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Versions: (draft-fischl-sipping-media-dtls) 00 01 02 03 04 05 06 07 RFC 5763

SIP                                                            J. Fischl
Internet-Draft                               CounterPath Solutions, Inc.
Intended status:  Standards Track                          H. Tschofenig
Expires:  May 15, 2008                            Nokia Siemens Networks
                                                             E. Rescorla
                                                       Network Resonance
                                                       November 12, 2007


     Framework for Establishing an SRTP Security Context using DTLS
               draft-ietf-sip-dtls-srtp-framework-00.txt

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
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   This Internet-Draft will expire on May 15, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2007).

Abstract

   This document specifies how to use the Session Initiation Protocol
   (SIP) to establish an Secure Real-time Transport Protocol (SRTP)
   security context using the Datagram Transport Layer Security (DTLS)
   protocol.  It describes a mechanism of transporting a fingerprint
   attribute in the Session Description Protocol (SDP) that identifies



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   the key that will be presented during the DTLS handshake.  It relies
   on the SIP identity mechanism to ensure the integrity of the
   fingerprint attribute.  The key management exchange travels along the
   media path as opposed to the signaling path.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  5
   3.  Motivation . . . . . . . . . . . . . . . . . . . . . . . . . .  6
   4.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  7
   5.  Verifying Certificate Integrity  . . . . . . . . . . . . . . .  7
   6.  Miscellaneous Considerations . . . . . . . . . . . . . . . . .  8
     6.1.  Anonymous Calls  . . . . . . . . . . . . . . . . . . . . .  8
     6.2.  Early Media  . . . . . . . . . . . . . . . . . . . . . . .  9
     6.3.  Forking  . . . . . . . . . . . . . . . . . . . . . . . . .  9
     6.4.  Delayed Offer Calls  . . . . . . . . . . . . . . . . . . .  9
     6.5.  Session Modification . . . . . . . . . . . . . . . . . . . 10
     6.6.  UDP Payload De-multiplex . . . . . . . . . . . . . . . . . 10
     6.7.  Rekeying . . . . . . . . . . . . . . . . . . . . . . . . . 10
     6.8.  Conference Servers and Shared Encryptions Contexts . . . . 11
     6.9.  Media over SRTP  . . . . . . . . . . . . . . . . . . . . . 11
     6.10. Best Effort Encryption . . . . . . . . . . . . . . . . . . 11
   7.  Example Message Flow . . . . . . . . . . . . . . . . . . . . . 11
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 16
     8.1.  UPDATE . . . . . . . . . . . . . . . . . . . . . . . . . . 17
     8.2.  SIPS . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     8.3.  S/MIME . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     8.4.  Single-sided Verification  . . . . . . . . . . . . . . . . 18
     8.5.  Continuity of Authentication . . . . . . . . . . . . . . . 18
     8.6.  Short Authentication String  . . . . . . . . . . . . . . . 19
     8.7.  Perfect Forward Secrecy  . . . . . . . . . . . . . . . . . 19
   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 19
   10. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 20
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 20
     11.2. Informational References . . . . . . . . . . . . . . . . . 21
   Appendix A.  Requirements Analysis . . . . . . . . . . . . . . . . 23
     A.1.  Forking and retargeting (R1, R2, R3) . . . . . . . . . . . 23
     A.2.  Reusage of a Security Context (R4), (R11)  . . . . . . . . 23
     A.3.  Clipping (R5)  . . . . . . . . . . . . . . . . . . . . . . 23
     A.4.  Passive Attacks on the Media Path (R6) . . . . . . . . . . 24
     A.5.  Passive Attacks on the Signaling Path (R7) . . . . . . . . 24
     A.6.  Perfect Forward Secrecy (R8) . . . . . . . . . . . . . . . 24
     A.7.  Algorithm Negotiation (R9) . . . . . . . . . . . . . . . . 24
     A.8.  RTP Validity Check (R10) . . . . . . . . . . . . . . . . . 24
     A.9.  3rd Party Certificates (R12, R18)  . . . . . . . . . . . . 24



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     A.10. FIPS 140-2 (R13) . . . . . . . . . . . . . . . . . . . . . 24
     A.11. Linkage between Keying Exchange and SIP Signaling (R14)  . 24
     A.12. Start with RTP and Upgrade to SRTP (R15) . . . . . . . . . 25
     A.13. Denial of Service Vulnerability (R16)  . . . . . . . . . . 25
     A.14. Adversary Model (R17)  . . . . . . . . . . . . . . . . . . 25
     A.15. Crypto-Agility (R19) . . . . . . . . . . . . . . . . . . . 25
     A.16. Downgrading Protection (R20) . . . . . . . . . . . . . . . 25
     A.17. Media Security Negotation (R21)  . . . . . . . . . . . . . 25
     A.18. Signaling Protocol Independence (R22)  . . . . . . . . . . 25
     A.19. Media Recording (R23)  . . . . . . . . . . . . . . . . . . 25
     A.20. Lawful Interception (R24)  . . . . . . . . . . . . . . . . 26
     A.21. Interworking with Intermediaries (R25) . . . . . . . . . . 26
     A.22. PSTN Gateway Termination (R26) . . . . . . . . . . . . . . 26
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 26
   Intellectual Property and Copyright Statements . . . . . . . . . . 28




































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1.  Introduction

   The Session Initiation Protocol (SIP) [RFC3261] and the Session
   Description Protocol (SDP) [RFC4566] are used to set up multimedia
   sessions or calls.  SDP is also used to set up TCP [RFC4145] and
   additionally TCP/TLS connections for usage with media sessions
   [RFC4572].  The Real-Time Protocol (RTP) [RFC3550] is used to
   transmit real time media on top of UDP, TCP [RFC4571], and TLS
   [RFC4572].  Datagram TLS [RFC4347] was introduced to allow TLS
   functionality to be applied to datagram transport protocols, such as
   UDP and DCCP.  This draft provides guidelines on how to use and to
   support for (a) transmission of media over DTLS and (b) to establish
   SRTP security using extensions to DTLS (see
   [I-D.ietf-avt-dtls-srtp]).

   The goal of this work is to provide a key negotiation technique that
   allows encrypted communication between devices with no prior
   relationships.  It also does not require the devices to trust every
   call signaling element that was involved in routing or session setup.
   This approach does not require any extra effort by end users and does
   not require deployment of certificates to all devices that are signed
   by a well-known certificate authority.

   The media is transported over a mutually authenticated DTLS session
   where both sides have certificates.  The certificate fingerprints are
   sent in SDP over SIP as part of the offer/answer exchange.  The SIP
   Identity mechanism [RFC4474] is used to provide integrity for the
   fingerprints.  It is very important to note that certificates are
   being used purely as a carrier for the public keys of the peers.
   This is required because DTLS does not have a mode for carrying bare
   keys, but it is purely an issue of formatting.  The certificates can
   be self-signed and completely self-generated.  All major TLS stacks
   have the capability to generate such certificates on demand.
   However, third party certificates MAY also be used for extra
   security.

   This approach differs from previous attempts to secure media traffic
   where the authentication and key exchange protocol (e.g., MIKEY
   [RFC3830]) is piggybacked in the signaling message exchange.  With
   this approach, establishing the protection of the media traffic
   between the endpoints is done by the media endpoints without
   involving the SIP/SDP communication.  It allows RTP and SIP to be
   used in the usual manner when there is no encrypted media.

   In SIP, typically the caller sends an offer and the callee may
   subsequently send one-way media back to the caller before a SIP
   answer is received by the caller.  The approach in this
   specification, where the media key negotiation is decoupled from the



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   SIP signaling, allows the early media to be set up before the SIP
   answer is received while preserving the important security property
   of allowing the media sender to choose some of the keying material
   for the media.  This also allows the media sessions to be changed,
   re-keyed, and otherwise modified after the initial SIP signaling
   without any additional SIP signaling.

   Design decisions that influence the applicability of this
   specification are discussed in Section 3.


2.  Overview

   Endpoints wishing to set up an RTP media session do so by exchanging
   offers and answers in SDP messages over SIP.  In a typical use case,
   two endpoints would negotiate to transmit audio data over RTP using
   the UDP protocol.

   Figure 1 shows a typical message exchange in the SIP Trapezoid.

                 +-----------+            +-----------+
                 |SIP        |   SIP/SDP  |SIP        |
         +------>|Proxy      |<---------->|Proxy      |<------+
         |       |Server X   | (+finger-  |Server Y   |       |
         |       +-----------+   print,   +-----------+       |
         |                      +auth.id.)                    |
         | SIP/SDP                              SIP/SDP       |
         | (+fingerprint)                       (+fingerprint,|
         |                                       +auth.id.)   |
         |                                                    |
         v                                                    v
     +-----------+          Datagram TLS               +-----------+
     |SIP        | <---------------------------------> |SIP        |
     |User Agent |               Media                 |User Agent |
     |Alice@X    | <=================================> |Bob@Y      |
     +-----------+                                     +-----------+

     Legend:
     <--->: Signaling Traffic
     <===>: Data Traffic

                 Figure 1: DTLS Usage in the SIP Trapezoid

   Consider Alice wanting to set up an encrypted audio session with Bob.
   Both Bob and Alice could use public-key based authentication in order
   to establish a confidentiality protected channel using DTLS.

   Since providing mutual authentication between two arbitrary end



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   points on the Internet using public key based cryptography tends to
   be problematic, we consider more deployment friendly alternatives.
   This document uses one approach and several others are discussed in
   Section 8.

   Alice sends an SDP offer to Bob over SIP.  If Alice uses only self-
   signed certificates for the communication with Bob, a fingerprint is
   included in the SDP offer/answer exchange.  This fingerprint is
   integrity protected using the identity mechanism defined in
   Enhancements for Authenticated Identity Management in SIP [RFC4474].
   When Bob receives the offer, Bob establishes a mutually authenticated
   DTLS connection with Alice.  At this point Bob can begin sending
   media to Alice.  Once Bob accepts Alice's offer and sends an SDP
   answer to Alice, Alice can begin sending confidential media to Bob.


3.  Motivation

   Although there is already prior work in this area (e.g., Secure
   Descriptions for SDP [RFC4568], Key Management Extensions [RFC4567]
   combined with MIKEY [RFC3830] for authentication and key exchange),
   this specification is motivated as follows:

   o  TLS will be used to offer security for connection-oriented media.
      The design of TLS is well-known and implementations are widely
      available.
   o  This approach deals with forking and early media without requiring
      support for PRACK [RFC3262] while preserving the important
      security property of allowing the offerer to choose keying
      material for encrypting the media.
   o  The establishment of security protection for the media path is
      also provided along the media path and not over the signaling
      path.  In many deployment scenarios, the signaling and media
      traffic travel along a different path through the network.
   o  This solution works even when the SIP proxies downstream of the
      identity service are not trusted.  There is no need to reveal keys
      in the SIP signaling or in the SDP message exchange.  In order for
      SDES and MIKEY to provide this security property, they require
      distribution of certificates to the endpoints that are signed by
      well known certificate authorities.  SDES further requires that
      the endpoints employ S/MIME to encrypt the keying material.
   o  In this method, SSRC collisions do not result in any extra SIP
      signaling.
   o  Many SIP endpoints already implement TLS.  The changes to existing
      SIP and RTP usage are minimal even when DTLS-SRTP
      [I-D.ietf-avt-dtls-srtp] is used.





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4.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   DTLS/TLS uses the term "session" to refer to a long-lived set of
   keying material that spans associations.  In this document,
   consistent with SIP/SDP usage, we use it to refer to a multimedia
   session and use the term "TLS session" to refer to the TLS construct.
   We use the term "association" to refer to a particular DTLS
   ciphersuite and keying material set.  For consistency with other SIP/
   SDP usage, we use the term "connection" when what's being referred to
   is a multimedia stream that is not specifically DTLS/TLS.

   In this document, the term "Mutual DTLS" indicates that both the DTLS
   client and server present certificates even if one or both
   certificates are self-signed.


5.  Verifying Certificate Integrity

   The offer/answer model, defined in [RFC3264], is used by protocols
   like the Session Initiation Protocol (SIP) [RFC3261] to set up
   multimedia sessions.  In addition to the usual contents of an SDP
   [RFC4566] message, each 'm' line will also contain several attributes
   as specified in [I-D.fischl-mmusic-sdp-dtls], [RFC4145] and
   [RFC4572].

   The endpoint MUST use the setup and connection attributes defined in
   [RFC4145].  A setup:active endpoint will act as a DTLS client and a
   setup:passive endpoint will act as a DTLS server.  The connection
   attribute indicates whether or not to reuse an existing DTLS
   association.

   The endpoint MUST use the certificate fingerprint attribute as
   specified in [RFC4572].

   The setup:active endpoint establishes a DTLS association with the
   setup:passive endpoint [RFC4145].  Typically, the receiver of the SIP
   INVITE request containing an offer will take the setup:active role.

   The certificate presented during the DTLS handshake MUST match the
   fingerprint exchanged via the signaling path in the SDP.  The
   security properties of this mechanism are described in Section 8.

   If the fingerprint does not match the hashed certificate then the
   endpoint MUST tear down the media session immediately.



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   When an endpoint wishes to set up a secure media session with another
   endpoint it sends an offer in a SIP message to the other endpoint.
   This offer includes, as part of the SDP payload, the fingerprint of
   the certificate that the endpoint wants to use.  The SIP message
   containing the offer is sent to the offerer's sip proxy over an
   integrity protected channel which will add an identity header
   according to the procedures outlined in [RFC4474].  When the far
   endpoint receives the SIP message it can verify the identity of the
   sender using the identity header.  Since the identity header is a
   digital signature across several SIP headers, in addition to the
   bodies of the SIP message, the receiver can also be certain that the
   message has not been tampered with after the digital signature was
   applied and added to the SIP message.

   The far endpoint (answerer) may now establish a mutually
   authenticated DTLS association to the offerer.  After completing the
   DTLS handshake, information about the authenticated identities,
   including the certificates, are made available to the endpoint
   application.  The answerer is then able to verify that the offerer's
   certificate used for authentication in the DTLS handshake can be
   associated to the certificate fingerprint contained in the offer in
   the SDP.  At this point the answerer may indicate to the end user
   that the media is secured.  The offerer may only tentatively accept
   the answerer's certificate since it may not yet have the answerer's
   certificate fingerprint.

   When the answerer accepts the offer, it provides an answer back to
   the offerer containing the answerer's certificate fingerprint.  At
   this point the offerer can definitively accept or reject the peer's
   certificate and the offerer can indicate to the end user that the
   media is secured.

   Note that the entire authentication and key exchange for securing the
   media traffic is handled in the media path through DTLS.  The
   signaling path is only used to verify the peers' certificate
   fingerprints.


6.  Miscellaneous Considerations

6.1.  Anonymous Calls

   When making anonymous calls, a new self-signed certificate SHOULD be
   used for each call so that the calls can not be correlated as to
   being from the same caller.  In situations where some degree of
   correlation is acceptable, the same certificate SHOULD be used for a
   number of calls in order to enable continuity of authentication, see
   Section 8.5.



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   Additionally, it MUST be ensured that the Privacy header [RFC3325] is
   used in conjunction with the SIP identity mechanism to ensure that
   the identity of the user is not asserted when enabling anonymous
   calls.  Furthermore, the content of the subjectAltName attribute
   inside the certificate MUST NOT contain information that either
   allows correlation or identification of the user that wishes to place
   an anonymous call.

6.2.  Early Media

   If an offer is received by an endpoint that wishes to provide early
   media, it MUST take the setup:active role and can immediately
   establish a DTLS association with the other endpoint and begin
   sending media.  The setup:passive endpoint may not yet have validated
   the fingerprint of the active endpoint's certificate.  The security
   aspects of media handling in this situation are discussed in
   Section 8.

6.3.  Forking

   In SIP, it is possible for a request to fork to multiple endpoints.
   Each forked request can result in a different answer.  Assuming that
   the requester provided an offer, each of the answerers' will provide
   a unique answer.  Each answerer will create a DTLS association with
   the offerer.  The offerer can then correlate the SDP answer received
   in the SIP message by comparing the fingerprint in the answer to the
   hashed certificate for each DTLS association.

   Note that in the situation where a request forks to multiple
   endpoints that all share the same certificate, there is no way for
   the caller to correlate the DTLS associations with the SIP dialogs.
   Practically, this is not a problem, since the callees will terminate
   the unused associations.  No new security problem is introduced here
   since endpoints which share the same certificate are assumed to
   represent the same user.

6.4.  Delayed Offer Calls

   An endpoint may send a SIP INVITE request with no offer in it.  When
   this occurs, the receiver(s) of the INVITE will provide the offer in
   the response and the originator will provide the answer in the
   subsequent ACK request or in the PRACK request [RFC3262] if both
   endpoints support reliable provisional responses.  In any event, the
   active endpoint still establishes the DTLS association with the
   passive endpoint as negotiated in the offer/answer exchange.






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6.5.  Session Modification

   Once an answer is provided to the offerer, either endpoint MAY
   request a session modification which MAY include an updated offer.
   This session modification can be carried in either an INVITE or
   UPDATE request.  In this case, it is RECOMMENDED that the offerer
   indicate a request to reuse the existing association (using the
   connection attribute) as described in Connection-Oriented Media
   [RFC4145].  Once the answer is received, the active endpoint will
   either reuse the existing association or establish a new one, tearing
   down the existing association as soon as the offer/answer exchange is
   completed.  The exact association/connection reuse behavior is
   specified in RFC 4145 [RFC4145].

6.6.  UDP Payload De-multiplex

   Interactive Connectivity Establishment (ICE), as specified in
   [I-D.ietf-mmusic-ice], provides a methodology of allowing
   participants in multi-media sessions to verify mutual connectivity.
   In order to make ICE work with this specification the endpoints MUST
   be able to demultiplex STUN packets from DTLS packets.  STUN
   [RFC3489] packets MUST NOT be sent over DTLS.

   The first byte of a STUN message is 0 or 1 and it is reasonable to
   expect it to remain 0 or 1 for the near future.  The first byte of a
   DTLS packet is "Type" which can currently have values of 20, 21, 22,
   and 23 as defined in ContentType declaration in [RFC4346].  It is
   reasonable to expect the first byte to remain under 64 and greater
   than 1.  For RTP the first byte has a value that is 196 or above.  A
   viable demultiplexing strategy would be to look at the first byte of
   the UDP payload and if the value is less than 2, assume STUN, if
   greater or equal to 196 assume RTP, otherwise assume DTLS.

6.7.  Rekeying

   As with TLS, DTLS endpoints can rekey at any time by redoing the DTLS
   handshake.  While the rekey is under way, the endpoints continue to
   use the previously established keying material for usage with DTLS.
   Once the new session keys are established the session can switch to
   using these and abandon the old keys.  This ensures that latency is
   not introduced during the rekeying process.

   Further considerations regarding rekeying in case the SRTP security
   context is established with DTLS can be found in Section 3.7 of
   [I-D.ietf-avt-dtls-srtp].






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6.8.  Conference Servers and Shared Encryptions Contexts

   It has been proposed that conference servers might use the same
   encryption context for all of the participants in a conference.  The
   advantage of this approach is that the conference server only needs
   to encrypt the output for all speakers instead of once per
   participant.

   This shared encryption context approach is not possible under this
   specification.  However, it is argued that the effort to encrypt each
   RTP packet is small compared to the other tasks performed by the
   conference server such as the codec processing.

   Future extensions such as [I-D.mcgrew-srtp-ekt] could be used to
   provide this functionality in concert with the mechanisms described
   in this specification.

6.9.  Media over SRTP

   Because DTLS's data transfer protocol is generic, it is less highly
   optimized for use with RTP than is SRTP [RFC3711], which has been
   specifically tuned for that purpose.  DTLS-SRTP
   [I-D.ietf-avt-dtls-srtp], has been defined to provide for the
   negotiation of SRTP transport using a DTLS connection, thus allowing
   the performance benefits of SRTP with the easy key management of
   DTLS.  The ability to reuse existing SRTP software and hardware
   implementations may in some environments another important motivation
   for using DTLS-SRTP instead of RTP over DTLS.  Implementations of
   this specification SHOULD support DTLS-SRTP [I-D.ietf-avt-dtls-srtp].

6.10.  Best Effort Encryption

   [I-D.ietf-sip-media-security-requirements] describes a requirement
   for best effort encryption where SRTP is used where both endpoints
   support it and key negotiation succeeds otherwise RTP is used.

   [I-D.ietf-mmusic-sdp-capability-negotiation] describes a mechanism
   which can signal both RTP and SRTP as an alternative.  RTP is the
   default and will be understood by endpoints that do not understand
   SRTP or this key exchange mechanism but SRTP is preferred.


7.  Example Message Flow

   Prior to establishing the session, both Alice and Bob generate self-
   signed certificates which are used for a single session or, more
   likely, reused for multiple sessions.  In this example, Alice calls
   Bob. In this example we assume that Alice and Bob share the same



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   proxy.

   The example shows the SIP message flows where Alice acts as the
   passive endpoint and Bob acts as the active endpoint meaning that as
   soon as Bob receives the INVITE from Alice, with DTLS specified in
   the 'm' line of the offer, Bob will begin to negotiate a DTLS
   association with Alice for both RTP and RTCP streams.  Early media
   (RTP and RTCP) starts to flow from Bob to Alice as soon as Bob sends
   the DTLS finished message to Alice.  Bi-directional media (RTP and
   RTCP) can flow after Bob sends the SIP 200 response and once Alice
   has sent the DTLS finished message.

   The SIP signaling from Alice to her proxy is transported over TLS to
   ensure an integrity protected channel between Alice and her identity
   service.  Note that all other signaling is transported over TCP in
   this example although it could be done over any supported transport.


   Alice            Proxies             Bob
     |(1) INVITE       |                  |
     |---------------->|                  |
     |                 |(2) INVITE        |
     |                 |----------------->|
     |                 |        (3) hello |
     |<-----------------------------------|
     |(4) hello        |                  |
     |----------------------------------->|
     |                 |     (5) finished |
     |<-----------------------------------|
     |                 |     (6) media    |
     |<-----------------------------------|
     |(7) finished     |                  |
     |----------------------------------->|
     |                 |     (8) 200 OK   |
     |<-----------------------------------|
     |                 |     (9) media    |
     |----------------------------------->|
     |(10) ACK         |                  |
     |----------------------------------->|

   Message (1):  INVITE Alice -> Proxy


      This shows the initial INVITE from Alice to Bob carried over the
      TLS transport protocol to ensure an integrity protected channel
      between Alice and her proxy which acts as Alice's identity
      service.  Note that Alice has requested to be the passive endpoint
      which means that it will act as the DTLS server and Bob will



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      initiate the session.  Also note that there is a fingerprint
      attribute on the 'c' line of the SDP.  This is computed from Bob's
      self-signed certificate.


      [[ NOTE:  This example is not completely correct because the exact
      syntax of the SDP is not yet determined.  The MMUSIC working group
      is currently working on standardizing mechanisms for SDP
      capability negotiation which will enable this sort of best-effort
      encryption.  When that work is finished, this draft will be
      harmonized with it.]]

   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.101:5060;branch=z9hG4bK-0e53sadfkasldkfj
   Max-Forwards: 70
   Contact: <sip:alice@192.168.1.103:6937;transport=TLS>
   To: <sip:bob@example.com>
   From: "Alice"<sip:alice@example.com>;tag=843c7b0b
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
   Content-Type: application/sdp
   Content-Length: xxxx

   v=0
   o=- 1181923068 1181923196 IN IP4 192.168.1.103
   s=example1
   c=IN IP4 192.168.1.103
   a=setup:passive
   a=connection:new
   a=fingerprint: \
     SHA-1 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
   t=0 0
   m=audio 6056 RTP/AVP 0
   a=sendrecv
   a=tcap:1 UDP/TLS/RTP/AVP RTP/AVP
   a=pcfg:1 t=1



   Message (2):  INVITE Proxy -> Bob


      This shows the INVITE being relayed to Bob from Alice (and Bob's)
      proxy.  Note that Alice's proxy has inserted an Identity and
      Identity-Info header.  This example only shows one element for
      both proxies for the purposes of simplification.  Bob verifies the
      identity provided with the INVITE.  Note that this offer includes



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      a default m-line offering RTP in case the answerer does not
      support SRTP.  However, the potential configuration utilizing a
      transport of SRTP is preferred.  See
      [I-D.ietf-mmusic-sdp-capability-negotiation] for more details on
      the details of SDP capability negotiation.



   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.101:5060;branch=z9hG4bK-0e53sadfkasldkfj
   Via: SIP/2.0/TCP 192.168.1.100:5060;branch=z9hG4bK-0e53244234324234
   Via: SIP/2.0/TCP 192.168.1.103:6937;branch=z9hG4bK-0e5b7d3edb2add32
   Max-Forwards: 70
   Contact: <sip:alice@192.168.1.103:6937;transport=TLS>
   To: <sip:bob@example.com>
   From: "Alice"<sip:alice@example.com>;tag=843c7b0b
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 1 INVITE
   Identity: CyI4+nAkHrH3ntmaxgr01TMxTmtjP7MASwliNRdupRI1vpkXRvZXx1ja9k
             3W+v1PDsy32MaqZi0M5WfEkXxbgTnPYW0jIoK8HMyY1VT7egt0kk4XrKFC
             HYWGCl0nB2sNsM9CG4hq+YJZTMaSROoMUBhikVIjnQ8ykeD6UXNOyfI=
   Identity-Info: https://example.com/cert
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
   Content-Type: application/sdp
   Content-Length: xxxx

   v=0
   o=- 1181923068 1181923196 IN IP4 192.168.1.103
   s=example1
   c=IN IP4 192.168.1.103
   a=setup:passive
   a=connection:new
   a=fingerprint: \
     SHA-1 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
   t=0 0
   m=audio 6056 RTP/AVP 0
   a=sendrecv
   a=tcap:1 UDP/TLS/RTP/SAVP RTP/AVP
   a=pcfg:1 t=1


   Message (3):  ClientHello Bob -> Alice


      Assuming that Alice's identity is valid, Message 3 shows Bob
      sending a DTLS ClientHello directly to Alice for each 'm' line in
      the SDP.  In this case two DTLS ClientHello messages are sent to
      Alice.  Bob sends a DTLS ClientHello to 192.168.1.103:6056 for RTP



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      and another to port 6057 for RTCP.

   Message (4):  ServerHello+Certificate Alice -> Bob


      Alice sends back a ServerHello, Certificate, ServerHelloDone for
      both RTP and RTCP associations.  Note that the same certificate is
      used for both the RTP and RTCP associations.  If RTP/RTCP
      multiplexing [I-D.ietf-avt-rtp-and-rtcp-mux] were being used only
      a single association would be required.

   Message (5):  Certificate Bob -> Alice


      Bob sends a Certificate, ClientKeyExchange, CertificateVerify,
      change_cipher_spec and Finished for both RTP and RTCP
      associations.  Again note that Bob uses the same server
      certificate for both associations.

   Message (6):  Early Media Bob -> Alice


      At this point, Bob can begin sending early media (RTP and RTCP) to
      Alice.  Note that Alice can't yet trust the media since the
      fingerprint has not yet been received.  This lack of trusted,
      secure media is indicated to Alice.

   Message (7):  Finished Alice -> Bob


      After Message 5 is received by Bob, Alice sends change_cipher_spec
      and Finished.

   Message (8):  200 OK Bob -> Alice


      When Bob answers the call, Bob sends a 200 OK SIP message which
      contains the fingerprint for Bob's certificate.  When Alice
      receives the message and validates the certificate presented in
      Message 5.  The endpoint now shows Alice that the call as secured.











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   SIP/2.0 200 OK

   To: <sip:bob@example.com>;tag=6418913922105372816
   From: "Alice" <sip:alice@example.com>;tag=843c7b0b
   Via: SIP/2.0/TCP 192.168.1.103:6937;branch=z9hG4bK-0e5b7d3edb2add32
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 1 INVITE
   Contact: <sip:192.168.1.104:5060;transport=TCP>
   Content-Type: application/sdp
   Content-Length: xxxx

   v=0
   o=- 6418913922105372816 2105372818 IN IP4 192.168.1.104
   s=example2
   c=IN IP4 192.168.1.104
   a=setup:active
   a=connection:new
   a=fingerprint:\
     SHA-1 FF:FF:FF:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
   t=0 0
   m=audio 12000 UDP/TLS/RTP/SAVP 0
   a=rtpmap:0 PCMU/8000/1
   a=acfg:1 t=1



   Message (9):  RTP+RTCP Alice -> Bob


      At this point, Alice can also start sending RTP and RTCP to Bob.
      Note that in this case, Bob signals the actual transport protocol
      configuration of SRTP over DTLS in the acfg parameter.

   Message 10:  ACK Alice -> Bob


      Finally, Alice sends the SIP ACK to Bob.


8.  Security Considerations

   DTLS or TLS media signalled with SIP requires a way to ensure that
   the communicating peers' certificates are correct.

   The standard TLS/DTLS strategy for authenticating the communicating
   parties is to give the server (and optionally the client) a PKIX
   [RFC3280] certificate.  The client then verifies the certificate and
   checks that the name in the certificate matches the server's domain



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   name.  This works because there are a relatively small number of
   servers with well-defined names; a situation which does not usually
   occur in the VoIP context.

   The design described in this document is intended to leverage the
   authenticity of the signaling channel (while not requiring
   confidentiality).  As long as each side of the connection can verify
   the integrity of the SDP INVITE then the DTLS handshake cannot be
   hijacked via a man-in-the-middle attack.  This integrity protection
   is easily provided by the caller to the callee (see Alice to Bob in
   Section 7) via the SIP Identity [RFC4474] mechanism.  However, it is
   less straightforward for the responder.

   Ideally Alice would want to know that Bob's SDP had not been tampered
   with and who it was from so that Alice's User Agent could indicate to
   Alice that there was a secure phone call to Bob. This is known as the
   SIP connected party problem and is still a topic of ongoing work in
   the SIP community.  In the meantime, there are several approaches
   that can be used to mitigate this problem:  Use UPDATE, Use SIPS, Use
   S/MIME, Single Sided Verification, or use human-read Short
   Authentication String (SAS) to validate the certificates.  Each one
   is discussed here followed by the security implications of that
   approach.

8.1.  UPDATE

   [RFC4916] defines an approach for a UA to supply its identity to its
   peer UA and for this identity to be signed by an authentication
   service.  For example, using this approach, Bob sends an answer, then
   immediately follows up with an UPDATE that includes the fingerprint
   and uses the SIP Identity mechanism to assert that the message is
   from Bob@example.com.  The downside of this approach is that it
   requires the extra round trip of the UPDATE.  However, it is simple
   and secure even when not all of the proxies are trusted.  In this
   example, Bob only needs to trust his proxy.

   [[OPEN ISSUE:  Note that there is a window of vulnerability during
   the early media phase of this operation before Alice receives the
   UPDATE (which immediately follows the SDP answer).  During this
   window, Alice cannot be sure of Bob's identity.  This risk might be
   mitigated by including a secret in the offer which must be used to
   establish the DTLS association, for instance via TLS PSK [RFC4279].
   We are still studying this issue.  Obviously, this is more attractive
   if SIPS is used.]]







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8.2.  SIPS

   In this approach, the signaling is protected by TLS from hop to hop.
   As long as all proxies are trusted, this provides integrity for the
   fingerprint.  It does not provide a strong assertion of who Alice is
   communicating with.  However, as much as the target domain can be
   trusted to correctly populate the From header field value, Alice can
   use that.  The security issue with this approach is that if one of
   the Proxies wished to mount a man-in-the-middle attack, it could
   convince Alice that she was talking to Bob when really the media was
   flowing through a man in the middle media relay.  However, this
   attack could not convince Bob that he was taking to Alice.

8.3.  S/MIME

   RFC 3261 [RFC3261] defines a S/MIME security mechanism for SIP that
   could be used to sign that the fingerprint was from Bob. This would
   be secure.  However, so far there have been no deployments of S/MIME
   for SIP.

8.4.  Single-sided Verification

   In this approach, no integrity is provided for the fingerprint from
   Bob to Alice.  In this approach, an attacker that was on the
   signaling path could tamper with the fingerprint and insert
   themselves as a man-in-the-middle on the media.  Alice would know
   that she had a secure call with someone but would not know if it was
   with Bob or a man-in-the-middle.  Bob would know that an attack was
   happening.  The fact that one side can detect this attack means that
   in most cases where Alice and Bob both wish the communications to be
   encrypted there is not a problem.  Keep in mind that in any of the
   possible approaches Bob could always reveal the media that was
   received to anyone.  We are making the assumption that Bob also wants
   secure communications.  In this do nothing case, Bob knows the media
   has not been tampered with or intercepted by a third party and that
   it is from Alice@example.com.  Alice knows that she is talking to
   someone and that whoever that is has probably checked that the media
   is not being intercepted or tampered with.  This approach is
   certainly less than ideal but very usable for many situations.

8.5.  Continuity of Authentication

   One desirable property of a secure media system is to provide
   continuity of authentication:  being able to ensure cryptographically
   that you are talking to the same person as before.  With DTLS,
   continuity of authentication is achieved by having each side use the
   same public key/self-signed certificate for each connection (at least
   with a given peer entity).  It then becomes possible to cache the



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   credential (or its hash) and verify that it is unchanged.  Thus, once
   a single secure connection has been established, an implementation
   can establish a future secure channel even in the face of future
   insecure signalling.

   In order to enable continuity of authentication, implementations
   SHOULD attempt to keep a constant long-term key.  Verifying
   implementations SHOULD maintain a cache of the key used for each peer
   identity and alert the user if that key changes.

8.6.  Short Authentication String

   An alternative available to Alice and Bob is to use human speech to
   verify each others' identity and then to verify each others'
   fingerprints also using human speech.  Assuming that it is difficult
   to impersonate another's speech and seamlessly modify the audio
   contents of a call, this approach is relatively safe.  It would not
   be effective if other forms of communication were being used such as
   video or instant messaging.  DTLS supports this mode of operation.
   The minimal secure fingerprint length is around 64 bits.

   ZRTP [I-D.zimmermann-avt-zrtp] includes Short Authentication String
   mode in which a unique per-connection bitstring is generated as part
   of the cryptographic handshake.  The SAS can be as short as 25 bits
   and so is somewhat easier to read.  DTLS does not natively support
   this mode, however it would be straightforward to add one as a TLS
   extension [RFC3546].

8.7.  Perfect Forward Secrecy

   One concern about the use of a long-term key is that compromise of
   that key may lead to compromise of past communications.  In order to
   prevent this attack, DTLS supports modes with Perfect Forward Secrecy
   using Diffie-Hellman and Elliptic-Curve Diffie-Hellman cipher suites.
   When these modes are in use, the system is secure against such
   attacks.  Note that compromise of a long-term key may still lead to
   future active attacks.  If this is a concern, a backup authentication
   channel such as manual fingerprint establishment or a short
   authentication string should be used.


9.  IANA Considerations

   This specification does not require any IANA actions.







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10.  Acknowledgments

   Cullen Jennings contributed substantial text and comments to this
   document.  This document benefited from discussions with Francois
   Audet, Nagendra Modadugu, and Dan Wing.  Thanks also for useful
   comments by Flemming Andreasen, Rohan Mahy, David McGrew, Miguel
   Garcia, Steffen Fries, Brian Stucker, and David Oran.

   We would like to thank Thomas Belling, Guenther Horn, Steffen Fries,
   Brian Stucker, Francois Audet, Dan Wing, Jari Arkko, and Vesa
   Lehtovirta for their input regarding traversal of SBCs.


11.  References

11.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3280]  Housley, R., Polk, W., Ford, W., and D. Solo, "Internet
              X.509 Public Key Infrastructure Certificate and
              Certificate Revocation List (CRL) Profile", RFC 3280,
              April 2002.

   [RFC3325]  Jennings, C., Peterson, J., and M. Watson, "Private
              Extensions to the Session Initiation Protocol (SIP) for
              Asserted Identity within Trusted Networks", RFC 3325,
              November 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.




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   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              September 2005.

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, April 2006.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4572]  Lennox, J., "Connection-Oriented Media Transport over the
              Transport Layer Security (TLS) Protocol in the Session
              Description Protocol (SDP)", RFC 4572, July 2006.

   [I-D.fischl-mmusic-sdp-dtls]
              Fischl, J. and H. Tschofenig, "Session Description
              Protocol (SDP) Indicators for Datagram Transport Layer
              Security (DTLS)", draft-fischl-mmusic-sdp-dtls-03 (work in
              progress), July 2007.

11.2.  Informational References

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, July 2006.

   [I-D.ietf-mmusic-ice]
              Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address  Translator (NAT)
              Traversal for Offer/Answer Protocols",
              draft-ietf-mmusic-ice-19 (work in progress), October 2007.

   [RFC4567]  Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
              Carrara, "Key Management Extensions for Session
              Description Protocol (SDP) and Real Time Streaming
              Protocol (RTSP)", RFC 4567, July 2006.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

   [RFC4346]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.1", RFC 4346, April 2006.




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   [I-D.zimmermann-avt-zrtp]
              Zimmermann, P., "ZRTP: Media Path Key Agreement for Secure
              RTP", draft-zimmermann-avt-zrtp-04 (work in progress),
              July 2007.

   [I-D.mcgrew-srtp-ekt]
              McGrew, D., "Encrypted Key Transport for Secure RTP",
              draft-mcgrew-srtp-ekt-03 (work in progress), July 2007.

   [I-D.ietf-avt-dtls-srtp]
              McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for  Secure
              Real-time Transport Protocol (SRTP)",
              draft-ietf-avt-dtls-srtp-00 (work in progress), July 2007.

   [I-D.ietf-sip-media-security-requirements]
              Wing, D., Fries, S., Tschofenig, H., and F. Audet,
              "Requirements and Analysis of Media Security Key
              Management Protocols",
              draft-ietf-sip-media-security-requirements-00 (work in
              progress), September 2007.

   [I-D.ietf-mmusic-sdp-capability-negotiation]
              Andreasen, F., "SDP Capability Negotiation",
              draft-ietf-mmusic-sdp-capability-negotiation-07 (work in
              progress), October 2007.

   [I-D.ietf-avt-rtp-and-rtcp-mux]
              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.

   [RFC3262]  Rosenberg, J. and H. Schulzrinne, "Reliability of
              Provisional Responses in Session Initiation Protocol
              (SIP)", RFC 3262, June 2002.

   [RFC3546]  Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J.,
              and T. Wright, "Transport Layer Security (TLS)
              Extensions", RFC 3546, June 2003.

   [RFC4916]  Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", RFC 4916, June 2007.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.




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   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC4279]  Eronen, P. and H. Tschofenig, "Pre-Shared Key Ciphersuites
              for Transport Layer Security (TLS)", RFC 4279,
              December 2005.

   [I-D.wing-sipping-srtp-key]
              Wing, D., "Disclosing Secure RTP (SRTP) Session Keys with
              a SIP Event Package", draft-wing-sipping-srtp-key-01 (work
              in progress), July 2007.


Appendix A.  Requirements Analysis

   [I-D.ietf-sip-media-security-requirements] describes security
   requirements for media keying.  This section evaluates this proposal
   with respect to each requirement.

A.1.  Forking and retargeting (R1, R2, R3)

   In this draft, the SDP offer (in the INVITE) is simply an
   advertisement of the capability to do security.  This advertisement
   does not depend on the identity of the communicating peer, so forking
   and retargeting work work when all the endpoints will do SRTP.  When
   a mix of SRTP and non-SRTP endpoints are present, we expect to use
   the SDP capabilities mechanism currently being defined
   [I-D.ietf-mmusic-sdp-capability-negotiation] to transparently
   negotiate security where possible.  Because DTLS establishes a new
   key for each session, only the entity with which the call is finally
   established gets the media encryption keys (R3).

A.2.  Reusage of a Security Context (R4), (R11)

   DTLS allows sessions to be resumed with the 'TLS session resumption'
   functionality.  This feature can be used to lower the amount of
   cryptographic computation that needs to be done when two peers re-
   initiates the communication.

A.3.  Clipping (R5)

   Because the key establishment occurs in the media plane, media need
   not be clipped before the receipt of the SDP answer.







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A.4.  Passive Attacks on the Media Path (R6)

   The public key algorithms used by DTLS ciphersuites, such as RSA,
   Diffie-Hellman, and Elliptic Curve Diffie-Hellman, are secure against
   passive attacks.

A.5.  Passive Attacks on the Signaling Path (R7)

   DTLS provides protection against passive attacks by adversaries on
   the signaling path since only a fingerprint is exchanged using SIP
   signaling.

A.6.  Perfect Forward Secrecy (R8)

   DTLS supports Diffie-Hellman and Elliptic Curve Diffie-Hellman cipher
   suites which provide PFS.

A.7.  Algorithm Negotiation (R9)

   DTLS negotiates cipher suites before performing significant
   cryptographic computation and therefore supports algorithm
   negotiation and multiple cipher suites without additional
   computational expense.

A.8.  RTP Validity Check (R10)

   TBD

A.9.  3rd Party Certificates (R12, R18)

   Third party certificates are not required.  However, if the parties
   share an authentication infrastructure that is compatible with TLS
   (3rd party certificates or shared keys) it can be used.

A.10.  FIPS 140-2 (R13)

   TLS implementations already may be FIPS 140-2 approved and the
   algorithms used here are consistent with the approval of DTLS and
   DTLS-SRTP.

A.11.  Linkage between Keying Exchange and SIP Signaling (R14)

   The signaling exchange is linked to the key management exchange using
   the fingerprints carried in SIP and the certificates are exchanged in
   DTLS.






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A.12.  Start with RTP and Upgrade to SRTP (R15)

   DTLS-SRTP as described in this framework does not require an SRTP
   security context to be established as part of the initial
   communication setup.  Instead, the DTLS handshake can be initiated
   later during on ongoing session.

A.13.  Denial of Service Vulnerability (R16)

   DTLS offers some degree of DoS protection particuarly as a built-in
   feature.

A.14.  Adversary Model (R17)

   DTLS-SRTP requires that an adversary is at least able to intercept
   the fingerprint exchange along the SIP signaling path (i.e., active
   attack) and to intercept the DTLS handshake by acting as a man-in-
   the-middle adversary (i.e., active attack).

A.15.  Crypto-Agility (R19)

   DTLS allows ciphersuites to be negotiated and hence new algorithms
   can be incrementally deployed.  Work on replacing the fixed MD5/SHA-1
   key derivation function is ongoing.

A.16.  Downgrading Protection (R20)

   DTLS provides protection against downgrading attacks since the
   selection of the offered ciphersuites is confirmed in a later stage
   of the handshake.  This protection is efficient unless an adversary
   is able to break a ciphersuite in real-time.

A.17.  Media Security Negotation (R21)

   DTLS allows a User Agent to negotiate media security parameters for
   each individual session.

A.18.  Signaling Protocol Independence (R22)

   The DTLS-SRTP framework does not rely on SIP; every protocol that is
   capable of exchanging a fingerprint and the media description can be
   secured.

A.19.  Media Recording (R23)

   An extension, see [I-D.wing-sipping-srtp-key], has been specified to
   support media recording that does not require intermediaries to act
   as a MITM.



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   When media recording is done by intermediaries then they need to act
   as a MITM.

A.20.  Lawful Interception (R24)

   Lawful interception requires an active MITM who is located along the
   signaling and the data path.

A.21.  Interworking with Intermediaries (R25)

   A description of the interworking with Session Border Controllers is
   described in this document.

A.22.  PSTN Gateway Termination (R26)

   The DTLS-SRTP framework allows the media security to terminate at a
   PSTN gateway.  [Editor's Note:  A detailed description will be
   provided in a future version of this document.]


Authors' Addresses

   Jason Fischl
   CounterPath Solutions, Inc.
   Suite 300, One Bentall Centre, 505 Burrard Street
   Vancouver, BC  V7X 1M3
   Canada

   Phone:  +1 604 320-3340
   Email:  jason@counterpath.com


   Hannes Tschofenig
   Nokia Siemens Networks
   Otto-Hahn-Ring 6
   Munich, Bavaria  81739
   Germany

   Email:  Hannes.Tschofenig@nsn.com
   URI:    http://www.tschofenig.com











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   Eric Rescorla
   Network Resonance
   2483 E. Bayshore #212
   Palo Alto, CA  94303
   USA

   Email:  ekr@networkresonance.com












































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Full Copyright Statement

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