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PROPOSED STANDARD
Errata Exist
Internet Engineering Task Force (IETF)                         J. Fischl
Request for Comments: 5763                                   Skype, Inc.
Category: Standards Track                                  H. Tschofenig
ISSN: 2070-1721                                   Nokia Siemens Networks
                                                             E. Rescorla
                                                              RTFM, Inc.
                                                                May 2010


Framework for Establishing a Secure Real-time Transport Protocol (SRTP)
    Security Context Using Datagram Transport Layer Security (DTLS)

Abstract

   This document specifies how to use the Session Initiation Protocol
   (SIP) to establish a 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
   the key that will be presented during the DTLS handshake.  The key
   exchange travels along the media path as opposed to the signaling
   path.  The SIP Identity mechanism can be used to protect the
   integrity of the fingerprint attribute from modification by
   intermediate proxies.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc5763.













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Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
   modifications of such material outside the IETF Standards Process.
   Without obtaining an adequate license from the person(s) controlling
   the copyright in such materials, this document may not be modified
   outside the IETF Standards Process, and derivative works of it may
   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.

Table of Contents

   1. Introduction ....................................................4
   2. Overview ........................................................5
   3. Motivation ......................................................7
   4. Terminology .....................................................8
   5. Establishing a Secure Channel ...................................8
   6. Miscellaneous Considerations ...................................10
      6.1. Anonymous Calls ...........................................10
      6.2. Early Media ...............................................11
      6.3. Forking ...................................................11
      6.4. Delayed Offer Calls .......................................11
      6.5. Multiple Associations .....................................11
      6.6. Session Modification ......................................12
      6.7. Middlebox Interaction .....................................12
           6.7.1. ICE Interaction ....................................12
           6.7.2. Latching Control without ICE .......................13
      6.8. Rekeying ..................................................13
      6.9. Conference Servers and Shared Encryptions Contexts ........13
      6.10. Media over SRTP ..........................................14
      6.11. Best Effort Encryption ...................................14



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   7. Example Message Flow ...........................................14
      7.1. Basic Message Flow with Early Media and SIP Identity ......14
      7.2. Basic Message Flow with Connected Identity (RFC 4916) .....19
      7.3. Basic Message Flow with STUN Check for NAT Case ...........23
   8. Security Considerations ........................................25
      8.1. Responder Identity ........................................25
      8.2. SIPS ......................................................26
      8.3. S/MIME ....................................................26
      8.4. Continuity of Authentication ..............................26
      8.5. Short Authentication String ...............................27
      8.6. Limits of Identity Assertions .............................27
      8.7. Third-Party Certificates ..................................29
      8.8. Perfect Forward Secrecy ...................................29
   9. Acknowledgments ................................................29
   10. References ....................................................30
      10.1. Normative References .....................................30
      10.2. Informative References ...................................31
   Appendix A.  Requirements Analysis ................................33
      A.1.  Forking and Retargeting (R-FORK-RETARGET,
            R-BEST-SECURE, R-DISTINCT) ...............................33
      A.2.  Distinct Cryptographic Contexts (R-DISTINCT) .............33
      A.3.  Reusage of a Security Context (R-REUSE) ..................33
      A.4.  Clipping (R-AVOID-CLIPPING) ..............................33
      A.5.  Passive Attacks on the Media Path (R-PASS-MEDIA) .........33
      A.6.  Passive Attacks on the Signaling Path (R-PASS-SIG) .......34
      A.7.  (R-SIG-MEDIA, R-ACT-ACT) .................................34
      A.8.  Binding to Identifiers (R-ID-BINDING) ....................34
      A.9.  Perfect Forward Secrecy (R-PFS) ..........................34
      A.10. Algorithm Negotiation (R-COMPUTE) ........................35
      A.11. RTP Validity Check (R-RTP-VALID) .........................35
      A.12. Third-Party Certificates (R-CERTS, R-EXISTING) ...........35
      A.13. FIPS 140-2 (R-FIPS) ......................................35
      A.14. Linkage between Keying Exchange and SIP Signaling
            (R-ASSOC) ................................................35
      A.15. Denial-of-Service Vulnerability (R-DOS) ..................35
      A.16. Crypto-Agility (R-AGILITY) ...............................35
      A.17. Downgrading Protection (R-DOWNGRADE) .....................36
      A.18. Media Security Negotiation (R-NEGOTIATE) .................36
      A.19. Signaling Protocol Independence (R-OTHER-SIGNALING) ......36
      A.20. Media Recording (R-RECORDING) ............................36
      A.21. Interworking with Intermediaries (R-TRANSCODER) ..........36
      A.22. PSTN Gateway Termination (R-PSTN) ........................36
      A.23. R-ALLOW-RTP ..............................................36
      A.24. R-HERFP ..................................................37







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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 Transport Protocol (RTP) [RFC3550] is used
   to transmit real-time media on top of UDP and TCP [RFC4571].
   Datagram TLS [RFC4347] was introduced to allow TLS functionality to
   be applied to datagram transport protocols, such as UDP and DCCP.
   This document provides guidelines on how to establish SRTP [RFC3711]
   security over UDP using an extension to DTLS (see [RFC5764]).

   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 that are signed by a well-
   known certificate authority to all devices.

   The media is transported over a mutually authenticated DTLS session
   where both sides have certificates.  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 if the peers have them (thus reducing the need to trust
   intermediaries).  The certificate fingerprints are sent in SDP over
   SIP as part of the offer/answer exchange.

   The fingerprint mechanism allows one side of the connection to verify
   that the certificate presented in the DTLS handshake matches the
   certificate used by the party in the signaling.  However, this
   requires some form of integrity protection on the signaling.  S/MIME
   signatures, as described in RFC 3261, or SIP Identity, as described
   in [RFC4474], provide the highest level of security because they are
   not susceptible to modification by malicious intermediaries.
   However, even hop-by-hop security, such as provided by SIPS, offers
   some protection against modification by attackers who are not in
   control of on-path signaling elements.  Because DTLS-SRTP only
   requires message integrity and not confidentiality for the signaling,
   the number of elements that must have credentials and be trusted is
   significantly reduced.  In particular, if RFC 4474 is used, only the
   Authentication Service need have a certificate and be trusted.
   Intermediate elements cannot undetectably modify the message and



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   therefore cannot mount a man-in-the-middle (MITM) attack.  By
   comparison, because SDESCRIPTIONS [RFC4568] requires confidentiality
   for the signaling, all intermediate elements must be trusted.

   This approach differs from previous attempts to secure media traffic
   where the authentication and key exchange protocol (e.g., Multimedia
   Internet KEYing (MIKEY) [RFC3830]) is piggybacked in the signaling
   message exchange.  With DTLS-SRTP, establishing the protection of the
   media traffic between the endpoints is done by the media endpoints
   with only a cryptographic binding of the media keying to 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
   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,
   rekeyed, 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.
















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

     Legend:
     ------>: Signaling Traffic
     <-+-+->: Key Management 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 endpoints
   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 binds
   the DTLS key exchange in the media plane to the signaling plane.

   The fingerprint alone protects against active attacks on the media
   but not active attacks on the signaling.  In order to prevent active
   attacks on the signaling, "Enhancements for Authenticated Identity
   Management in the Session Initiation Protocol (SIP)" [RFC4474] may be
   used.  When Bob receives the offer, the peers establish some number
   of DTLS connections (depending on the number of media sessions) with
   mutual DTLS authentication (i.e., both sides provide certificates).
   At this point, Bob can verify that Alice's credentials offered in TLS
   match the fingerprint in the SDP offer, and Bob can begin sending



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   media to Alice.  Once Bob accepts Alice's offer and sends an SDP
   answer to Alice, Alice can begin sending confidential media to Bob
   over the appropriate streams.  Alice and Bob will verify that the
   fingerprints from the certificates received over the DTLS handshakes
   match with the fingerprints received in the SDP of the SIP signaling.
   This provides the security property that Alice knows that the media
   traffic is going to Bob and vice versa without necessarily requiring
   global Public Key Infrastructure (PKI) certificates for Alice and
   Bob.  (See Section 8 for detailed security analysis.)

3.  Motivation

   Although there is already prior work in this area (e.g., Security
   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 Provisional Response ACKnowledgement (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  When RFC 4474 is used, this solution works even when the SIP
      proxies downstream of the authentication service are not trusted.
      There is no need to reveal keys in the SIP signaling or in the SDP
      message exchange, as is done in SDESCRIPTIONS [RFC4568].
      Retargeting of a dialog-forming request (changing the value of the
      Request-URI), the User Agent (UA) that receives it (the User Agent
      Server, UAS) can have a different identity from that in the To
      header field.  When RFC 4916 is used, then it is possible to
      supply its identity to the peer UA by means of a request in the
      reverse direction, and for that identity to be signed by an
      Authentication Service.

   o  In this method, synchronization source (SSRC) collisions do not
      result in any extra SIP signaling.






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   o  Many SIP endpoints already implement TLS.  The changes to existing
      SIP and RTP usage are minimal even when DTLS-SRTP [RFC5764] is
      used.

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 cipher
   suite and keying material set that is associated with a single host/
   port quartet.  The same DTLS/TLS session can be used to establish the
   keying material for multiple associations.  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.  Establishing a Secure Channel

   The two endpoints in the exchange present their identities as part of
   the DTLS handshake procedure using certificates.  This document uses
   certificates in the same style as described in "Connection-Oriented
   Media Transport over the Transport Layer Security (TLS) Protocol in
   the Session Description Protocol (SDP)" [RFC4572].

   If self-signed certificates are used, the content of the
   subjectAltName attribute inside the certificate MAY use the uniform
   resource identifier (URI) of the user.  This is useful for debugging
   purposes only and is not required to bind the certificate to one of
   the communication endpoints.  The integrity of the certificate is
   ensured through the fingerprint attribute in the SDP.  The
   subjectAltName is not an important component of the certificate
   verification.

   The generation of public/private key pairs is relatively expensive.
   Endpoints are not required to generate certificates for each session.

   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



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   [RFC4566] message, each media description ("m=" line and associated
   parameters) will also contain several attributes as specified in
   [RFC5764], [RFC4145], and [RFC4572].

   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 endpoint SHOULD
   send the SIP message containing the offer to the offerer's SIP proxy
   over an integrity protected channel.  The proxy SHOULD add an
   Identity header field according to the procedures outlined in
   [RFC4474].  The SIP message containing the offer SHOULD be sent to
   the offerer's SIP proxy over an integrity protected channel.  When
   the far endpoint receives the SIP message, it can verify the identity
   of the sender using the Identity header field.  Since the Identity
   header field is a digital signature across several SIP header fields,
   in addition to the body 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 DTLS association with
   the offerer.  Alternately, it can indicate in its answer that the
   offerer is to initiate the TLS association.  In either case, mutual
   DTLS certificate-based authentication will be used.  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 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.







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   The offer and answer MUST conform to the following requirements.

   o  The endpoint MUST use the setup attribute defined in [RFC4145].
      The endpoint that is the offerer MUST use the setup attribute
      value of setup:actpass and be prepared to receive a client_hello
      before it receives the answer.  The answerer MUST use either a
      setup attribute value of setup:active or setup:passive.  Note that
      if the answerer uses setup:passive, then the DTLS handshake will
      not begin until the answerer is received, which adds additional
      latency. setup:active allows the answer and the DTLS handshake to
      occur in parallel.  Thus, setup:active is RECOMMENDED.  Whichever
      party is active MUST initiate a DTLS handshake by sending a
      ClientHello over each flow (host/port quartet).

   o  The endpoint MUST NOT use the connection attribute defined in
      [RFC4145].

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

   o  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.

   o  If the fingerprint does not match the hashed certificate, then the
      endpoint MUST tear down the media session immediately.  Note that
      it is permissible to wait until the other side's fingerprint has
      been received before establishing the connection; however, this
      may have undesirable latency effects.

6.  Miscellaneous Considerations

6.1.  Anonymous Calls

   The use of DTLS-SRTP does not provide anonymous calling; however, it
   also does not prevent it.  However, if care is not taken when
   anonymous calling features, such as those described in [RFC3325] or
   [RFC5767] are used, DTLS-SRTP may allow deanonymizing an otherwise
   anonymous call.  When anonymous calls are being made, the following
   procedures SHOULD be used to prevent deanonymization.

   When making anonymous calls, a new self-signed certificate SHOULD be
   used for each call so that the calls cannot 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 8.4.




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   Additionally, note that in networks that deploy [RFC3325], RFC 3325
   requires that the Privacy header field value defined in [RFC3323]
   needs to be set to 'id'.  This 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.  Note that following
   this recommendation is not sufficient to provide anonymization.

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 form a DTLS association with the
   offerer.  The offerer can then securely correlate the SDP answer
   received in the SIP message by comparing the fingerprint in the
   answer to the hashed certificate for each DTLS association.

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.

6.5.  Multiple Associations

   When there are multiple flows (e.g., multiple media streams, non-
   multiplexed RTP and RTCP, etc.) the active side MAY perform the DTLS
   handshakes in any order.  Appendix B of [RFC5764] provides some
   guidance on the performance of parallel DTLS handshakes.  Note that
   if the answerer ends up being active, it may only initiate handshakes
   on some subset of the potential streams (e.g., if audio and video are



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   offered but it only wishes to do audio).  If the offerer ends up
   being active, the complete answer will be received before the offerer
   begins initiating handshakes.

6.6.  Session Modification

   Once an answer is provided to the offerer, either endpoint MAY
   request a session modification that MAY include an updated offer.
   This session modification can be carried in either an INVITE or
   UPDATE request.  The peers can reuse the existing associations if
   they are compatible (i.e., they have the same key fingerprints and
   transport parameters), or establish a new one following the same
   rules are for initial exchanges, tearing down the existing
   association as soon as the offer/answer exchange is completed.  Note
   that if the active/passive status of the endpoints changes, a new
   connection MUST be established.

6.7.  Middlebox Interaction

   There are a number of potentially bad interactions between DTLS-SRTP
   and middleboxes, as documented in [MMUSIC-MEDIA], which also provides
   recommendations for avoiding such problems.

6.7.1.  ICE Interaction

   Interactive Connectivity Establishment (ICE), as specified in
   [RFC5245], provides a methodology of allowing participants in
   multimedia sessions to verify mutual connectivity.  When ICE is being
   used, the ICE connectivity checks are performed before the DTLS
   handshake begins.  Note that if aggressive nomination mode is used,
   multiple candidate pairs may be marked valid before ICE finally
   converges on a single candidate pair.  Implementations MUST treat all
   ICE candidate pairs associated with a single component as part of the
   same DTLS association.  Thus, there will be only one DTLS handshake
   even if there are multiple valid candidate pairs.  Note that this may
   mean adjusting the endpoint IP addresses if the selected candidate
   pair shifts, just as if the DTLS packets were an ordinary media
   stream.

   Note that Simple Traversal of the UDP Protocol through NAT (STUN)
   packets are sent directly over UDP, not over DTLS.  [RFC5764]
   describes how to demultiplex STUN packets from DTLS packets and SRTP
   packets.








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6.7.2.  Latching Control without ICE

   If ICE is not being used, then there is potential for a bad
   interaction with Session Border Controllers (SBCs) via "latching", as
   described in [MMUSIC-MEDIA].  In order to avoid this issue, if ICE is
   not being used and the DTLS handshake has not completed upon
   receiving the other side's SDP, then the passive side MUST do a
   single unauthenticated STUN [RFC5389] connectivity check in order to
   open up the appropriate pinhole.  All implementations MUST be
   prepared to answer this request during the handshake period even if
   they do not otherwise do ICE.  However, the active side MUST proceed
   with the DTLS handshake as appropriate even if no such STUN check is
   received and the passive MUST NOT wait for a STUN answer before
   sending its ServerHello.

6.8.  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
   [RFC5764].

6.9.  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 because each DTLS handshake establishes fresh keys that
   are not completely under the control of either side.  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 [SRTP-EKT] or [KEY-TRANSPORT], could be
   used to provide this functionality in concert with the mechanisms
   described in this specification.





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RFC 5763                   DTLS-SRTP Framework                  May 2010


6.10.  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 [RFC5764] 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
   provide another important motivation for using DTLS-SRTP instead of
   RTP over DTLS.  Implementations of this specification MUST support
   DTLS-SRTP [RFC5764].

6.11.  Best Effort Encryption

   [RFC5479] describes a requirement for best-effort encryption where
   SRTP is used and where both endpoints support it and key negotiation
   succeeds, otherwise RTP is used.

   [MMUSIC-SDP] describes a mechanism that can signal both RTP and SRTP
   as an alternative.  This allows an offerer to express a preference
   for SRTP, but RTP is the default and will be understood by endpoints
   that do not understand SRTP or this key exchange mechanism.
   Implementations of this document MUST support [MMUSIC-SDP].

7.  Example Message Flow

   Prior to establishing the session, both Alice and Bob generate self-
   signed certificates that 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
   proxy.

7.1.  Basic Message Flow with Early Media and SIP Identity

   This 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 Alice receives the SIP 200 response and once
   Alice has sent the DTLS finished message.







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   The SIP signaling from Alice to her proxy is transported over TLS to
   ensure an integrity protected channel between Alice and her identity
   service.  Transport between proxies should also be protected somehow,
   especially if SIP Identity is not in use.

   Alice            Proxies             Bob
     |(1) INVITE       |                  |
     |---------------->|                  |
     |                 |(2) INVITE        |
     |                 |----------------->|
     |                 |(3) hello         |
     |<-----------------------------------|
     |(4) hello        |                  |
     |----------------------------------->|
     |                 |(5) finished      |
     |<-----------------------------------|
     |                 |(6) media         |
     |<-----------------------------------|
     |(7) finished     |                  |
     |----------------------------------->|
     |                 |(8)  200 OK       |
     |                 <------------------|
     |(9)  200 OK      |                  |
     |<----------------|                  |
     |                 |(10) media        |
     |<---------------------------------->|
     |(11) 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 that acts as Alice's identity service.
      Alice has requested to be either the active or passive endpoint by
      specifying a=setup:actpass in the SDP.  Bob chooses to act as the
      DTLS client and will initiate the session.  Also note that there
      is a fingerprint attribute in the SDP.  This is computed from
      Alice's self-signed certificate.

      This offer includes 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
      [MMUSIC-SDP] for more details on the details of SDP capability
      negotiation.






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RFC 5763                   DTLS-SRTP Framework                  May 2010


   INVITE sip:bob@example.com SIP/2.0
   To: <sip:bob@example.com>
   From: "Alice"<sip:alice@example.com>;tag=843c7b0b
   Via: SIP/2.0/TLS ua1.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   Contact: <sip:alice@ua1.example.com>
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE
   Max-Forwards: 70
   Content-Type: application/sdp
   Content-Length: xxxx
   Supported: from-change

   v=0
   o=- 1181923068 1181923196 IN IP4 ua1.example.com
   s=example1
   c=IN IP4 ua1.example.com
   a=setup:actpass
   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 (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.


















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   INVITE sip:bob@ua2.example.com SIP/2.0
   To: <sip:bob@example.com>
   From: "Alice"<sip:alice@example.com>;tag=843c7b0b
   Via: SIP/2.0/TLS proxy.example.com;branch=z9hG4bK-0e53sadfkasldk
   Via: SIP/2.0/TLS ua1.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   Record-Route: <sip:proxy.example.com;lr>
   Contact: <sip:alice@ua1.example.com>
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE
   Max-Forwards: 69
   Identity: CyI4+nAkHrH3ntmaxgr01TMxTmtjP7MASwliNRdupRI1vpkXRvZXx1ja9k
             3W+v1PDsy32MaqZi0M5WfEkXxbgTnPYW0jIoK8HMyY1VT7egt0kk4XrKFC
             HYWGCl0nB2sNsM9CG4hq+YJZTMaSROoMUBhikVIjnQ8ykeD6UXNOyfI=
   Identity-Info: https://example.com/cert
   Content-Type: application/sdp
   Content-Length: xxxx
   Supported: from-change

   v=0
   o=- 1181923068 1181923196 IN IP4 ua1.example.com
   s=example1
   c=IN IP4 ua1.example.com
   a=setup:actpass
   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, Line 3 shows Bob sending
      a DTLS ClientHello(s) directly to Alice.  In this case, two DTLS
      ClientHello messages would be sent to Alice: one to
      ua1.example.com:6056 for RTP and another to port 6057 for RTCP,
      but only one arrow is drawn for compactness of the figure.

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

      Alice sends back a ServerHello, Certificate, and 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 [RFC5761] were being used only a single
      association would be required.




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   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 via the UA user interface.

   Message (7):  Finished Alice -> Bob

      After Message 7 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 that
      contains the fingerprint for Bob's certificate.  Bob signals the
      actual transport protocol configuration of SRTP over DTLS in the
      acfg parameter.

   SIP/2.0 200 OK
   To: <sip:bob@example.com>;tag=6418913922105372816
   From: "Alice" <sip:alice@example.com>;tag=843c7b0b
   Via: SIP/2.0/TLS proxy.example.com:5061;branch=z9hG4bK-0e53sadfkasldk
   Via: SIP/2.0/TLS ua1.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   Record-Route: <sip:proxy.example.com;lr>
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 1 INVITE
   Contact: <sip:bob@ua2.example.com>
   Content-Type: application/sdp
   Content-Length: xxxx
   Supported: from-change













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   v=0
   o=- 6418913922105372816 2105372818 IN IP4 ua2.example.com
   s=example2
   c=IN IP4 ua2.example.com
   a=setup:active
   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=acfg:1 t=1

   Message (9):  200 OK Proxy -> Alice

      Alice receives the message from her proxy and validates the
      certificate presented in Message 7.  The endpoint now shows Alice
      that the call as secured.

   Message (10):  RTP+RTCP Alice -> Bob

      At this point, Alice can also start sending RTP and RTCP to Bob.

   Message (11):  ACK Alice -> Bob

      Finally, Alice sends the SIP ACK to Bob.

7.2.  Basic Message Flow with Connected Identity (RFC 4916)

   The previous example did not show the use of RFC 4916 for connected
   identity.  The following example does:






















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   Alice            Proxies             Bob
     |(1) INVITE       |                  |
     |---------------->|                  |
     |                 |(2) INVITE        |
     |                 |----------------->|
     |                 |(3) hello         |
     |<-----------------------------------|
     |(4) hello        |                  |
     |----------------------------------->|
     |                 |(5) finished      |
     |<-----------------------------------|
     |                 |(6) media         |
     |<-----------------------------------|
     |(7) finished     |                  |
     |----------------------------------->|
     |                 |(8)  200 OK       |
     |<-----------------------------------|
     |(9) ACK          |                  |
     |----------------------------------->|
     |                 |(10)  UPDATE      |
     |                 |<-----------------|
     |(11) UPDATE      |                  |
     |<----------------|                  |
     |(12) 200 OK      |                  |
     |---------------->|                  |
     |                 |(13) 200 OK       |
     |                 |----------------->|
     |                 |(14) media        |
     |<---------------------------------->|

   The first 9 messages of this example are the same as before.
   However, Messages 10-13, performing the RFC 4916 UPDATE, are new.

   Message (10):  UPDATE Bob -> Proxy

      Bob sends an RFC 4916 UPDATE towards Alice.  This update contains
      his fingerprint.  Bob's UPDATE contains the same session
      information that he provided in his 200 OK (Message 8).  Note that
      in principle an UPDATE here can be used to modify session
      parameters.  However, in this case it's being used solely to
      confirm the fingerprint.










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RFC 5763                   DTLS-SRTP Framework                  May 2010


   UPDATE sip:alice@ua1.example.com SIP/2.0
   Via: SIP/2.0/TLS ua2.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   To: "Alice" <sip:alice@example.com>;tag=843c7b0b
   From <sip:bob@example.com>;tag=6418913922105372816
   Route: <sip:proxy.example.com;lr>
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 2 UPDATE
   Contact: <sip:ua2.example.com>
   Content-Type: application/sdp
   Content-Length: xxxx
   Supported: from-change
   Max-Forwards: 70

   v=0
   o=- 6418913922105372816 2105372818 IN IP4 ua2.example.com
   s=example2
   c=IN IP4 ua2.example.com
   a=setup:active
   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=acfg:1 t=1

   Message (11):  UPDATE Proxy -> Alice

      This shows the UPDATE being relayed to Alice from Bob (and Alice's
      proxy).  Note that Bob's proxy has inserted an Identity and
      Identity-Info header.  As above, we only show one element for both
      proxies for purposes of simplification.  Alice verifies the
      identity provided.  (Note: the actual identity signatures here are
      incorrect and provided merely as examples.)



















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   UPDATE sip:alice@ua1.example.com SIP/2.0
   Via: SIP/2.0/TLS proxy.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   Via: SIP/2.0/TLS ua2.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   To: "Alice" <sip:alice@example.com>;tag=843c7b0b
   From <sip:bob@example.com>;tag=6418913922105372816
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 2 UPDATE
   Contact: <sip:bob@ua2.example.com>
   Content-Type: application/sdp
   Content-Length: xxxx
   Supported: from-change
   Max-Forwards: 69
   Identity: CyI4+nAkHrH3ntmaxgr01TMxTmtjP7MASwliNRdupRI1vpkXRvZXx1ja9k
             3W+v1PDsy32MaqZi0M5WfEkXxbgTnPYW0jIoK8HMyY1VT7egt0kk4XrKFC
             HYWGCl0nB2sNsM9CG4hq+YJZTMaSROoMUBhikVIjnQ8ykeD6UXNOyfI=
   Identity-Info: https://example.com/cert

   v=0
   o=- 6418913922105372816 2105372818 IN IP4 ua2.example.com
   s=example2
   c=IN IP4 ua2.example.com
   a=setup:active
   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=acfg:1 t=1

   Message (12):  200 OK Alice -> Bob

      This shows Alice's 200 OK response to Bob's UPDATE.  Because Bob
      has merely sent the same session parameters he sent in his 200 OK,
      Alice can simply replay her view of the session parameters as
      well.

















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   SIP/2.0 200 OK
   To: "Alice" <sip:alice@example.com>;tag=843c7b0b
   From <sip:bob@example.com>;tag=6418913922105372816
   Via: SIP/2.0/TLS proxy.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   Via: SIP/2.0/TLS ua2.example.com;branch=z9hG4bK-0e53sadfkasldkfj
   Call-ID: 6076913b1c39c212@REVMTEpG
   CSeq: 2 UPDATE
   Contact: <sip:bob@ua2.example.com>
   Content-Type: application/sdp
   Content-Length: xxxx
   Supported: from-change

   v=0
   o=- 1181923068 1181923196 IN IP4 ua2.example.com
   s=example1
   c=IN IP4 ua2.example.com
   a=setup:actpass
   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

7.3.  Basic Message Flow with STUN Check for NAT Case

   In the previous examples, the DTLS handshake has already completed by
   the time Alice receives Bob's 200 OK (8).  Therefore, no STUN check
   is sent.  However, if Alice had a NAT, then Bob's ClientHello might
   get blocked by that NAT, in which case Alice would send the STUN
   check described in Section 6.7.1 upon receiving the 200 OK, as shown
   below:


















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   Alice            Proxies             Bob
     |(1) INVITE       |                  |
     |---------------->|                  |
     |                 |(2) INVITE        |
     |                 |----------------->|
     |                 |(3) hello         |
     |                 X<-----------------|
     |                 |(4)  200 OK       |
     |<-----------------------------------|
     | (5) conn-check  |                  |
     |----------------------------------->|
     |                 |(6) conn-response |
     |<-----------------------------------|
     |                 |(7) hello (rtx)   |
     |<-----------------------------------|
     |(8) hello        |                  |
     |----------------------------------->|
     |                 |(9) finished      |
     |<-----------------------------------|
     |                 |(10) media        |
     |<-----------------------------------|
     |(11) finished    |                  |
     |----------------------------------->|
     |                 |(11) media        |
     |----------------------------------->|
     |(12) ACK         |                  |
     |----------------------------------->|

   The messages here are the same as in the first example (for
   simplicity this example omits an UPDATE), with the following three
   new messages:

   Message (5):  STUN connectivity-check Alice -> Bob

      Section 6.7.1 describes an approach to avoid an SBC interaction
      issue where the endpoints do not support ICE.  Alice (the passive
      endpoint) sends a STUN connectivity check to Bob.  This opens a
      pinhole in Alice's NAT/firewall.

   Message (6):  STUN connectivity-check response Bob -> Alice

      Bob (the active endpoint) sends a response to the STUN
      connectivity check (Message 3) to Alice.  This tells Alice that
      her connectivity check has succeeded and she can stop the
      retransmit state machine.






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   Message (7):  Hello (retransmit) Bob -> Alice

      Bob retransmits his DTLS ClientHello, which now passes through the
      pinhole created in Alice's firewall.  At this point, the DTLS
      handshake proceeds as before.

8.  Security Considerations

   DTLS or TLS media signaled 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
   [RFC5280] certificate.  The client then verifies the certificate and
   checks that the name in the certificate matches the server's domain
   name.  This works because there are a relatively small number of
   servers with well-defined names; a situation that 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 received from the other side, 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.  Other mechanisms, such as the S/MIME mechanism described
   in RFC 3261, or perhaps future mechanisms yet to be defined could
   also serve this purpose.

   While this mechanism can still be used without such integrity
   mechanisms, the security provided is limited to defense against
   passive attack by intermediaries.  An active attack on the signaling
   plus an active attack on the media plane can allow an attacker to
   attack the connection (R-SIG-MEDIA in the notation of [RFC5479]).

8.1.  Responder Identity

   SIP Identity does not support signatures in responses.  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.  [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



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   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.  Offerers SHOULD support this
   mechanism and answerers SHOULD use it.

   In some cases, answerers will not send an UPDATE and in many calls,
   some media will be sent before the UPDATE is received.  In these
   cases, 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 for 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.

8.2.  SIPS

   If SIP Identity is not used, but the signaling is protected by SIPS,
   the security guarantees are weaker.  Some security is still provided
   as long as all proxies are trusted.  This provides integrity for the
   fingerprint in a chain-of-trust security model.  Note, however, that
   if the proxies are not trusted, then the level of security provided
   is limited.

8.3.  S/MIME

   RFC 3261 [RFC3261] defines an S/MIME security mechanism for SIP that
   could be used to sign that the fingerprint was from Bob.  This would
   be secure.

8.4.  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 signaling.

   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.

8.5.  Short Authentication String

   An alternative available to Alice and Bob is to use human speech to
   verify each other's identity and then to verify each other's
   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 [AVT-ZRTP] includes Short Authentication String (SAS) 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.  Based on the level of deployment interest, a TLS extension
   [RFC5246] could provide support for it.  Note that SAS schemes only
   work well when the endpoints recognize each other's voices, which is
   not true in many settings (e.g., call centers).

8.6.  Limits of Identity Assertions

   When RFC 4474 is used to bind the media keying material to the SIP
   signaling, the assurances about the provenance and security of the
   media are only as good as those for the signaling.  There are two
   important cases to note here:

   o  RFC 4474 assumes that the proxy with the certificate "example.com"
      controls the namespace "example.com".  Therefore, the RFC 4474
      authentication service that is authoritative for a given namespace
      can control which user is assigned each name.  Thus, the
      authentication service can take an address formerly assigned to
      Alice and transfer it to Bob.  This is an intentional design
      feature of RFC 4474 and a direct consequence of the SIP namespace
      architecture.






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   o  When phone number URIs (e.g.,
      'sip:+17005551008@chicago.example.com' or
      'sip:+17005551008@chicago.example.com;user=phone') are used, there
      is no structural reason to trust that the domain name is
      authoritative for a given phone number, although individual
      proxies and UAs may have private arrangements that allow them to
      trust other domains.  This is a structural issue in that Public
      Switched Telephone Network (PSTN) elements are trusted to assert
      their phone number correctly and that there is no real concept of
      a given entity being authoritative for some number space.

   In both of these cases, the assurances that DTLS-SRTP provides in
   terms of data origin integrity and confidentiality are necessarily no
   better than SIP provides for signaling integrity when RFC 4474 is
   used.  Implementors should therefore take care not to indicate
   misleading peer identity information in the user interface.  That is,
   if the peer's identity is sip:+17005551008@chicago.example.com, it is
   not sufficient to display that the identity of the peer as
   +17005551008, unless there is some policy that states that the domain
   "chicago.example.com" is trusted to assert the E.164 numbers it is
   asserting.  In cases where the UA can determine that the peer
   identity is clearly an E.164 number, it may be less confusing to
   simply identify the call as encrypted but to an unknown peer.

   In addition, some middleboxes (back-to-back user agents (B2BUAs) and
   Session Border Controllers) are known to modify portions of the SIP
   message that are included in the RFC 4474 signature computation, thus
   breaking the signature.  This sort of man-in-the-middle operation is
   precisely the sort of message modification that RFC 4474 is intended
   to detect.  In cases where the middlebox is itself permitted to
   generate valid RFC 4474 signatures (e.g., it is within the same
   administrative domain as the RFC 4474 authentication service), then
   it may generate a new signature on the modified message.
   Alternately, the middlebox may be able to sign with some other
   identity that it is permitted to assert.  Otherwise, the recipient
   cannot rely on the RFC 4474 Identity assertion and the UA MUST NOT
   indicate to the user that a secure call has been established to the
   claimed identity.  Implementations that are configured to only
   establish secure calls SHOULD terminate the call in this case.

   If SIP Identity or an equivalent mechanism is not used, then only
   protection against attackers who cannot actively change the signaling
   is provided.  While this is still superior to previous mechanisms,
   the security provided is inferior to that provided if integrity is
   provided for the signaling.






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8.7.  Third-Party Certificates

   This specification does not depend on the certificates being held by
   endpoints being independently verifiable (e.g., being issued by a
   trusted third party).  However, there is no limitation on such
   certificates being used.  Aside from the difficulty of obtaining such
   certificates, it is not clear what identities those certificates
   would contain -- RFC 3261 specifies a convention for S/MIME
   certificates that could also be used here, but that has seen only
   minimal deployment.  However, in closed or semi-closed contexts where
   such a convention can be established, third-party certificates can
   reduce the reliance on trusting even proxies in the endpoint's
   domains.

8.8.  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.  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, Jonathan Rosenberg, Rohan Mahy, David
   McGrew, Miguel Garcia, Steffen Fries, Brian Stucker, Robert Gilman,
   David Oran, and Peter Schneider.

   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.













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10.  References

10.1.  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.

   [RFC5280]  Cooper, D., Santesson, S., Farrell, S., Boeyen, S.,
              Housley, R., and W. Polk, "Internet X.509 Public Key
              Infrastructure Certificate and Certificate Revocation List
              (CRL) Profile", RFC 5280, May 2008.

   [RFC3323]  Peterson, J., "A Privacy Mechanism for the Session
              Initiation Protocol (SIP)", RFC 3323, November 2002.

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

   [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.






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   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

10.2.  Informative References

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

   [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.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

   [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.

   [AVT-ZRTP] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Secure RTP", Work in Progress,
              March 2009.

   [SRTP-EKT] McGrew, D., Andreasen, F., and L. Dondeti, "Encrypted Key
              Transport for Secure RTP", Work in Progress, March 2009.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

   [RFC5479]  Wing, D., Fries, S., Tschofenig, H., and F. Audet,
              "Requirements and Analysis of Media Security Management
              Protocols", RFC 5479, March 2009.








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   [MMUSIC-SDP]
              Andreasen, F., "SDP Capability Negotiation", Work
              in Progress, February 2010.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

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

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

   [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.

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

   [SIPPING-SRTP]
              Wing, D., Audet, F., Fries, S., Tschofenig, H., and A.
              Johnston, "Secure Media Recording and Transcoding with the
              Session Initiation Protocol", Work in Progress,
              October 2008.

   [KEY-TRANSPORT]
              Wing, D., "DTLS-SRTP Key Transport (KTR)", Work
              in Progress, March 2009.

   [MMUSIC-MEDIA]
              Stucker, B. and H. Tschofenig, "Analysis of Middlebox
              Interactions for Signaling Protocol Communication along
              the Media Path", Work in Progress, March 2009.

   [RFC5767]  Munakata, M., Schubert, S., and T. Ohba, "User-Agent-
              Driven Privacy Mechanism for SIP", RFC 5767, April 2010.









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Appendix A.  Requirements Analysis

   [RFC5479] describes security requirements for media keying.  This
   section evaluates this proposal with respect to each requirement.

A.1.  Forking and Retargeting (R-FORK-RETARGET, R-BEST-SECURE,
      R-DISTINCT)

   In this document, 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 when all the endpoints will do SRTP.  When a mix
   of SRTP and non-SRTP endpoints are present, we use the SDP
   capabilities mechanism currently being defined [MMUSIC-SDP] 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.  Distinct Cryptographic Contexts (R-DISTINCT)

   DTLS performs a new DTLS handshake with each endpoint, which
   establishes distinct keys and cryptographic contexts for each
   endpoint.

A.3.  Reusage of a Security Context (R-REUSE)

   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-initiate the communication.  See [RFC5764] for more on session
   resumption in this context.

A.4.  Clipping (R-AVOID-CLIPPING)

   Because the key establishment occurs in the media plane, media need
   not be clipped before the receipt of the SDP answer.  Note, however,
   that only confidentiality is provided until the offerer receives the
   answer: the answerer knows that they are not sending data to an
   attacker but the offerer cannot know that they are receiving data
   from the answerer.

A.5.  Passive Attacks on the Media Path (R-PASS-MEDIA)

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





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A.6.  Passive Attacks on the Signaling Path (R-PASS-SIG)

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

A.7.  (R-SIG-MEDIA, R-ACT-ACT)

   An attacker who controls the media channel but not the signaling
   channel can perform a MITM attack on the DTLS handshake but this will
   change the certificates that will cause the fingerprint check to
   fail.  Thus, any successful attack requires that the attacker modify
   the signaling messages to replace the fingerprints.

   If RFC 4474 Identity or an equivalent mechanism is used, an attacker
   who controls the signaling channel at any point between the proxies
   performing the Identity signatures cannot modify the fingerprints
   without invalidating the signature.  Thus, even an attacker who
   controls both signaling and media paths cannot successfully attack
   the media traffic.  Note that the channel between the UA and the
   authentication service MUST be secured and the authentication service
   MUST verify the UA's identity in order for this mechanism to be
   secure.

   Note that an attacker who controls the authentication service can
   impersonate the UA using that authentication service.  This is an
   intended feature of SIP Identity -- the authentication service owns
   the namespace and therefore defines which user has which identity.

A.8.  Binding to Identifiers (R-ID-BINDING)

   When an end-to-end mechanism such as SIP-Identity [RFC4474] and SIP-
   Connected-Identity [RFC4916] or S/MIME are used, they bind the
   endpoint's certificate fingerprints to the From: address in the
   signaling.  The fingerprint is covered by the Identity signature.
   When other mechanisms (e.g., SIPS) are used, then the binding is
   correspondingly weaker.

A.9.  Perfect Forward Secrecy (R-PFS)

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









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A.10.  Algorithm Negotiation (R-COMPUTE)

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

A.11.  RTP Validity Check (R-RTP-VALID)

   DTLS packets do not pass the RTP validity check.  The first byte of a
   DTLS packet is the content type and all current DTLS content types
   have the first two bits set to zero, resulting in a version of zero;
   thus, failing the first validity check.  DTLS packets can also be
   distinguished from STUN packets.  See [RFC5764] for details on
   demultiplexing.

A.12.  Third-Party Certificates (R-CERTS, R-EXISTING)

   Third-party certificates are not required because signaling (e.g.,
   [RFC4474]) is used to authenticate the certificates used by DTLS.
   However, if the parties share an authentication infrastructure that
   is compatible with TLS (third-party certificates or shared keys) it
   can be used.

A.13.  FIPS 140-2 (R-FIPS)

   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.14.  Linkage between Keying Exchange and SIP Signaling (R-ASSOC)

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

A.15.  Denial-of-Service Vulnerability (R-DOS)

   DTLS offers some degree of Denial-of-Service (DoS) protection as a
   built-in feature (see Section 4.2.1 of [RFC4347]).

A.16.  Crypto-Agility (R-AGILITY)

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





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A.17.  Downgrading Protection (R-DOWNGRADE)

   DTLS provides protection against downgrading attacks since the
   selection of the offered cipher suites is confirmed in a later stage
   of the handshake.  This protection is efficient unless an adversary
   is able to break a cipher suite in real-time.  RFC 4474 is able to
   prevent an active attacker on the signaling path from downgrading the
   call from SRTP to RTP.

A.18.  Media Security Negotiation (R-NEGOTIATE)

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

A.19.  Signaling Protocol Independence (R-OTHER-SIGNALING)

   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.20.  Media Recording (R-RECORDING)

   An extension, see [SIPPING-SRTP], has been specified to support media
   recording that does not require intermediaries to act as an MITM.

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

A.21.  Interworking with Intermediaries (R-TRANSCODER)

   In order to interface with any intermediary that transcodes the
   media, the transcoder must have access to the keying material and be
   treated as an endpoint for the purposes of this document.

A.22.  PSTN Gateway Termination (R-PSTN)

   The DTLS-SRTP framework allows the media security to terminate at a
   PSTN gateway.  This does not provide end-to-end security, but is
   consistent with the security goals of this framework because the
   gateway is authorized to speak for the PSTN namespace.

A.23.  R-ALLOW-RTP

   DTLS-SRTP allows RTP media to be received by the calling party until
   SRTP has been negotiated with the answerer, after which SRTP is
   preferred over RTP.





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A.24.  R-HERFP

   The Heterogeneous Error Response Forking Problem (HERFP) is not
   applicable to DTLS-SRTP since the key exchange protocol will be
   executed along the media path and hence error messages are
   communicated along this path and proxies do not need to progress
   them.

Authors' Addresses

   Jason Fischl
   Skype, Inc.
   2145 Hamilton Ave.
   San Jose, CA  95135
   USA

   Phone: +1-415-692-1760
   EMail: jason.fischl@skype.net


   Hannes Tschofenig
   Nokia Siemens Networks
   Linnoitustie 6
   Espoo,   02600
   Finland

   Phone: +358 (50) 4871445
   EMail: Hannes.Tschofenig@gmx.net
   URI:   http://www.tschofenig.priv.at


   Eric Rescorla
   RTFM, Inc.
   2064 Edgewood Drive
   Palo Alto, CA  94303
   USA

   EMail: ekr@rtfm.com













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