RFC5479: Requirements and Analysis of Media Security Management Protocols

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Related keywords:  (keying) (Secure RTP) (srtp)





Network Working Group                                       D. Wing, Ed.
Request for Comments: 5479                                         Cisco
Category: Informational                                         S. Fries
                                                              Siemens AG
                                                           H. Tschofenig
                                                  Nokia Siemens Networks
                                                                F. Audet
                                                                  Nortel
                                                              April 2009


    Requirements and Analysis of Media Security Management Protocols

Status of This Memo

   This memo provides information for the Internet community.  It does
   not specify an Internet standard of any kind.  Distribution of this
   memo is unlimited.

Copyright Notice

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

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents in effect on the date of
   publication of this document (http://trustee.ietf.org/license-info).
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.

Abstract

   This document describes requirements for a protocol to negotiate a
   security context for SIP-signaled Secure RTP (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.













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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Attack Scenarios . . . . . . . . . . . . . . . . . . . . . . .  5
   4.  Call Scenarios and Requirements Considerations . . . . . . . .  7
     4.1.  Clipping Media before Signaling Answer . . . . . . . . . .  7
     4.2.  Retargeting and Forking  . . . . . . . . . . . . . . . . .  8
     4.3.  Recording  . . . . . . . . . . . . . . . . . . . . . . . . 11
     4.4.  PSTN Gateway . . . . . . . . . . . . . . . . . . . . . . . 12
     4.5.  Call Setup Performance . . . . . . . . . . . . . . . . . . 12
     4.6.  Transcoding  . . . . . . . . . . . . . . . . . . . . . . . 13
     4.7.  Upgrading to SRTP  . . . . . . . . . . . . . . . . . . . . 13
     4.8.  Interworking with Other Signaling Protocols  . . . . . . . 14
     4.9.  Certificates . . . . . . . . . . . . . . . . . . . . . . . 14
   5.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 14
     5.1.  Key Management Protocol Requirements . . . . . . . . . . . 15
     5.2.  Security Requirements  . . . . . . . . . . . . . . . . . . 16
     5.3.  Requirements outside of the Key Management Protocol  . . . 19
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 20
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 20
     8.2.  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  . . . . . . . . . . . . . . . . . . . . . . 25
       A.1.4.  MIKEY-RSA-R  . . . . . . . . . . . . . . . . . . . . . 25
       A.1.5.  MIKEY-DHSIGN . . . . . . . . . . . . . . . . . . . . . 26
       A.1.6.  MIKEY-DHHMAC . . . . . . . . . . . . . . . . . . . . . 26
       A.1.7.  MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC)  . . . . . . . 26
       A.1.8.  SDP Security Descriptions with SIPS  . . . . . . . . . 26
       A.1.9.  SDP Security Descriptions with S/MIME  . . . . . . . . 27
       A.1.10. SDP-DH (Expired) . . . . . . . . . . . . . . . . . . . 27
       A.1.11. MIKEYv2 in SDP (Expired) . . . . . . . . . . . . . . . 27
     A.2.  Media Path Keying Technique  . . . . . . . . . . . . . . . 27
       A.2.1.  ZRTP . . . . . . . . . . . . . . . . . . . . . . . . . 27
     A.3.  Signaling and Media Path Keying Techniques . . . . . . . . 28
       A.3.1.  EKT  . . . . . . . . . . . . . . . . . . . . . . . . . 28
       A.3.2.  DTLS-SRTP  . . . . . . . . . . . . . . . . . . . . . . 28
       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 . . . . . . . . . . . 31
       A.4.3.  SSRC and ROC . . . . . . . . . . . . . . . . . . . . . 33



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     A.5.  Evaluation Criteria - Security . . . . . . . . . . . . . . 35
       A.5.1.  Distribution and Validation of Persistent Public
               Keys and Certificates  . . . . . . . . . . . . . . . . 35
       A.5.2.  Perfect Forward Secrecy  . . . . . . . . . . . . . . . 38
       A.5.3.  Best Effort Encryption . . . . . . . . . . . . . . . . 39
       A.5.4.  Upgrading Algorithms . . . . . . . . . . . . . . . . . 40
   Appendix B.  Out-of-Scope  . . . . . . . . . . . . . . . . . . . . 42
     B.1.  Shared Key Conferencing  . . . . . . . . . . . . . . . . . 42

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, and Section 5 lists requirements for media
   security.  The main part of the document concludes with the security
   considerations Section 6, and acknowledgements in Section 7.
   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.




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   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.

   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 requirements, where XXX is the
   requirement, which needs to be met.








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3.  Attack Scenarios

   The discussion in this section relates to requirements R-ASSOC
   (specified in Section 5.1) R-PASS-MEDIA, R-PASS-SIG, R-SIG-MEDIA,
   R-ACT-ACT, and R-ID-BINDING (specified in Section 5.2).

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

   1.  only to the media path,

   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 only to 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.



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   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.

   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 endpoints.

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

   As another example, SDP Security Descriptions [RFC4568], when
   protected by TLS (as it is commonly implemented and deployed), belong
   in the passive-signaling-passive-media category since the adversary
   needs to learn the SDP 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 (Datagram Transport Layer Security
   Extension for 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 RFC-4474 signature protects both the offer and the
   answer, and such a system would then belong to the active-signaling-
   active-media-detect category (provided, of course, the signaling path
   to the RFC-4474 authenticator and verifier is secured as per RFC
   4474, and the RFC-4474 authenticator and verifier are behaving as per
   RFC 4474).



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   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 a 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, 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 explained in detail in Section 5.  They are listed
   according to their association to the key management protocol, to
   attack scenarios, and requirements that 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 requirements R-AVOID-
   CLIPPING and R-ALLOW-RTP.

   Per the Session Description Protocol (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.





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   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 endpoint support Provisional Response
   ACKnowledgement (PRACK) [RFC3262], use TCP across all SIP proxies,
   implement Security Preconditions [RFC5027], or both ends implement
   Interactive Connectivity Establishment [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.

   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 [MIDDLEBOX]).  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.














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                                +-----+
                                |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
   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 Public Switched Telephone Network
   (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 retargeted.

   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., Multimedia
   Internet KEYing (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



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   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.

                               +-----+
                               |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 [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 the security of the media traffic, only the endpoint that
   answers the call should know the SRTP keys for the session.  Forked
   and retargeted 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 rekey immediately after forking or



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   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
   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 brokerages, 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., [SRTP-KEY]).







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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 Voice over IP (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
   that converts the VoIP conversation to a format understood by the
   other network.  An example of such a 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 truly 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
   reuse 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 Modules (ISIMs), and PSTN gateways.
   PSTN gateways typically utilize a Digital Signal Processor (DSP) that
   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.





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   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, it is necessary for network equipment to
   transcode from one codec (e.g., a highly compressed codec that makes
   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
   back-to-back user agent (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 [EARLY-MEDIA].

   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.









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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 that 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.

   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 that can be met inside the key
   management protocol or outside of the key management protocol
   (Section 5.3).



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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.

   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
             reuse of a previously established security context.

         Note: reuse of the security context does not imply reuse 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.




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   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 endpoints, 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 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).







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   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 [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 that 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).








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   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 that 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.

   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.

         Note: This allows domains to deploy SIP Identity [RFC4474].







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   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).

         Note: Failing to meet R-ACT-ACT indicates the protocol cannot
               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 endpoints of a forked or retargeted
                   call are incapable of using SRTP, a solution MUST be
                   described that 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, Media Gateway Control Protocol (MGCP).

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

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

   R-ALLOW-RTP:  A solution SHOULD be described that allows RTP media to
                 be received by the calling party until SRTP has been
                 negotiated with the answerer, after which SRTP is
                 preferred over RTP.







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6.  Security Considerations

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

7.  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, Frank Shearar, Srinath Thiruvengadam, Dave Ward, Dan York,
   and Phil Zimmermann.

8.  References

8.1.  Normative References

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





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

8.2.  Informative References

   [AISITSEC]     Bundesamt fuer Sicherheit in der Informationstechnik
                  [Federal Office of Information Security - Germany],
                  "Anwendungshinweise und Interpretationen (AIS) zu
                  ITSEC", January 2002,
                  <http://www.bsi.de/zertifiz/zert/interpr/
                  aisitsec.htm>.

   [DTLS-SRTP]    McGrew, D. and E. Rescorla, "Datagram Transport Layer
                  Security (DTLS) Extension to Establish Keys for Secure
                  Real-time Transport Protocol (SRTP)", Work
                  in Progress, October 2008.

   [EARLY-MEDIA]  Stucker, B., "Coping with Early Media in the Session
                  Initiation Protocol (SIP)", Work in Progress,
                  October 2006.

   [EKT]          McGrew, D., "Encrypted Key Transport for Secure RTP",
                  Work in Progress, July 2007.

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

   [ICE]          Rosenberg, J., "Interactive Connectivity Establishment
                  (ICE): A Protocol for Network Address  Translator
                  (NAT) Traversal for Offer/Answer Protocols", Work
                  in Progress, October 2007.

   [MIDDLEBOX]    Stucker, B. and H. Tschofenig, "Analysis of Middlebox
                  Interactions for Signaling Protocol Communication
                  along the Media Path", Work in Progress, July 2008.

   [MIKEY-ECC]    Milne, A., "ECC Algorithms for MIKEY", Work
                  in Progress, June 2007.

   [MIKEYv2]      Dondeti, L., "MIKEYv2: SRTP Key Management using
                  MIKEY, revisited", Work in Progress, March 2007.

   [MULTIPART]    Wing, D. and C. Jennings, "Session Initiation Protocol
                  (SIP) Offer/Answer with Multipart Alternative", Work
                  in Progress, March 2006.





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   [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.

   [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.






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   [RFC4949]      Shirey, R., "Internet Security Glossary, Version 2",
                  FYI 36, 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.

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

   [SDP-CAP]      Andreasen, F., "SDP Capability Negotiation", Work
                  in Progress, July 2008.

   [SDP-DH]       Baugher, M. and D. McGrew, "Diffie-Hellman Exchanges
                  for Multimedia Sessions", Work in Progress,
                  February 2006.

   [SIP-CERTS]    Jennings, C. and J. Fischl, "Certificate Management
                  Service for The Session Initiation Protocol (SIP)",
                  Work in Progress, November 2008.

   [SIP-DTLS]     Fischl, J., "Datagram Transport Layer Security (DTLS)
                  Protocol for Protection of Media Traffic Established
                  with the Session Initiation Protocol", Work
                  in Progress, July 2007.

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

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












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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.
   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
   that 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.




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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 Secure/Multipurpose Internet Mail Extensions
   (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.

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.





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A.1.5.  MIKEY-DHSIGN

   In MIKEY-DHSIGN [RFC3830], the offerer and answerer derive the key
   from a Diffie-Hellman (DH) 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
   additional media path messages.

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

   ECC Algorithms For MIKEY [MIKEY-ECC] describes how ECC can be used
   with MIKEY-RSA (using Elliptic Curve Digital Signature Algorithm
   (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.  SDP Security Descriptions with SIPS

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






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   SDP 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.  SDP 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 [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
   Connected 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 [MIKEYv2] adds mode negotiation to MIKEYv1 and removes the
   time synchronization requirement.  It 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.

   Furthermore, 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 [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



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   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 four media path messages (Hello,
   Commit, DHPart1, and DHPart2) to establish the SRTP key, and three
   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 [EKT] relies on another SRTP key exchange protocol, such as SDP
   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
   such as SDP Security Descriptions or MIKEY.  Each answerer sends an
   SRTCP message that 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 [DTLS-SRTP] exchanges public key fingerprints in SDP
   [SIP-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 four media path messages to establish the
   SRTP key.






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   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 [RFC5246].

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 document 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.

      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 where 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.



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      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.

      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.

      SDP 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.

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




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         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
         Secure Forking: 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 SDP 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.

         Secure Retargeting: Inherited from the bootstrapping mechanism
         (the specific MIKEY mode or SDP 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



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   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.

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

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

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

      SDP 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 initially uses RTP.  While RTP
         is flowing, both ends negotiate SRTP keys in the media path and
         then switch to using SRTP.








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      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 [MIDDLEBOX].

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

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 SDP 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.



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   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.

      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.

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

      SDP 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.



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      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.

      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 to 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




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   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 [SIP-CERTS] might be used to obtain the certificate of a peer.

      Note: Even if sip-certs [SIP-CERTS] were deployed, the retargeting
      problem (Appendix A.4.1) would still prevent successful deployment
      of keying techniques which require the 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 [SIP-CERTS] might be
         viable in the future.

         The offer may also contain a certificate for the offerer, 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.






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      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.

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

      SDP 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.







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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.

      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.

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

      SDP Security Descriptions with S/MIME
         Not applicable; SDP 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.




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      EKT
         No PFS.

      DTLS-SRTP
         PFS is provided if the negotiated cipher suite uses ephemeral
         keys (e.g., Diffie-Hellman (DHE_RSA [RFC5246]) or Elliptic
         Curve Diffie-Hellman [RFC4492]).

      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
      [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.







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   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-SRT-aware endpoints, but doesn't use the RTP/SAVP protocol
      registered by SRTP [RFC3711].

   SDP Capability Negotiation
      SDP Capability Negotiation [SDP-CAP] provides a backwards-
      compatible mechanism to allow offering both SRTP and RTP in a
      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 [SDP-CAP], 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 [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 computationally expensive because the offer needs to contain all
   combinations of the key exchange mechanisms (all MIKEY modes, SDP
   Security Descriptions), 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 that 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.








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   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.

      MIKEY-RSA
         For each offered SRTP crypto suite, the offerer has to perform
         RSA operation to encrypt the TGK (TEK Generation Key).

      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.

      SDP Security Descriptions with SIPS
         No significant computational expense.

      SDP 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".






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      EKT
         The offerer's computational expense depends entirely on the EKT
         bootstrapping mechanism selected (one or more MIKEY modes or
         SDP 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.

      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 the
   confidentiality of such conferences where listeners share a common
   key, all listeners must rekeyed when a listener joins or leaves a
   conference.















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   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.

   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 endpoint 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



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   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).








































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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
   Linnoitustie 6
   Espoo,   02600
   Finland

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


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

   EMail: audet@nortel.com













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