Network Working Group                                          D. McGrew
Internet-Draft                                             Cisco Systems
Intended status:  Standards Track                            E. Rescorla
Expires:  August 28, 2008                              Network Resonance
                                                       February 25, 2008

Datagram Transport Layer Security (DTLS) Extension to Establish Keys for
               Secure Real-time Transport Protocol (SRTP)

Status of this Memo

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   have been or will be disclosed, and any of which he or she becomes
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Copyright Notice

   Copyright (C) The IETF Trust (2008).


   This document describes a Datagram Transport Layer Security (DTLS)
   extension to establish keys for secure RTP (SRTP) and secure RTP
   Control Protocol (SRTCP) flows.  DTLS keying happens on the media
   path, independent of any out-of-band signalling channel present.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Conventions Used In This Document  . . . . . . . . . . . . . .  3
   3.  Overview of DTLS-SRTP Operation  . . . . . . . . . . . . . . .  4
   4.  DTLS Extensions for SRTP Key Establishment . . . . . . . . . .  5
     4.1.  The use_srtp Extension . . . . . . . . . . . . . . . . . .  5
       4.1.1.  use_srtp Extension Definition  . . . . . . . . . . . .  6
       4.1.2.  SRTP Protection Profiles . . . . . . . . . . . . . . .  7
       4.1.3.  srtp_mki value . . . . . . . . . . . . . . . . . . . .  9
     4.2.  Key Derivation . . . . . . . . . . . . . . . . . . . . . . 10
     4.3.  Key Scope  . . . . . . . . . . . . . . . . . . . . . . . . 12
     4.4.  Key Usage Limitations  . . . . . . . . . . . . . . . . . . 12
   5.  Use of RTP and RTCP over a DTLS-SRTP Channel . . . . . . . . . 12
     5.1.  Data Protection  . . . . . . . . . . . . . . . . . . . . . 12
       5.1.1.  Transmission . . . . . . . . . . . . . . . . . . . . . 12
       5.1.2.  Reception  . . . . . . . . . . . . . . . . . . . . . . 13
     5.2.  Rehandshake and Re-key . . . . . . . . . . . . . . . . . . 15
   6.  Multi-party RTP Sessions . . . . . . . . . . . . . . . . . . . 16
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 16
     7.1.  Security of Negotiation  . . . . . . . . . . . . . . . . . 16
     7.2.  Framing Confusion  . . . . . . . . . . . . . . . . . . . . 17
     7.3.  Sequence Number Interactions . . . . . . . . . . . . . . . 17
       7.3.1.  Alerts . . . . . . . . . . . . . . . . . . . . . . . . 17
       7.3.2.  Renegotiation  . . . . . . . . . . . . . . . . . . . . 17
     7.4.  Decryption Cost  . . . . . . . . . . . . . . . . . . . . . 18
   8.  Session Description for RTP/SAVP over DTLS . . . . . . . . . . 18
   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 19
   10. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 19
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 19
     11.2. Informational References . . . . . . . . . . . . . . . . . 20
   Appendix A.  Performance of Multiple DTLS Handshakes . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23
   Intellectual Property and Copyright Statements . . . . . . . . . . 24

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1.  Introduction

   The Secure RTP profile (SRTP) [6] can provide confidentiality,
   message authentication, and replay protection to RTP data and RTP
   Control (RTCP) traffic.  SRTP does not provide key management
   functionality, but instead depends on external key management to
   exchange secret master keys, and to negotiate the algorithms and
   parameters for use with those keys.

   Datagram Transport Layer Security (DTLS) [6] is a channel security
   protocol that offers integrated key management, parameter
   negotiation, and secure data transfer.  Because DTLS's data transfer
   protocol is generic, it is less highly optimized for use with RTP
   than is SRTP, which has been specifically tuned for that purpose.

   This document describes DTLS-SRTP, an SRTP extension for DTLS which
   combine the performance and encryption flexibility benefits of SRTP
   with the flexibility and convenience of DTLS's integrated key and
   association management.  DTLS-SRTP can be viewed in two equivalent
   ways:  as a new key management method for SRTP, and a new RTP-
   specific data format for DTLS.

   The key points of DTLS-SRTP are that:
   o  application data is protected using SRTP,
   o  the DTLS handshake is used to establish keying material,
      algorithms, and parameters for SRTP,
   o  a DTLS extension used to negotiate SRTP algorithms, and
   o  other DTLS record layer content types are protected using the
      ordinary DTLS record format.

   The remainder of this memo is structured as follows.  Section 2
   describes conventions used to indicate normative requirements.
   Section 3 provides an overview of DTLS-SRTP operation.  Section 4
   specifies the DTLS extensions, while Section 5 discusses how RTP and
   RTCP are transported over a DTLS-SRTP channel.  Section 6 describes
   use with multi-party sessions.  Section 7 and Section 9 describe
   Security and IANA considerations.

2.  Conventions Used In This Document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in [1].

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3.  Overview of DTLS-SRTP Operation

   DTLS-SRTP is defined for point-to-point media sessions, in which
   there are exactly two participants.  Each DTLS-SRTP session contains
   a single DTLS association (called a "connection" in TLS jargon), and
   an SRTP context.  A single DTLS-SRTP session only protects data
   carried over a single UDP source and destination port pair.

   The general pattern of DTLS-SRTP is as follows.  For each RTP or RTCP
   flow the peers do a DTLS handshake on the same source and destination
   port pair to establish a DTLS association.  The keying material from
   that handshake is fed into the SRTP stack.  Once that association is
   established, RTP packets are protected (becoming SRTP) using that
   keying material.

   RTP and RTCP traffic is usually sent on two separate UDP ports.  When
   symmetric RTP [10] is used, two bidirectional DTLS-SRTP sessions are
   needed, one for the RTP port, one for the RTCP port.  When RTP flows
   are not symmetric, four unidirectional DTLS-SRTP sessions are needed
   (for inbound and outbound RTP, and inbound and outbound RTCP).

   Symmetric RTP [10] is the case in which there are two RTP sessions
   that have their source and destination ports and addresses reversed,
   in a manner similar to the way that a TCP connection uses its ports.
   Each participant has an inbound RTP session and an outbound RTP
   session.  When symmetric RTP is used, a single DTLS-SRTP session can
   protect both of the RTP sessions.

   RTP and RTCP traffic MAY be multiplexed on a single UDP port [7].  In
   this case, both RTP and RTCP packets may be sent over the same DTLS-
   SRTP session, halving the number of DTLS-SRTP sessions needed.  It is
   RECOMMENDED that symmetric RTP is used, with RTP and RTCP multiplexed
   on a single UDP port; this requires only a single DTLS-SRTP session.

   Between a single pair of participants, there may be multiple media
   sessions.  There MUST be a separate DTLS-SRTP session for each
   distinct pair of source and destination ports used by a media session
   (though the sessions can share a single DTLS session and hence
   amortize the initial public key handshake!).

   A DTLS-SRTP session MAY be indicated by an external signaling
   protocol like SIP.  When the signaling exchange is integrity-
   protected (e.g when SIP Identity protection via digital signatures is
   used), DTLS-SRTP can leverage this integrity guarantee to provide
   complete security of the media stream.  A description of how to
   indicate DTLS-SRTP sessions in SIP and SDP, and how to authenticate
   the endpoints using fingerprints can be found in [13].

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   In a naive implementation, when there are multiple media sessions,
   there is a new DTLS session establishment (complete with public key
   cryptography) for each media channel.  For example, a videophone may
   be sending both an audio stream and a video stream, each of which
   would use a separate DTLS session establishment exchange, which would
   proceed in parallel.  As an optimization, the DTLS-SRTP
   implementation SHOULD use the following strategy:  a single DTLS
   association is established, and all other DTLS associations wait
   until that connection is established before proceeding with their
   handshakes establishment exchanges.  This strategy allows the later
   sessions to use DTLS session resumption, which allows the
   amortization of the expensive public key cryptography operations over
   multiple DTLS handshakes.

   The SRTP keys used to protect packets originated by the client are
   distinct from the SRTP keys used to protect packets originated by the
   server.  All of the RTP sources originating on the client use the
   same SRTP keys, and similarly, all of the RTP sources originating on
   the server over the same channel use the same SRTP keys.  The SRTP
   implementation MUST ensure that all of the SSRC values for all of the
   RTP sources originating from the same device over the same channel
   are distinct, in order to avoid the "two-time pad" problem (as
   described in Section 9.1 of RFC 3711).

4.  DTLS Extensions for SRTP Key Establishment

4.1.  The use_srtp Extension

   In order to negotiate the use of SRTP data protection, clients
   include an extension of type "use_srtp" in the DTLS extended client
   hello.  This extension MUST only be used when the data being
   transported is RTP and RTCP [4].  The "extension_data" field of this
   extension contains the list of acceptable SRTP protection profiles,
   as indicated below.

   Servers that receive an extended hello containing a "use_srtp"
   extension can agree to use SRTP by including an extension of type
   "use_srtp", with the chosen protection profile in the extended server
   hello.  This process is shown below.

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         Client                                               Server

         ClientHello + use_srtp       -------->
                                              ServerHello + use_srtp
                                      <--------      ServerHelloDone
         Finished                     -------->
                                      <--------             Finished
         SRTP packets                 <------->      SRTP packets

   Note that '*' indicates messages which are not always sent in DTLS.
   The CertificateRequest, client Certificate, and CertificateVerify
   will be sent in DTLS-SRTP.

   Once the "use_srtp" extension is negotiated, the RTP or RTCP
   application data is protected solely using SRTP.  Application data is
   never sent in DTLS record-layer "application_data" packets.  Rather,
   complete RTP or RTCP packets are passed to the DTLS stack which
   passes them to the SRTP stack which protects them appropriately.
   Note that if RTP/RTCP multiplexing [12] is in use, this means that
   RTP and RTCP packets may both be passed to the DTLS stack.  Because
   the DTLS layer does not process the packets, it does need to
   distinguish them.  The SRTP stack can use the procedures of [12] to
   distinguish RTP from RTCP.

   When the "use_srtp" extension is in effect, implementations MUST NOT
   place more than one application data "record" per datagram.  (This is
   only meaningful from the perspective of DTLS because SRTP is
   inherently oriented towards one payload per packet, but is stated
   purely for clarification.)

   Records of type other than "application_data" MUST use ordinary DTLS

4.1.1.  use_srtp Extension Definition

   The client MUST fill the extension_data field of the "use_srtp"
   extension with an UseSRTPData value (see Section 9 for the

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      uint8 SRTPProtectionProfile[2];

      struct {
         SRTPProtectionProfiles SRTPProtectionProfiles;
         uint8 srtp_mki<0..255>;
      } UseSRTPData;

      SRTPProtectionProfile SRTPProtectionProfiles<2^16-1>;

   The SRTPProtectionProfiles list indicates the SRTP protection
   profiles that the client is willing to support, listed in descending
   order of preference.  The srtp_mki value contains the SRTP
   MasterKeyIdentifier (MKI) value (if any) which the client will use
   for his SRTP messages.  If this field is of zero length, then no MKI
   will be used.

   If the server is willing to accept the use_srtp extension, it MUST
   respond with its own "use_srtp" extension in the ExtendedServerHello.
   The extension_data field MUST contain a UseSRTPData value with a
   single SRTPProtectionProfile value which the server has chosen for
   use with this connection.  The server MUST NOT select a value which
   the client has not offered.  If there is no shared profile, the
   server SHOULD not return the use_srtp extension at which point the
   connection falls back to the negotiated DTLS cipher suite.  If that
   is not acceptable the server SHOULD return an appropriate DTLS alert.

4.1.2.  SRTP Protection Profiles

   A DTLS-SRTP SRTP Protection Profile defines the parameters and
   options that are in effect for the SRTP processing.  This document
   defines the following SRTP protection profiles.

      SRTPProtectionProfile SRTP_AES128_CM_SHA1_80 = {0x00, 0x01};
      SRTPProtectionProfile SRTP_AES128_CM_SHA1_32 = {0x00, 0x02};
      SRTPProtectionProfile SRTP_AES256_CM_SHA1_80 = {0x00, 0x03};
      SRTPProtectionProfile SRTP_AES256_CM_SHA1_32 = {0x00, 0x04};
      SRTPProtectionProfile SRTP_NULL_SHA1_80      = {0x00, 0x05};
      SRTPProtectionProfile SRTP_NULL_SHA1_32      = {0x00, 0x06};

   The following list indicates the SRTP transform parameters for each
   protection profile.  The parameters cipher_key_length,
   cipher_salt_length, auth_key_length, and auth_tag_length express the
   number of bits in the values to which they refer.  The
   maximum_lifetime parameter indicates the maximum number of packets
   that can be protected with each single set of keys when the parameter
   profile is in use.  All of these parameters apply to both RTP and
   RTCP, unless the RTCP parameters are separately specified.

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   All of the crypto algorithms in these profiles are from [8], except
   for the AES256_CM cipher, which is specified in [15].

         cipher:  AES_128_CM
         cipher_key_length:  128
         cipher_salt_length:  112
         maximum_lifetime:  2^31
         auth_function:  HMAC-SHA1
         auth_key_length:  160
         auth_tag_length:  80
         cipher:  AES_128_CM
         cipher_key_length:  128
         cipher_salt_length:  112
         maximum_lifetime:  2^31
         auth_function:  HMAC-SHA1
         auth_key_length:  160
         auth_tag_length:  32
         RTCP auth_tag_length:  80
         cipher:  AES_256_CM
         cipher_key_length:  256
         cipher_salt_length:  112
         maximum_lifetime:  2^31
         auth_function:  HMAC-SHA1
         auth_key_length:  160
         auth_tag_length:  80
         cipher:  AES_256_CM
         cipher_key_length:  256
         cipher_salt_length:  112
         maximum_lifetime:  2^31
         auth_function:  HMAC-SHA1
         auth_key_length:  160
         auth_tag_length:  32
         RTCP auth_tag_length:  80
         cipher:  NULL
         cipher_key_length:  0
         cipher_salt_length:  0
         maximum_lifetime:  2^31
         auth_function:  HMAC-SHA1
         auth_key_length:  160

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         auth_tag_length:  80
         cipher:  NULL
         cipher_key_length:  0
         cipher_salt_length:  0
         maximum_lifetime:  2^31
         auth_function:  HMAC-SHA1
         auth_key_length:  160
         auth_tag_length:  32
         RTCP auth_tag_length:  80

   With all of these SRTP Parameter profiles, the following SRTP options
   are in effect:

   o  The TLS Key Derivation Function (KDF) is used to generate keys to
      feed into the SRTP KDF.
   o  The Key Derivation Rate (KDR) is equal to zero.  Thus, keys are
      not re-derived based on the SRTP sequence number.
   o  For all other parameters, the default values are used.

   All SRTP parameters that are not determined by the SRTP Protection
   Profile MAY be established via the signaling system.  In particular,
   the relative order of Forward Error Correction and SRTP processing,
   and a suggested SRTP replay window size SHOULD be established in this
   manner.  An example of how these parameters can be defined for SDP by
   is contained in [9].  If they are not otherwise signalled, they take
   on their default values from [8].

   Applications using DTLS-SRTP SHOULD coordinate the SRTP Protection
   Profiles between the DTLS-SRTP session that protects an RTP flow and
   the DTLS-SRTP session that protects the associated RTCP flow (in
   those case in which the RTP and RTCP are not multiplexed over a
   common port).  In particular, identical ciphers SHOULD be used.

   New SRTPProtectionProfile values must be defined by RFC 2434
   Standards Action.  See Section 9 for IANA Considerations.

4.1.3.  srtp_mki value

   The srtp_mki value MAY be used to indicate the capability and desire
   to use the SRTP Master Key Indicator (MKI) field in the SRTP and
   SRTCP packets.  The MKI field indicates to an SRTP receiver which key
   was used to protect the packet that contains that field.  The
   srtp_mki field contains the value of the SRTP MKI which is associated
   with the SRTP master keys derived from this handshake.  Each SRTP
   session MUST have exactly one master key that is used to protect
   packets at any given time.  The client MUST choose the MKI value so
   that it is distinct from the last MKI value that was used, and it

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   SHOULD make these values unique.

   Upon receipt of a "use_srtp" extension containing a "srtp_mki" field,
   the server MUST either (assuming it accepts the extension at all):

   1.  include a matching "srtp_mki" value in its "use_srtp" extension
       to indicate that it will make use of the MKI, or
   2.  return an empty "srtp_mki" value to indicate that it cannot make
       use of the MKI.

   If the client detects a nonzero-length MKI in the server's response
   that is different than the one the client offered MUST abort the
   handshake and SHOULD send an invalid_parameter alert.  If the client
   and server agree on an MKI, all SRTP packets protected under the new
   security parameters MUST contain that MKI.

4.2.  Key Derivation

   When SRTP mode is in effect, different keys are used for ordinary
   DTLS record protection and SRTP packet protection.  These keys are
   generated using a TLS extractor [11] to generate 2 *
   (SRTPSecurityParams.master_key_len +
   SRTPSecurityParams.master_salt_len) bytes of data, which are assigned
   as shown below.


   The extractor label for this usage is "EXTRACTOR-dtls_srtp".

   The four keying material values are provided as inputs to the SRTP
   key derivation mechanism, as shown in Figure 5 and detailed below.
   By default, the mechanism defined in Section 4.3 of [8] is used,
   unless another key derivation mechanism is specified as part of an
   SRTP Protection Profile.

   The client_write_SRTP_master_key and client_write_SRTP_master_salt
   are provided to one invocation of the SRTP key derivation function,
   to generate the SRTP keys used to encrypt and authenticate packets
   sent by the client.  The server MUST only use these keys to decrypt
   and to check the authenticity of inbound packets.

   The server_write_SRTP_master_key and server_write_SRTP_master_salt
   are provided to one invocation of the SRTP key derivation function,
   to generate the SRTP keys used to encrypt and authenticate packets
   sent by the server.  The client MUST only use these keys to decrypt

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   and to check the authenticity of inbound packets.

       +------- TLS master secret
       v    +-> client_write_MAC_secret
   +-----+  |
   | TLS |--+-> server_write_MAC_secret
   | KDF |  |
   +-----+  +-> client_write_key
            +-> server_write_key
            |                                       +------+   SRTP
            +-> client_write_SRTP_master_key  ----->| SRTP |-> client
            |                                  +--->|  KDF |   write
            +-> server_write_SRTP_master_key --|-+  +------+   keys
            |                                  | |
            +-> client_write_SRTP_master_salt -+ |  +------+   SRTP
            |                                    +->| SRTP |-> server
            +-> server_write_SRTP_master_salt ----->|  KDF |   write
                                                    +------+   keys

                Figure 5: The derivation of the SRTP keys.

   When both RTCP and RTP use the same source and destination ports,
   then both the SRTP and SRTCP keys are need.  Otherwise, there are two
   DTLS-SRTP sessions, one of which protects the RTP packets and one of
   which protects the RTCP packets; each DTLS-SRTP session protects the
   part of an SRTP session that passes over a single source/destination
   transport address pair, as shown in Figure 6.  When a DTLS-SRTP
   session is protecting RTP, the SRTCP keys derived from the DTLS
   handshake are not needed and are discarded.  When a DTLS-SRTP session
   is protecting RTCP, the SRTP keys derived from the DTLS handshake are
   not needed and are discarded.

      Client            Server
     (Sender)         (Receiver)
   (1)   <----- DTLS ------>    src/dst = a/b and b/a
         ------ SRTP ------>    src/dst = a/b, uses client write keys

   (2)   <----- DTLS ------>    src/dst = c/d and d/c
         ------ SRTCP ----->    src/dst = c/d, uses client write keys
         <----- SRTCP ------    src/dst = d/c, uses server write keys

     Figure 6: A DTLS-SRTP session protecting RTP (1) and another one
    protecting RTCP (2), showing the transport addresses and keys used.

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4.3.  Key Scope

   Because of the possibility of packet reordering, DTLS-SRTP
   implementations SHOULD store multiple SRTP keys sets during a re-key
   in order to avoid the need for receivers to drop packets for which
   they lack a key.

4.4.  Key Usage Limitations

   The maximum_lifetime parameter in the SRTP protection profile
   indicates the maximum number of packets that can be protected with
   each single encryption and authentication key.  (Note that, since RTP
   and RTCP are protected with independent keys, those protocols are
   counted separately for the purposes of determining when a key has
   reached the end of its lifetime.)  Each profile defines its own
   limit.  When this limit is reached, a new DTLS session SHOULD be used
   to establish replacement keys, and SRTP implementations MUST NOT use
   the existing keys for the processing of either outbound or inbound

5.  Use of RTP and RTCP over a DTLS-SRTP Channel

5.1.  Data Protection

   Once the DTLS handshake has completed the peers can send RTP or RTCP
   over the newly created channel.  We describe the transmission process
   first followed by the reception process.

   Within each RTP session, SRTP processing MUST NOT take place before
   the DTLS handshake completes.

5.1.1.  Transmission

   DTLS and TLS define a number of record content types.  In ordinary
   TLS/DTLS, all data is protected using the same record encoding and
   mechanisms.  When the mechanism described in this document is in
   effect, this is modified so that data of type "application_data"
   (used to transport data traffic) is encrypted using SRTP rather than
   the standard TLS record encoding.

   When a user of DTLS wishes to send an RTP packet in SRTP mode it
   delivers it to the DTLS implementation as a single write of type
   "application_data".  The DTLS implementation then invokes the
   processing described in RFC 3711 Sections 3 and 4.  The resulting
   SRTP packet is then sent directly on the wire as a single datagram
   with no DTLS framing.  This provides an encapsulation of the data
   that conforms to and interoperates with SRTP.  Note that the RTP

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   sequence number rather than the DTLS sequence number is used for
   these packets.

5.1.2.  Reception

   When DTLS-SRTP is used to protect an RTP session, the RTP receiver
   needs to demultiplex packets that are arriving on the RTP port.
   Arriving packets may be of types RTP, DTLS, or STUN[14].  The type of
   a packet can be determined by looking at its first byte.

   The process for demultiplexing a packet is as follows.  The receiver
   looks at the first byte of the packet.  If the value of this byte is
   0 or 1, then the packet is STUN.  If the value is in between 128 and
   191 (inclusive), then the packet is RTP (or RTCP, if both RTCP and
   RTP are being multiplexed over the same destination port).  If the
   value is between 20 and 63 (inclusive), the packet is DTLS.  This
   processes is summarized in Figure 7.

                   | 127 < B < 192 -+--> forward to RTP
                   |                |
       packet -->  |  19 < B < 64  -+--> forward to DTLS
                   |                |
                   |       B < 2   -+--> forward to STUN

         Figure 7: The DTLS-SRTP receiver's packet demultiplexing
   algorithm.  Here the field B denotes the leading byte of the packet.

   In some cases there will be multiple DTLS-SRTP associations for a
   given SRTP endpoint.  For instance, if Alice makes a call which is
   SIP forked to both Bob and Charlie, she will use the same local host/
   port pair for both of them, as shown in Figure 8.  (The SSRCs shown
   are the ones for data flowing to Alice).

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                                          Bob (
                                       / SSRC=1
                                      /  DTLS-SRTP=XXX
               Alice (
                                      \  SSRC=2
                                       \ DTLS-SRTP=YYY
                                          Charlie (

                  Figure 8: RTP sessions with SIP forking

   Because DTLS operates on the host/port quartet, the DTLS association
   will still complete correctly, with the foreign host/port pair being
   used to distinguish the associations.  However, in RTP the source
   host/port is not used and sessions are identified by the destination
   host/port and the SSRC.  Thus, some mechanism is needed to determine
   which SSRCs correspond to which DTLS associations.  The following
   method SHOULD be used.

   For each local host/port pair, the DTLS-SRTP implementation maintains
   a table listing all the SSRCs it knows about and the DTLS-SRTP
   associations they correspond to.  Initially, this table is empty.
   When an SRTP packet is received, the following procedure is used:

   1.  If the SSRC is already known, then the corresponding DTLS-SRTP
       association and its keying material is used to decrypt the
   2.  If the SSRC is not known, then the receiver tries to decrypt it
       with the keying material corresponding to each DTLS-SRTP
   3.  If the decryption succeeds (the authentication tag verifies) then
       an entry is placed in the table mapping the SSRC to that
   4.  If the decryption fails, then the packet is silently discarded.
   5.  When a DTLS-SRTP association is closed (for instance because the
       fork is abandoned) its entries MUST be removed from the mapping

   The average cost of this algorithm for a single SSRC is the
   decryption time of a single packet times the number valid DTLS-SRTP
   associations corresponding to a single receiving port on the host.

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   In practice, this means the number of forks, so in the case shown in
   Figure 8, that would be two.  This cost is only incurred once for any
   given SSRC, since afterwards that SSRC is placed in the map table and
   looked up immediately.  As with normal RTP, this algorithm allows new
   SSRCs to be introduced by the source at any time.  They will
   automatically be mapped to the correct DTLS association.

   Note that this algorithm explicitly allows multiple SSRCs to be sent
   from the same address/port pair.  One way in which this can happen is
   an RTP translator.  This algorithm will automatically assign the
   SSRCs to the correct associations.  Note that because the SRTP
   packets are cryptographically protected, such a translator must
   either share keying material with one endpoint in or refrain from
   modifying the packets in a way which would cause the integrity check
   to fail.  This is a general property of SRTP and is not specific to

   There are two error cases that should be considered.  First, if an
   SSRC collision occurs, then only the packets from the first source
   will be processed.  When the packets from the second source arrive,
   the DTLS association with the first source will be used for
   decryption, which will fail, and the packet will be discarded.  This
   is consistent with [4], which permits the receiver to keep the
   packets from one source and discard those from the other.  Of course
   the RFC 3550 SSRC collision detection and handling procedures MUST
   also be followed.

   Second, there may be cases where a malfunctioning source is sending
   corrupt packets which cannot be decrypted.  In this case, the SSRC
   will never be entered into the mapping table, because the decryption
   always fails.  Receivers MAY keep records of unmapped SSRCs which
   consistently fail decryption and abandon attempts to decrypt them
   once they reach some limit.  That limit MUST be large enough to
   account for the effects of transmission errors.

5.2.  Rehandshake and Re-key

   Rekeying in DTLS is accomplished by performing a new handshake over
   the existing DTLS channel.  This handshake can be performed in
   parallel with data transport, so no interruption of the data flow is
   required.  Once the handshake is finished, the newly derived set of
   keys is used to protect all outbound packets, both DTLS and SRTP.

   Because of packet reordering, packets protected by the previous set
   of keys can appear on the wire after the handshake has completed.  To
   compensate for this fact, receivers SHOULD maintain both sets of keys
   for some time in order to be able to decrypt and verify older
   packets.  The keys should be maintained for the duration of the

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   maximum segment lifetime (MSL).

   If an MKI is used, then the receiver should use the corresponding set
   of keys to process an incoming packet.  Otherwise, when a packet
   arrives after the handshake completed, a receiver SHOULD use the
   newly derived set of keys to process that packet unless there is an
   MKI (If the packet was protected with the older set of keys, this
   fact will become apparent to the receiver as an authentication
   failure will occur.)  If the authentication check on the packet fails
   and no MKI is being used, then the receiver MAY process the packet
   with the older set of keys.  If that authentication check indicates
   that the packet is valid, the packet should be accepted; otherwise,
   the packet MUST be discarded and rejected.

   Receivers MAY use the SRTP packet sequence number to aid in the
   selection of keys.  After a packet has been received and
   authenticated with the new key set, any packets with sequence numbers
   that are greater will also have been protected with the new key set.

6.  Multi-party RTP Sessions

   Since DTLS is a point-to-point protocol, DTLS-SRTP is intended only
   to protect unicast RTP sessions.  This does not preclude its use with
   RTP mixers.  For example, a conference bridge may use DTLS-SRTP to
   secure the communication to and from each of the participants in a
   conference.  However, because each flow between an endpoint and a
   mixer has its own key, the mixer has to decrypt and then reencrypt
   the traffic for each recipient.

   A future specification may describe methods for sharing a single key
   between multiple DTLS-SRTP associations which would allow
   conferencing systems to avoid the decrypt/reencrypt stage.  However,
   any system in which the media is modified (e.g., for level balancing
   or transcoding) will generally need to be performed on the plaintext
   and will certainly break the authentication tag and therefore will
   require a decrypt/reencrypt stage.

7.  Security Considerations

   The use of multiple data protection framings negotiated in the same
   handshake creates some complexities, which are discussed here.

7.1.  Security of Negotiation

   One concern here is that attackers might be able to implement a bid-
   down attack forcing the peers to use ordinary DTLS rather than SRTP.

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   However, because the negotiation of this extension is performed in
   the DTLS handshake, it is protected by the Finished messages.
   Therefore, any bid-down attack is automatically detected, which
   reduces this to a denial of service attack - which any attacker who
   can control the channel can always mount.

7.2.  Framing Confusion

   Because two different framing formats are used, there is concern that
   an attacker could convince the receiver to treat an SRTP-framed RTP
   packet as a DTLS record (e.g., a handshake message) or vice versa.
   This attack is prevented by using different keys for MAC verification
   for each type of data.  Therefore, this type of attack reduces to
   being able to forge a packet with a valid MAC, which violates a basic
   security invariant of both DTLS and SRTP.

   As an additional defense against injection into the DTLS handshake
   channel, the DTLS record type is included in the MAC.  Therefore, an
   SRTP record would be treated as an unknown type and ignored.  (See
   Section 6 of [5]).

7.3.  Sequence Number Interactions

   As described in Section 5.1.1, the SRTP and DTLS sequence number
   spaces are distinct.  This means that it is not possible to
   unambiguously order a given DTLS control record with respect to an
   SRTP packet.  In general, this is relevant in two situations:  alerts
   and rehandshake.

7.3.1.  Alerts

   Because DTLS handshake and change_cipher_spec messages share the same
   sequence number space as alerts, they can be ordered correctly.
   Because DTLS alerts are inherently unreliable and SHOULD NOT be
   generated as a response to data packets, reliable sequencing between
   SRTP packets and DTLS alerts is not an important feature.  However,
   implementations which wish to use DTLS alerts to signal problems with
   the SRTP encoding SHOULD simply act on alerts as soon as they are
   received and assume that they refer to the temporally contiguous
   stream.  Such implementations MUST check for alert retransmission and
   discard retransmitted alerts to avoid overreacting to replay attacks.

7.3.2.  Renegotiation

   Because the rehandshake transition algorithm specified in Section
   Section 5.2 requires trying multiple sets of keys if no MKI is used,
   it slightly weakens the authentication.  For instance, if an n-bit
   MAC is used and k different sets of keys are present, then the MAC is

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   weakened by log_2(k) bits to n - log_2(k).  In practice, since the
   number of keys used will be very small and the MACs in use are
   typically strong (the default for SRTP is 80 bits) the decrease in
   security involved here is minimal.

   Another concern here is that this algorithm slightly increases the
   work factor on the receiver because it needs to attempt multiple
   validations.  However, again, the number of potential keys will be
   very small (and the attacker cannot force it to be larger) and this
   technique is already used for rollover counter management, so the
   authors do not consider this to be a serious flaw.

7.4.  Decryption Cost

   An attacker can impose computational costs on the receiver by sending
   superficially valid SRTP packets which do not decrypt correctly.  In
   general, encryption algorithms are so fast that this cost is
   extremely small compared to the bandwidth consumed.  The SSRC-DTLS
   mapping algorithm described in Section 5.1.2 gives the attacker a
   slight advantage here because he can force the receiver to do more
   then one decryption per packet.  However, this advantage is modest
   because the number of decryptions that the receiver does is limited
   by the number of associations he has corresponding to a given
   destination host/port, which is typically quite small.  For
   comparison, a single 1024-bit RSA private key operation (the typical
   minimum cost to establish a DTLS-SRTP association) is hundreds of
   times as expensive as decrypting an SRTP packet.

   Implementations can detect this form of attack by keeping track of
   the number of SRTP packets observed with unknown SSRCs which fail the
   authentication tag check.  If under such attack, implementations
   SHOULD prioritize decryption and verification of packets which either
   have known SSRCs or come from source addresses which match those of
   peers with which it has DTLS-SRTP associations.

8.  Session Description for RTP/SAVP over DTLS

   This specification defines new tokens to describe the protocol used
   in SDP media descriptions ('m' lines and their associated
   parameters).  The new values defined for the proto field are:
   o  When a RTP/SAVP stream is transported over DTLS with DCCP, then
      the token SHALL be DCCP/TLS/RTP/SAVP.
   o  When a RTP/SAVP stream is transported over DTLS with UDP, the
      token SHALL be UDP/TLS/RTP/SAVP.
   o  When a RTP/SAVP stream is transported over DTLS with TCP, the
      token SHALL be TCP/TLS/RTP/SAVP.  Note that even though TCP is
      being used, the PDUs carried over the TCP connection are the same

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      as would be carried over DCCP or UDP.

   The "fmt" parameter SHALL be as defined for RTP/SAVP.

9.  IANA Considerations

   This document a new extension for DTLS, in accordance with [7]:
        enum { use_srtp (??) } ExtensionType;

   [[ NOTE:  This value needs to be assigned by IANA ]]

   This extension MUST only be used with DTLS, and not with TLS.

   Section 4.1.2 requires that all SRTPProtectionProfile values be
   defined by RFC 2434 Standards Action.  IANA SHOULD create a DTLS
   SRTPProtectionProfile registry initially populated with values from
   Section 4.1.2 of this document.  Future values MUST be allocated via
   Standards Action as described in [3]

   This specification updates the "Session Description Protocol (SDP)
   Parameters" registry as defined in Appendix B of RFC 2327 [2].
   Specifically it adds the following values to the table for the
   "proto" field.

           Type            SDP Name                     Reference
           ----            ------------------           ---------
           proto           TCP/TLS/RTP/SAVP             [RFC-XXXX]
           proto           UDP/TLS/RTP/SAVP             [RFC-XXXX]
           proto           DCCP/TLS/RTP/SAVP            [RFC-XXXX]

   Note to RFC Editor:  Please replace RFC-XXXX with the RFC number of
   this specification.

10.  Acknowledgments

   Special thanks to Flemming Andreasen, Francois Audet, Jason Fischl,
   Cullen Jennings, Colin Perkins, and Dan Wing, for input, discussions,
   and guidance.

11.  References

11.1.  Normative References

   [1]   Bradner, S., "Key words for use in RFCs to Indicate Requirement
         Levels", BCP 14, RFC 2119, March 1997.

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   [2]   Handley, M. and V. Jacobson, "SDP: Session Description
         Protocol", RFC 2327, April 1998.

   [3]   Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
         Considerations Section in RFCs", BCP 26, RFC 2434,
         October 1998.

   [4]   Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
         "RTP: A Transport Protocol for Real-Time Applications", STD 64,
         RFC 3550, July 2003.

   [5]   Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS)
         Protocol Version 1.1", RFC 4346, April 2006.

   [6]   Rescorla, E. and N. Modadugu, "Datagram Transport Layer
         Security", RFC 4347, April 2006.

   [7]   Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J., and
         T. Wright, "Transport Layer Security (TLS) Extensions",
         RFC 4366, April 2006.

   [8]   Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
         Norrman, "The Secure Real-time Transport Protocol (SRTP)",
         RFC 3711, March 2004.

   [9]   Andreasen, F., Baugher, M., and D. Wing, "Session Description
         Protocol (SDP) Security Descriptions for Media Streams",
         RFC 4568, July 2006.

   [10]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
         BCP 131, RFC 4961, July 2007.

   [11]  Rescorla, E., "Keying Material Extractors for Transport Layer
         Security (TLS)", draft-ietf-tls-extractor-01 (work in
         progress), February 2008.

11.2.  Informational References

   [12]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
         Control Packets on a Single Port",
         draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress),
         August 2007.

   [13]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for
         Establishing an SRTP Security Context using DTLS",
         draft-ietf-sip-dtls-srtp-framework-01 (work in progress),
         February 2008.

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   [14]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
         Traversal Utilities for (NAT) (STUN)",
         draft-ietf-behave-rfc3489bis-15 (work in progress),
         February 2008.

   [15]  McGrew, D., "The use of AES-192 and AES-256 in Secure RTP",
         draft-mcgrew-srtp-big-aes-00 (work in progress), April 2006.

Appendix A.  Performance of Multiple DTLS Handshakes

   Standard practice for security protocols such as TLS, DTLS, and SSH
   which do inline key management is to create a separate security
   association for each underlying network channel (TCP connection, UDP
   host/port quartet, etc.).  This has dual advantages of simplicity and
   independence of the security contexts for each channel.

   Three concerns have been raised about the overhead of this strategy
   in the context of RTP security.  The first concern is the additional
   performance overhead of doing a separate public key operation for
   each channel.  The conventional procedure here (used in TLS and DTLS)
   is to establish a master context which can then be used to derive
   fresh traffic keys for new associations.  In TLS/DTLS this is called
   "session resumption" and can be transparently negotiated between the

   The second concern is network bandwidth overhead for the
   establishment of subsequent connections and for rehandshake (for
   rekeying) for existing connections.  In particular, there is a
   concern that the channels will have very narrow capacity requirements
   allocated entirely to media which will be overflowed by the
   rehandshake.  Measurements of the size of the rehandshake (with
   resumption) in TLS indicate that it is about 300-400 bytes if a full
   selection of cipher suites is offered. (the size of a full handshake
   is approximately 1-2k larger because of the certificate and keying
   material exchange).

   The third concern is the additional round-trips associated with
   establishing the 2nd, 3rd, ... channels.  In TLS/DTLS these can all
   be done in parallel but in order to take advantage of session
   resumption they should be done after the first channel is
   established.  For two channels this provides a ladder diagram
   something like this (parenthetical #s are media channel #s)

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   Alice                                   Bob
                      <-       ClientHello (1)
   ServerHello (1)    ->
   Certificate (1)
   ServerHelloDone (1)
                      <- ClientKeyExchange (1)
                          ChangeCipherSpec (1)
                                  Finished (1)
   ChangeCipherSpec (1)->
   Finished         (1)->
                                                <--- Channel 1 ready

                      <-       ClientHello (2)
   ServerHello (2)    ->
   Finished(2)        ->
                      <-  ChangeCipherSpec (2)
                                  Finished (2)
                                                <--- Channel 2 ready

   So, there is an additional 1 RTT after Channel 1 is ready before
   Channel 2 is ready.  If the peers are potentially willing to forego
   resumption they can interlace the handshakes, like so:

   Alice                                   Bob
                      <-       ClientHello (1)
   ServerHello (1)    ->
   Certificate (1)
   ServerHelloDone (1)
                      <- ClientKeyExchange (1)
                          ChangeCipherSpec (1)
                                  Finished (1)
                      <-       ClientHello (2)
   ChangeCipherSpec (1)->
   Finished         (1)->
                                                <--- Channel 1 ready
   ServerHello (2)    ->
   Finished(2)        ->
                      <-  ChangeCipherSpec (2)
                                  Finished (2)
                                                <--- Channel 2 ready

   In this case the channels are ready contemporaneously, but if a
   message in handshake (1) is lost then handshake (2) requires either a
   full rehandshake or that Alice be clever and queue the resumption

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   attempt until the first handshake completes.  Note that just dropping
   the packet works as well since Bob will retransmit.

Authors' Addresses

   David McGrew
   Cisco Systems
   510 McCarthy Blvd.
   Milpitas, CA  95305


   Eric Rescorla
   Network Resonance
   2064 Edgewood Drive
   Palo Alto, CA  94303


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Full Copyright Statement

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