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Versions: 00 01 02 draft-ietf-perc-private-media-framework

Network Working Group                                     P. Jones (Ed.)
Internet Draft                                                 N. Ismail
Intended status: Standards Track                               D. Benham
Expires: April 9, 2016                                     Cisco Systems
                                                         October 9, 2015



      A Solution Framework for Private Media in Privacy Enhanced RTP
                               Conferencing
                draft-jones-perc-private-media-framework-01


Abstract

   This document describes a solution framework for ensuring that media
   confidentiality and integrity are maintained end-to-end within the
   context of a switched conferencing environment where media
   distribution devices are not trusted with the end-to-end media
   encryption keys.  The solution aims to build upon existing security
   mechanisms defined for the real-time transport protocol (RTP).

Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 9, 2016.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors. All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1. Introduction...................................................2
   2. Requirements Language..........................................3
   3. Private Media Trust Model......................................3
   4. Solution Framework Overview....................................3
      4.1. End-to-End Media Privacy..................................4
      4.2. Hop-by-Hop Operations.....................................4
   5. Private Media Packet Format....................................5
   6. SRTP Cryptographic Context.....................................7
   7. Cryptographic Operations.......................................8
      7.1. Hop-by-Hop Authentication and Optional Encryption.........8
      7.2. End-to-End Media Payload Encryption and Authentication....8
         7.2.1. End-to-End Cryptographic Context Considerations......9
   8. Key Exchange..................................................11
      8.1. Session Signaling........................................11
      8.2. Negotiating SRTP Protection Profiles and Key Exchange....12
         8.2.1. Endpoint and KMF....................................12
         8.2.2. MDD and KMF.........................................15
   9. Changing Media Forwarded and EKT Field........................16
   10. To-Do List...................................................16
      10.1. What is needed to realize this Framework................16
      10.2. Other Considerations for this Framework.................17
   11. IANA Considerations..........................................17
   12. Security Considerations......................................17
   13. References...................................................17
      13.1. Normative References....................................17
      13.2. Informative References..................................18
   14. Acknowledgments..............................................19
   Authors' Addresses...............................................19


1. Introduction

   Switched conferencing is an increasingly popular model for multimedia
   conferences with multiple participants using a combination of audio,
   video, text, and other media types.  With this model, real-time media
   flows from conference participants are not mixed, transcoded,
   transrated, recomposed, or otherwise manipulated by a media
   distribution device (MDD), as might be the case with a traditional
   media server or multipoint control unit (MCU).  Instead, media flows
   transmitted by conference participants are simply forwarded by the
   MDD to each of the other participants, often forwarding only a subset
   of flows based on voice activity detection or other criteria.  In
   some instances, the switching MDDs may make limited modifications to
   RTP [RFC3550] headers, for example, but the actual media content
   (e.g., voice or video data) is unaltered.




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   An advantage of switched conferencing is that MDDs can be deployed on
   general-purpose computing hardware.  This, in turn, means that it is
   possible to deploy switching MDDs in virtualized environments,
   including private and public clouds. Deploying conference resource in
   a cloud environment might introduce a higher security risk.  Whereas
   traditional conference resources were usually deployed in private
   networks that were protected, cloud-based conference resources might
   be viewed as less secure since they are not always physically
   controlled by those who use the hardware.  Additionally, there are
   usually several ports open to the public in cloud deployments, such
   as for remote administration, and so on.

   Recognizing the need to improve the way in which media
   confidentiality is ensured, requirements for private media were
   specified in [I.D-draft-jones-perc-private-media-reqts].  Attempting
   to meet those requirements, this document defines a solution
   framework wherein privacy is ensured by making it impossible for an
   MDD to gain access to keys needed to decrypt or authenticate the
   actual media content sent between conference participants.  At the
   same time, the framework allows for the switching MDD to modify
   certain RTP headers; add, remove, encrypt, or decrypt RTP header
   extensions; and encrypt and decrypt RTCP packets.  The framework also
   prevents replay attacks by authenticating each packet transmitted
   between a given participant and the switching MDD by using a key that
   is independent from the media encryption and authentication key(s)
   and is unique to the participating endpoint and the switching MDD.

   A goal of this framework is to meet the referenced requirements and
   stated objectives by utilizing existing security procedures defined
   for RTP with minimal extensions.

2. Requirements Language

   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 RFC 2119 [RFC2119]
   when they appear in ALL CAPS.  These words may also appear in this
   document in lower case as plain English words, absent their normative
   meanings.

3. Private Media Trust Model

   The private media trust model is specified in [I.D-draft-jones-perc-
   private-media-reqts].

4. Solution Framework Overview

   The purpose for this framework is to define a means through which
   media privacy can be ensured when communicating within a switched
   conferencing environment consisting of one or more centrally located
   media distribution devices.  This framework specifies the reuse of



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   several technologies, including SRTP [RFC3711], EKT [I.D-draft-ietf-
   avtcore-srtp-ekt], and DTLS-SRTP [RFC5764].  For the purposes of this
   document, a conference refers to any session with two or more
   participants - endpoints or other trusted devices - exchanging RTP
   flows through media distribution devices.

4.1. End-to-End Media Privacy

   This framework does not attempt to hide the fact that communication
   between parties takes place.  Rather, it only addresses the end-to-
   end confidentiality and integrity of the actual media content.

   To ensure the confidentiality and integrity of RTP media packets,
   endpoints will utilize an EKT key - known to all conference
   participants - to encrypt the SRTP key that is used to encrypt the
   media (i.e., the RTP payload) via authenticated encryption.

   Note that this EKT key may need to change from time-to-time during
   the life of a conference, such as when a new participant joins or
   leaves a conference.  Dictating when a conference is to be re-keyed
   is outside the scope of this document, but this framework does enable
   re-keying of the conference.

   Endpoints MUST maintain a list of SSRCs, track received sequence
   number values relating to those SSRCs, and maintain associated SRTP
   master keys for those SSRCs.  All of this information SHOULD be
   retained for some reasonable period of time and SHOULD be discarded
   shortly after the EKT key for the conference is changed and upon
   leaving the conference.  However, following a change of the EKT key,
   old key material SHOULD be retained long enough to ensure that late-
   arriving or out-of-order packets can be successfully played.

4.2. Hop-by-Hop Operations

   To ensure the integrity of transmitted media packets, this framework
   requires that every packet be authenticated hop-by-hop.  The
   authentication key used for hop-by-hop authentication is derived from
   an SRTP master key shared only on the respective hop between the
   endpoint and the MDD to which it is attached.  If MDDs are cascaded,
   then there will also be an SRTP master key and derived authentication
   key shared between the cascaded servers.  Importantly, each of these
   keys is distinct per hop and no two hops ever intentionally use the
   same SRTP master key.

   MDDs may find it necessary to change certain parts of the RTP packet
   header, add or remove RTP header extensions, etc.  By using hop-by-
   hop authentication, the MDD is given liberty to change certain values
   present in the RTP header, such as the payload type value.

   If there is a desire to encrypt RTP header extensions, an encryption
   key is derived from the hop-by-hop SRTP master key to encrypt header



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   extensions as per [RFC6904].  This will give the switching MDD
   visibility into header extensions, such as the one used to determine
   audio level [RFC6464] of conference participants.  Note that allowing
   RTP header extensions to be encrypted requires that all hops decrypt
   and re-encrypt any encrypted header extensions.

   RTCP is optionally encrypted and mandatorily authenticated hop-by-hop
   using the encryption and authentication keys derived from the SRTP
   master key for the hop.  This gives the switching MDD the flexibility
   of either forwarding RTCP packets unchanged, transmit compound RTCP
   packets, or to create RTCP packets to report statistics or for
   conference control.

   One of the reasons for performing hop-by-hop authentication is to
   provide replay protection.  If a media packet is replayed to the
   switching MDD, it will be detected and rejected.  Likewise, the
   endpoint can detect replayed packets originally sent by the MDD.
   Packets received by an endpoint that were originally sent to a
   different endpoint will fail to pass authentication checks.

5. Private Media Packet Format

   Since the RTP packet payload is encrypted and authenticated end-to-
   end, extensions optionally encrypted hop-by-hop, and the entire RTP
   packet is authenticated hop-by-hop, it may be useful to see the
   entire RTP packet similarly to what is shown in [RFC3711].




























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      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   A |V=2|P|X|  CC   |M|     PT      |       sequence number         | |
   A +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   A |                           timestamp                           | |
   A +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   A |           synchronization source (SSRC) identifier            | |
     +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
     |            contributing source (CSRC) identifiers             | |
     |                               ....                            | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                   RTP extension (OPTIONAL*)                   | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |                          payload  ...                         | |
   | |                               +-------------------------------+ |
   | |                               | RTP padding   | RTP pad count | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |                        EKT ciphertext ...                     | |
   | |                               +-------------------------------+ |
   | |                               |  SRTP ROC (upper 16 bits)     | |
   | +-------------------------------+-------------------------------+ |
   | |  SRTP ROC (lower 16 bits)     |  Security Parameter Index   |1| |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   | :                 authentication tag (MANDATORY)                : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   |                                                                   |
   +-- Authenticated Encryption                Authenticated Portion --+
       End-to-End (A = Associated Data)         using Hop-by-Hop Key

       * Header extensions are optionally Encrypted Hop-by-Hop

          Figure 1 - Private Media SRTP Packet with Full EKT Field

   The rollover counter (ROC) value is shown and transmitted as
   plaintext.  This is necessary since a switching MDD may not transmit
   media from a "silent" participant's endpoint to others in the
   conference for a long period of time.  When media from that
   previously "silent" participant is later forwarded to others, the
   receiving endpoint(s) would not otherwise know the value of the
   rollover counter.  Further, this value is needed so that the correct
   authentication tag can be generated hop-by-hop.  Since the expected
   length of the EKT Field might not be known to the MDD that is only
   authenticating packets, the ROC field is placed as shown to ensure
   that its location can be consistently determined.

   The EKT Field shown in Figure 1 is the "Full EKT Field".  The "Short
   EKT Field" may also be present in its place.  When the short EKT
   Field is used, the packet format looks like the one shown in Figure
   2.




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      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   A |V=2|P|X|  CC   |M|     PT      |       sequence number         | |
   A +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   A |                           timestamp                           | |
   A +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   A |           synchronization source (SSRC) identifier            | |
     +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
     |            contributing source (CSRC) identifiers             | |
     |                               ....                            | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                   RTP extension (OPTIONAL*)                   | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |                          payload  ...                         | |
   | |                                                               | |
   | |-------------------------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | | RTP padding   | RTP pad count |  Security Parameter Index   |0| |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | :                 authentication tag (MANDATORY)                : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   |                                                                   |
   +-- Authenticated Encryption                Authenticated Portion --+
       End-to-End (A = Associated Data)         using Hop-by-Hop Key

       * Header extensions are optionally Encrypted Hop-by-Hop

          Figure 2 - Private Media SRTP Packet with Short EKT Field

   Note that the ROC field is absent when the short EKT Field is
   transmitted.  The assumption is that endpoints will transmit the Full
   EKT Field regularly and frequently (e.g., every 100ms) to ensure that
   media transmitted by a previously "silent" participant's endpoint can
   be properly decrypted by other endpoints within a period of time that
   is not noticeable to the human user.

6. SRTP Cryptographic Context

   For any given media source identified by its SSRC, there is a single
   SRTP cryptographic context as described in Section 3.2 of [RFC3711]
   used in this framework.

   For end-to-end encryption, this framework extends the parameter set
   of the cryptographic context by adding an identifier for the end-to-
   end authenticated encryption algorithm.  That parameter has
   associated with it an EKT key (and associated EKT information, such
   as master salt, key length, etc.), one or more SRTP master keys, and
   as outlined in Section 3.2.1 of [RFC3711], other associated values
   that relate to the master keys (e.g., master salt and key length
   values).




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   For hop-by-hop encryption, the existing parameters in the SRTP
   cryptographic context are used, including for the optional encryption
   of RTP header extensions, authentication tag generation, etc.

7. Cryptographic Operations

7.1. Hop-by-Hop Authentication and Optional Encryption

   For operations that occur hop-by-hop, the cryptographic transforms
   defined in SRTP [RFC3711] (or other standardized transforms) may be
   used in order optionally encrypt RTP header extensions, authenticate
   the RTP packet, optionally encrypt the RTCP packet, and to
   authenticate the RTCP packet.

   The encryption and authentication of the RTP payload (media content)
   itself is not a hop-by-hop operation, as explained in the next
   section.

   The procedures for optionally encrypting RTP header extensions is
   define in [RFC6904] and MUST be used when encrypting header
   extensions using the hop-by-hop SRTP master key to derive the k_he
   and k_hs values.

   The procedures for authenticating the RTP packet, optionally
   encrypting the RTCP packet, and for authenticating the RTCP packet
   shall follow the procedures defined in [RFC3711] using the hop-by-hop
   SRTP master key and master salt to derive additional keys as
   specified in that specification.

7.2. End-to-End Media Payload Encryption and Authentication

   This section covers the encryption and authentication of the RTP
   payload (i.e., media content) using the SRTP master key(s) created by
   the endpoint and securely conveyed in the EKT Field using the EKT
   key(s) shared with each of the endpoints in the conference.

   This framework requires that the end-to-end cryptographic transforms
   use authenticated encryption with associated data (AEAD) algorithms.
   Specifically, the transforms defined in [I.D-draft-ietf-avtcore-srtp-
   aes-gcm] are used as the mandatory transforms in this framework.

   The procedures followed to encrypt the payload are those described in
   [I.D-draft-ietf-avtcore-srtp-aes-gcm], except that the associated
   data used specified in Section 8.2 is redefined as follows:

     Associated Data: The version V (2 bits), padding flag P (1 bit),
                      CSRC count CC (4 bits), marker M (1 bit), the
                      sequence number (16 bits), timestamp (32 bits),
                      SSRC (32 bits), and optional contributing source
                      identifiers (CSRCs, 32 bits each).




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   To consume the associated data, the endpoint copies the RTP header
   minus any header extensions and sets the payload type (PT) and
   extension (X) bits to zero.

   The authentication tag for the end-to-end encrypted payload
   immediately follows the encrypted payload in the defined packet
   format (see section 5).

   Note that RTP header extensions are not encrypted as a part of the
   end-to-end operations.  Rather, they are optionally encrypted as a
   hop-by-hop operation as explained in the previous section.

   Only a part of the RTP header is authenticated with the above
   definition of "Associated Data" since there is a desire to allow
   switching MDDs to make changes to certain parts of the RTP header.
   For this reason, there is still a need for an authentication tag as
   defined in [RFC3711] to be placed at the end of the RTP packet.  This
   authentication tag is provided via the hop-by-hop authentication
   operation as discussed in the previous section.  Note that this is
   also a deviation from what [I.D-draft-ietf-avtcore-srtp-aes-gcm]
   recommends, but is necessary to allow the switching MDD to make
   changes to certain fields that would otherwise be protected and to
   ensure that the RTP header extension and EKT Field are not
   manipulated in transit without detection.

7.2.1. End-to-End Cryptographic Context Considerations

7.2.1.1. Initialization Vector Formation

   SRTP defines the following Initialization Vector (IV) as part of the
   context for the AES Counter Mode cipher when encrypting RTP packets:

   SRTP IV =
       (SALT << 16) XOR (SSRC << 64) XOR (ROC << 32) XOR (SEQ << 16)

   Following similar logic, [I.D-draft-ietf-avtcore-srtp-aes-gcm]
   defines an Initialization Vector for encrypting RTP as follows:

   SRTP IV =
       SALT XOR (0x00 || 0x00 || SSRC ||ROC || SEQ)

   Since this context includes and makes use of the SSRC, SEQ, and ROC,
   these parameters must be preserved end-to-end for proper cipher
   operation.  In some RTP topologies, for example, a Selective
   Forwarding Middlebox (SFM) with a common SSRC space, these parameters
   are preserved end-to-end because RTP middleboxes do not alter them.
   In other RTP topologies, such as a Media Switching Mixer or a SFM
   with separate SSRC spaces, the RTP middleboxes may need to alter them
   for proper operation.  When any of these parameters is altered, the
   original parameters must still be preserved elsewhere in the packet
   since they are essential parts of the cipher context.  An RTP header



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   extension and EKT Security Parameter Index prefix for End-to-End IV
   (EEIV) is defined for forwarding this original context in the
   following sections.

7.2.1.2. End-to-End IV (EEIV) RTP Header Extension

   The End-to-End IV (EEIV) RTP Header Extension [RFC5285] conveys some
   or all of the original end-to-end cipher context parameters: SSRC,
   SEQ, and ROC.  The extension has the following format:

       EEIV = SSRC || SEQ

   It is not necessary that both the SSRC and SEQ values be present in
   the header extension if only one was changed in the header.  The
   components present can be determined by the length of the EEIV
   extension.  A length of two octets indicates the original SEQ is in
   the field.  A length of four octets indicates that the original SSRC
   is present.  Finally, a length of six octets indicates that both the
   original SSRC and SEQ are present.

   The extension MUST be added (if absent) by any MDD that alters these
   parameters in the RTP header.  It MUST NOT be added, removed or
   altered if already present, except when a secondary MDD changes
   either the SSRC or SEQ and the original value is not already present
   in the EEIV extension, it must add that changed value to the
   extension to form a complete and proper EEIV value.  Endpoints MAY
   add this extension to operate with a RTP MDD that can forward the
   extension, but not add such an extension itself.

   When an endpoint receives this extension in an SRTP packet, the
   endpoint MUST use these values as the SSRC and RTP sequence number
   when performing the authenticated decryption step as opposed to the
   values found in the RTP header.

   Note that he ROC does not need to be present in the header extension
   since it is included in the payload as proposed in Section 5.

7.2.1.3. End-to-End IV (EEIV) EKT Security Parameter Index Prefix

   As an alternative to an RTP header extension, the EEIV is placed in
   the payload of the RTP packet just before the Security Parameter
   Index (SPI).  Just before the SPI would be a single EEIV length
   indicator.  This value would be zero if the EEIV is not present, 2 if
   the SEQ is present, four if the SSRC is present, and six if both the
   SSRC and SEQ are present.

   Except for the location of the EEIV, the rules for the MDD and
   endpoint receiving the EEIV would be the same as described in the
   previous section.





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   As an alternative to those procedures, we could require that the
   sender always insert the complete EEIV into every packet.  This would
   remove the need to have a length octet, as the length would always be
   six octets.  It also removed the need for an MDD to perform any
   special operations with respect to the SSRC or RTP sequence number.

   The recipient of the packet would use the RTP header to perform hop-
   by-hop operations (e.g., packet authentication and SRTCP
   authentication and decryption), but would use the EEIV in the payload
   to decrypt the actual media content.

7.2.1.4. Simplification of the EKT Field

   If we allow MDDs to alter the original SSRC value, there is no value
   in carrying that original value inside the Full EKT Field. Thus, we
   would suggest removing it from there, carrying it only in the EEIV.

8. Key Exchange

   Within this framework, there are various keys each endpoint needs:
   those for end-to-end encryption/authentication and those for hop-by-
   hop authentication, optional encryption of RTP header extensions,
   SRTCP authentication, and optional SRTCP encryption.  Likewise, the
   MDD needs a hop-by-hop key when communicating with an endpoint or
   cascaded conference server.  The challenge is in securely exchanging
   these keys between the appropriate entities.

   To facilitate key exchange, this framework utilizes DTLS-SRTP and
   procedures defined in EKT.  This is explained further in the
   following sub-sections.

8.1. Session Signaling

   The session signaling protocol is not significant to this
   specification, since the call processing functions are not assumed to
   be trusted.  Signaling might be via SIP [RFC3261] or a proprietary
   signaling between a browser and a server, as examples.  What is
   important is that the signaling convey, in some manner, the
   fingerprint of the endpoint's certificate that will be used with
   DTLS-SRTP.  For the sake of providing a more concrete discussion, it
   is assumed that SIP is used and SDP [RFC4566] conveys the fingerprint
   information per [RFC5763].

   The endpoint ("User Agent" in SIP terminology) will send an INVITE
   message containing SDP for the media session along with the
   endpoint's certificate fingerprint.  This message or part thereof
   MUST be cryptographically signed so as to prevent unauthorized,
   undetectable modification of the fingerprint value, or the message
   MUST be sent to a trusted element over a secure connection.





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   For this example, it is assumed that the endpoint sends a message to
   a call processing function (e.g., a B2BUA) over a TLS connection.
   The B2BUA might sign the message using the procedures described in
   [RFC4474] for the benefit of forwarding the message to other
   entities.  It's important to note, however, that this does not lend
   to the security of media, as the call processing function is not
   assumed to be trusted.

   An associated Key Management Function (KMF) needs to receive
   information about the call and the endpoint(s).  This might be
   performed via an interface between the endpoint and the KMF, the call
   processing function and the KMF, or it might be via a signaling
   interface between the MDD and the KMF (see Figure 6 in [I.D-draft-
   jones-perc-private-media-reqts]).

   Regardless of the exact method, it is important that the endpoint's
   certificate fingerprint and a participant identifier (a random value
   created by the endpoint and provided to the KMF for each RTP session)
   are securely conveyed to the KMF.  The client certificate and
   participant identifier will allow the KMF to associate the DTLS
   connection to the specific endpoint and RTP session for the
   conference.

   Ultimately, a call is established to a conference and the endpoint
   receives address information to which it may establish one or more
   RTP sessions through to a MDD.

   Call signaling going back to the endpoint might contain the
   certificate fingerprint of the KMF that will process DTLS-SRTP
   messages.  Alternatively, the endpoint might already know the
   certificate fingerprint.  Whatever mechanism is employed, it is
   extremely vital that the endpoint be able to fully trust the validity
   of the fingerprint information for the KMF.

8.2. Negotiating SRTP Protection Profiles and Key Exchange

8.2.1. Endpoint and KMF

   There is a need for an SRTP master key and STRP master salt for hop-
   by-hop authentication and optional encryption known to the endpoint
   and the MDD.  Additionally, there is a need to exchange an EKT master
   key and EKT master salt for the end-to-end encryption of the media
   content that is known to all participants in the conference, but not
   known to the switching MDD.

   To convey keys, the endpoint uses the procedures defined in [I.D-
   draft-ietf-avtcore-srtp-ekt] for DTLS-SRTP over the media ports for
   the RTP session.  However, the switching MDD does not terminate the
   DTLS signaling.  Rather, DTLS packets received by the switching MDD
   are forwarded to the KMF and vice versa.  The figure below depicts
   this.



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                                  +----------------------------+
          +-----+                 |       Switching MDD        |
          |     |                 |                            |
          | KMF |<--------------->|<------------+ (Tunnels     |
          |     |     DTLS-       |             v  DTLS-SRTP)  |
          +-----+     SRTP        +----------------------------+
                      Tunnel                    ^
                                                |
                                                | DTLS-SRTP
                                                |
                                                v
                                           +----------+
                                           | Endpoint |
                                           +----------+

                    Figure 3 - DTLS-SRTP Tunneled to KMF

   Through this tunneled DTLS-SRTP exchange, an EKT master key and EKT
   master salt are conveyed from the KMF to the endpoint, which the
   endpoint will use when conveying SRTP keys and encrypt and
   authenticate the media content in SRTP packets.  The DTLS-SRTP
   message exchanges between the switching MDD and KMF are encapsulated
   in a second DTLS connection wherein the KMF also provides the MDD
   with the hop-by-hop key material.

   The KMF is described as a logical function in this document where the
   functionality needed might be provided by one or more physical or
   virtual entities.  For example, there would obviously be a device
   needed to terminate the DTLS-SRTP signaling.  However, that device
   may or may not be in possession of the EKT key used for the
   conference.  There DTLS-SRTP termination function might interface
   with a Key Management Server (KMS), such as the one described in
   [I.D-draft-abiggs-saag-key-management-service].

   {Editor's Note: a DTLS encapsulation protocol has been selected, but
   has not been published in a separate draft.  If there is no objection
   to this approach, a proposal draft for the tunneling protocol will be
   prepared.}

   The endpoint does not transmit media encryption keys to the KMF.  The
   endpoint will follow the procedures specified in the EKT
   specification to generate an SRTP master key and convey this
   information to conference participants periodically (and anytime an
   I-Frame is explicitly requested) via the "Full EKT Field."

   This tunneling approach also needs an extension to EKT in order to
   negotiate the SRTP Protection Profile used for end-to-end encryption
   and authentication.  The RECOMMENDED default protection profile is
   AEAD_AES_128_GCM [I.D-draft-ietf-avtcore-srtp-aes-gcm].





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   The DTLS-SRTP procedures will result in the determination of an SRTP
   master key and master salt, along with an SRTP Protection Profile.
   This information is used for the hop-by-hop operations.

   During the lifetime of a conference, the KMF MAY send a new EKT
   message to endpoints providing a new EKT key to use from that point
   forward, which might be desired when an endpoint leaves a conference
   for example.

   If a new endpoint joins a conference and does not support the same
   SRTP Protection Profile in use, the KMF must initiate a new DTLS-SRTP
   handshake with all conference participants to negotiate a new
   security profile and to re-key the conference.  This may cause some
   disruption to conference.  Therefore, it is recommended that only a
   small number of protection profiles be required to implement by all
   endpoints.

   To help in understanding better the sequence of messages and the
   relationship between the endpoint, MDD, and KMF, consider the
   following figure:


































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   Endpoint                    KMF                        MDD
      |                         |                          |
      | External Signaling      |                          |
      | to exchange Cert and    |                          |
      | and participant ID      |                          |
      | ----------------------> |                          |
      |                         |                          |
      | DTLS connection         |                          |
      | -------------------------------------------------> | \
      |                         | <======================= | /
      |                         |              DTLS Tunnel |
      |                         |                          |
      | For brevity, === is used for tunneled DTLS messages to KMF
      |                         |                          |
      | DTLS-SRTP and EKT       |                          |
      | ======================> |                          |
      |  (Participant ID, cert, |                          |
      |   security profiles,    |                          |
      |   etc.)                 |                          |
      |                         |                          |
      | <=====================> |                          |
      |         SRTP Master key | -----------------------> |
      |     and salt determined | SRTP Master keys         |
      |          for hop-by-hop | and salts conveyed       |
      |                         | for hop-by-hop           |
      | <====================== | (interface and           |
      |        EKT key conveyed |  endpoint/conference     |
      |          for end-to-end |  association TBD)        |
      |                         |                          |
      |                         |                          |

                      Figure 4 - Key Exchange Procedure

   Following the key exchange, the endpoint will be able to encrypt
   media end-to-end and authenticate packets hop-by-hop.  Likewise, the
   conference server will be able to authenticate the received packet at
   the hop, but will have no visibility into the encrypted media
   content.

8.2.2. MDD and KMF

   The DTLS tunnel between the MDD and the KMF used to encapsulate the
   DTLS-SRTP signaling will also be used to convey the hop-by-hop
   encryption keys, salt, and protection profile information.  In this
   way, no additional messages or interfaces are required in order for
   the switching MDD to receive the required security parameters.







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9. Changing Media Forwarded and EKT Field

   Endpoints transmit media to the MDD as they would to a traditional
   conference server, except that media is encrypted and authenticated
   with different keys as outlined in this framework.  Each media source
   within an RTP session has a distinct SSRC and endpoints work to
   address SSRC collisions when they occur (see Section 8.2 of
   [RFC3550]).  From the endpoint's perspective, what is particularly
   unique about the model described in this document is how the RTP
   payload (media content) is encrypted and authenticated end-to-end,
   while other security procedures are performed hop-by-hop.

   To ensure a speedy decoder synchronization in receivers when
   transitioning from forwarding one active speaker's media to the next,
   a switching MDD will send a request for Full Intra-frame Request
   (FIR) [RFC5104] (also known as a "video fast update" in [H.323]
   systems) when a decision is made to switch active video flows.  When
   the endpoint receives this request, it would transmit the video frame
   as requested and include with that initial packet the current "Full
   EKT Field" so that recipients will be able to decrypt the media flow.
   Additionally, a "Full EKT Field" should be transmitted about every
   100ms to ensure that conference participants can decrypt the media
   transmitted.

   It is not possible to request a "Full EKT Field" for audio flows.
   For this reason, it is RECOMMENDED that a "Full EKT Field" be
   included in audio packets about every 100ms to smooth the transition
   of the active speaker's audio forwarded by the server.

   Endpoints SHOULD NOT include the "Full EKT Field" more frequently
   than specified herein, rather opting for the "Short EKT Field" when
   sending most packets to reduce the bandwidth consumed on the wire.

   Endpoints MUST implement [RFC6464] in order for an MDD to determine
   which endpoint(s) have an active speaker as no other method requiring
   access to decrypted media can be used by an untrusted MDD.

10. To-Do List

10.1. What is needed to realize this Framework

      - Endpoint must securely convey its certificate information to
        the KMF so the KMF can recognize a valid endpoint.

      - A means through EKT or another mechanism to negotiate the SRTP
        security profiles for end-to-end encryption/authentication
        (e.g., proposing to negotiate AEAD_AES_128_GCM for end-to-end
        security)

      - A means through EKT or another extension of sending the
        participant identifier (the participant identifier could



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        implicitly identify the conference) so the KMF will know which
        keys to provide for a given conference and RTP sessions related
        to that conference.  Alternatively, this could be an element of
        the tunneling protocol, wherein the MDD indicates the
        associated identifiers.

      - A change to EKT such that the ROC is transmitted in the clear,
        with integrity check performed by XORing the ROC with the IV
        used in AES Key Wrap

      - (Conditional) Removal of the SSRC from the Full EKT Field if we
        allow MDDs to modify the original SSRC value; having it in the
        Full EKT Field would serve no value in this case.

      - A change to EKT to use neither the ISN (per IETF 93) nor MKI
        (per IETF 93)

      - A means of conveying per-hop SRTP master key and salt
        information to the switching MDD (which can be accomplished
        using the DTLS-SRTP tunneling protocol)

10.2. Other Considerations for this Framework

      - Investigate adding ability to enable one-way media from a non-
        trusted device (e.g., announcements).  While not specified as a
        requirement, it was mentioned during a previous IETF meeting
        and may be worth considering.

11. IANA Considerations

   There are no IANA considerations for this document.

12. Security Considerations

   [TBD]

13. References

13.1. Normative References

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

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

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




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

   [I.D-draft-ietf-avtcore-srtp-ekt]
               Mattson, J., McGrew, D., Wing, D., and F. Andreasen,
               "Encrypted Key Transport for Secure RTP", Work in
               Progress, October 2014.

   [RFC6904]   J. Lennox, "Encryption of Header Extensions in the Secure
               Real-time Transport Protocol (SRTP)", RFC 6904, December
               2013.

   [I.D-draft-ietf-avtcore-srtp-aes-gcm]
               McGrew, D. and K. Igoe, "AES-GCM Authenticated Encryption
               in Secure RTP (SRTP)", Work in Progress, June 2015.

   [RFC5763]   Fischl, J. Tschofenig, H., and E. Rescorla, "Framework
               for Establishing a Secure Real-time Transport Protocol
               (SRTP) Security Context Using Datagram Transport Layer
               Security (DTLS)", RFC 5763, May 2010.

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

   [RFC5104]   Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
               "Codec Control Messages in the RTP Audio-Visual Profile
               with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC6464]   Lennox, J., Ivov, E., and E. Marocco, "A Real-time
               Transport Protocol (RTP) Header Extension for Client-to-
               Mixer Audio Level Indication", RFC 6464, December 2011.

   [RFC5285]   Singer, D. and H. Desineni, "A General Mechanism for RTP
               Header Extensions", RFC 5285, July 2008.

   [I.D-draft-abiggs-saag-key-management-service]
               Biggs, A. and S. Cooley, "Key Management Service
               Architecture", Work in Progress, July 2016.

13.2. Informative References

   [I.D-draft-jones-perc-private-media-reqts]
               Jones, P. et al., "Private Media Requirements in Privacy
               Enhanced RTP Conferencing", Work in Progress, July 2015.

   [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.



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   [H.323]     Recommendation ITU-T H.323, "Packet-based multimedia
               communications systems", December 2009.

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

14. Acknowledgments

   The authors would like to thank Mo Zanaty and Christian Oien for
   invaluable input on this document.

Authors' Addresses

   Paul E. Jones
   Cisco Systems, Inc.
   7025 Kit Creek Rd.
   Research Triangle Park, NC 27709
   USA

   Phone: +1 919 476 2048
   Email: paulej@packetizer.com


   Nermeen Ismail
   Cisco Systems, Inc.
   170 W Tasman Dr.
   San Jose
   USA

   Email: nermeen@cisco.com


   David Benham
   Cisco Systems, Inc.
   170 W Tasman Dr.
   San Jose
   USA

   Email: dbenham@cisco.com














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