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Versions: 00 01 02 03 04 05 draft-ietf-sip-media-security-requirements

Network Working Group                                            D. Wing
Internet-Draft                                                     Cisco
Intended status: Informational                                  S. Fries
Expires: December 27, 2007                                    Siemens AG
                                                           H. Tschofenig
                                                  Nokia Siemens Networks
                                                           June 25, 2007


       Requirements for a Media Security Key Management Protocol
             draft-wing-media-security-requirements-04.txt

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   This Internet-Draft will expire on December 27, 2007.

Copyright Notice

   Copyright (C) The IETF Trust (2007).

Abstract

   A number of proposals have been published to address the need of
   securing media traffic.  Different assumptions, requirements, and
   usage environments justify every one of them.  This document aims to
   summarize the discussed media security requirements in order progress
   the work on identifying a small subset applicable to a large range of



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   deployment environments.

   This document is discussed on the RTPSEC mailing list,
   http://www.imc.org/ietf-rtpsec.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Discussion of Call Scenarios . . . . . . . . . . . . . . . . .  3
     3.1.  Clipping Media Before Signaling Answer . . . . . . . . . .  4
     3.2.  Retargeting and Forking  . . . . . . . . . . . . . . . . .  4
     3.3.  Shared Key Conferencing  . . . . . . . . . . . . . . . . .  7
   4.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  8
   5.  Requirements Classification  . . . . . . . . . . . . . . . . . 12
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   8.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 15
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 15
   Appendix A.  Out-of-Scope  . . . . . . . . . . . . . . . . . . . . 16
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
   Intellectual Property and Copyright Statements . . . . . . . . . . 18


























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1.  Introduction

   The work on media security started a long time ago where the
   capability of the Session Initiation Protocol (SIP) was still at 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 extend, considered necessary protocol work to ensure that the
   developed protocols indeed meet the previously envisioned needs for
   the users in the Internet.

   This document aims to summarize the discussed media security
   requirements, i.e., requirements for mechanisms that negotiate keys
   for SRTP.  Once the list of requirements and architectural aspects
   have been investigated, the work on the protocol proposals can be
   progressed by identifying a small number of soltuions and to complete
   the protocol work.

   This document is organized as follows.  Section 2 introduces
   terminology, Section 3 provides an overview about possible call
   scenarios, Section 4 lists requirements for media security, Section 5
   will provide a clustering of requirements to certain deployment
   environments to adress the problem that there might not be a single
   solution with universal applicability and Appendix A provides out-of-
   scope items and aspects for further discussion.  The document
   concludes with a security considerations Section 6, IANA
   considerations Section 7 and an acknowledgement section in Section 8.


2.  Terminology

   The keywords "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
   (referred as ', not its implementation or application.


3.  Discussion of Call Scenarios

   The following subsections describe call scenarios, which have been
   discussed elaborately.  These call scenarios pose the most challenge
   to the key management for media data in cooperation with SIP
   signaling.  The scenarios have already been described as part of the
   key management evaluation draft [I-D.wing-rtpsec-keying-eval], and
   are stated here to give a better insight in these discussion.



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3.1.  Clipping Media Before Signaling Answer

   Per the SDP Offer/Answer Model [RFC3264],

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

   To meet this requirement with SRTP, the offerer needs to know the
   SRTP key for arriving media.  If encrypted SRTP media arrives before
   the associated SRTP key, the offerer cannot play the media -- causing
   clipping.

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

   Requirements are created due to early media, in the sense of pre-
   offer/answer media, as described in [I-D.barnes-sip-em-ps-req-sol].
   Fixes to early media might make the requirements to become obsolete.

3.2.  Retargeting and Forking

   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

   The mechanism used by SIP for identifying the calling party is SIP
   Identity [RFC3261].  However, due to SIP retargeting issues
   [I-D.peterson-sipping-retarget], 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.  However, those built-in identity mechanism
   suffer from the same SIP retargeting problem described in the above
   draft.  Going forward, it is anticipated that Connected Identity
   [I-D.ietf-sip-connected-identity] may allow identifying the called
   party.  This is also described as the 'retargeting identity' problem.

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

















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

                             Figure 2: Forking

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

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

   To maintain security of the media traffic, only the end point that
   answers the call should know the SRTP keys for the session.  This is
   only an issue with public key encryption and not with DH-based
   approaches.  For key exchange mechanisms that do not provide secure
   forking or secure retargeting, one workaround is to re-key
   immediately after forking or retargeting (that is, perform a re-
   Invite).  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 Invite messages processed
   by the network.

   Retargeting securely introduces a more significant problem.  With
   retargeting, the actual recipient of the request is not the original
   recipient.  This means that if the offerer encrypted material (such
   as the session key or the SDP) using the original recipient's public
   key, the actual recipient will not be able to decrypt that material
   because the recipient won't have the original recipient's private
   key.  In some cases, this is the intended behavior, i.e., you wanted



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   to establish a secure connection with a specific individual.  In
   other cases, it is not intended behavior (you want all voice media to
   be encrypted, regardless of who answers).

   For some forms of key management the calling party needs to know in
   advance the certificate or shared secret of the called party, and
   retargeting can interfere with this.

   Further compounding this problem is a particularity 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)
   [I-D.mahy-sipping-herfp-fix], is difficult to solve in SIP.

3.3.  Shared Key Conferencing

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

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


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

                       Figure 3: Centralized Keying




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   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 end point and can set up a point-to-point secure
   session between the participant and the mixer.  This is the simplest
   situation, but is computationally wasteful, since SRTP processing has
   to be done independently for each participant.  The use of multiple
   inbound sessions (scenario a) doesn't waste computational resources,
   though it does consume additional cryptographic context on the mixer
   for each participant and has the advantage of non-repudiation of the
   originator of the incoming stream.

   To support a single outbound session (scenario d), the mixer has to
   dictate its encryption key to the participants.  Some keying
   mechanisms allow the transmitter to determine its own key, and others
   allow the offerer to determine the key for the offerer and answerer.
   Depending on how the call is established, the offerer might be a
   participant (such as a participant dialing into a conference bridge)
   or the offerer might be the mixer (such as a conference bridge
   calling a participant).  The use of offerless Invites may help some
   keying mechanisms reverse the role of offerer/answerer.  A
   difficulty, however, is knowing a priori if the role should be
   reversed for a particular call.


4.  Requirements

   R1:    Negotiation of SRTP keys MUST NOT cause the call setup to fail
          in forked and retargeted calls where all end points are
          willing to use SRTP, unless the execution of the
          authentication and key exchange protocol leads to a failure
          (e.g., an unsuccessful authentication attempt).



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   R2:    Even when some end points of a forked or retargeted call are
          incapable of using SRTP, the key management protocol MUST
          allow the establishment of SRTP associations with SRTP-capable
          endpoints and / or RTP associations with non-SRTP-capable
          endpoints.

   R3:    Forked end points MUST NOT know the SRTP key of any call
          established with another forked end point.

   R4:    The media security key management protocol MAY support the
          ability to utilize an initially established security context
          that was established as part of the first call setup with a
          remote end point.

          Specialized devices may need to avoid public key operations or
          Diffie-Hellman operations as much as possible because of the
          computational cost or because of the additional call setup
          delay.  For example, it can take a second or two to perform a
          Diffie-Hellman operation in certain devices.  Examples of
          these specialized devices would include some handsets,
          intelligent SIMs, and PSTN gateways.  For the typical case
          because a phone call has not yet been established, ancillary
          processing cycles can be utilized to perform the PK or DH
          operation; for example, in a PSTN gateway the DSP, which is
          not yet involved with typical DSP operations, could be used to
          perform the calculation, so as to avoid having the central
          host processor perform the calculation.  Some devices, such as
          handsets, and intelligent SIMs do not have such ancillary
          processing capability.

   R5:    The media security key management protocol SHOULD avoid
          clipping media before SDP answer without requiring PRACK
          [RFC3262].

   R6:    The media security key management protocol MUST provide
          protection against passive attacks on the media path.

   R7:    The media security key management protocol MUST provide
          protection against passive attacks of a SIP proxy that is
          legitimately routing SIP messages.

   R8:    The media security key management protocol MUST be able to
          support perfect forward secrecy (or PFS).  The term PFS is 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.





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   R9:    The media security key management protocol MUST support
          negotiation of SRTP cipher suites without incurring per-
          algorithm computational expense.  This allows an offer to be
          built without incurring computational expense for each
          algorithm.

   R10:   If SRTP keying is performed over the media path, the keying
          packets MUST NOT pass the RTP validity check defined in
          Appendix A.1 of [RFC3550].

   R11:   The media security key management protocol that utilizes
          expensive cryptographic computations SHOULD offer the ability
          to resume previous sessions in an efficient way.

   R12:   The media security key management protocol MUST NOT require
          3rd parties to sign certificates.

          This requirement points to the fact that a global PKI cannot
          be assumed and opportunistic security approaches should be
          considered as part of the solution.

   R13:   The media security key management protocol SHOULD use
          algorithms that allow FIPS 140-2 [FIPS-140-2] certification.

          Note that 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."[cryptval]

          Some commercial organizations, such as banks and defense
          contractors, also require or prefer equipment which has
          validated by the FIPS-140 process.

   R14:   The media security key management protocol SHOULD be able to
          associate the signaling exchange with the media traffic.

   R15:   For example, if using a Diffie-Hellman keying technique with
          security preconditions that forks to 20 end points, the call
          initiator would get 20 provisional responses containing 20
          signed Diffie-Hellman key pairs.  Calculating 20 DH secrets
          and validating signatures can be a difficult task depending on
          the device capabilities.  Hence, in the case of forking, it is



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          not desirable to perform a DH or PK 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 needs to be completed.  This might be
          done by using the transport address indicated in the SDP,
          although NATs can complicate this association.

   R14:   The media security key management protocol SHOULD allow to
          start with RTP and then upgrade to SRTP.

   R15:   The media security key management protocol SHOULD NOT
          introduce new denial of service vulnerabilities.

   R16:   The media security key management protocol SHOULD require the
          adversary to have access to the data as well as the signaling
          path for a successful attack to be launched.  An adversary
          that is located only along the data or only along the
          signaling path MUST NOT be able to successfully mount an
          attack.  A successful attack refers to the ability for the
          adversary to obtain keying material to decrypt the SRTP
          encrypted media traffic.

   R17:   If two parties share an authentication infrastructure that has
          3rd parties signing certificates, they SHOULD be able to make
          use of it.

   R18:   The media security key management protocol MUST provide
          crypto-agility.

   R19:   The media security key management protocol MUST protect cipher
          suite negotiation against downgrading attacks.

   R20:   The media security key management protocol MUST allow a SIP
          User Agent to negotiate media security parameters for each
          individual session.

   R21:   The media security key management protocol SHOULD allow
          establishing SRTP keying between different call signaling
          protocols (e.g., between Jabber, SIP, H.323, MGCP)

   R22:   The media security key management protocol SHOULD support
          recording of decrypted media.

          Media recording may be realized by an intermediate nodes.  An
          example for those intermediate nodes are devices, which could
          be used in banking applications or for quality monitoring in
          call centers.  Here, it must be ensured, that the media



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          security is ensured by the intermediate nodes on the
          connections to the involved endpoints as originally
          negotiated.  The endpoints need to be informed that there is
          an intermediate device and need to cooperate.  A solution
          approach for this is described in [I-D.wing-sipping-srtp-key].

   R23:   The media security key management protocol SHOULD NOT allow
          end users to determine whether their end-to-end interaction is
          subject to lawful interception.

   R24:   The media security key management protocol MUST work when
          there are intermediate nodes, terminating or processing media,
          between the end points.

   R25:   The media security key management protocol MUST consider
          termination of media security in a PSTN gateway.

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

          If media security (e.g., SRTP protection) is employed in this
          kind of gateway-setting, then media security and the related
          key management needs to be terminated at the 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 end-
          to-end between the communicating entities.


5.  Requirements Classification

   An adversary might be located along

   1.  the media path,

   2.  the signaling path,

   3.  the media and the signaling path.

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



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   Furthermore, it is reasonable to consider the capabilities of the
   adversary.  We also have different types of adversaries, namely

   a.  active adversary

   b.  passive adversary

   Note that the adversary model for (a) and (b) also assumes the
   attacker being able to control SIP signaling entities.

   With respect to item (a) an adversary may need to be active with
   regard to the key exchange relevant information traveling along the
   data or the signaling path.

   Some of the deployment variants of the media security key management
   proposals under considerations do not provide protection against man-
   in-the-middle adversaries under certain conditions, for example when
   SIP signaling entities are compromised, when a global PKI is missing
   or pre-shared secrets are not exchanged between the end points prior
   to the protocol exchange.

   Based on the above-mentioned considerations the following
   classifications can be made:

   Class I:

      Passive attack on the signaling and the data path sufficient to
      reveal the content of the media traffic.


   Class II:

      Active attack on the signaling path and passive attack on the data
      path to reveal the content of the media traffic.


   Class III:

      Active attack on the signaling and the data path necessary to
      reveal the content of the media traffic.


   Class IV:

      Active attack is required and will be detected by the end points
      when adversary tampers with the messages.

   For example, SDES falls into Class I since the adversary needs to



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   learn the SDES key by progressing a signaling message at a SIP proxy
   (assuming that the adversary is in control of the SIP proxy).
   Subsequent media traffic can be decrypted with the help of the
   learned key.

   As another example, DTLS-RTP falls into Class III when DTLS is used a
   public key based ciphersuite with self-signed certificates and
   without SIP Identity.  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.

   An attack is not successful when SIP Identity is used, the adversary
   is not between the SIP UA and its Authentication Service (or at the
   Authentication Service), both end points are able to verify the
   digital signature (of the SIP Identity) and are able to validate the
   corresponding certificates.


6.  Security Considerations

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


7.  IANA Considerations

   This document does not require actions by IANA.


8.  Acknowledgements

   The authors would like to thank the active participants of the RTPSEC
   BoF and on the RTPSEC mailing list.  The authors would furthermore
   like to thank Wolfgang Buecker, Guenther Horn, Peter Howard, Hans-
   Heinrich Grusdt, Srinath Thiruvengadam, Martin Euchner, Eric
   Rescorla, Matt Lepinski, Dan York, Werner Dittmann, Richard Barnes,
   Vesa Lehtovirta, Colin Perkins, Peter Schneider, and Christer
   Holmberg for their feedback to this document.  We would like to
   particularly thank Francois Audet for his work on the evaluation of
   various media security key exchange proposals.


9.  References







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9.1.  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.

   [cryptval]
              NIST, "Cryptographic Module Validation Program",
              December 2006,
              <http://csrc.nist.gov/cryptval/140-2APP.htm>.

9.2.  Informative References

   [I-D.barnes-sip-em-ps-req-sol]
              Barnes, R. and M. Lepinski, "Early Media in SIP: Problem
              Statement, Requirements, and Analysis of  Solutions",
              draft-barnes-sip-em-ps-req-sol-00 (work in progress),
              February 2007.

   [I-D.ietf-mmusic-ice]
              Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address  Translator (NAT)
              Traversal for Offer/Answer Protocols",
              draft-ietf-mmusic-ice-16 (work in progress), June 2007.

   [I-D.ietf-mmusic-securityprecondition]
              Andreasen, F. and D. Wing, "Security Preconditions for
              Session Description Protocol (SDP) Media  Streams",
              draft-ietf-mmusic-securityprecondition-03 (work in
              progress), October 2006.




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   [I-D.ietf-sip-connected-identity]
              Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", draft-ietf-sip-connected-identity-05
              (work in progress), February 2007.

   [I-D.mahy-sipping-herfp-fix]
              Mahy, R., "A Solution to the Heterogeneous Error Response
              Forking Problem (HERFP) in  the Session Initiation
              Protocol (SIP)", draft-mahy-sipping-herfp-fix-01 (work in
              progress), March 2006.

   [I-D.peterson-sipping-retarget]
              Peterson, J., "Retargeting and Security in SIP: A
              Framework and Requirements",
              draft-peterson-sipping-retarget-00 (work in progress),
              February 2005.

   [I-D.wing-rtpsec-keying-eval]
              Audet, F. and D. Wing, "Evaluation of SRTP Keying with
              SIP", draft-wing-rtpsec-keying-eval-02 (work in progress),
              February 2007.

   [I-D.wing-sipping-srtp-key]
              Wing, D., "Disclosing Secure RTP (SRTP) Session Keys with
              a SIP Event Package", draft-wing-sipping-srtp-key-00 (work
              in progress), February 2007.

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


Appendix A.  Out-of-Scope

   Discussions concluded that key management for shared-key encryption
   of conferencing is outside the scope of this document.  As the
   priority is point-to-point unicast SRTP session keying, resolving
   shared-key SRTP session keying is deferred to later and left as an
   item for future investigations.












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Authors' Addresses

   Dan Wing
   Cisco
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Email: dwing@cisco.com


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

   Email: steffen.fries@siemens.com


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

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























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