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Requirements and Analysis of Media Security Management Protocols
draft-ietf-sip-media-security-requirements-09

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This is an older version of an Internet-Draft that was ultimately published as RFC 5479.
Authors Steffen Fries , Hannes Tschofenig , Dan Wing , Francois Audet
Last updated 2020-01-21 (Latest revision 2009-01-09)
Replaces draft-wing-rtpsec-keying-eval, draft-wing-media-security-requirements
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draft-ietf-sip-media-security-requirements-09
SIP Working Group                                           D. Wing, Ed.
Internet-Draft                                                     Cisco
Intended status: Informational                                  S. Fries
Expires: July 13, 2009                                        Siemens AG
                                                           H. Tschofenig
                                                  Nokia Siemens Networks
                                                                F. Audet
                                                                  Nortel
                                                         January 9, 2009

    Requirements and Analysis of Media Security Management Protocols
             draft-ietf-sip-media-security-requirements-09

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

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   This Internet-Draft will expire on July 13, 2009.

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   Copyright (c) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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Abstract

   This document describes requirements for a protocol to negotiate a
   security context for SIP-signaled SRTP media.  In addition to the
   natural security requirements, this negotiation protocol must
   interoperate well with SIP in certain ways.  A number of proposals
   have been published and a summary of these proposals is in the
   appendix of this document.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Attack Scenarios . . . . . . . . . . . . . . . . . . . . . . .  5
   4.  Call Scenarios and Requirements Considerations . . . . . . . .  8
     4.1.  Clipping Media Before Signaling Answer . . . . . . . . . .  8
     4.2.  Retargeting and Forking  . . . . . . . . . . . . . . . . .  9
     4.3.  Recording  . . . . . . . . . . . . . . . . . . . . . . . . 12
     4.4.  PSTN gateway . . . . . . . . . . . . . . . . . . . . . . . 12
     4.5.  Call Setup Performance . . . . . . . . . . . . . . . . . . 13
     4.6.  Transcoding  . . . . . . . . . . . . . . . . . . . . . . . 13
     4.7.  Upgrading to SRTP  . . . . . . . . . . . . . . . . . . . . 14
     4.8.  Interworking with Other Signaling Protocols  . . . . . . . 14
     4.9.  Certificates . . . . . . . . . . . . . . . . . . . . . . . 15
   5.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 15
     5.1.  Key Management Protocol Requirements . . . . . . . . . . . 15
     5.2.  Security Requirements  . . . . . . . . . . . . . . . . . . 17
     5.3.  Requirements Outside of the Key Management Protocol  . . . 19
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 20
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 20
   8.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 20
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 21
   Appendix A.  Overview and Evaluation of Existing Keying
                Mechanisms  . . . . . . . . . . . . . . . . . . . . . 24
     A.1.  Signaling Path Keying Techniques . . . . . . . . . . . . . 25
       A.1.1.  MIKEY-NULL . . . . . . . . . . . . . . . . . . . . . . 25
       A.1.2.  MIKEY-PSK  . . . . . . . . . . . . . . . . . . . . . . 25
       A.1.3.  MIKEY-RSA  . . . . . . . . . . . . . . . . . . . . . . 26
       A.1.4.  MIKEY-RSA-R  . . . . . . . . . . . . . . . . . . . . . 26
       A.1.5.  MIKEY-DHSIGN . . . . . . . . . . . . . . . . . . . . . 26
       A.1.6.  MIKEY-DHHMAC . . . . . . . . . . . . . . . . . . . . . 26
       A.1.7.  MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC)  . . . . . . . 27
       A.1.8.  Security Descriptions with SIPS  . . . . . . . . . . . 27
       A.1.9.  Security Descriptions with S/MIME  . . . . . . . . . . 27
       A.1.10. SDP-DH (expired) . . . . . . . . . . . . . . . . . . . 27

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       A.1.11. MIKEYv2 in SDP (expired) . . . . . . . . . . . . . . . 27
     A.2.  Media Path Keying Technique  . . . . . . . . . . . . . . . 28
       A.2.1.  ZRTP . . . . . . . . . . . . . . . . . . . . . . . . . 28
     A.3.  Signaling and Media Path Keying Techniques . . . . . . . . 28
       A.3.1.  EKT  . . . . . . . . . . . . . . . . . . . . . . . . . 28
       A.3.2.  DTLS-SRTP  . . . . . . . . . . . . . . . . . . . . . . 29
       A.3.3.  MIKEYv2 Inband (expired) . . . . . . . . . . . . . . . 29
     A.4.  Evaluation Criteria - SIP  . . . . . . . . . . . . . . . . 29
       A.4.1.  Secure Retargeting and Secure Forking  . . . . . . . . 29
       A.4.2.  Clipping Media Before SDP Answer . . . . . . . . . . . 32
       A.4.3.  SSRC and ROC . . . . . . . . . . . . . . . . . . . . . 34
     A.5.  Evaluation Criteria - Security . . . . . . . . . . . . . . 36
       A.5.1.  Distribution and Validation of Persistent Public
               Keys and Certificates  . . . . . . . . . . . . . . . . 36
       A.5.2.  Perfect Forward Secrecy  . . . . . . . . . . . . . . . 38
       A.5.3.  Best Effort Encryption . . . . . . . . . . . . . . . . 40
       A.5.4.  Upgrading Algorithms . . . . . . . . . . . . . . . . . 41
   Appendix B.  Out-of-Scope  . . . . . . . . . . . . . . . . . . . . 43
     B.1.  Shared Key Conferencing  . . . . . . . . . . . . . . . . . 43
   Appendix C.  Requirement renumbering in -02  . . . . . . . . . . . 44
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 46

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1.  Introduction

   The work on media security started when the Session Initiation
   Protocol (SIP) was still in its infancy.  With the increased SIP
   deployment and the availability of new SIP extensions and related
   protocols, the need for end-to-end security was re-evaluated.  The
   procedure of re-evaluating prior protocol work and design decisions
   is not an uncommon strategy and, to some extent, considered necessary
   to ensure that the developed protocols indeed meet the previously
   envisioned needs for the users on the Internet.

   This document summarizes media security requirements, i.e.,
   requirements for mechanisms that negotiate security context such as
   cryptographic keys and parameters for SRTP.

   The organization of this document is as follows: Section 2 introduces
   terminology, Section 3 describes various attack scenarios against the
   signaling path and media path, Section 4 provides an overview about
   possible call scenarios, Section 5 lists requirements for media
   security.  The main part of the document concludes with the security
   considerations Section 6, IANA considerations Section 7 and an
   acknowledgement section in Section 8.  Appendix A lists and compares
   available solution proposals.  The following Appendix A.4 compares
   the different approaches regarding their suitability for the SIP
   signaling scenarios described in Appendix A, while Appendix A.5
   provides a comparison regarding security aspects.  Appendix B lists
   non-goals for this document.

2.  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], with the
   important qualification that, unless otherwise stated, these terms
   apply to the design of the media security key management protocol,
   not its implementation or application.

   Furthermore, the terminology described in SIP ([RFC3261]) regarding
   functions and components are used throughout the document

   Additionally, the following items are used in this document:

   AOR (Address-of-Record):   A SIP or SIPS URI that points to a domain
      with a location service that can map the URI to another URI where
      the user might be available.  Typically, the location service is
      populated through registrations.  An AOR is frequently thought of
      as the "public address" of the user.

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   SSRC:  The 32-bit value that defines the synchronization source, used
      in RTP.  These are generally unique, but collisions can occur.

   two-time pad:  The use of the same key and the same keystream to
      encrypt different data.  For SRTP, a two-time pad occurs if two
      senders are using the same key and the same RTP SSRC value.

   Perfect Forward Secrecy (PFS):  The property that disclosure of the
      long-term secret keying material that is used to derive an agreed
      ephemeral key does not compromise the secrecy of agreed keys from
      earlier runs.

   active adversary:  An active adversary is able to alter data
      communication to affect its operation (see also [RFC4949]).

   passive adversary:  A passive adversary is able to learn information
      from data communication, but not alter that data communication
      (see also[RFC4949]).

   signaling path:  The signaling path is the route taken by SIP
      signaling messages transmitted between the calling and called user
      agents.  This can be either direct signaling between the calling
      and called user agents or, more commonly involves the SIP proxy
      servers that were involved in the call setup.

   media path:  The media path is the route taken by media packets
      exchanged by the endpoints.  In the simplest case, the endpoints
      exchange media directly, and the "media path" is defined by a
      quartet of IP addresses and TCP/UDP ports, along with an IP route.
      In other cases, this path may include RTP relays, mixers,
      transcoders, session border controllers, NATs, or media gateways.

   Moreover, as this document discusses requirements for media security,
   the nomenclature R-XXX is used to mark requrements, were XXX is the
   requirement, which needs to be met.

3.  Attack Scenarios

   The discussion in this section relates to requirements R-PASS-MEDIA,
   R-PASS-SIG, R-ASSOC, R-SIG-MEDIA, R-ACT-ACT, and R-ID-BINDING.

   This document classifies adversaries according to their access and
   their capabilities.  An adversary might have access:

   1.  only to the media path,

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   2.  only to the signaling path,

   3.  to the media path and to the signaling path.

   An attacker that can solely be located along the signaling path, and
   does not have access to media (item 2), is not considered in this
   document.

   There are two different types of adversaries, active and passive.  An
   active adversary may need to be active with regard to the key
   exchange relevant information traveling along the media path or
   traveling along the signaling path.

   Based on their robustness against the adversary capabilities
   described above, we can group security mechanisms using the following
   labels.  This list is generally ordered from easiest to compromise
   (at the top) to more difficult to compromise:

    +---------------+---------+--------------------------------------+
    | SIP signaling |  media  |             abbreviation             |
    +---------------+---------+--------------------------------------+
    |      none     | passive |      no-signaling-passive-media      |
    |      none     |  active |       no-signaling-active-media      |
    |    passive    | passive |    passive-signaling-passive-media   |
    |    passive    |  active |    passive-signaling-active-media    |
    |     active    | passive |    active-signaling-passive-media    |
    |     active    |  active |     active-signaling-active-media    |
    |     active    |  active | active-signaling-active-media-detect |
    +---------------+---------+--------------------------------------+

   no-signaling-passive-media:
      Access to only the media path is sufficient to reveal the content
      of the media traffic.

   passive-signaling-passive-media:
      Passive attack on the signaling and passive attack on the media
      path is necessary to reveal the content of the media traffic.

   passive-signaling-active-media:
      Passive attack on the signaling and active attack on the media
      path is necessary to reveal the content of the media traffic.

   active-signaling-passive-media:
      Active attack on the signaling path and passive attack on the
      media path is necessary to reveal the content of the media
      traffic.

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   no-signaling-active-media:
      Active attack on the media path is sufficient to reveal the
      content of the media traffic.

   active-signaling-active-media:
      Active attack on both the signaling path and the media path is
      necessary to reveal the content of the media traffic.

   active-signaling-active-media-detect:
      Active attack on both signaling and media path is necessary to
      reveal the content of the media traffic (as with active-signaling-
      active-media), and the attack is detectable by protocol messages
      exchanged between the end points.

   For example, unencrypted RTP is vulnerable to no-signaling-passive-
   media.

   As another example, Security Descriptions [RFC4568], when protected
   by TLS (as it is commonly implemented and deployed), belongs in the
   passive-signaling-passive-media category since the adversary needs to
   learn the Security Descriptions key by seeing the SIP signaling
   message at a SIP proxy (assuming that the adversary is in control of
   the SIP proxy).  The media traffic can be decrypted using that
   learned key.

   As another example, DTLS-SRTP falls into active-signaling-active-
   media category when DTLS-SRTP is used with a public key based
   ciphersuite with self-signed certificates and without SIP-Identity
   [RFC4474].  An adversary would have to modify the fingerprint that is
   sent along the signaling path and subsequently to modify the
   certificates carried in the DTLS handshake that travel along the
   media path.  If DTLS-SRTP is used with both SIP Identity [RFC4474]
   and SIP Connected Identity [RFC4916], the RFC4474 signature protects
   both the offer and the answer, and such a system would then belong to
   the active-signaling-active-attack-detect category (provided, of
   course, the signaling path to the RFC4474 authenticator and verifier
   is secured as per RFC4474 and the RFC4474 authenticator and verifier
   are behaving as per RFC4474).

   The above discussion of DTLS-SRTP demonstrates how a single security
   protocol can be in different classes depending on the mode in which
   it is operated.  Other protocols can achieve similar effect by adding
   functions outside of the on-the-wire key management protocol itself.
   Although it may be appropriate to deploy lower-classed mechanisms in
   some cases, the ultimate security requirement for a media security
   negotiation protocol is that it have a mode of operation available in
   which is detect-attack, which provides protection against the passive
   and active attacks and provides detection of such attacks.  That is,

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   there must be a way to use the protocol so that an active attack is
   required against both the signaling and media paths, and so that such
   attacks are detectable by the endpoints.

4.  Call Scenarios and Requirements Considerations

   The following subsections describe call scenarios that pose the most
   challenge to the key management system for media data in cooperation
   with SIP signaling.

   Throughout the subsections requirements are stated by using the
   nomenclature R- to state an explicit requirement.  All of the stated
   requirements are explanied in detail in section Section 5.  The
   requirements in section Section 5 are listed according their
   association to the key management protocol, to attack scenarios, and
   requirements which can be met inside the key management protocol or
   outside of the key management protocol.

4.1.  Clipping Media Before Signaling Answer

   The discussion in this section relates to requirement R-AVOID-
   CLIPPING and R-ALLOW-RTP.

   Per the SDP Offer/Answer Model [RFC3264],

      "Once the offerer has sent the offer, it MUST be prepared to
      receive media for any recvonly streams described by that offer.
      It MUST be prepared to send and receive media for any sendrecv
      streams in the offer, and send media for any sendonly streams in
      the offer (of course, it cannot actually send until the peer
      provides an answer with the needed address and port information)."

   To meet this requirement with SRTP, the offerer needs to know the
   SRTP key for arriving media.  If either endpoint receives encrypted
   media before it has access to the associated SRTP key, it cannot play
   the media -- causing clipping.

   For key exchange mechanisms that send the answerer's key in SDP, a
   SIP provisional response [RFC3261], such as 183 (session progress),
   is useful.  However, the 183 messages are not reliable unless both
   the calling and called end point support PRACK [RFC3262], use TCP
   across all SIP proxies, implement Security Preconditions [RFC5027],
   or the both ends implement ICE [I-D.ietf-mmusic-ice] and the answerer
   implements the reliable provisional response mechanism described in
   ICE.  Unfortunately, there is not wide deployment of any of these
   techniques and there is industry reluctance to require these
   techniques to avoid the problems described in this section.

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   Note that the receipt of an SDP answer is not always sufficient to
   allow media to be played to the offerer.  Sometimes, the offerer must
   send media in order to open up firewall holes or NAT bindings before
   media can be received (for details see
   [I-D.ietf-mmusic-media-path-middleboxes]).  In this case, even a
   solution that makes the key available before the SDP answer arrives
   will not help.

   Preventing the arrival of early media (i.e., media that arrives at
   the SDP offerer before the SDP answer arrives) might obsolete the
   R-AVOID-CLIPPING requirement, but at the time of writing such early
   media exists in many normal call scenarios.

4.2.  Retargeting and Forking

   The discussion in this section relates to requirements R-FORK-
   RETARGET, R-DISTINCT, R-HERFP, and R-BEST-SECURE.

   In SIP, a request sent to a specific AOR but delivered to a different
   AOR is called a "retarget".  A typical scenario is a "call
   forwarding" feature.  In Figure 1 Alice sends an INVITE in step 1
   that is sent to Bob in step 2.  Bob responds with a redirect (SIP
   response code 3xx) pointing to Carol in step 3.  This redirect
   typically does not propagate back to Alice but only goes to a proxy
   (i.e., the retargeting proxy) that sends the original INVITE to Carol
   in step 4.

                                    +-----+
                                    |Alice|
                                    +--+--+
                                       |
                                       | INVITE (1)
                                       V
                                  +----+----+
                                  |  proxy  |
                                  ++-+-----++
                                   | ^     |
                        INVITE (2) | |     | INVITE (4)
                    & redirect (3) | |     |
                                   V |     V
                                  ++-++   ++----+
                                  |Bob|   |Carol|
                                  +---+   +-----+

                           Figure 1: Retargeting

   Using retargeting might lead to situations where the User Agent

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   Client (UAC) does not know where its request will be going.  This
   might not immediately seem like a serious problem; after all, when
   one places a telephone call on the PSTN, one never really knows if it
   will be forwarded to a different number, who will pick up the line
   when it rings, and so on.  However, when considering SIP mechanisms
   for authenticating the called party, this function can also make it
   difficult to differentiate an intermediary that is behaving
   legitimately from an attacker.  From this perspective, the main
   problems with retargeting are:

   Not detectable by the caller:   The originating user agent has no
      means of anticipating that the condition will arise, nor any means
      of determining that it has occurred until the call has already
      been set up.

   Not preventable by the caller:  There is no existing mechanism that
      might be employed by the originating user agent in order to
      guarantee that the call will not be re-targeted.

   The mechanism used by SIP for identifying the calling party is SIP
   Identity [RFC4474].  However, due to the nature of retargeting SIP
   Identity can only identify the calling party (that is, the party that
   initiated the SIP request).  Some key exchange mechanisms predate SIP
   Identity and include their own identity mechanism (e.g., MIKEY).
   However, those built-in identity mechanism also suffer from the SIP
   retargeting problem.  While Connected Identity [RFC4916] allows
   positive identification of the called party, the primary difficulty
   still remains that the calling party does not know if a mismatched
   called party is legitimate (i.e., due to authorized retargeting) or
   illegitimate (i.e., due to unauthorized retargeting by an attacker
   above to modify SIP signaling).

   In SIP, 'forking' is the delivery of a request to multiple locations.
   This happens when a single AOR is registered more than once.  An
   example of forking is when a user has a desk phone, PC client, and
   mobile handset all registered with the same AOR.

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                                  +-----+
                                  |Alice|
                                  +--+--+
                                     |
                                     | INVITE
                                     V
                               +-----+-----+
                               |   proxy   |
                               ++---------++
                                |         |
                         INVITE |         | INVITE
                                V         V
                             +--+--+   +--+--+
                             |Bob-1|   |Bob-2|
                             +-----+   +-----+

                             Figure 2: Forking

   With forking, both Bob-1 and Bob-2 might send back SDP answers in SIP
   responses.  Alice will see those intermediate (18x) and final (200)
   responses.  It is useful for Alice to be able to associate the SIP
   response with the incoming media stream.  Although this association
   can be done with ICE [I-D.ietf-mmusic-ice], and ICE is useful to make
   this association with RTP, it is not desirable to require ICE to
   accomplish this association.

   Forking and retargeting are often used together.  For example, a boss
   and secretary might have both phones ring (forking) and rollover to
   voice mail if neither phone is answered (retargeting).

   To maintain security of the media traffic, only the end point that
   answers the call should know the SRTP keys for the session.  Forked
   and re-targeted calls only reveal sensitive information to non-
   responders when the signaling messages contain sensitive information
   (e.g., SRTP keys) that is accessible by parties that receive the
   offer, but may not respond (i.e., the original recipients in a
   retargeted call, or non-answering endpoints in a forked call).  For
   key exchange mechanisms that do not provide secure forking or secure
   retargeting, one workaround is to re-key immediately after forking or
   retargeting.  However, because the originator may not be aware that
   the call forked this mechanism requires rekeying immediately after
   every session is established.  This doubles the number of messages
   processed by the network.

   Further compounding this problem is a unique feature of SIP that when
   forking is used, there is always only one final error response
   delivered to the sender of the request: the forking proxy is
   responsible for choosing which final response to choose in the event

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   where forking results in multiple final error responses being
   received by the forking proxy.  This means that if a request is
   rejected, say with information that the keying information was
   rejected and providing the far end's credentials, it is very possible
   that the rejection will never reach the sender.  This problem, called
   the Heterogeneous Error Response Forking Problem (HERFP) [RFC3326],
   is difficult to solve in SIP.  Because we expect the HERFP to
   continue to be a problem in SIP for the foreseeable future, a media
   security system should function even in the presence of HERFP
   behavior.

4.3.  Recording

   The discussion in this section relates to requirement R-RECORDING.

   Some business environments, such as stock brokers, banks, and catalog
   call centers, require recording calls with customers.  This is the
   familiar "this call is being recorded for quality purposes" heard
   during calls to these sorts of businesses.  In these environments,
   media recording is typically performed by an intermediate device
   (with RTP, this is typically implemented in a 'sniffer').

   When performing such call recording with SRTP, the end-to-end
   security is compromised.  This is unavoidable, but necessary because
   the operation of the business requires such recording.  It is
   desirable that the media security is not unduly compromised by the
   media recording.  The endpoint within the organization needs to be
   informed that there is an intermediate device and needs to cooperate
   with that intermediate device.

   This scenario does not place a requirement directly on the key
   management protocol.  The requirement could be met directly by the
   key management protocol (e.g., MIKEY-NULL or [RFC4568]) or through an
   external out-of-band-mechanism (e.g., [I-D.wing-sipping-srtp-key]).

4.4.  PSTN gateway

   The discussion in this section relates to requirement R-PSTN.

   It is desirable, even when one leg of a call is on the PSTN, that the
   IP leg of the call be protected with SRTP.

   A typical case of using media security where two entities are having
   a VoIP conversation over IP capable networks.  However, there are
   cases where the other end of the communication is not connected to an
   IP capable network.  In this kind of setting, there needs to be some
   kind of gateway at the edge of the IP network which converts the VoIP
   conversation to format understood by the other network.  An example

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   of such gateway is a PSTN gateway sitting at the edge of IP and PSTN
   networks (such as the architecture described in [RFC3372]).

   If media security (e.g., SRTP protection) is employed in this kind of
   gateway-setting, then media security and the related key management
   is terminated at the PSTN gateway.  The other network (e.g., PSTN)
   may have its own measures to protect the communication, but this
   means that from media security point of view the media security is
   not employed truely end-to-end between the communicating entities.

4.5.  Call Setup Performance

   The discussion in this section relates to requirement R-REUSE.

   Some devices lack sufficient processing power to perform public key
   operations or Diffie-Hellman operations for each call, or prefer to
   avoid performing those operations on every call.  The ability to re-
   use previous public key or Diffie-Hellman operations can vastly
   decrease the call setup delay and processing requirements for such
   devices.

   In certain devices, it can take a second or two to perform a Diffie-
   Hellman operation.  Examples of these devices include handsets, IP
   Multimedia Services Identity Module (ISIMs), and PSTN gateways.  PSTN
   gateways typically utilize a Digital Signal Processor (DSP) which is
   not yet involved with typical DSP operations at the beginning of a
   call, thus the DSP could be used to perform the calculation, so as to
   avoid having the central host processor perform the calculation.
   However, not all PSTN gateways use DSPs (some have only central
   processors or their DSPs are incapable of performing the necessary
   public key or Diffie-Hellman operation), and handsets lack a
   separate, unused processor to perform these operations.

   Two scenarios where R-REUSE is useful are calls between an endpoint
   and its voicemail server or its PSTN gateway.  In those scenarios
   calls are made relatively often and it can be useful for the
   voicemail server or PSTN gateway to avoid public key operations for
   subsequent calls.

   Storing keys across sessions often interferes with perfect forward
   secrecy (R-PFS).

4.6.  Transcoding

   The discussion in this section relates to requirement R-TRANSCODER.

   In some environments is is necessary for network equipment to
   transcode from one codec (e.g., a highly compressed codec which makes

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   efficient use of wireless bandwidth) to another codec (e.g., a
   standardized codec to a SIP peering interface).  With RTP, a
   transcoding function can be performed with the combination of a SIP
   B2BUA (to modify the SDP) and a processor to perform the transcoding
   between the codecs.  However, with end-to-end secured SRTP, a
   transcoding function implemented the same way is a man in the middle
   attack, and the key management system prevents its use.

   However, such a network-based transcoder can still be realized with
   the cooperation and approval of the endpoint, and can provide end-to-
   transcoder and transcoder-to-end security.

4.7.  Upgrading to SRTP

   The discussion in this section relates to the requirement R-ALLOW-
   RTP.

   Legitimate RTP media can be sent to an endpoint for announcements,
   colorful ringback tones (e.g., music), advertising, or normal call
   progress tones.  The RTP may be received before an associated SDP
   answer.  For details on various scenarios, see
   [I-D.stucker-sipping-early-media-coping].

   While receiving such RTP exposes the calling party to a risk of
   receiving malicious RTP from an attacker, SRTP endpoints will need to
   receive and play out RTP media in order to be compatible with
   deployed systems that send RTP to calling parties.

4.8.  Interworking with Other Signaling Protocols

   The discussion in this section relates to the requirement R-OTHER-
   SIGNALING.

   In many environments, some devices are signaled with protocols other
   than SIP which do not share SIP's offer/answer model (e.g., [H.248.1]
   or do not utilize SDP (e.g., H.323).  In other environments, both
   endpoints may be SIP, but may use different key management systems
   (e.g., one uses MIKEY-RSA, the other MIKEY-RSA-R).

   In these environments, it is desirable to have SRTP -- rather than
   RTP -- between the two endpoints.  It is always possible, although
   undesirable, to interwork those disparate signaling systems or
   disparate key management systems by decrypting and re-encrypting each
   SRTP packet in a device in the middle of the network (often the same
   device performing the signaling interworking).  This is undesirable
   due to the cost and increased attack area, as such an SRTP/SRTP
   interworking device is a valuable attack target.

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   At the time of this writing, interworking is considered important.
   Interworking without decryption/encryption of the SRTP, while useful,
   is not yet deemed critical because the scale of such SRTP deployments
   is, to date, relatively small.

4.9.  Certificates

   The discussion in this section relates to R-CERTS.

   On the Internet and on some private networks, validating another
   peer's certificate is often done through a trust anchor -- a list of
   Certificate Authorities that are trusted.  It can be difficult or
   expensive for a peer to obtain these certificates.  In all cases,
   both parties to the call would need to trust the same trust anchor
   (i.e., "certificate authority").  For these reasons, it is important
   that the media plane key management protocol offer a mechanism that
   allows end-users who have no prior association to authenticate to
   each other without acquiring credentials from a third party trust
   point.  Note that this does not rule out mechanisms in which servers
   have certificates and attest to the identities of end-users.

5.  Requirements

   This section is divided into several parts: requirements specific to
   the key management protocol (Section 5.1), attack scenarios
   (Section 5.2), and requirements which can be met inside the key
   management protocol or outside of the key management protocol
   (Section 5.3).

5.1.  Key Management Protocol Requirements

   SIP Forking and Retargeting, from Section 4.2:

   R-FORK-RETARGET:
         The media security key management protocol MUST securely
         support forking and retargeting when all endpoints are willing
         to use SRTP without causing the call setup to fail.  This
         requirement means the endpoints that did not answer the call
         MUST NOT learn the SRTP keys (in either direction) used by the
         answering endpoint.

   R-DISTINCT:
         The media security key management protocol MUST be capable of
         creating distinct, independent cryptographic contexts for each
         endpoint in a forked session.

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   R-HERFP:
         The media security key management protocol MUST function
         securely even in the presence of HERFP behavior, i.e., the
         rejection of key information does not reach the sender.

   Performance considerations:

   R-REUSE:
         The media security key management protocol MAY support the re-
         use of a previously established security context.

               Note: re-use of the security context does not imply re-
               use of RTP parameters (e.g., payload type or SSRC).

   Media considerations:

   R-AVOID-CLIPPING:
         The media security key management protocol SHOULD avoid
         clipping media before SDP answer without requiring Security
         Preconditions [RFC5027].  This requirement comes from
         Section 4.1.

   R-RTP-CHECK:
         If SRTP key negotiation is performed over the media path (i.e.,
         using the same UDP/TCP ports as media packets), the key
         negotiation packets MUST NOT pass the RTP validity check
         defined in Appendix A.1 of [RFC3550], so that SRTP negotiation
         packets can be differentiated from RTP packets.

   R-ASSOC:
         The media security key management protocol SHOULD include a
         mechanism for associating key management messages with both the
         signaling traffic that initiated the session and with protected
         media traffic.  It is useful to associate key management
         messages with call signaling messages, as this allows the SDP
         offerer to avoid performing CPU-consuming operations (e.g.,
         Diffie-Hellman or public key operations) with attackers that
         have not seen the signaling messages.

         For example, if using a Diffie-Hellman keying technique with
         security preconditions that forks to 20 end points, the call
         initiator would get 20 provisional responses containing 20
         signed Diffie-Hellman key pairs.  Calculating 20 Diffie-Hellman
         secrets and validating signatures can be a difficult task for
         some devices.  Hence, in the case of forking, it is not
         desirable to perform a Diffie-Hellman operation with every
         party, but rather only with the party that answers the call
         (and incur some media clipping).  To do this, the signaling and

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         media need to be associated so the calling party knows which
         key management exchange needs to be completed.  This might be
         done by using the transport address indicated in the SDP,
         although NATs can complicate this association.

               Note: due to RTP's design requirements, it is expected
               that SRTP receivers will have to perform authentication
               of any received SRTP packets.

   R-NEGOTIATE:
         The media security key management protocol MUST allow a SIP
         User Agent to negotiate media security parameters for each
         individual session.  Such negotiation MUST NOT cause a two-time
         pad (Section 9.1 of [RFC3711]).

   R-PSTN:
         The media security key management protocol MUST support
         termination of media security in a PSTN gateway.  This
         requirement is from Section 4.4.

5.2.  Security Requirements

   This section describes overall security requirements and specific
   requirements from the attack scenarios (Section 3).

   Overall security requirements:

   R-PFS:
         The media security key management protocol MUST be able to
         support perfect forward secrecy.

   R-COMPUTE:
         The media security key management protocol MUST support
         offering additional SRTP cipher suites without incurring
         significant computational expense.

   R-CERTS:
         The key management protocol MUST NOT require that end-users
         obtain credentials (certificates or private keys) from a third-
         party trust anchor.

   R-FIPS:
         The media security key management protocol SHOULD use
         algorithms that allow FIPS 140-2 [FIPS-140-2] certification or
         similar country-specific certification (e.g., [AISITSEC]).

         The United States Government can only purchase and use crypto
         implementations that have been validated by the FIPS-140

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         [FIPS-140-2] process:

               "The FIPS-140 standard is applicable to all Federal
               agencies that use cryptographic-based security systems to
               protect sensitive information in computer and
               telecommunication systems, including voice systems.  The
               adoption and use of this standard is available to private
               and commercial organizations."

         Some commercial organizations, such as banks and defense
         contractors, require or prefer equipment which has received the
         same validation.

   R-DOS:
         The media security key management protocol MUST NOT introduce
         any new significant denial of service vulnerabilities (e.g.,
         the protocol should not request the endpoint to perform CPU-
         intensive operations without the client being able to validate
         or authorize the request).

   R-EXISTING:
         The media security key management protocol SHOULD allow
         endpoints to authenticate using pre-existing cryptographic
         credentials, e.g., certificates or pre-shared keys.

   R-AGILITY:
         The media security key management protocol MUST provide crypto-
         agility, i.e., the ability to adapt to evolving cryptography
         and security requirements (update of cryptographic algorithms
         without substantial disruption to deployed implementations)

   R-DOWNGRADE:
         The media security key management protocol MUST protect cipher
         suite negotiation against downgrading attacks.

   R-PASS-MEDIA:
         The media security key management protocol MUST have a mode
         which prevents a passive adversary with access to the media
         path from gaining access to keying material used to protect
         SRTP media packets.

   R-PASS-SIG:
         The media security key management protocol MUST have a mode in
         which it prevents a passive adversary with access to the
         signaling path from gaining access to keying material used to
         protect SRTP media packets.

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   R-SIG-MEDIA:
         The media security key management protocol MUST have a mode in
         which it defends itself from an attacker that is solely on the
         media path and from an attacker that is solely on the signaling
         path.  A successful attack refers to the ability for the
         adversary to obtain keying material to decrypt the SRTP
         encrypted media traffic.

   R-ID-BINDING:
         The media security key management protocol MUST enable the
         media security keys to be cryptographically bound to an
         identity of the endpoint.

               This allows domains to deploy SIP Identity [RFC4474].

   R-ACT-ACT:
         The media security key management protocol MUST support a mode
         of operation that provides active-signaling-active-media-detect
         robustness, and MAY support modes of operation that provide
         lower levels of robustness (as described in Section 3).

               Failing to meet R-ACT-ACT indicates the protocol can not
               provide secure end-to-end media.

5.3.  Requirements Outside of the Key Management Protocol

   The requirements in this section are for an overall VoIP security
   system.  These requirements can be met within the key management
   protocol itself, or can be solved outside of the key management
   protocol itself (e.g., solved in SIP or in SDP).

   R-BEST-SECURE:
         Even when some end points of a forked or retargeted call are
         incapable of using SRTP, a solution MUST be described which
         allows the establishment of SRTP associations with SRTP-capable
         endpoints and / or RTP associations with non-SRTP-capable
         endpoints.

   R-OTHER-SIGNALING:
         A solution SHOULD be able to negotiate keys for SRTP sessions
         created via different call signaling protocols (e.g., between
         Jabber, SIP, H.323, MGCP).

   R-RECORDING:
         A solution SHOULD be described which supports recording of
         decrypted media.  This requirement comes from Section 4.3.

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   R-TRANSCODER:
         A solution SHOULD be described which supports intermediate
         nodes (e.g., transcoders), terminating or processing media,
         between the end points.

   R-ALLOW-RTP:  A solution SHOULD be described which 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.

6.  Security Considerations

   This document lists requirements for securing media traffic.  As
   such, it addresses security throughout the document.

7.  IANA Considerations

   This document does not require actions by IANA.

8.  Acknowledgements

   For contributions to the requirements portion of this document, the
   authors would like to thank the active participants of the RTPSEC BoF
   and on the RTPSEC mailing list, and a special thanks to Steffen Fries
   and Dragan Ignjatic for their excellent MIKEY comparison [RFC5197]
   document.

   The authors would furthermore like to thank the following people for
   their review, suggestions, and comments: Flemming Andreasen, Richard
   Barnes, Mark Baugher, Wolfgang Buecker, Werner Dittmann, Lakshminath
   Dondeti, John Elwell, Martin Euchner, Hans-Heinrich Grusdt, Christer
   Holmberg, Guenther Horn, Peter Howard, Leo Huang, Dragan Ignjatic,
   Cullen Jennings, Alan Johnston, Vesa Lehtovirta, Matt Lepinski, David
   McGrew, David Oran, Colin Perkins, Eric Raymond, Eric Rescorla, Peter
   Schneider, Srinath Thiruvengadam, Dave Ward, Dan York, and Phil
   Zimmermann.

9.  References

9.1.  Normative References

   [FIPS-140-2]
              NIST, "Security Requirements for Cryptographic Modules",
              June 2005, <http://csrc.nist.gov/publications/fips/

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              fips140-2/fips1402.pdf>.

   [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.

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

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

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

9.2.  Informative References

   [AISITSEC]
              "Anwendungshinweise und Interpretationen (AIS) zu ITSEC",
              January 2002,
              <http://www.bsi.de/zertifiz/zert/interpr/aisitsec.htm>.

   [H.248.1]  ITU, "Gateway control protocol", June 2000,
              <http://www.itu.int/rec/T-REC-H.248/e>.

   [I-D.baugher-mmusic-sdp-dh]
              Baugher, M. and D. McGrew, "Diffie-Hellman Exchanges for
              Multimedia Sessions", draft-baugher-mmusic-sdp-dh-00 (work
              in progress), February 2006.

   [I-D.dondeti-msec-rtpsec-mikeyv2]
              Dondeti, L., "MIKEYv2: SRTP Key Management using MIKEY,
              revisited", draft-dondeti-msec-rtpsec-mikeyv2-01 (work in
              progress), March 2007.

   [I-D.fischl-sipping-media-dtls]
              Fischl, J., "Datagram Transport Layer Security (DTLS)
              Protocol for Protection of Media  Traffic Established with
              the Session Initiation Protocol",
              draft-fischl-sipping-media-dtls-03 (work in progress),
              July 2007.

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   [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-06 (work in progress),
              October 2008.

   [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.

   [I-D.ietf-mmusic-media-path-middleboxes]
              Stucker, B. and H. Tschofenig, "Analysis of Middlebox
              Interactions for Signaling Protocol Communication  along
              the Media Path",
              draft-ietf-mmusic-media-path-middleboxes-01 (work in
              progress), July 2008.

   [I-D.ietf-mmusic-sdp-capability-negotiation]
              Andreasen, F., "SDP Capability Negotiation",
              draft-ietf-mmusic-sdp-capability-negotiation-09 (work in
              progress), July 2008.

   [I-D.ietf-msec-mikey-ecc]
              Milne, A., "ECC Algorithms for MIKEY",
              draft-ietf-msec-mikey-ecc-03 (work in progress),
              June 2007.

   [I-D.ietf-sip-certs]
              Jennings, C. and J. Fischl, "Certificate Management
              Service for The Session Initiation Protocol (SIP)",
              draft-ietf-sip-certs-07 (work in progress), November 2008.

   [I-D.ietf-tls-rfc4346-bis]
              Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", draft-ietf-tls-rfc4346-bis-10
              (work in progress), March 2008.

   [I-D.jennings-sipping-multipart]
              Wing, D. and C. Jennings, "Session Initiation Protocol
              (SIP) Offer/Answer with Multipart Alternative",
              draft-jennings-sipping-multipart-02 (work in progress),
              March 2006.

   [I-D.mcgrew-srtp-ekt]
              McGrew, D., "Encrypted Key Transport for Secure RTP",

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              draft-mcgrew-srtp-ekt-03 (work in progress), July 2007.

   [I-D.stucker-sipping-early-media-coping]
              Stucker, B., "Coping with Early Media in the Session
              Initiation Protocol (SIP)",
              draft-stucker-sipping-early-media-coping-03 (work in
              progress), October 2006.

   [I-D.wing-sipping-srtp-key]
              Wing, D., Audet, F., Fries, S., Tschofenig, H., and A.
              Johnston, "Secure Media Recording and Transcoding with the
              Session Initiation  Protocol",
              draft-wing-sipping-srtp-key-04 (work in progress),
              October 2008.

   [I-D.zimmermann-avt-zrtp]
              Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Secure RTP",
              draft-zimmermann-avt-zrtp-11 (work in progress),
              November 2008.

   [RFC3326]  Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
              Header Field for the Session Initiation Protocol (SIP)",
              RFC 3326, December 2002.

   [RFC3372]  Vemuri, A. and J. Peterson, "Session Initiation Protocol
              for Telephones (SIP-T): Context and Architectures",
              BCP 63, RFC 3372, September 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.

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

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

   [RFC4492]  Blake-Wilson, S., Bolyard, N., Gupta, V., Hawk, C., and B.
              Moeller, "Elliptic Curve Cryptography (ECC) Cipher Suites
              for Transport Layer Security (TLS)", RFC 4492, May 2006.

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

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   [RFC4650]  Euchner, M., "HMAC-Authenticated Diffie-Hellman for
              Multimedia Internet KEYing (MIKEY)", RFC 4650,
              September 2006.

   [RFC4738]  Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
              RSA-R: An Additional Mode of Key Distribution in
              Multimedia Internet KEYing (MIKEY)", RFC 4738,
              November 2006.

   [RFC4771]  Lehtovirta, V., Naslund, M., and K. Norrman, "Integrity
              Transform Carrying Roll-Over Counter for the Secure Real-
              time Transport Protocol (SRTP)", RFC 4771, January 2007.

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

   [RFC4949]  Shirey, R., "Internet Security Glossary, Version 2",
              RFC 4949, August 2007.

   [RFC5027]  Andreasen, F. and D. Wing, "Security Preconditions for
              Session Description Protocol (SDP) Media Streams",
              RFC 5027, October 2007.

   [RFC5197]  Fries, S. and D. Ignjatic, "On the Applicability of
              Various Multimedia Internet KEYing (MIKEY) Modes and
              Extensions", RFC 5197, June 2008.

Appendix A.  Overview and Evaluation of Existing Keying Mechanisms

   Based on how the SRTP keys are exchanged, each SRTP key exchange
   mechanism belongs to one general category:

   signaling path:
        All the keying is carried in the call signaling (SIP or SDP)
        path.

   media path:
        All the keying is carried in the SRTP/SRTCP media path, and no
        signaling whatsoever is carried in the call signaling path.

   signaling and media path:
        Parts of the keying are carried in the SRTP/SRTCP media path,
        and parts are carried in the call signaling (SIP or SDP) path.

   One of the significant benefits of SRTP over other end-to-end
   encryption mechanisms, such as for example IPsec, is that SRTP is
   bandwidth efficient and SRTP retains the header of RTP packets.

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   Bandwidth efficiency is vital for VoIP in many scenarios where access
   bandwidth is limited or expensive, and retaining the RTP header is
   important for troubleshooting packet loss, delay, and jitter.

   Related to SRTP's characteristics is a goal that any SRTP keying
   mechanism to also be efficient and not cause additional call setup
   delay.  Contributors to additional call setup delay include network
   or database operations: retrieval of certificates and additional SIP
   or media path messages, and computational overhead of establishing
   keys or validating certificates.

   When examining the choice between keying in the signaling path,
   keying in the media path, or keying in both paths, it is important to
   realize the media path is generally 'faster' than the SIP signaling
   path.  The SIP signaling path has computational elements involved
   which parse and route SIP messages.  The media path, on the other
   hand, does not normally have computational elements involved, and
   even when computational elements such as firewalls are involved, they
   cause very little additional delay.  Thus, the media path can be
   useful for exchanging several messages to establish SRTP keys.  A
   disadvantage of keying over the media path is that interworking
   different key exchange requires the interworking function be in the
   media path, rather than just in the signaling path; in practice this
   involvement is probably unavoidable anyway.

A.1.  Signaling Path Keying Techniques

A.1.1.  MIKEY-NULL

   MIKEY-NULL [RFC3830] has the offerer indicate the SRTP keys for both
   directions.  The key is sent unencrypted in SDP, which means the SDP
   must be encrypted hop-by-hop (e.g., by using TLS (SIPS)) or end-to-
   end (e.g., by using S/MIME).

   MIKEY-NULL requires one message from offerer to answerer (half a
   round trip), and does not add additional media path messages.

A.1.2.  MIKEY-PSK

   MIKEY-PSK (pre-shared key) [RFC3830] requires that all endpoints
   share one common key.  MIKEY-PSK has the offerer encrypt the SRTP
   keys for both directions using this pre-shared key.

   MIKEY-PSK requires one message from offerer to answerer (half a round
   trip), and does not add additional media path messages.

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A.1.3.  MIKEY-RSA

   MIKEY-RSA [RFC3830] has the offerer encrypt the keys for both
   directions using the intended answerer's public key, which is
   obtained from a mechanism outside of MIKEY.

   MIKEY-RSA requires one message from offerer to answerer (half a round
   trip), and does not add additional media path messages.  MIKEY-RSA
   requires the offerer to obtain the intended answerer's certificate.

A.1.4.  MIKEY-RSA-R

   MIKEY-RSA-R [RFC4738] is essentially the same as MIKEY-RSA but
   reverses the role of the offerer and the answerer with regards to
   providing the keys.  That is, the answerer encrypts the keys for both
   directions using the offerer's public key.  Both the offerer and
   answerer validate each other's public keys using a standard X.509
   validation techniques.  MIKEY-RSA-R also enables sending certificates
   in the MIKEY message.

   MIKEY-RSA-R requires one message from offerer to answer, and one
   message from answerer to offerer (full round trip), and does not add
   additional media path messages.  MIKEY-RSA-R requires the offerer
   validate the answerer's certificate.

A.1.5.  MIKEY-DHSIGN

   In MIKEY-DHSIGN [RFC3830] the offerer and answerer derive the key
   from a Diffie-Hellman exchange.  In order to prevent an active man-
   in-the-middle the DH exchange itself is signed using each endpoint's
   private key and the associated public keys are validated using
   standard X.509 validation techniques.

   MIKEY-DHSIGN requires one message from offerer to answerer, and one
   message from answerer to offerer (full round trip), and does not add
   additional media path messages.  MIKEY-DHSIGN requires the offerer
   and answerer to validate each other's certificates.  MIKEY-DHSIGN
   also enables sending the answerer's certificate in the MIKEY message.

A.1.6.  MIKEY-DHHMAC

   MIKEY-DHHMAC [RFC4650] uses a pre-shared secret to HMAC the Diffie-
   Hellman exchange, essentially combining aspects of MIKEY-PSK with
   MIKEY-DHSIGN, but without MIKEY-DHSIGN's need for certificate
   authentication.

   MIKEY-DHHMAC requires one message from offerer to answerer, and one
   message from answerer to offerer (full round trip), and does not add

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   additional media path messages.

A.1.7.  MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC)

   ECC Algorithms For MIKEY [I-D.ietf-msec-mikey-ecc] describes how ECC
   can be used with MIKEY-RSA (using ECDSA signature) and with MIKEY-
   DHSIGN (using a new DH-Group code), and also defines two new ECC-
   based algorithms, Elliptic Curve Integrated Encryption Scheme (ECIES)
   and Elliptic Curve Menezes-Qu-Vanstone (ECMQV) .

   With this proposal, the ECDSA signature, MIKEY-ECIES, and MIKEY-ECMQV
   function exactly like MIKEY-RSA, and the new DH-Group code function
   exactly like MIKEY-DHSIGN.  Therefore these ECC mechanisms are not
   discussed separately in this document.

A.1.8.  Security Descriptions with SIPS

   Security Descriptions [RFC4568] has each side indicate the key it
   will use for transmitting SRTP media, and the keys are sent in the
   clear in SDP.  Security Descriptions relies on hop-by-hop (TLS via
   "SIPS:") encryption to protect the keys exchanged in signaling.

   Security Descriptions requires one message from offerer to answerer,
   and one message from answerer to offerer (full round trip), and does
   not add additional media path messages.

A.1.9.  Security Descriptions with S/MIME

   This keying mechanism is identical to Appendix A.1.8, except that
   rather than protecting the signaling with TLS, the entire SDP is
   encrypted with S/MIME.

A.1.10.  SDP-DH (expired)

   SDP Diffie-Hellman [I-D.baugher-mmusic-sdp-dh] exchanges Diffie-
   Hellman messages in the signaling path to establish session keys.  To
   protect against active man-in-the-middle attacks, the Diffie-Hellman
   exchange needs to be protected with S/MIME, SIPS, or SIP Identity
   [RFC4474] and SIP Conected Identity [RFC4916].

   SDP-DH requires one message from offerer to answerer, and one message
   from answerer to offerer (full round trip), and does not add
   additional media path messages.

A.1.11.  MIKEYv2 in SDP (expired)

   MIKEYv2 [I-D.dondeti-msec-rtpsec-mikeyv2] adds mode negotiation to
   MIKEYv1 and removes the time synchronization requirement.  It

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   therefore now takes 2 round-trips to complete.  In the first round
   trip, the communicating parties learn each other's identities, agree
   on a MIKEY mode, crypto algorithm, SRTP policy, and exchanges nonces
   for replay protection.  In the second round trip, they negotiate
   unicast and/or group SRTP context for SRTP and/or SRTCP.

   Furthemore, MIKEYv2 also defines an in-band negotiation mode as an
   alternative to SDP (see Appendix A.3.3).

A.2.  Media Path Keying Technique

A.2.1.  ZRTP

   ZRTP [I-D.zimmermann-avt-zrtp] does not exchange information in the
   signaling path (although it's possible for endpoints to exchange a
   hash of the ZRTP Hello message with "a=zrtp-hash" in the initial
   Offer if sent over an integrity-protected signaling channel.  This
   provides some useful correlation between the signaling and media
   layers).  In ZRTP the keys are exchanged entirely in the media path
   using a Diffie-Hellman exchange.  The advantage to this mechanism is
   that the signaling channel is used only for call setup and the media
   channel is used to establish an encrypted channel -- much like
   encryption devices on the PSTN.  ZRTP uses voice authentication of
   its Diffie-Hellman exchange by having each person read digits or
   words to the other person.  Subsequent sessions with the same ZRTP
   endpoint can be authenticated using the stored hash of the previously
   negotiated key rather than voice authentication.  ZRTP uses 4 media
   path messages (Hello, Commit, DHPart1, and DHPart2) to establish the
   SRTP key, and 3 media path confirmation messages.  These initial
   messages are all sent as non-RTP packets.

      Note that when ZRTP probing is used, unencrypted RTP can be
      exchanged until the SRTP keys are established.

A.3.  Signaling and Media Path Keying Techniques

A.3.1.  EKT

   EKT [I-D.mcgrew-srtp-ekt] relies on another SRTP key exchange
   protocol, such as Security Descriptions or MIKEY, for bootstrapping.
   In the initial phase, each member of a conference uses an SRTP key
   exchange protocol to establish a common key encryption key (KEK).
   Each member may use the KEK to securely transport its SRTP master key
   and current SRTP rollover counter (ROC), via RTCP, to the other
   participants in the session.

   EKT requires the offerer to send some parameters (EKT_Cipher, KEK,
   and security parameter index (SPI)) via the bootstrapping protocol

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   such as Security Descriptions or MIKEY.  Each answerer sends an SRTCP
   message which contains the answerer's SRTP Master Key, rollover
   counter, and the SRTP sequence number.  Rekeying is done by sending a
   new SRTCP message.  For reliable transport, multiple RTCP messages
   need to be sent.

A.3.2.  DTLS-SRTP

   DTLS-SRTP [I-D.ietf-avt-dtls-srtp] exchanges public key fingerprints
   in SDP [I-D.fischl-sipping-media-dtls] and then establishes a DTLS
   session over the media channel.  The endpoints use the DTLS handshake
   to agree on crypto suites and establish SRTP session keys.  SRTP
   packets are then exchanged between the endpoints.

   DTLS-SRTP requires one message from offerer to answerer (half round
   trip), and one message from the answerer to offerer (full round trip)
   so the offerer can correlate the SDP answer with the answering
   endpoint.  DTLS-SRTP uses 4 media path messages to establish the SRTP
   key.

   This document assumes DTLS will use TLS_RSA_WITH_AES_128_CBC_SHA as
   its cipher suite, which is the mandatory-to-implement cipher suite in
   TLS [I-D.ietf-tls-rfc4346-bis].

A.3.3.  MIKEYv2 Inband (expired)

   As defined in Appendix A.1.11, MIKEYv2 also defines an in-band
   negotiation mode as an alternative to SDP (see Appendix A.3.3).  The
   details are not sorted out in the draft yet on what in-band actually
   means (i.e., UDP, RTP, RTCP, etc.).

A.4.  Evaluation Criteria - SIP

   This section considers how each keying mechanism interacts with SIP
   features.

A.4.1.  Secure Retargeting and Secure Forking

   Retargeting and forking of signaling requests is described within
   Section 4.2.  The following builds upon this description.

   The following list compares the behavior of secure forking, answering
   association, two-time pads, and secure retargeting for each keying
   mechanism.

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      MIKEY-NULL  Secure Forking: No, all AORs see offerer's and
         answerer's keys.  Answer is associated with media by the SSRC
         in MIKEY.  Additionally, a two-time pad occurs if two branches
         choose the same 32-bit SSRC and transmit SRTP packets.

         Secure Retargeting: No, all targets see offerer's and
         answerer's keys.  Suffers from retargeting identity problem.

      MIKEY-PSK
         Secure Forking: No, all AORs see offerer's and answerer's keys.
         Answer is associated with media by the SSRC in MIKEY.  Note
         that all AORs must share the same pre-shared key in order for
         forking to work at all with MIKEY-PSK.  Additionally, a two-
         time pad occurs if two branches choose the same 32-bit SSRC and
         transmit SRTP packets.

         Secure Retargeting: Not secure.  For retargeting to work, the
         final target must possess the correct PSK.  As this is likely
         in scenarios were the call is targeted to another device
         belonging to the same user (forking), it is very unlikely that
         other users will possess that PSK and be able to successfully
         answer that call.

      MIKEY-RSA
         Secure Forking: No, all AORs see offerer's and answerer's keys.
         Answer is associated with media by the SSRC in MIKEY.  Note
         that all AORs must share the same private key in order for
         forking to work at all with MIKEY-RSA.  Additionally, a two-
         time pad occurs if two branches choose the same 32-bit SSRC and
         transmit SRTP packets.

         Secure Retargeting: No.

      MIKEY-RSA-R
         Secure Forking: Yes. Answer is associated with media by the
         SSRC in MIKEY.

         Secure Retargeting: Yes.

      MIKEY-DHSIGN
         Secure Forking: Yes, each forked endpoint negotiates unique
         keys with the offerer for both directions.  Answer is
         associated with media by the SSRC in MIKEY.

         Secure Retargeting: Yes, each target negotiates unique keys
         with the offerer for both directions.

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      MIKEYv2 in SDP
         The behavior will depend on which mode is picked.

      MIKEY-DHHMAC
         Secure Forking: Yes, each forked endpoint negotiates unique
         keys with the offerer for both directions.  Answer is
         associated with media by the SSRC in MIKEY.

         Secure Retargeting: Yes, each target negotiates unique keys
         with the offerer for both directions.  Note that for the keys
         to be meaningful, it would require the PSK to be the same for
         all the potential intermediaries, which would only happen
         within a single domain.

      Security Descriptions with SIPS
         Secure Forking: No.  Each forked endpoint sees the offerer's
         key.  Answer is not associated with media.

         Secure Retargeting: No.  Each target sees the offerer's key.

      Security Descriptions with S/MIME
         Secure Forking: No.  Each forked endpoint sees the offerer's
         key.  Answer is not associated with media.

         Secure Retargeting: No.  Each target sees the offerer's key.
         Suffers from retargeting identity problem.

      SDP-DH
         Secure Forking: Yes. Each forked endpoint calculates a unique
         SRTP key.  Answer is not associated with media.

         Secure Retargeting: Yes. The final target calculates a unique
         SRTP key.

      ZRTP
         Yes. Each forked endpoint calculates a unique SRTP key.  With
         the "a=zrtp-hash" attribute, the media can be associated with
         an answer.

         Secure Retargeting: Yes. The final target calculates a unique
         SRTP key.

      EKT
         Secure Forking: Inherited from the bootstrapping mechanism (the
         specific MIKEY mode or Security Descriptions).  Answer is
         associated with media by the SPI in the EKT protocol.  Answer
         is associated with media by the SPI in the EKT protocol.

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         Secure Retargeting: Inherited from the bootstrapping mechanism
         (the specific MIKEY mode or Security Descriptions).

      DTLS-SRTP
         Secure Forking: Yes. Each forked endpoint calculates a unique
         SRTP key.  Answer is associated with media by the certificate
         fingerprint in signaling and certificate in the media path.

         Secure Retargeting: Yes. The final target calculates a unique
         SRTP key.

      MIKEYv2 Inband
         The behavior will depend on which mode is picked.

A.4.2.  Clipping Media Before SDP Answer

   Clipping media before receiving the signaling answer is described
   within Section 4.1.  The following builds upon this description.

   Furthermore, the problem of clipping gets compounded when forking is
   used.  For example, if using a Diffie-Hellman keying technique with
   security preconditions that forks to 20 endpoints, the call initiator
   would get 20 provisional responses containing 20 signed Diffie-
   Hellman half keys.  Calculating 20 DH secrets and validating
   signatures can be a difficult task depending on the device
   capabilities.

   The following list compares the behavior of clipping before SDP
   answer for each keying mechanism.

      MIKEY-NULL
         Not clipped.  The offerer provides the answerer's keys.

      MIKEY-PSK
         Not clipped.  The offerer provides the answerer's keys.

      MIKEY-RSA
         Not clipped.  The offerer provides the answerer's keys.

      MIKEY-RSA-R
         Clipped.  The answer contains the answerer's encryption key.

      MIKEY-DHSIGN
         Clipped.  The answer contains the answerer's Diffie-Hellman
         response.

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      MIKEY-DHHMAC
         Clipped.  The answer contains the answerer's Diffie-Hellman
         response.

      MIKEYv2 in SDP
         The behavior will depend on which mode is picked.

      Security Descriptions with SIPS
         Clipped.  The answer contains the answerer's encryption key.

      Security Descriptions with S/MIME
         Clipped.  The answer contains the answerer's encryption key.

      SDP-DH
         Clipped.  The answer contains the answerer's Diffie-Hellman
         response.

      ZRTP
         Not clipped because the session intially uses RTP.  While RTP
         is flowing, both ends negotiate SRTP keys in the media path and
         then switch to using SRTP.

      EKT
         Not clipped, as long as the first RTCP packet (containing the
         answerer's key) is not lost in transit.  The answerer sends its
         encryption key in RTCP, which arrives at the same time (or
         before) the first SRTP packet encrypted with that key.

            Note: RTCP needs to work, in the answerer-to-offerer
            direction, before the offerer can decrypt SRTP media.

      DTLS-SRTP
         No clipping after the DTLS-SRTP handshake has completed.  SRTP
         keys are exchanged in the media path.  Need to wait for SDP
         answer to ensure DTLS-SRTP handshake was done with an
         authorized party.

            If a middlebox interferes with the media path, there can be
            clipping [I-D.ietf-mmusic-media-path-middleboxes].

      MIKEYv2 Inband
         Not clipped.  Keys are exchanged in the media path without
         relying on the signaling path.

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A.4.3.  SSRC and ROC

   In SRTP, a cryptographic context is defined as the SSRC, destination
   network address, and destination transport port number.  Whereas RTP,
   a flow is defined as the destination network address and destination
   transport port number.  This results in a problem -- how to
   communicate the SSRC so that the SSRC can be used for the
   cryptographic context.

   Two approaches have emerged for this communication.  One, used by all
   MIKEY modes, is to communicate the SSRCs to the peer in the MIKEY
   exchange.  Another, used by Security Descriptions, is to apply "late
   binding" -- that is, any new packet containing a previously-unseen
   SSRC (which arrives at the same destination network address and
   destination transport port number) will create a new cryptographic
   context.  Another approach, common amongst techniques with media-path
   SRTP key establishment, is to require a handshake over that media
   path before SRTP packets are sent.  MIKEY's approach changes RTP's
   SSRC collision detection behavior by requiring RTP to pre-establish
   the SSRC values for each session.

   Another related issue is that SRTP introduces a rollover counter
   (ROC), which records how many times the SRTP sequence number has
   rolled over.  As the sequence number is used for SRTP's default
   ciphers, it is important that all endpoints know the value of the
   ROC.  The ROC starts at 0 at the beginning of a session.

   Some keying mechanisms cause a two-time pad to occur if two endpoints
   of a forked call have an SSRC collision.

   Note: A proposal has been made to send the ROC value on every Nth
   SRTP packet[RFC4771].  This proposal has not yet been incorporated
   into this document.

   The following list examines handling of SSRC and ROC:

      MIKEY-NULL
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

      MIKEY-PSK
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

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      MIKEY-RSA
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

      MIKEY-RSA-R
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

      MIKEY-DHSIGN
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

      MIKEY-DHHMAC
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

      MIKEYv2 in SDP
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

      Security Descriptions with SIPS
         Neither SSRC nor ROC are signaled.  SSRC 'late binding' is
         used.

      Security Descriptions with S/MIME
         Neither SSRC nor ROC are signaled.  SSRC 'late binding' is
         used.

      SDP-DH
         Neither SSRC nor ROC are signaled.  SSRC 'late binding' is
         used.

      ZRTP
         Neither SSRC nor ROC are signaled.  SSRC 'late binding' is
         used.

      EKT
         The SSRC of the SRTCP packet containing an EKT update
         corresponds to the SRTP master key and other parameters within
         that packet.

      DTLS-SRTP
         Neither SSRC nor ROC are signaled.  SSRC 'late binding' is
         used.

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      MIKEYv2 Inband
         Each endpoint indicates a set of SSRCs and the ROC for SRTP
         packets it transmits.

A.5.  Evaluation Criteria - Security

   This section evaluates each keying mechanism on the basis of their
   security properties.

A.5.1.  Distribution and Validation of Persistent Public Keys and
        Certificates

   Using persistent public keys for confidentiality and authentication
   can introduce requirements for two types of systems, often
   implemented using certificates: (1) a system to distribute those
   persistent public keys certificates, and (2) a system for validating
   those persistent public keys.  We refer to the former as a key
   distribution system and the latter as an authentication
   infrastructure.  In many cases, a monolithic public key
   infrastructure (PKI) is used for fulfill both of these roles.
   However, these functions can be provided by many other systems.  For
   instance, key distribution may be accomplished by any public
   repository of keys.  Any system in which the two endpoints have
   access to trust anchors and intermediate CA certificates that can be
   used to validate other endpoints' certificates (including a system of
   self-signed certificates) can be used to support certificate
   validation in the below schemes.

   With real-time communications it is desirable to avoid fetching or
   validating certificates that delay call setup.  Rather, it is
   preferable to fetch or validate certificates in such a way that call
   setup is not delayed.  For example, a certificate can be validated
   while the phone is ringing or can be validated while ring-back tones
   are being played or even while the called party is answering the
   phone and saying "hello".  Even better is to avoid fetching or
   validating persistent public keys at all.

   SRTP key exchange mechanisms that require a particular authentication
   infrastructure to operate (whether for distribution or validation)
   are gated on the deployment of a such an infrastructure available to
   both endpoints.  This means that no media security is achievable
   until such an infrastructure exists.  For SIP, something like sip-
   certs [I-D.ietf-sip-certs] might be used to obtain the certificate of
   a peer.

      Note: Even if sip-certs [I-D.ietf-sip-certs] was deployed, the
      retargeting problem (Appendix A.4.1) would still prevent
      successful deployment of keying techniques which require the

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      offerer to obtain the actual target's public key.

   The following list compares the requirements introduced by the use of
   public-key cryptography in each keying mechanism, both for public key
   distribution and for certificate validation.

      MIKEY-NULL
         Public-key cryptography is not used.

      MIKEY-PSK
         Public-key cryptography is not used.  Rather, all endpoints
         must have some way to exchange per-endpoint or per-system pre-
         shared keys.

      MIKEY-RSA
         The offerer obtains the intended answerer's public key before
         initiating the call.  This public key is used to encrypt the
         SRTP keys.  There is no defined mechanism for the offerer to
         obtain the answerer's public key, although [I-D.ietf-sip-certs]
         might be viable in the future.

         The offer may also contain a certificate for the offeror, which
         would require an authentication infrastructure in order to be
         validated by the receiver.

      MIKEY-RSA-R
         The offer contains the offerer's certificate, and the answer
         contains the answerer's certificate.  The answerer uses the
         public key in the certificate to encrypt the SRTP keys that
         will be used by the offerer and the answerer.  An
         authentication infrastructure is necessary to validate the
         certificates.

      MIKEY-DHSIGN
         An authentication infrastructure is used to authenticate the
         public key that is included in the MIKEY message.

      MIKEY-DHHMAC
         Public-key cryptography is not used.  Rather, all endpoints
         must have some way to exchange per-endpoint or per-system pre-
         shared keys.

      MIKEYv2 in SDP
         The behavior will depend on which mode is picked.

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      Security Descriptions with SIPS
         Public-key cryptography is not used.

      Security Descriptions with S/MIME
         Use of S/MIME requires that the endpoints be able to fetch and
         validate certificates for each other.  The offerer must obtain
         the intended target's certificate and encrypts the SDP offer
         with the public key contained in target's certificate.  The
         answerer must obtain the offerer's certificate and encrypt the
         SDP answer with the public key contained in the offerer's
         certificate.

      SDP-DH
         Public-key cryptography is not used.

      ZRTP
         Public-key cryptography is used (Diffie-Hellman), but without
         dependence on persistent public keys.  Thus, certificates are
         not fetched or validated.

      EKT
         Public-key cryptography is not used by itself, but might be
         used by the EKT bootstrapping keying mechanism (such as certain
         MIKEY modes).

      DTLS-SRTP
         Remote party's certificate is sent in media path, and a
         fingerprint of the same certificate is sent in the signaling
         path.

      MIKEYv2 Inband
         The behavior will depend on which mode is picked.

A.5.2.  Perfect Forward Secrecy

   In the context of SRTP, Perfect Forward Secrecy is the property that
   SRTP session keys that protected a previous session are not
   compromised if the static keys belonging to the endpoints are
   compromised.  That is, if someone were to record your encrypted
   session content and later acquires either party's private key, that
   encrypted session content would be safe from decryption if your key
   exchange mechanism had perfect forward secrecy.

   The following list describes how each key exchange mechanism provides
   PFS.

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      MIKEY-NULL
         Not applicable; MIKEY-NULL does not have a long-term secret.

      MIKEY-PSK
         No PFS.

      MIKEY-RSA
         No PFS.

      MIKEY-RSA-R
         No PFS.

      MIKEY-DHSIGN
         PFS is provided with the Diffie-Hellman exchange.

      MIKEY-DHHMAC
         PFS is provided with the Diffie-Hellman exchange.

      MIKEYv2 in SDP
         The behavior will depend on which mode is picked.

      Security Descriptions with SIPS
         Not applicable; Security Descriptions does not have a long-term
         secret.

      Security Descriptions with S/MIME
         Not applicable; Security Descriptions does not have a long-term
         secret.

      SDP-DH
         PFS is provided with the Diffie-Hellman exchange.

      ZRTP
         PFS is provided with the Diffie-Hellman exchange.

      EKT
         No PFS.

      DTLS-SRTP
         PFS is provided if the negotiated cipher suite uses ephemeral
         keys (e.g., Diffie-Hellman (DHE_RSA [I-D.ietf-tls-rfc4346-bis])
         or Elliptic Curve Diffie-Hellman [RFC4492]).

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      MIKEYv2 Inband
         The behavior will depend on which mode is picked.

A.5.3.  Best Effort Encryption

   With best effort encryption, SRTP is used with endpoints that support
   SRTP, otherwise RTP is used.

   SIP needs a backwards-compatible best effort encryption in order for
   SRTP to work successfully with SIP retargeting and forking when there
   is a mix of forked or retargeted devices that support SRTP and don't
   support SRTP.

      Consider the case of Bob, with a phone that only does RTP and a
      voice mail system that supports SRTP and RTP.  If Alice calls Bob
      with an SRTP offer, Bob's RTP-only phone will reject the media
      stream (with an empty "m=" line) because Bob's phone doesn't
      understand SRTP (RTP/SAVP).  Alice's phone will see this rejected
      media stream and may terminate the entire call (BYE) and re-
      initiate the call as RTP-only, or Alice's phone may decide to
      continue with call setup with the SRTP-capable leg (the voice mail
      system).  If Alice's phone decided to re-initiate the call as RTP-
      only, and Bob doesn't answer his phone, Alice will then leave
      voice mail using only RTP, rather than SRTP as expected.

   Currently, several techniques are commonly considered as candidates
   to provide opportunistic encryption:

   multipart/alternative
      [I-D.jennings-sipping-multipart] describes how to form a
      multipart/alternative body part in SIP.  The significant issues
      with this technique are (1) that multipart MIME is incompatible
      with existing SIP proxies, firewalls, Session Border Controllers,
      and endpoints and (2) when forking, the Heterogeneous Error
      Response Forking Problem (HERFP) [RFC3326] causes problems if such
      non-multipart-capable endpoints were involved in the forking.

   session attribute
      With this technique, the endpoints signal their desire to do SRTP
      by signaling RTP (RTP/AVP), and using an attribute ("a=") in the
      SDP.  This technique is entirely backwards compatible with non-
      SRTP-aware endpoints, but doesn't use the RTP/SAVP protocol
      registered by SRTP [RFC3711].

   SDP Capability Negotiation
      SDP Capability Negotiation
      [I-D.ietf-mmusic-sdp-capability-negotiation] provides a backwards-
      compatible mechanism to allow offering both SRTP and RTP in a

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      single offer.  This is the preferred technique.

   Probing
      With this technique, the endpoints first establish an RTP session
      using RTP (RTP/AVP).  The endpoints send probe messages, over the
      media path, to determine if the remote endpoint supports their
      keying technique.  A disadvantage of probing is an active attacker
      can interfere with probes, and until probing completes (and SRTP
      is established) the media is in the clear.

   The preferred technique, SDP Capability Negotiation
   [I-D.ietf-mmusic-sdp-capability-negotiation], can be used with all
   key exchange mechanisms.  What remains unique is ZRTP, which can also
   accomplish its best effort encryption by probing (sending ZRTP
   messages over the media path) or by session attribute (see "a=zrtp-
   hash" in [I-D.zimmermann-avt-zrtp]).  Current implementations of ZRTP
   use probing.

A.5.4.  Upgrading Algorithms

   It is necessary to allow upgrading SRTP encryption and hash
   algorithms, as well as upgrading the cryptographic functions used for
   the key exchange mechanism.  With SIP's offer/answer model, this can
   be computionally expensive because the offer needs to contain all
   combinations of the key exchange mechanisms (all MIKEY modes,
   Security Descriptions) and all SRTP cryptographic suites (AES-128,
   AES-256) and all SRTP cryptographic hash functions (SHA-1, SHA-256)
   that the offerer supports.  In order to do this, the offerer has to
   expend CPU resources to build an offer containing all of this
   information which becomes computationally prohibitive.

   Thus, it is important to keep the offerer's CPU impact fixed so that
   offering multiple new SRTP encryption and hash functions incurs no
   additional expense.

   The following list describes the CPU effort involved in using each
   key exchange technique.

      MIKEY-NULL
         No significant computational expense.

      MIKEY-PSK
         No significant computational expense.

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      MIKEY-RSA
         For each offered SRTP crypto suite, the offerer has to perform
         RSA operation to encrypt the TGK

      MIKEY-RSA-R
         For each offered SRTP crypto suite, the offerer has to perform
         public key operation to sign the MIKEY message.

      MIKEY-DHSIGN
         For each offered SRTP crypto suite, the offerer has to perform
         Diffie-Hellman operation, and a public key operation to sign
         the Diffie-Hellman output.

      MIKEY-DHHMAC
         For each offered SRTP crypto suite, the offerer has to perform
         Diffie-Hellman operation.

      MIKEYv2 in SDP
         The behavior will depend on which mode is picked.

      Security Descriptions with SIPS
         No significant computational expense.

      Security Descriptions with S/MIME
         S/MIME requires the offerer and the answerer to encrypt the SDP
         with the other's public key, and to decrypt the received SDP
         with their own private key.

      SDP-DH
         For each offered SRTP crypto suite, the offerer has to perform
         a Diffie-Hellman operation.

      ZRTP
         The offerer has no additional computational expense at all, as
         the offer contains no information about ZRTP or might contain
         "a=zrtp-hash".

      EKT
         The offerer's Computational expense depends entirely on the EKT
         bootstrapping mechanism selected (one or more MIKEY modes or
         Security Descriptions).

      DTLS-SRTP
         The offerer has no additional computational expense at all, as
         the offer contains only a fingerprint of the certificate that
         will be presented in the DTLS exchange.

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      MIKEYv2 Inband
         The behavior will depend on which mode is picked.

Appendix B.  Out-of-Scope

   The compromise of an endpoint that has access to decrypted media
   (e.g., SIP user agent, transcoder, recorder) is out of scope of this
   document.  Such a compromise might be via privilege escalation,
   installation of a virus or trojan horse, or similar attacks.

B.1.  Shared Key Conferencing

   The consensus on the RTPSEC mailing list was to concentrate on
   unicast, point-to-point sessions.  Thus, there are no requirements
   related to shared key conferencing.  This section is retained for
   informational purposes.

   For efficient scaling, large audio and video conference bridges
   operate most efficiently by encrypting the current speaker once and
   distributing that stream to the conference attendees.  Typically,
   inactive participants receive the same streams -- they hear (or see)
   the active speaker(s), and the active speakers receive distinct
   streams that don't include themselves.  In order to maintain
   confidentiality of such conferences where listeners share a common
   key, all listeners must rekeyed when a listener joins or leaves a
   conference.

   An important use case for mixers/translators is a conference bridge:

                                         +----+
                             A --- 1 --->|    |
                               <-- 2 ----| M  |
                                         | I  |
                             B --- 3 --->| X  |
                               <-- 4 ----| E  |
                                         | R  |
                             C --- 5 --->|    |
                               <-- 6 ----|    |
                                         +----+

                       Figure 3: Centralized Keying

   In the figure above, 1, 3, and 5 are RTP media contributions from
   Alice, Bob, and Carol, and 2, 4, and 6 are the RTP flows to those
   devices carrying the 'mixed' media.

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   Several scenarios are possible:

   a.  Multiple inbound sessions: 1, 3, and 5 are distinct RTP sessions,

   b.  Multiple outbound sessions: 2, 4, and 6 are distinct RTP
       sessions,

   c.  Single inbound session: 1, 3, and 5 are just different sources
       within the same RTP session,

   d.  Single outbound session: 2, 4, and 6 are different flows of the
       same (multi-unicast) RTP session

   If there are multiple inbound sessions and multiple outbound sessions
   (scenarios a and b), then every keying mechanism behaves as if the
   mixer were an end point and can set up a point-to-point secure
   session between the participant and the mixer.  This is the simplest
   situation, but is computationally wasteful, since SRTP processing has
   to be done independently for each participant.  The use of multiple
   inbound sessions (scenario a) doesn't waste computational resources,
   though it does consume additional cryptographic context on the mixer
   for each participant and has the advantage of data origin
   authentication.

   To support a single outbound session (scenario d), the mixer has to
   dictate its encryption key to the participants.  Some keying
   mechanisms allow the transmitter to determine its own key, and others
   allow the offerer to determine the key for the offerer and answerer.
   Depending on how the call is established, the offerer might be a
   participant (such as a participant dialing into a conference bridge)
   or the offerer might be the mixer (such as a conference bridge
   calling a participant).  The use of offerless INVITEs may help some
   keying mechanisms reverse the role of offerer/answerer.  A
   difficulty, however, is knowing a priori if the role should be
   reversed for a particular call.  The significant advantage of a
   single outbound session is the number of SRTP encryption operations
   remains constant even as the number of participants increases.
   However, a disadvantage is that data origin authentication is lost,
   allowing any participant to spoof the sender (because all
   participants know the sender's SRTP key).

Appendix C.  Requirement renumbering in -02

   [[RFC Editor: Please delete this section prior to publication.]]

   Previous versions of this document used requirement numbers, which
   were changed to mnemonics as follows:

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   R1    R-FORK-RETARGET

   R2    R-BEST-SECURE

   R3    R-DISTINCT

   R4    R-REUSE; changed from 'MAY' to 'protocol MUST support, and
         SHOULD implement'

   R5    R-AVOID-CLIPPING

   R6    R-PASS-MEDIA

   R7    R-PASS-SIG

   R8    R-PFS

   R9    R-COMPUTE

   R10   R-RTP-CHECK

   R11   (folded into R4; was reuse previous session)

   R12   R-CERTS

   R13   R-FIPS

   R14   R-ASSOC

   R15   R-ALLOW-RTP

   R16   R-DOS

   R17   R-SIG-MEDIA

   R18   R-EXISTING

   R19   R-AGILITY

   R20   R-DOWNGRADE

   R21   R-NEGOTIATE

   R23   R-OTHER-SIGNALING

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   R23   R-RECORDING (R23 was duplicated in previous versions of the
         document)

   R24   (deleted; was lawful intercept)

   R25   R-TRANSCODER

   R26   R-PSTN

   R27   R-ID-BINDING

   R28   R-ACT-ACT

Authors' Addresses

   Dan Wing (editor)
   Cisco Systems, Inc.
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Email: dwing@cisco.com

   Steffen Fries
   Siemens AG
   Otto-Hahn-Ring 6
   Munich, Bavaria  81739
   Germany

   Email: steffen.fries@siemens.com

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

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

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   Francois Audet
   Nortel
   4655 Great America Parkway
   Santa Clara, CA  95054
   USA

   Email: audet@nortel.com

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