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RFC 3711 - The Secure Real-time Transport Protocol (SRTP)

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Network Working Group                                         M. Baugher
Request for Comments: 3711                                     D. McGrew
Category: Standards Track                            Cisco Systems, Inc.
                                                              M. Naslund
                                                              E. Carrara
                                                              K. Norrman
                                                       Ericsson Research
                                                              March 2004

             The Secure Real-time Transport Protocol (SRTP)

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2004).  All Rights Reserved.


   This document describes the Secure Real-time Transport Protocol
   (SRTP), a profile of the Real-time Transport Protocol (RTP), which
   can provide confidentiality, message authentication, and replay
   protection to the RTP traffic and to the control traffic for RTP, the
   Real-time Transport Control Protocol (RTCP).

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
       1.1.  Notational Conventions . . . . . . . . . . . . . . . . .  3
   2.  Goals and Features . . . . . . . . . . . . . . . . . . . . . .  4
       2.1.  Features . . . . . . . . . . . . . . . . . . . . . . . .  5
   3.  SRTP Framework . . . . . . . . . . . . . . . . . . . . . . . .  5
       3.1.  Secure RTP . . . . . . . . . . . . . . . . . . . . . . .  6
       3.2.  SRTP Cryptographic Contexts. . . . . . . . . . . . . . .  7
             3.2.1.  Transform-independent parameters . . . . . . . .  8
             3.2.2.  Transform-dependent parameters . . . . . . . . . 10
             3.2.3.  Mapping SRTP Packets to Cryptographic Contexts . 10
       3.3.  SRTP Packet Processing . . . . . . . . . . . . . . . . . 11
             3.3.1.  Packet Index Determination, and ROC, s_l Update. 13
             3.3.2.  Replay Protection. . . . . . . . . . . . . . . . 15
      3.4.  Secure RTCP . . . . . . . . . . . . . . . . . . . . . . . 15

   4.  Pre-Defined Cryptographic Transforms . . . . . . . . . . . . . 19
       4.1.  Encryption . . . . . . . . . . . . . . . . . . . . . . . 19
             4.1.1.  AES in Counter Mode. . . . . . . . . . . . . . . 21
             4.1.2.  AES in f8-mode . . . . . . . . . . . . . . . . . 22
             4.1.3.  NULL Cipher. . . . . . . . . . . . . . . . . . . 25
       4.2.  Message Authentication and Integrity . . . . . . . . . . 25
             4.2.1.  HMAC-SHA1. . . . . . . . . . . . . . . . . . . . 25
       4.3.  Key Derivation . . . . . . . . . . . . . . . . . . . . . 26
             4.3.1.  Key Derivation Algorithm . . . . . . . . . . . . 26
             4.3.2.  SRTCP Key Derivation . . . . . . . . . . . . . . 28
             4.3.3.  AES-CM PRF . . . . . . . . . . . . . . . . . . . 28
   5.  Default and mandatory-to-implement Transforms. . . . . . . . . 28
       5.1.  Encryption: AES-CM and NULL. . . . . . . . . . . . . . . 29
       5.2.  Message Authentication/Integrity: HMAC-SHA1. . . . . . . 29
       5.3.  Key Derivation: AES-CM PRF . . . . . . . . . . . . . . . 29
   6.  Adding SRTP Transforms . . . . . . . . . . . . . . . . . . . . 29
   7.  Rationale. . . . . . . . . . . . . . . . . . . . . . . . . . . 30
       7.1.  Key derivation . . . . . . . . . . . . . . . . . . . . . 30
       7.2.  Salting key. . . . . . . . . . . . . . . . . . . . . . . 30
       7.3.  Message Integrity from Universal Hashing . . . . . . . . 31
       7.4.  Data Origin Authentication Considerations. . . . . . . . 31
       7.5.  Short and Zero-length Message Authentication . . . . . . 32
   8.  Key Management Considerations. . . . . . . . . . . . . . . . . 33
       8.1.  Re-keying  . . . . . . . . . . . . . . . . . . . . . . . 34
             8.1.1.  Use of the <From, To> for re-keying. . . . . . . 34
       8.2.  Key Management parameters. . . . . . . . . . . . . . . . 35
   9.  Security Considerations. . . . . . . . . . . . . . . . . . . . 37
       9.1.  SSRC collision and two-time pad. . . . . . . . . . . . . 37
       9.2.  Key Usage. . . . . . . . . . . . . . . . . . . . . . . . 38
       9.3.  Confidentiality of the RTP Payload . . . . . . . . . . . 39
       9.4.  Confidentiality of the RTP Header. . . . . . . . . . . . 40
       9.5.  Integrity of the RTP payload and header. . . . . . . . . 40
             9.5.1. Risks of Weak or Null Message Authentication. . . 42
             9.5.2.  Implicit Header Authentication . . . . . . . . . 43
   10.  Interaction with Forward Error Correction mechanisms. . . . . 43
   11.  Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . 43
       11.1. Unicast. . . . . . . . . . . . . . . . . . . . . . . . . 43
       11.2. Multicast (one sender) . . . . . . . . . . . . . . . . . 44
       11.3. Re-keying and access control . . . . . . . . . . . . . . 45
       11.4. Summary of basic scenarios . . . . . . . . . . . . . . . 46
   12. IANA Considerations. . . . . . . . . . . . . . . . . . . . . . 46
   13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 47
   14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 47
       14.1. Normative References . . . . . . . . . . . . . . . . . . 47
       14.2. Informative References . . . . . . . . . . . . . . . . . 48
   Appendix A: Pseudocode for Index Determination . . . . . . . . . . 51
   Appendix B: Test Vectors . . . . . . . . . . . . . . . . . . . . . 51
       B.1.  AES-f8 Test Vectors. . . . . . . . . . . . . . . . . . . 51

       B.2.  AES-CM Test Vectors. . . . . . . . . . . . . . . . . . . 52
       B.3.  Key Derivation Test Vectors. . . . . . . . . . . . . . . 53
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 55
   Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 56

1.  Introduction

   This document describes the Secure Real-time Transport Protocol
   (SRTP), a profile of the Real-time Transport Protocol (RTP), which
   can provide confidentiality, message authentication, and replay
   protection to the RTP traffic and to the control traffic for RTP,
   RTCP (the Real-time Transport Control Protocol) [RFC3350].

   SRTP provides a framework for encryption and message authentication
   of RTP and RTCP streams (Section 3).  SRTP defines a set of default
   cryptographic transforms (Sections 4 and 5), and it allows new
   transforms to be introduced in the future (Section 6).  With
   appropriate key management (Sections 7 and 8), SRTP is secure
   (Sections 9) for unicast and multicast RTP applications (Section 11).

   SRTP can achieve high throughput and low packet expansion.  SRTP
   proves to be a suitable protection for heterogeneous environments
   (mix of wired and wireless networks).  To get such features, default
   transforms are described, based on an additive stream cipher for
   encryption, a keyed-hash based function for message authentication,
   and an "implicit" index for sequencing/synchronization based on the
   RTP sequence number for SRTP and an index number for Secure RTCP

1.1.  Notational Conventions

   The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in [RFC2119].  The
   terminology conforms to [RFC2828] with the following exception.  For
   simplicity we use the term "random" throughout the document to denote
   randomly or pseudo-randomly generated values.  Large amounts of
   random bits may be difficult to obtain, and for the security of SRTP,
   pseudo-randomness is sufficient [RFC1750].

   By convention, the adopted representation is the network byte order,
   i.e., the left most bit (octet) is the most significant one.  By XOR
   we mean bitwise addition modulo 2 of binary strings, and || denotes
   concatenation.  In other words, if C = A || B, then the most
   significant bits of C are the bits of A, and the least significant
   bits of C equal the bits of B.  Hexadecimal numbers are prefixed by

   The word "encryption" includes also use of the NULL algorithm (which
   in practice does leave the data in the clear).

   With slight abuse of notation, we use the terms "message
   authentication" and "authentication tag" as is common practice, even
   though in some circumstances, e.g., group communication, the service
   provided is actually only integrity protection and not data origin

2.  Goals and Features

   The security goals for SRTP are to ensure:

   *  the confidentiality of the RTP and RTCP payloads, and

   *  the integrity of the entire RTP and RTCP packets, together with
      protection against replayed packets.

   These security services are optional and independent from each other,
   except that SRTCP integrity protection is mandatory (malicious or
   erroneous alteration of RTCP messages could otherwise disrupt the
   processing of the RTP stream).

   Other, functional, goals for the protocol are:

   *  a framework that permits upgrading with new cryptographic

   *  low bandwidth cost, i.e., a framework preserving RTP header
      compression efficiency,

   and, asserted by the pre-defined transforms:

   *  a low computational cost,

   *  a small footprint (i.e., small code size and data memory for
      keying information and replay lists),

   *  limited packet expansion to support the bandwidth economy goal,

   *  independence from the underlying transport, network, and physical
      layers used by RTP, in particular high tolerance to packet loss
      and re-ordering.

   These properties ensure that SRTP is a suitable protection scheme for
   RTP/RTCP in both wired and wireless scenarios.

2.1.  Features

   Besides the above mentioned direct goals, SRTP provides for some
   additional features.  They have been introduced to lighten the burden
   on key management and to further increase security.  They include:

   *  A single "master key" can provide keying material for
      confidentiality and integrity protection, both for the SRTP stream
      and the corresponding SRTCP stream.  This is achieved with a key
      derivation function (see Section 4.3), providing "session keys"
      for the respective security primitive, securely derived from the
      master key.

   *  In addition, the key derivation can be configured to periodically
      refresh the session keys, which limits the amount of ciphertext
      produced by a fixed key, available for an adversary to

   *  "Salting keys" are used to protect against pre-computation and
      time-memory tradeoff attacks [MF00] [BS00].

   Detailed rationale for these features can be found in Section 7.

3.  SRTP Framework

   RTP is the Real-time Transport Protocol [RFC3550].  We define SRTP as
   a profile of RTP.  This profile is an extension to the RTP
   Audio/Video Profile [RFC3551].  Except where explicitly noted, all
   aspects of that profile apply, with the addition of the SRTP security
   features.  Conceptually, we consider SRTP to be a "bump in the stack"
   implementation which resides between the RTP application and the
   transport layer.  SRTP intercepts RTP packets and then forwards an
   equivalent SRTP packet on the sending side, and intercepts SRTP
   packets and passes an equivalent RTP packet up the stack on the
   receiving side.

   Secure RTCP (SRTCP) provides the same security services to RTCP as
   SRTP does to RTP.  SRTCP message authentication is MANDATORY and
   thereby protects the RTCP fields to keep track of membership, provide
   feedback to RTP senders, or maintain packet sequence counters.  SRTCP
   is described in Section 3.4.

3.1.  Secure RTP

      The format of an SRTP packet is illustrated in Figure 1.

        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  ...                         | |
   | |                               +-------------------------------+ |
   | |                               | RTP padding   | RTP pad count | |
   | ~                     SRTP MKI (OPTIONAL)                       ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | :                 authentication tag (RECOMMENDED)              : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   |                                                                   |
   +- Encrypted Portion*                      Authenticated Portion ---+

   Figure 1.  The format of an SRTP packet.  *Encrypted Portion is the
   same size as the plaintext for the Section 4 pre-defined transforms.

   The "Encrypted Portion" of an SRTP packet consists of the encryption
   of the RTP payload (including RTP padding when present) of the
   equivalent RTP packet.  The Encrypted Portion MAY be the exact size
   of the plaintext or MAY be larger.  Figure 1 shows the RTP payload
   including any possible padding for RTP [RFC3550].

   None of the pre-defined encryption transforms uses any padding; for
   these, the RTP and SRTP payload sizes match exactly.  New transforms
   added to SRTP (following Section 6) may require padding, and may
   hence produce larger payloads.  RTP provides its own padding format
   (as seen in Fig. 1), which due to the padding indicator in the RTP
   header has merits in terms of compactness relative to paddings using
   prefix-free codes.  This RTP padding SHALL be the default method for
   transforms requiring padding.  Transforms MAY specify other padding
   methods, and MUST then specify the amount, format, and processing of
   their padding.  It is important to note that encryption transforms

   that use padding are vulnerable to subtle attacks, especially when
   message authentication is not used [V02].  Each specification for a
   new encryption transform needs to carefully consider and describe the
   security implications of the padding that it uses.  Message
   authentication codes define their own padding, so this default does
   not apply to authentication transforms.

   The OPTIONAL MKI and the RECOMMENDED authentication tag are the only
   fields defined by SRTP that are not in RTP.  Only 8-bit alignment is

      MKI (Master Key Identifier): configurable length, OPTIONAL.  The
              MKI is defined, signaled, and used by key management.  The
              MKI identifies the master key from which the session
              key(s) were derived that authenticate and/or encrypt the
              particular packet.  Note that the MKI SHALL NOT identify
              the SRTP cryptographic context, which is identified
              according to Section 3.2.3.  The MKI MAY be used by key
              management for the purposes of re-keying, identifying a
              particular master key within the cryptographic context
              (Section 3.2.1).

      Authentication tag: configurable length, RECOMMENDED.  The
              authentication tag is used to carry message authentication
              data.  The Authenticated Portion of an SRTP packet
              consists of the RTP header followed by the Encrypted
              Portion of the SRTP packet.  Thus, if both encryption and
              authentication are applied, encryption SHALL be applied
              before authentication on the sender side and conversely on
              the receiver side.  The authentication tag provides
              authentication of the RTP header and payload, and it
              indirectly provides replay protection by authenticating
              the sequence number.  Note that the MKI is not integrity
              protected as this does not provide any extra protection.

3.2.  SRTP Cryptographic Contexts

   Each SRTP stream requires the sender and receiver to maintain
   cryptographic state information.  This information is called the
   "cryptographic context".

   SRTP uses two types of keys: session keys and master keys.  By a
   "session key", we mean a key which is used directly in a
   cryptographic transform (e.g., encryption or message authentication),
   and by a "master key", we mean a random bit string (given by the key
   management protocol) from which session keys are derived in a

   cryptographically secure way.  The master key(s) and other parameters
   in the cryptographic context are provided by key management
   mechanisms external to SRTP, see Section 8.

3.2.1.  Transform-independent parameters

   Transform-independent parameters are present in the cryptographic
   context independently of the particular encryption or authentication
   transforms that are used.  The transform-independent parameters of
   the cryptographic context for SRTP consist of:

   *  a 32-bit unsigned rollover counter (ROC), which records how many
      times the 16-bit RTP sequence number has been reset to zero after
      passing through 65,535.  Unlike the sequence number (SEQ), which
      SRTP extracts from the RTP packet header, the ROC is maintained by
      SRTP as described in Section 3.3.1.

      We define the index of the SRTP packet corresponding to a given
      ROC and RTP sequence number to be the 48-bit quantity

            i = 2^16 * ROC + SEQ.

   *  for the receiver only, a 16-bit sequence number s_l, which can be
      thought of as the highest received RTP sequence number (see
      Section 3.3.1 for its handling), which SHOULD be authenticated
      since message authentication is RECOMMENDED,

   *  an identifier for the encryption algorithm, i.e., the cipher and
      its mode of operation,

   *  an identifier for the message authentication algorithm,

   *  a replay list, maintained by the receiver only (when
      authentication and replay protection are provided), containing
      indices of recently received and authenticated SRTP packets,

   *  an MKI indicator (0/1) as to whether an MKI is present in SRTP and
      SRTCP packets,

   *  if the MKI indicator is set to one, the length (in octets) of the
      MKI field, and (for the sender) the actual value of the currently
      active MKI (the value of the MKI indicator and length MUST be kept
      fixed for the lifetime of the context),

   *  the master key(s), which MUST be random and kept secret,

   *  for each master key, there is a counter of the number of SRTP
      packets that have been processed (sent) with that master key
      (essential for security, see Sections 3.3.1 and 9),

   *  non-negative integers n_e, and n_a, determining the length of the
      session keys for encryption, and message authentication.

   In addition, for each master key, an SRTP stream MAY use the
   following associated values:

   *  a master salt, to be used in the key derivation of session keys.
      This value, when used, MUST be random, but MAY be public.  Use of
      master salt is strongly RECOMMENDED, see Section 9.2.  A "NULL"
      salt is treated as 00...0.

   *  an integer in the set {1,2,4,...,2^24}, the "key_derivation_rate",
      where an unspecified value is treated as zero.  The constraint to
      be a power of 2 simplifies the session-key derivation
      implementation, see Section 4.3.

   *  an MKI value,

   *  <From, To> values, specifying the lifetime for a master key,
      expressed in terms of the two 48-bit index values inside whose
      range (including the range end-points) the master key is valid.
      For the use of <From, To>, see Section 8.1.1.  <From, To> is an
      alternative to the MKI and assumes that a master key is in one-
      to-one correspondence with the SRTP session key on which the
      <From, To> range is defined.

   SRTCP SHALL by default share the crypto context with SRTP, except:

   *  no rollover counter and s_l-value need to be maintained as the
      RTCP index is explicitly carried in each SRTCP packet,

   *  a separate replay list is maintained (when replay protection is

   *  SRTCP maintains a separate counter for its master key (even if the
      master key is the same as that for SRTP, see below), as a means to
      maintain a count of the number of SRTCP packets that have been
      processed with that key.

   Note in particular that the master key(s) MAY be shared between SRTP
   and the corresponding SRTCP, if the pre-defined transforms (including
   the key derivation) are used but the session key(s) MUST NOT be so

   In addition, there can be cases (see Sections 8 and 9.1) where
   several SRTP streams within a given RTP session, identified by their
   synchronization source (SSRCs, which is part of the RTP header),
   share most of the crypto context parameters (including possibly
   master and session keys).  In such cases, just as in the normal
   SRTP/SRTCP parameter sharing above, separate replay lists and packet
   counters for each stream (SSRC) MUST still be maintained.  Also,
   separate SRTP indices MUST then be maintained.

   A summary of parameters, pre-defined transforms, and default values
   for the above parameters (and other SRTP parameters) can be found in
   Sections 5 and 8.2.

3.2.2.  Transform-dependent parameters

   All encryption, authentication/integrity, and key derivation
   parameters are defined in the transforms section (Section 4).
   Typical examples of such parameters are block size of ciphers,
   session keys, data for the Initialization Vector (IV) formation, etc.
   Future SRTP transform specifications MUST include a section to list
   the additional cryptographic context's parameters for that transform,
   if any.

3.2.3.  Mapping SRTP Packets to Cryptographic Contexts

   Recall that an RTP session for each participant is defined [RFC3550]
   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 session could include an audio RTP session, a video RTP
   session, and a text RTP session.

   A cryptographic context SHALL be uniquely identified by the triplet
   context identifier:

   context id = <SSRC, destination network address, destination
   transport port number>

   where the destination network address and the destination transport
   port are the ones in the SRTP packet.  It is assumed that, when
   presented with this information, the key management returns a context
   with the information as described in Section 3.2.

   As noted above, SRTP and SRTCP by default share the bulk of the
   parameters in the cryptographic context.  Thus, retrieving the crypto
   context parameters for an SRTCP stream in practice may imply a
   binding to the correspondent SRTP crypto context.  It is up to the
   implementation to assure such binding, since the RTCP port may not be

   directly deducible from the RTP port only.  Alternatively, the key
   management may choose to provide separate SRTP- and SRTCP- contexts,
   duplicating the common parameters (such as master key(s)).  The
   latter approach then also enables SRTP and SRTCP to use, e.g.,
   distinct transforms, if so desired.  Similar considerations arise
   when multiple SRTP streams, forming part of one single RTP session,
   share keys and other parameters.

   If no valid context can be found for a packet corresponding to a
   certain context identifier, that packet MUST be discarded.

3.3.  SRTP Packet Processing

   The following applies to SRTP.  SRTCP is described in Section 3.4.

   Assuming initialization of the cryptographic context(s) has taken
   place via key management, the sender SHALL do the following to
   construct an SRTP packet:

   1. Determine which cryptographic context to use as described in
      Section 3.2.3.

   2. Determine the index of the SRTP packet using the rollover counter,
      the highest sequence number in the cryptographic context, and the
      sequence number in the RTP packet, as described in Section 3.3.1.

   3. Determine the master key and master salt.  This is done using the
      index determined in the previous step or the current MKI in the
      cryptographic context, according to Section 8.1.

   4. Determine the session keys and session salt (if they are used by
      the transform) as described in Section 4.3, using master key,
      master salt, key_derivation_rate, and session key-lengths in the
      cryptographic context with the index, determined in Steps 2 and 3.

   5. Encrypt the RTP payload to produce the Encrypted Portion of the
      packet (see Section 4.1, for the defined ciphers).  This step uses
      the encryption algorithm indicated in the cryptographic context,
      the session encryption key and the session salt (if used) found in
      Step 4 together with the index found in Step 2.

   6. If the MKI indicator is set to one, append the MKI to the packet.

   7. For message authentication, compute the authentication tag for the
      Authenticated Portion of the packet, as described in Section 4.2.
      This step uses the current rollover counter, the authentication

      algorithm indicated in the cryptographic context, and the session
      authentication key found in Step 4.  Append the authentication tag
      to the packet.

   8. If necessary, update the ROC as in Section 3.3.1, using the packet
      index determined in Step 2.

   To authenticate and decrypt an SRTP packet, the receiver SHALL do the

   1. Determine which cryptographic context to use as described in
      Section 3.2.3.

   2. Run the algorithm in Section 3.3.1 to get the index of the SRTP
      packet.  The algorithm uses the rollover counter and highest
      sequence number in the cryptographic context with the sequence
      number in the SRTP packet, as described in Section 3.3.1.

   3. Determine the master key and master salt.  If the MKI indicator in
      the context is set to one, use the MKI in the SRTP packet,
      otherwise use the index from the previous step, according to
      Section 8.1.

   4. Determine the session keys, and session salt (if used by the
      transform) as described in Section 4.3, using master key, master
      salt, key_derivation_rate and session key-lengths in the
      cryptographic context with the index, determined in Steps 2 and 3.

   5. For message authentication and replay protection, first check if
      the packet has been replayed (Section 3.3.2), using the Replay
      List and the index as determined in Step 2.  If the packet is
      judged to be replayed, then the packet MUST be discarded, and the
      event SHOULD be logged.

      Next, perform verification of the authentication tag, using the
      rollover counter from Step 2, the authentication algorithm
      indicated in the cryptographic context, and the session
      authentication key from Step 4.  If the result is "AUTHENTICATION
      FAILURE" (see Section 4.2), the packet MUST be discarded from
      further processing and the event SHOULD be logged.

   6. Decrypt the Encrypted Portion of the packet (see Section 4.1, for
      the defined ciphers), using the decryption algorithm indicated in
      the cryptographic context, the session encryption key and salt (if
      used) found in Step 4 with the index from Step 2.

   7. Update the rollover counter and highest sequence number, s_l, in
      the cryptographic context as in Section 3.3.1, using the packet
      index estimated in Step 2.  If replay protection is provided, also
      update the Replay List as described in Section 3.3.2.

   8. When present, remove the MKI and authentication tag fields from
      the packet.

3.3.1.  Packet Index Determination, and ROC, s_l Update

   SRTP implementations use an "implicit" packet index for sequencing,
   i.e., not all of the index is explicitly carried in the SRTP packet.
   For the pre-defined transforms, the index i is used in replay
   protection (Section 3.3.2), encryption (Section 4.1), message
   authentication (Section 4.2), and for the key derivation (Section

   When the session starts, the sender side MUST set the rollover
   counter, ROC, to zero.  Each time the RTP sequence number, SEQ, wraps
   modulo 2^16, the sender side MUST increment ROC by one, modulo 2^32
   (see security aspects below).  The sender's packet index is then
   defined as

      i = 2^16 * ROC + SEQ.

   Receiver-side implementations use the RTP sequence number to
   determine the correct index of a packet, which is the location of the
   packet in the sequence of all SRTP packets.  A robust approach for
   the proper use of a rollover counter requires its handling and use to
   be well defined.  In particular, out-of-order RTP packets with
   sequence numbers close to 2^16 or zero must be properly handled.

   The index estimate is based on the receiver's locally maintained ROC
   and s_l values.  At the setup of the session, the ROC MUST be set to
   zero.  Receivers joining an on-going session MUST be given the
   current ROC value using out-of-band signaling such as key-management
   signaling.  Furthermore, the receiver SHALL initialize s_l to the RTP
   sequence number (SEQ) of the first observed SRTP packet (unless the
   initial value is provided by out of band signaling such as key

   On consecutive SRTP packets, the receiver SHOULD estimate the index
         i = 2^16 * v + SEQ,

   where v is chosen from the set { ROC-1, ROC, ROC+1 } (modulo 2^32)
   such that i is closest (in modulo 2^48 sense) to the value 2^16 * ROC
   + s_l (see Appendix A for pseudocode).

   After the packet has been processed and authenticated (when enabled
   for SRTP packets for the session), the receiver MUST use v to
   conditionally update its s_l and ROC variables as follows.  If
   v=(ROC-1) mod 2^32, then there is no update to s_l or ROC.  If v=ROC,
   then s_l is set to SEQ if and only if SEQ is larger than the current
   s_l; there is no change to ROC.  If v=(ROC+1) mod 2^32, then s_l is
   set to SEQ and ROC is set to v.

   After a re-keying occurs (changing to a new master key), the rollover
   counter always maintains its sequence of values, i.e., it MUST NOT be
   reset to zero.

   As the rollover counter is 32 bits long and the sequence number is 16
   bits long, the maximum number of packets belonging to a given SRTP
   stream that can be secured with the same key is 2^48 using the pre-
   defined transforms.  After that number of SRTP packets have been sent
   with a given (master or session) key, the sender MUST NOT send any
   more packets with that key.  (There exists a similar limit for SRTCP,
   which in practice may be more restrictive, see Section 9.2.)  This
   limitation enforces a security benefit by providing an upper bound on
   the amount of traffic that can pass before cryptographic keys are
   changed.  Re-keying (see Section 8.1) MUST be triggered, before this
   amount of traffic, and MAY be triggered earlier, e.g., for increased
   security and access control to media.  Recurring key derivation by
   means of a non-zero key_derivation_rate (see Section 4.3), also gives
   stronger security but does not change the above absolute maximum

   On the receiver side, there is a caveat to updating s_l and ROC: if
   message authentication is not present, neither the initialization of
   s_l, nor the ROC update can be made completely robust.  The
   receiver's "implicit index" approach works for the pre-defined
   transforms as long as the reorder and loss of the packets are not too
   great and bit-errors do not occur in unfortunate ways.  In
   particular, 2^15 packets would need to be lost, or a packet would
   need to be 2^15 packets out of sequence before synchronization is
   lost.  Such drastic loss or reorder is likely to disrupt the RTP
   application itself.

   The algorithm for the index estimate and ROC update is a matter of
   implementation, and should take into consideration the environment
   (e.g., packet loss rate) and the cases when synchronization is likely
   to be lost, e.g., when the initial sequence number (randomly chosen
   by RTP) is not known in advance (not sent in the key management
   protocol) but may be near to wrap modulo 2^16.

   A more elaborate and more robust scheme than the one given above is
   the handling of RTP's own "rollover counter", see Appendix A.1 of

3.3.2.  Replay Protection

   Secure replay protection is only possible when integrity protection
   is present.  It is RECOMMENDED to use replay protection, both for RTP
   and RTCP, as integrity protection alone cannot assure security
   against replay attacks.

   A packet is "replayed" when it is stored by an adversary, and then
   re-injected into the network.  When message authentication is
   provided, SRTP protects against such attacks through a Replay List.
   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.  Packet indices which lag behind the packet index in the
   context by more than SRTP-WINDOW-SIZE can be assumed to have been
   received, where SRTP-WINDOW-SIZE is a receiver-side, implementation-
   dependent parameter and MUST be at least 64, but which MAY be set to
   a higher value.

   The receiver checks the index of an incoming packet against the
   replay list and the window.  Only packets with index ahead of the
   window, or, inside the window but not already received, SHALL be

   After the packet has been authenticated (if necessary the window is
   first moved ahead), the replay list SHALL be updated with the new

   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 [RFC2401].

3.4.  Secure RTCP

   Secure RTCP follows the definition of Secure RTP.  SRTCP adds three
   mandatory new fields (the SRTCP index, an "encrypt-flag", and the
   authentication tag) and one optional field (the MKI) to the RTCP
   packet definition.  The three mandatory fields MUST be appended to an
   RTCP packet in order to form an equivalent SRTCP packet.  The added
   fields follow any other profile-specific extensions.

   According to Section 6.1 of [RFC3550], there is a REQUIRED packet
   format for compound packets.  SRTCP MUST be given packets according
   to that requirement in the sense that the first part MUST be a sender
   report or a receiver report.  However, the RTCP encryption prefix (a
   random 32-bit quantity) specified in that Section MUST NOT be used
   since, as is stated there, it is only applicable to the encryption
   method specified in [RFC3550] and is not needed by the cryptographic
   mechanisms used in SRTP.

      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 or RR   |             length          | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                         SSRC of sender                        | |
   +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | ~                          sender info                          ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                         report block 1                        ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                         report block 2                        ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                              ...                              ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |V=2|P|    SC   |  PT=SDES=202  |             length            | |
   | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | |                          SSRC/CSRC_1                          | |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                           SDES items                          ~ |
   | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | ~                              ...                              ~ |
   +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | |E|                         SRTCP index                         | |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   | ~                     SRTCP MKI (OPTIONAL)                      ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | :                     authentication tag                        : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   |                                                                   |
   +-- Encrypted Portion                    Authenticated Portion -----+

   Figure 2.  An example of the format of a Secure RTCP packet,
   consisting of an underlying RTCP compound packet with a Sender Report
   and SDES packet.

   The Encrypted Portion of an SRTCP packet consists of the encryption
   (Section 4.1) of the RTCP payload of the equivalent compound RTCP
   packet, from the first RTCP packet, i.e., from the ninth (9) octet to
   the end of the compound packet.  The Authenticated Portion of an
   SRTCP packet consists of the entire equivalent (eventually compound)
   RTCP packet, the E flag, and the SRTCP index (after any encryption
   has been applied to the payload).

   The added fields are:

   E-flag: 1 bit, REQUIRED
            The E-flag indicates if the current SRTCP packet is
            encrypted or unencrypted.  Section 9.1 of [RFC3550] allows
            the split of a compound RTCP packet into two lower-layer
            packets, one to be encrypted and one to be sent in the
            clear.  The E bit set to "1" indicates encrypted packet, and
            "0" indicates non-encrypted packet.

   SRTCP index: 31 bits, REQUIRED
            The SRTCP index is a 31-bit counter for the SRTCP packet.
            The index is explicitly included in each packet, in contrast
            to the "implicit" index approach used for SRTP.  The SRTCP
            index MUST be set to zero before the first SRTCP packet is
            sent, and MUST be incremented by one, modulo 2^31, after
            each SRTCP packet is sent.  In particular, after a re-key,
            the SRTCP index MUST NOT be reset to zero again.

   Authentication Tag: configurable length, REQUIRED
            The authentication tag is used to carry message
            authentication data.

   MKI: configurable length, OPTIONAL
            The MKI is the Master Key Indicator, and functions according
            to the MKI definition in Section 3.

   SRTCP uses the cryptographic context parameters and packet processing
   of SRTP by default, with the following changes:

   *  The receiver does not need to "estimate" the index, as it is
      explicitly signaled in the packet.

   *  Pre-defined SRTCP encryption is as specified in Section 4.1, but
      using the definition of the SRTCP Encrypted Portion given in this
      section, and using the SRTCP index as the index i.  The encryption
      transform and related parameters SHALL by default be the same
      selected for the protection of the associated SRTP stream(s),
      while the NULL algorithm SHALL be applied to the RTCP packets not
      to be encrypted.  SRTCP may have a different encryption transform

      than the one used by the corresponding SRTP.  The expected use for
      this feature is when the former has NULL-encryption and the latter
      has a non NULL-encryption.

   The E-flag is assigned a value by the sender depending on whether the
   packet was encrypted or not.

   *  SRTCP decryption is performed as in Section 4, but only if the E
      flag is equal to 1.  If so, the Encrypted Portion is decrypted,
      using the SRTCP index as the index i.  In case the E-flag is 0,
      the payload is simply left unmodified.

   *  SRTCP replay protection is as defined in Section 3.3.2, but using
      the SRTCP index as the index i and a separate Replay List that is
      specific to SRTCP.

   *  The pre-defined SRTCP authentication tag is specified as in
      Section 4.2, but with the Authenticated Portion of the SRTCP
      packet given in this section (which includes the index).  The
      authentication transform and related parameters (e.g., key size)
      SHALL by default be the same as selected for the protection of the
      associated SRTP stream(s).

   *  In the last step of the processing, only the sender needs to
      update the value of the SRTCP index by incrementing it modulo 2^31
      and for security reasons the sender MUST also check the number of
      SRTCP packets processed, see Section 9.2.

   Message authentication for RTCP is REQUIRED, as it is the control
   protocol (e.g., it has a BYE packet) for RTP.

   Precautions must be taken so that the packet expansion in SRTCP (due
   to the added fields) does not cause SRTCP messages to use more than
   their share of RTCP bandwidth.  To avoid this, the following two
   measures MUST be taken:

   1. When initializing the RTCP variable "avg_rtcp_size" defined in
      chapter 6.3 of [RFC3550], it MUST include the size of the fields
      that will be added by SRTCP (index, E-bit, authentication tag, and
      when present, the MKI).

   2. When updating the "avg_rtcp_size" using the variable "packet_size"
      (section 6.3.3 of [RFC3550]), the value of "packet_size" MUST
      include the size of the additional fields added by SRTCP.

   With these measures in place the SRTCP messages will not use more
   than the allotted bandwidth.  The effect of the size of the added
   fields on the SRTCP traffic will be that messages will be sent with
   longer packet intervals.  The increase in the intervals will be
   directly proportional to size of the added fields.  For the pre-
   defined transforms, the size of the added fields will be at least 14
   octets, and upper bounded depending on MKI and the authentication tag

4.  Pre-Defined Cryptographic Transforms

   While there are numerous encryption and message authentication
   algorithms that can be used in SRTP, below we define default
   algorithms in order to avoid the complexity of specifying the
   encodings for the signaling of algorithm and parameter identifiers.
   The defined algorithms have been chosen as they fulfill the goals
   listed in Section 2.  Recommendations on how to extend SRTP with new
   transforms are given in Section 6.

4.1.  Encryption

   The following parameters are common to both pre-defined, non-NULL,
   encryption transforms specified in this section.

   *  BLOCK_CIPHER-MODE indicates the block cipher used and its mode of
   *  n_b is the bit-size of the block for the block cipher
   *  k_e is the session encryption key
   *  n_e is the bit-length of k_e
   *  k_s is the session salting key
   *  n_s is the bit-length of k_s
   *  SRTP_PREFIX_LENGTH is the octet length of the keystream prefix, a
      non-negative integer, specified by the message authentication code
      in use.

   The distinct session keys and salts for SRTP/SRTCP are by default
   derived as specified in Section 4.3.

   The encryption transforms defined in SRTP map the SRTP packet index
   and secret key into a pseudo-random keystream segment.  Each
   keystream segment encrypts a single RTP packet.  The process of
   encrypting a packet consists of generating the keystream segment
   corresponding to the packet, and then bitwise exclusive-oring that
   keystream segment onto the payload of the RTP packet to produce the
   Encrypted Portion of the SRTP packet.  In case the payload size is
   not an integer multiple of n_b bits, the excess (least significant)
   bits of the keystream are simply discarded.  Decryption is done the
   same way, but swapping the roles of the plaintext and ciphertext.

   +----+   +------------------+---------------------------------+
   | KG |-->| Keystream Prefix |          Keystream Suffix       |---+
   +----+   +------------------+---------------------------------+   |
                               +---------------------------------+   v
                               |     Payload of RTP Packet       |->(*)
                               +---------------------------------+   |
                               +---------------------------------+   |
                               | Encrypted Portion of SRTP Packet|<--+

   Figure 3: Default SRTP Encryption Processing.  Here KG denotes the
   keystream generator, and (*) denotes bitwise exclusive-or.

   The definition of how the keystream is generated, given the index,
   depends on the cipher and its mode of operation.  Below, two such
   keystream generators are defined.  The NULL cipher is also defined,
   to be used when encryption of RTP is not required.

   The SRTP definition of the keystream is illustrated in Figure 3.  The
   initial octets of each keystream segment MAY be reserved for use in a
   message authentication code, in which case the keystream used for
   encryption starts immediately after the last reserved octet.  The
   initial reserved octets are called the "keystream prefix" (not to be
   confused with the "encryption prefix" of [RFC3550, Section 6.1]), and
   the remaining octets are called the "keystream suffix".  The
   keystream prefix MUST NOT be used for encryption.  The process is
   illustrated in Figure 3.

   The number of octets in the keystream prefix is denoted as
   SRTP_PREFIX_LENGTH.  The keystream prefix is indicated by a positive,
   non-zero value of SRTP_PREFIX_LENGTH.  This means that, even if
   confidentiality is not to be provided, the keystream generator output
   may still need to be computed for packet authentication, in which
   case the default keystream generator (mode) SHALL be used.

   The default cipher is the Advanced Encryption Standard (AES) [AES],
   and we define two modes of running AES, (1) Segmented Integer Counter
   Mode AES and (2) AES in f8-mode.  In the remainder of this section,
   let E(k,x) be AES applied to key k and input block x.

4.1.1.  AES in Counter Mode

   Conceptually, counter mode [AES-CTR] consists of encrypting
   successive integers.  The actual definition is somewhat more
   complicated, in order to randomize the starting point of the integer
   sequence.  Each packet is encrypted with a distinct keystream
   segment, which SHALL be computed as follows.

   A keystream segment SHALL be the concatenation of the 128-bit output
   blocks of the AES cipher in the encrypt direction, using key k = k_e,
   in which the block indices are in increasing order.  Symbolically,
   each keystream segment looks like

      E(k, IV) || E(k, IV + 1 mod 2^128) || E(k, IV + 2 mod 2^128) ...

   where the 128-bit integer value IV SHALL be defined by the SSRC, the
   SRTP packet index i, and the SRTP session salting key k_s, as below.

      IV = (k_s * 2^16) XOR (SSRC * 2^64) XOR (i * 2^16)

   Each of the three terms in the XOR-sum above is padded with as many
   leading zeros as needed to make the operation well-defined,
   considered as a 128-bit value.

   The inclusion of the SSRC allows the use of the same key to protect
   distinct SRTP streams within the same RTP session, see the security
   caveats in Section 9.1.

   In the case of SRTCP, the SSRC of the first header of the compound
   packet MUST be used, i SHALL be the 31-bit SRTCP index and k_e, k_s
   SHALL be replaced by the SRTCP encryption session key and salt.

   Note that the initial value, IV, is fixed for each packet and is
   formed by "reserving" 16 zeros in the least significant bits for the
   purpose of the counter.  The number of blocks of keystream generated
   for any fixed value of IV MUST NOT exceed 2^16 to avoid keystream
   re-use, see below.  The AES has a block size of 128 bits, so 2^16
   output blocks are sufficient to generate the 2^23 bits of keystream
   needed to encrypt the largest possible RTP packet (except for IPv6
   "jumbograms" [RFC2675], which are not likely to be used for RTP-based
   multimedia traffic).  This restriction on the maximum bit-size of the
   packet that can be encrypted ensures the security of the encryption
   method by limiting the effectiveness of probabilistic attacks [BDJR].

   For a particular Counter Mode key, each IV value used as an input
   MUST be distinct, in order to avoid the security exposure of a two-
   time pad situation (Section 9.1).  To satisfy this constraint, an
   implementation MUST ensure that the combination of the SRTP packet

   index of ROC || SEQ, and the SSRC used in the construction of the IV
   are distinct for any particular key.  The failure to ensure this
   uniqueness could be catastrophic for Secure RTP.  This is in contrast
   to the situation for RTP itself, which may be able to tolerate such
   failures.  It is RECOMMENDED that, if a dedicated security module is
   present, the RTP sequence numbers and SSRC either be generated or
   checked by that module (i.e., sequence-number and SSRC processing in
   an SRTP system needs to be protected as well as the key).

4.1.2.  AES in f8-mode

   To encrypt UMTS (Universal Mobile Telecommunications System, as 3G
   networks) data, a solution (see [f8-a] [f8-b]) 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 also define here the use of
   AES as a block cipher to be used in what we shall call "f8-mode of
   operation" RTP encryption.  The AES f8-mode SHALL use the same
   default sizes for session key and salt as AES counter mode.

   Figure 4 shows the structure of block cipher, E, running in f8-mode.

                |      |
           +--->|  E   |
           |    +------+
           |        |
     m -> (*)       +-----------+-------------+--  ...     ------+
           |    IV' |           |             |                  |
           |        |   j=1 -> (*)    j=2 -> (*)   ...  j=L-1 ->(*)
           |        |           |             |                  |
           |        |      +-> (*)       +-> (*)   ...      +-> (*)
           |        |      |    |        |    |             |    |
           |        v      |    v        |    v             |    v
           |    +------+   | +------+    | +------+         | +------+
    k_e ---+--->|  E   |   | |  E   |    | |  E   |         | |  E   |
                |      |   | |      |    | |      |         | |      |
                +------+   | +------+    | +------+         | +------+
                    |      |    |        |    |             |    |
                    +------+    +--------+    +--  ...  ----+    |
                    |           |             |                  |
                    v           v             v                  v
                   S(0)        S(1)          S(2)  . . .       S(L-1)

   Figure 4.  f8-mode of operation (asterisk, (*), denotes bitwise XOR).
   The figure represents the KG in Figure 3, when AES-f8 is used.  f8 Keystream Generation

   The Initialization Vector (IV) SHALL be determined as described in
   Section (and in Section for SRTCP).

   Let IV', S(j), and m denote n_b-bit blocks.  The keystream,
   S(0) ||... || S(L-1), for an N-bit message SHALL be defined by
   setting IV' = E(k_e XOR m, IV), and S(-1) = 00..0.  For
   j = 0,1,..,L-1 where L = N/n_b (rounded up to nearest integer if it
   is not already an integer) compute

            S(j) = E(k_e, IV' XOR j XOR S(j-1))

   Notice that the IV is not used directly.  Instead it is fed through E
   under another key to produce an internal, "masked" value (denoted
   IV') to prevent an attacker from gaining known input/output pairs.

   The role of the internal counter, j, is to prevent short keystream
   cycles.  The value of the key mask m SHALL be

           m = k_s || 0x555..5,

   i.e., the session salting key, appended by the binary pattern 0101..
   to fill out the entire desired key size, n_e.

   The sender SHOULD NOT generate more than 2^32 blocks, which is
   sufficient to generate 2^39 bits of keystream.  Unlike counter mode,
   there is no absolute threshold above (below) which f8 is guaranteed
   to be insecure (secure).  The above bound has been chosen to limit,
   with sufficient security margin, the probability of degenerative
   behavior in the f8 keystream generation.  f8 SRTP IV Formation

   The purpose of the following IV formation is to provide a feature
   which we call implicit header authentication (IHA), see Section 9.5.

   The SRTP IV for 128-bit block AES-f8 SHALL be formed in the following

        IV = 0x00 || M || PT || SEQ || TS || SSRC || ROC

   M, PT, SEQ, TS, SSRC SHALL be taken from the RTP header; ROC is from
   the cryptographic context.

   The presence of the SSRC as part of the IV allows AES-f8 to be used
   when a master key is shared between multiple streams within the same
   RTP session, see Section 9.1.  f8 SRTCP IV Formation

   The SRTCP IV for 128-bit block AES-f8 SHALL be formed in the
   following way:

   IV= 0..0 || E || SRTCP index || V || P || RC || PT || length || SSRC

   where V, P, RC, PT, length, SSRC SHALL be taken from the first header
   in the RTCP compound packet.  E and SRTCP index are the 1-bit and
   31-bit fields added to the packet.

4.1.3.  NULL Cipher

   The NULL cipher is used when no confidentiality for RTP/RTCP is
   requested.  The keystream can be thought of as "000..0", i.e., the
   encryption SHALL simply copy the plaintext input into the ciphertext

4.2.  Message Authentication and Integrity

   Throughout this section, M will denote data to be integrity
   protected.  In the case of SRTP, M SHALL consist of the Authenticated
   Portion of the packet (as specified in Figure 1) concatenated with
   the ROC, M = Authenticated Portion || ROC; in the case of SRTCP, M
   SHALL consist of the Authenticated Portion (as specified in Figure 2)

   Common parameters:

   *  AUTH_ALG is the authentication algorithm
   *  k_a is the session message authentication key
   *  n_a is the bit-length of the authentication key
   *  n_tag is the bit-length of the output authentication tag
   *  SRTP_PREFIX_LENGTH is the octet length of the keystream prefix as
      defined above, a parameter of AUTH_ALG

   The distinct session authentication keys for SRTP/SRTCP are by
   default derived as specified in Section 4.3.

   The values of n_a, n_tag, and SRTP_PREFIX_LENGTH MUST be fixed for
   any particular fixed value of the key.

   We describe the process of computing authentication tags as follows.
   The sender computes the tag of M and appends it to the packet.  The
   SRTP receiver verifies a message/authentication tag pair by computing
   a new authentication tag over M using the selected algorithm and key,
   and then compares it to the tag associated with the received message.
   If the two tags are equal, then the message/tag pair is valid;
   otherwise, it is invalid and the error audit message "AUTHENTICATION
   FAILURE" MUST be returned.

4.2.1.  HMAC-SHA1

   The pre-defined authentication transform for SRTP is HMAC-SHA1
   [RFC2104].  With HMAC-SHA1, the SRTP_PREFIX_LENGTH (Figure 3) SHALL
   be 0.  For SRTP (respectively SRTCP), the HMAC SHALL be applied to
   the session authentication key and M as specified above, i.e.,
   HMAC(k_a, M).  The HMAC output SHALL then be truncated to the n_tag
   left-most bits.

4.3.  Key Derivation

4.3.1.  Key Derivation Algorithm

   Regardless of the encryption or message authentication transform that
   is employed (it may be an SRTP pre-defined transform or newly
   introduced according to Section 6), interoperable SRTP
   implementations MUST use the SRTP key derivation to generate session
   keys.  Once the key derivation rate is properly signaled at the start
   of the session, there is no need for extra communication between the
   parties that use SRTP key derivation.

                         packet index ---+
               +-----------+ master  +--------+ session encr_key
               | ext       | key     |        |---------->
               | key mgmt  |-------->|  key   | session auth_key
               | (optional |         | deriv  |---------->
               | rekey)    |-------->|        | session salt_key
               |           | master  |        |---------->
               +-----------+ salt    +--------+

   Figure 5: SRTP key derivation.

   At least one initial key derivation SHALL be performed by SRTP, i.e.,
   the first key derivation is REQUIRED.  Further applications of the
   key derivation MAY be performed, according to the
   "key_derivation_rate" value in the cryptographic context.  The key
   derivation function SHALL initially be invoked before the first
   packet and then, when r > 0, a key derivation is performed whenever
   index mod r equals zero.  This can be thought of as "refreshing" the
   session keys.  The value of "key_derivation_rate" MUST be kept fixed
   for the lifetime of the associated master key.

   Interoperable SRTP implementations MAY also derive session salting
   keys for encryption transforms, as is done in both of the pre-
   defined transforms.

   Let m and n be positive integers.  A pseudo-random function family is
   a set of keyed functions {PRF_n(k,x)} such that for the (secret)
   random key k, given m-bit x, PRF_n(k,x) is an n-bit string,
   computationally indistinguishable from random n-bit strings, see
   [HAC].  For the purpose of key derivation in SRTP, a secure PRF with
   m = 128 (or more) MUST be used, and a default PRF transform is
   defined in Section 4.3.3.

   Let "a DIV t" denote integer division of a by t, rounded down, and
   with the convention that "a DIV 0 = 0" for all a.  We also make the
   convention of treating "a DIV t" as a bit string of the same length
   as a, and thus "a DIV t" will in general have leading zeros.

   Key derivation SHALL be defined as follows in terms of <label>, an
   8-bit constant (see below), master_salt and key_derivation_rate, as
   determined in the cryptographic context, and index, the packet index
   (i.e., the 48-bit ROC || SEQ for SRTP):

   *  Let r = index DIV key_derivation_rate (with DIV as defined above).

   *  Let key_id = <label> || r.

   *  Let x = key_id XOR master_salt, where key_id and master_salt are
      aligned so that their least significant bits agree (right-

   <label> MUST be unique for each type of key to be derived.  We
   currently define <label> 0x00 to 0x05 (see below), and future
   extensions MAY specify new values in the range 0x06 to 0xff for other
   purposes.  The n-bit SRTP key (or salt) for this packet SHALL then be
   derived from the master key, k_master as follows:

      PRF_n(k_master, x).

   (The PRF may internally specify additional formatting and padding of
   x, see e.g., Section 4.3.3 for the default PRF.)

   The session keys and salt SHALL now be derived using:

   - k_e (SRTP encryption): <label> = 0x00, n = n_e.

   - k_a (SRTP message authentication): <label> = 0x01, n = n_a.

   - k_s (SRTP salting key): <label> = 0x02, n = n_s.

   where n_e, n_s, and n_a are from the cryptographic context.

   The master key and master salt MUST be random, but the master salt
   MAY be public.

   Note that for a key_derivation_rate of 0, the application of the key
   derivation SHALL take place exactly once.

   The definition of DIV above is purely for notational convenience.
   For a non-zero t among the set of allowed key derivation rates, "a
   DIV t" can be implemented as a right-shift by the base-2 logarithm of

   t.  The derivation operation is further facilitated if the rates are
   chosen to be powers of 256, but that granularity was considered too
   coarse to be a requirement of this specification.

   The upper limit on the number of packets that can be secured using
   the same master key (see Section 9.2) is independent of the key

4.3.2.  SRTCP Key Derivation

   SRTCP SHALL by default use the same master key (and master salt) as
   SRTP.  To do this securely, the following changes SHALL be done to
   the definitions in Section 4.3.1 when applying session key derivation
   for SRTCP.

   Replace the SRTP index by the 32-bit quantity: 0 || SRTCP index
   (i.e., excluding the E-bit, replacing it with a fixed 0-bit), and use
   <label> = 0x03 for the SRTCP encryption key, <label> = 0x04 for the
   SRTCP authentication key, and, <label> = 0x05 for the SRTCP salting

4.3.3.  AES-CM PRF

   The currently defined PRF, keyed by 128, 192, or 256 bit master key,
   has input block size m = 128 and can produce n-bit outputs for n up
   to 2^23.  PRF_n(k_master,x) SHALL be AES in Counter Mode as described
   in Section 4.1.1, applied to key k_master, and IV equal to (x*2^16),
   and with the output keystream truncated to the n first (left-most)
   bits.  (Requiring n/128, rounded up, applications of AES.)

5.  Default and mandatory-to-implement Transforms

   The default transforms also are mandatory-to-implement transforms in
   SRTP.  Of course, "mandatory-to-implement" does not imply
   "mandatory-to-use".  Table 1 summarizes the pre-defined transforms.
   The default values below are valid for the pre-defined transforms.

                         mandatory-to-impl.   optional     default

   encryption            AES-CM, NULL         AES-f8       AES-CM
   message integrity     HMAC-SHA1              -          HMAC-SHA1
   key derivation (PRF)  AES-CM                 -          AES-CM

   Table 1: Mandatory-to-implement, optional and default transforms in
   SRTP and SRTCP.

5.1.  Encryption: AES-CM and NULL

   AES running in Segmented Integer Counter Mode, as defined in Section
   4.1.1, SHALL be the default encryption algorithm.  The default key
   lengths SHALL be 128-bit for the session encryption key (n_e).  The
   default session salt key-length (n_s) SHALL be 112 bits.

   The NULL cipher SHALL also be mandatory-to-implement.

5.2.  Message Authentication/Integrity: HMAC-SHA1

   HMAC-SHA1, as defined in Section 4.2.1, SHALL be the default message
   authentication code.  The default session authentication key-length
   (n_a) SHALL be 160 bits, the default authentication tag length
   (n_tag) SHALL be 80 bits, and the SRTP_PREFIX_LENGTH SHALL be zero
   for HMAC-SHA1.  In addition, for SRTCP, the pre-defined HMAC-SHA1
   MUST NOT be applied with a value of n_tag, nor n_a, that are smaller
   than these defaults.  For SRTP, smaller values are NOT RECOMMENDED,
   but MAY be used after careful consideration of the issues in Section
   7.5 and 9.5.

5.3.  Key Derivation: AES-CM PRF

   The AES Counter Mode based key derivation and PRF defined in Sections
   4.3.1 to 4.3.3, using a 128-bit master key, SHALL be the default
   method for generating session keys.  The default master salt length
   SHALL be 112 bits and the default key-derivation rate SHALL be zero.

6.  Adding SRTP Transforms

   Section 4 provides examples of the level of detail needed for
   defining transforms.  Whenever a new transform is to be added to
   SRTP, a companion standard track RFC MUST be written to exactly
   define how the new transform can be used with SRTP (and SRTCP).  Such
   a companion RFC SHOULD avoid overlap with the SRTP protocol document.
   Note however, that it MAY be necessary to extend the SRTP or SRTCP
   cryptographic context definition with new parameters (including fixed
   or default values), add steps to the packet processing, or even add
   fields to the SRTP/SRTCP packets.  The companion RFC SHALL explain
   any known issues regarding interactions between the transform and
   other aspects of SRTP.

   Each new transform document SHOULD specify its key attributes, e.g.,
   size of keys (minimum, maximum, recommended), format of keys,
   recommended/required processing of input keying material,
   requirements/recommendations on key lifetime, re-keying and key
   derivation, whether sharing of keys between SRTP and SRTCP is allowed
   or not, etc.

   An added message integrity transform SHOULD define a minimum
   acceptable key/tag size for SRTCP, equivalent in strength to the
   minimum values as defined in Section 5.2.

7.  Rationale

   This section explains the rationale behind several important features
   of SRTP.

7.1.  Key derivation

   Key derivation reduces the burden on the key establishment.  As many
   as six different keys are needed per crypto context (SRTP and SRTCP
   encryption keys and salts, SRTP and SRTCP authentication keys), but
   these are derived from a single master key in a cryptographically
   secure way.  Thus, the key management protocol needs to exchange only
   one master key (plus master salt when required), and then SRTP itself
   derives all the necessary session keys (via the first, mandatory
   application of the key derivation function).

   Multiple applications of the key derivation function are optional,
   but will give security benefits when enabled.  They prevent an
   attacker from obtaining large amounts of ciphertext produced by a
   single fixed session key.  If the attacker was able to collect a
   large amount of ciphertext for a certain session key, he might be
   helped in mounting certain attacks.

   Multiple applications of the key derivation function provide
   backwards and forward security in the sense that a compromised
   session key does not compromise other session keys derived from the
   same master key.  This means that the attacker who is able to recover
   a certain session key, is anyway not able to have access to messages
   secured under previous and later session keys (derived from the same
   master key).  (Note that, of course, a leaked master key reveals all
   the session keys derived from it.)

   Considerations arise with high-rate key refresh, especially in large
   multicast settings, see Section 11.

7.2.  Salting key

   The master salt guarantees security against off-line key-collision
   attacks on the key derivation that might otherwise reduce the
   effective key size [MF00].

   The derived session salting key used in the encryption, has been
   introduced to protect against some attacks on additive stream
   ciphers, see Section 9.2.  The explicit inclusion method of the salt
   in the IV has been selected for ease of hardware implementation.

7.3.  Message Integrity from Universal Hashing

   The particular definition of the keystream given in Section 4.1 (the
   keystream prefix) is to give provision for particular universal hash
   functions, suitable for message authentication in the Wegman-Carter
   paradigm [WC81].  Such functions are provably secure, simple, quick,
   and especially appropriate for Digital Signal Processors and other
   processors with a fast multiply operation.

   No authentication transforms are currently provided in SRTP other
   than HMAC-SHA1.  Future transforms, like the above mentioned
   universal hash functions, MAY be added following the guidelines in
   Section 6.

7.4.  Data Origin Authentication Considerations

   Note that in pair-wise communications, integrity and data origin
   authentication are provided together.  However, in group scenarios
   where the keys are shared between members, the MAC tag only proves
   that a member of the group sent the packet, but does not prevent
   against a member impersonating another.  Data origin authentication
   (DOA) for multicast and group RTP sessions is a hard problem that
   needs a solution; while some promising proposals are being
   investigated [PCST1] [PCST2], more work is needed to rigorously
   specify these technologies.  Thus SRTP data origin authentication in
   groups is for further study.

   DOA can be done otherwise using signatures.  However, this has high
   impact in terms of bandwidth and processing time, therefore we do not
   offer this form of authentication in the pre-defined packet-integrity

   The presence of mixers and translators does not allow data origin
   authentication in case the RTP payload and/or the RTP header are
   manipulated.  Note that these types of middle entities also disrupt
   end-to-end confidentiality (as the IV formation depends e.g., on the
   RTP header preservation).  A certain trust model may choose to trust
   the mixers/translators to decrypt/re-encrypt the media (this would
   imply breaking the end-to-end security, with related security

7.5.  Short and Zero-length Message Authentication

   As shown in Figure 1, the authentication tag is RECOMMENDED in SRTP.
   A full 80-bit authentication-tag SHOULD be used, but a shorter tag or
   even a zero-length tag (i.e., no message authentication) MAY be used
   under certain conditions to support either of the following two
   application environments.

      1. Strong authentication can be impractical in environments where
         bandwidth preservation is imperative.  An important special
         case is wireless communication systems, in which bandwidth is a
         scarce and expensive resource.  Studies have shown that for
         certain applications and link technologies, additional bytes
         may result in a significant decrease in spectrum efficiency
         [SWO].  Considerable effort has been made to design IP header
         compression techniques to improve spectrum efficiency
         [RFC3095].  A typical voice application produces 20 byte
         samples, and the RTP, UDP and IP headers need to be jointly
         compressed to one or two bytes on average in order to obtain
         acceptable wireless bandwidth economy [RFC3095].  In this case,
         strong authentication would impose nearly fifty percent

      2. Authentication is impractical for applications that use data
         links with fixed-width fields that cannot accommodate the
         expansion due to the authentication tag.  This is the case for
         some important existing wireless channels.  For example, zero-
         byte header compression is used to adapt EVRC/SMV voice with
         the legacy IS-95 bearer channel in CDMA2000 VoIP services.  It
         was found that not a single additional octet could be added to
         the data, which motivated the creation of a zero-byte profile
         for ROHC [RFC3242].

   A short tag is secure for a restricted set of applications.  Consider
   a voice telephony application, for example, such as a G.729 audio
   codec with a 20-millisecond packetization interval, protected by a
   32-bit message authentication tag.  The likelihood of any given
   packet being successfully forged is only one in 2^32.  Thus an
   adversary can control no more than 20 milliseconds of audio output
   during a 994-day period, on average.  In contrast, the effect of a
   single forged packet can be much larger if the application is
   stateful.  A codec that uses relative or predictive compression
   across packets will propagate the maliciously generated state,
   affecting a longer duration of output.

   Certainly not all SRTP or telephony applications meet the criteria
   for short or zero-length authentication tags.  Section 9.5.1
   discusses the risks of weak or no message authentication, and section
   9.5 describes the circumstances when it is acceptable and when it is

8.  Key Management Considerations

   There are emerging key management standards [MIKEY] [KEYMGT] [SDMS]
   for establishing an SRTP cryptographic context (e.g., an SRTP master
   key).  Both proprietary and open-standard key management methods are
   likely to be used for telephony applications [MIKEY] [KINK] and
   multicast applications [GDOI].  This section provides guidance for
   key management systems that service SRTP session.

   For initialization, an interoperable SRTP implementation SHOULD be
   given the SSRC and MAY be given the initial RTP sequence number for
   the RTP stream by key management (thus, key management has a
   dependency on RTP operational parameters).  Sending the RTP sequence
   number in the key management may be useful e.g., when the initial
   sequence number is close to wrapping (to avoid synchronization
   problems), and to communicate the current sequence number to a
   joining endpoint (to properly initialize its replay list).

   If the pre-defined transforms are used, SRTP allows sharing of the
   same master key between SRTP/SRTCP streams belonging to the same RTP

   First, sharing between SRTP streams belonging to the same RTP session
   is secure if the design of the synchronization mechanism, i.e., the
   IV, avoids keystream re-use (the two-time pad, Section 9.1).  This is
   taken care of by the fact that RTP provides for unique SSRCs for
   streams belonging to the same RTP session.  See Section 9.1 for
   further discussion.

   Second, sharing between SRTP and the corresponding SRTCP is secure.
   The fact that an SRTP stream and its associated SRTCP stream both
   carry the same SSRC does not constitute a problem for the two-time
   pad due to the key derivation.  Thus, SRTP and SRTCP corresponding to
   one RTP session MAY share master keys (as they do by default).

   Note that message authentication also has a dependency on SSRC
   uniqueness that is unrelated to the problem of keystream reuse: SRTP
   streams authenticated under the same key MUST have a distinct SSRC in
   order to identify the sender of the message.  This requirement is
   needed because the SSRC is the cryptographically authenticated field

   used to distinguish between different SRTP streams.  Were two streams
   to use identical SSRC values, then an adversary could substitute
   messages from one stream into the other without detection.

   SRTP/SRTCP MUST NOT share master keys under any other circumstances
   than the ones given above, i.e., between SRTP and its corresponding
   SRTCP, and, between streams belonging to the same RTP session.

8.1.  Re-keying

   The recommended way for a particular key management system to provide
   re-key within SRTP is by associating a master key in a crypto context
   with an MKI.

   This provides for easy master key retrieval (see Scenarios in Section
   11), but has the disadvantage of adding extra bits to each packet.
   As noted in Section 7.5, some wireless links do not cater for added
   bits, therefore SRTP also defines a more economic way of triggering
   re-keying, via use of <From, To>, which works in some specific,
   simple scenarios (see Section 8.1.1).

   SRTP senders SHALL count the amount of SRTP and SRTCP traffic being
   used for a master key and invoke key management to re-key if needed
   (Section 9.2).  These interactions are defined by the key management
   interface to SRTP and are not defined by this protocol specification.

8.1.1.  Use of the <From, To> for re-keying

   In addition to the use of the MKI, SRTP defines another optional
   mechanism for master key retrieval, the <From, To>.  The <From, To>
   specifies the range of SRTP indices (a pair of sequence number and
   ROC) within which a certain master key is valid, and is (when used)
   part of the crypto context.  By looking at the 48-bit SRTP index of
   the current SRTP packet, the corresponding master key can be found by
   determining which From-To interval it belongs to.  For SRTCP, the
   most recently observed/used SRTP index (which can be obtained from
   the cryptographic context) is used for this purpose, even though
   SRTCP has its own (31-bit) index (see caveat below).

   This method, compared to the MKI, has the advantage of identifying
   the master key and defining its lifetime without adding extra bits to
   each packet.  This could be useful, as already noted, for some
   wireless links that do not cater for added bits.  However, its use
   SHOULD be limited to specific, very simple scenarios.  We recommend
   to limit its use when the RTP session is a simple unidirectional or
   bi-directional stream.  This is because in case of multiple streams,
   it is difficult to trigger the re-key based on the <From, To> of a
   single RTP stream. For example, if several streams share a master

   key, there is no simple one-to-one correspondence between the index
   sequence space of a certain stream, and the index sequence space on
   which the <From, To> values are based.  Consequently, when a master
   key is shared between streams, one of these streams MUST be
   designated by key management as the one whose index space defines the
   re-keying points.  Also, the re-key triggering on SRTCP is based on
   the correspondent SRTP stream, i.e., when the SRTP stream changes the
   master key, so does the correspondent SRTCP.  This becomes obviously
   more and more complex with multiple streams.

   The default values for the <From, To> are "from the first observed
   packet" and "until further notice".  However, the maximum limit of
   SRTP/SRTCP packets that are sent under each given master/session key
   (Section 9.2) MUST NOT be exceeded.

   In case the <From, To> is used as key retrieval, then the MKI is not
   inserted in the packet (and its indicator in the crypto context is
   zero).  However, using the MKI does not exclude using <From, To> key
   lifetime simultaneously.  This can for instance be useful to signal
   at the sender side at which point in time an MKI is to be made

8.2.  Key Management parameters

   The table below lists all SRTP parameters that key management can
   supply.  For reference, it also provides a summary of the default and
   mandatory-to-support values for an SRTP implementation as described
   in Section 5.

   Parameter                     Mandatory-to-support    Default
   ---------                     --------------------    -------

   SRTP and SRTCP encr transf.       AES_CM, NULL         AES_CM
   (Other possible values: AES_f8)

   SRTP and SRTCP auth transf.       HMAC-SHA1           HMAC-SHA1

   SRTP and SRTCP auth params:
     n_tag (tag length)                 80                 80
     SRTP prefix_length                  0                  0

   Key derivation PRF                 AES_CM              AES_CM

   Key material params
   (for each master key):
     master key length                 128                128
     n_e (encr session key length)     128                128
     n_a (auth session key length)     160                160
     master salt key
     length of the master salt         112                112
     n_s (session salt key length)     112                112
     key derivation rate                 0                  0

     key lifetime
        SRTP-packets-max-lifetime      2^48               2^48
        SRTCP-packets-max-lifetime     2^31               2^31
        from-to-lifetime <From, To>
     MKI indicator                       0                 0
     length of the MKI                   0                 0
     value of the MKI

   Crypto context index params:
     SSRC value
     SRTCP Index
     Transport address
     Port number

   Relation to other RTP profiles:
     sender's order between FEC and SRTP FEC-SRTP      FEC-SRTP
     (see Section 10)

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
   NULL) authentication.

   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
   Section 7.5).

   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
   individual bits.

   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.

11.  Scenarios

   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.

11.1.  Unicast

   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
   master key.

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.

13.  Acknowledgements

   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.  References

14.1.  Normative References

   [AES]     NIST, "Advanced Encryption Standard (AES)", FIPS PUB 197,

   [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,
             May 2000.

   [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/

   [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-
             403, 1997.

   [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
             FSE 2004.

   [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.
             255-271, 2001.

   [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,
             265-279, 1981.

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


   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

   j = 0
   IV' xor j           :   595b699bbd3bc0df26062093c1ad8f73
   S(-1)               :   00000000000000000000000000000000
   IV' xor S(-1) xor j :   595b699bbd3bc0df26062093c1ad8f73
   S(0)                :   71ef82d70a172660240709c7fbb19d8e
   plaintext           :   70736575646f72616e646f6d6e657373
   ciphertext          :   019ce7a26e7854014a6366aa95d4eefd

   j = 1
   IV' xor j           :   595b699bbd3bc0df26062093c1ad8f72
   S(0)                :   71ef82d70a172660240709c7fbb19d8e
   IV' xor S(0) xor j  :   28b4eb4cb72ce6bf020129543a1c12fc
   S(1)                :   3abd640a60919fd43bd289a09649b5fc
   plaintext           :   20697320746865206e65787420626573
   ciphertext          :   1ad4172a14f9faf455b7f1d4b62bd08f

   j = 2
   IV' xor j           :   595b699bbd3bc0df26062093c1ad8f71
   S(1)                :   3abd640a60919fd43bd289a09649b5fc
   IV' xor S(1) xor j  :   63e60d91ddaa5f0b1dd4a93357e43a8d
   S(2)                :   220c7a8715266565b09ecc8a2a62b11b
   plaintext           :   74207468696e67
   ciphertext          :   562c0eef7c4802

B.2.  AES-CM Test Vectors

    Keystream segment length: 1044512 octets (65282 AES blocks)
    Session Key:      2B7E151628AED2A6ABF7158809CF4F3C
    Rollover Counter: 00000000
    Sequence Number:  0000
    SSRC:             00000000
    Session Salt:     F0F1F2F3F4F5F6F7F8F9FAFBFCFD0000 (already shifted)
    Offset:           F0F1F2F3F4F5F6F7F8F9FAFBFCFD0000

    Counter                            Keystream

    F0F1F2F3F4F5F6F7F8F9FAFBFCFD0000   E03EAD0935C95E80E166B16DD92B4EB4
    F0F1F2F3F4F5F6F7F8F9FAFBFCFD0001   D23513162B02D0F72A43A2FE4A5F97AB
    F0F1F2F3F4F5F6F7F8F9FAFBFCFD0002   41E95B3BB0A2E8DD477901E4FCA894C0
    ...                                ...
    F0F1F2F3F4F5F6F7F8F9FAFBFCFDFF00   362B7C3C6773516318A077D7FC5073AE
    F0F1F2F3F4F5F6F7F8F9FAFBFCFDFF01   6A2CC3787889374FBEB4C81B17BA6C44

   Nota Bene: this test case is contrived so that the latter part of the
   keystream segment coincides with the test case in Section F.5.1 of

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
      label:                       00
      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
      label:                       02
      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
      label:                         01
      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
   CEBE321F6FF7716B6FD4AB49AF256A15   0EC675AD498AFEEAB6960B3AABE60000
   6D38BAA48F0A0ACF3C34E2359E6CDBCE   0EC675AD498AFEEAB6960B3AABE60001
   E049646C43D9327AD175578EF7227098   0EC675AD498AFEEAB6960B3AABE60002
   6371C10C9A369AC2F94A8C5FBCDDDC25   0EC675AD498AFEEAB6960B3AABE60003
   6D6E919A48B610EF17C2041E47403576   0EC675AD498AFEEAB6960B3AABE60004
   6B68642C59BBFC2F34DB60DBDFB2       0EC675AD498AFEEAB6960B3AABE60005

Authors' Addresses

   Questions and comments should be directed to the authors and

   Mark Baugher
   Cisco Systems, Inc.
   5510 SW Orchid Street
   Portland, OR 97219 USA

   Phone:  +1 408-853-4418
   EMail:  mbaugher@cisco.com

   Elisabetta Carrara
   Ericsson Research
   SE-16480 Stockholm

   Phone:  +46 8 50877040
   EMail:  elisabetta.carrara@ericsson.com

   David A. McGrew
   Cisco Systems, Inc.
   San Jose, CA 95134-1706

   Phone:  +1 301-349-5815
   EMail:  mcgrew@cisco.com

   Mats Naslund
   Ericsson Research
   SE-16480 Stockholm

   Phone:  +46 8 58533739
   EMail:  mats.naslund@ericsson.com

   Karl Norrman
   Ericsson Research
   SE-16480 Stockholm

   Phone:  +46 8 4044502
   EMail:  karl.norrman@ericsson.com

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