "Security Architecture for IP",
   RFC 2401, November 1998.

   [KBHHKR00] Krovetz, T., Black, J., Halevi, S., Hevia, A., Krawczyk,
   H., Rogaway, P., "UMAC: Message Authentication Code using Universal
   Hashing", Internet Draft, October 2000, <draft-krovetz-umac-01.txt>.

   [LRW00] Lipmaa, H., Rogaway, P., and Wagner, D., "Comments to NIST
   Concerning AES Modes of Operation: CTR-Mode Encryption", NIST
   Workshop on AES Modes of Operation,

   [M00]   McGrew, D., "Segmented Integer Counter Mode: Specification
   and Rationale", NIST Workshop on AES Modes of Operation,

Blom et al.                                                    [Page 24]

INTERNET-DRAFT                    SRTP                     February 2001

   [MF00]  McGrew, D., and Fluhrer, S., "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.

   [MF00b] McGrew, D., and Fluhrer, S., "The Stream Cipher LEVIATHAN:
   Specification and Supporting Documentation", Submission to the New
   European Schemes for Signatures, Integrity, and Encryption (NESSIE)
   Process, October, 2000http://www.cryptonessie.org/.

   [R92]   Rueppel, R., "Stream Ciphers", Chapter 2 of Simmons, G.,
   "Contemporary Cryptology: the Science of Information Integrity,"
   1992, IEEE Press.

   [RC94]  Rogaway, P. and Coppersmith, D., "A Software-Optimized
   Encryption Algorithm", Proceedings of the 1994 Fast Software
   Encryption Workshop, Lecture Notes In Computer Science, Volume 809,
   Springer-Verlag, 1994, pp. 56-63.

   [RC98]  Rogaway, P. and Coppersmith, D., "A Software-Optimized
   Encryption Algorithm", Journal of Cryptology, Volume 11, Number 4,
   Springer-Verlag, 1998, Pages 273-287.  Also available on the Internet
   at http://www.cs.ucdavis.edu/~rogaway/papers/seal-abstract.html.

   [RK99]  Rescorla, E., and Korver, B., "Guidelines for Writing RFC
   Text on Security Considerations," draft-rescorla-sec-cons-00.txt

   [S96]   Schneier, B. "Applied Cryptography: Protocols, Algorithms,
   and Source Code in C", Wiley, 1996.

   [sc-esp] McGrew, D., Fluhrer, S., Peyravian, M.,  "The Stream Cipher
   Encapsulating Security Payload", Internet Draft, July 2000

   [SCFJ96] Schulzrinne, H., Casner, S., Frederick, R., Jacobson, V.,
   "RTP: A Transport Protocol for Real-Time Applications", IETF Request
   For Comments RFC 1889.


A. Test vectors

   We include in the following some test vectors for f8-AES.


   salting key || 0x555... :

Blom et al.                                                    [Page 25]

INTERNET-DRAFT                    SRTP                     February 2001

   AES-internal expanded key:
     23482900 8467be18 6c3de14a ae72d62c
     62be58e4 e6d9e6fc 8ae407b6 2496d19a
     f080e0d2 1659062e 9cbd0198 b82bd002
     05f097be 13a99190 8f149008 373f400a
     78f9f024 6b5061b4 e444f1bc d37bb1b6
     4931be42 2261dff6 c6252e4a 155e9ffc
     31ea0e1b 138bd1ed d5aeffa7 c0f0605b
     fd3a37a1 eeb1e64c 3b1f19eb fbef79b0
     a28cd0ae 4c3d36e2 77222f09 8ccd56b9
     043d86ca 4800b028 3f229f21 b3efc998
     ede0c0a7 a5e0708f 9ac2efae 292d2636

   AES-internal expanded salting key || 555...:
     32f2870d 55555555 55555555 55555555
     cf0e7bf1 9a5b2ea4 cf0e7bf1 9a5b2ea4
     f43f3249 6e641ced a16a671c 3b3149b8
     37045eab 59604246 f80a255a c33b6ce2
     dd54c685 843484c3 7c3ea199 bf05cd7b
     a6e9e78d 22dd634e 5ee3c2d7 e1e60fac
     089f7675 2a42153b 74a1d7ec 9547d840
     e8fe7f5f c2bc6a64 b61dbd88 235a65c8
     d6b39779 140ffd1d a2124095 8148255d
     9f8cdb75 8b832668 299166fd a8d943a0
     9c963bb7 17151ddf 3e847b22 965d3882

   RTP-packet header fields:
     version      = 2
     padding      = 0
     extension    = 0
     CSRC count   = 0
     marker bit   = 0
     payload type = 6e
     sequence no. = 5cba
     timestamp    = 50681de5
     SSRC         = 5c621599

   Data from Cryptographic context:
   FLAG = 0
   Rollover counter = d462564a



   Encryption of bits 0 to 127:

Blom et al.                                                    [Page 26]

INTERNET-DRAFT                    SRTP                     February 2001

   ct: 0
   S(-1)                   : 00000000000000000000000000000000
   S(-1) XOR IV'           : 4fee844eedb458a3e2b0c7ed43888cc1
   S(-1) XOR IV' XOR ct    : 4fee844eedb458a3e2b0c7ed43888cc1
   plain text P[0..127]    : 6e915f07cd6f1c0d44afaab4961c7d31
   final keystream S(0)    : b2d3b3d7e16092de379e33b350582e63
   cipher text C[0..127]   : dc42ecd02c0f8ed373319907c6445352

   Encryption of bits 128 to 255:

   ct: 1
   S(0)                    : b2d3b3d7e16092de379e33b350582e63
   S(0) XOR IV'            : fd3d37990cd4ca7dd52ef45e13d0a2a2
   S(0) XOR IV' XOR ct     : fd3d37990cd4ca7dd52ef45e13d0a2a3
   plain text P[128..255]  : 7b9daad84352a6d4bcdf501a560832a0
   final keystream S(1)    : b1ce287dc53c1975de3d7d0500f780ba
   cipher text C[128..255] : ca5382a5866ebfa162e22d1f56ffb21a


   This Internet-Draft expires in July 2001.

Blom et al.                                                    [Page 27]

Html markup produced by rfcmarkup 1.129d, available from https://tools.ietf.org/tools/rfcmarkup/