Network Working Group                                     P. Jones (Ed.)
Internet Draft                                                 N. Ismail
Intended status: Standards Track                               D. Benham
Expires: September 7, 2015                                 Cisco Systems
                                                           March 7, 2015

     A Solution Framework for Private Media in a Switched Conferencing


   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 the switching
   conference server is not entrusted with the 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.

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   This Internet-Draft will expire on September 7, 2015.

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   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors. All rights reserved.

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   described in the Simplified BSD License.

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Table of Contents

   1. Introduction...................................................2
   2. Requirements Language..........................................3
   3. Private Media Trust Model......................................3
   4. Solution Framework Overview....................................3
      4.1. Switching Conference Server Behavior......................4
      4.2. End-to-End Media Privacy..................................5
      4.3. Hop-by-Hop Operations.....................................6
   5. Media Packet Format............................................6
   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
   8. Key Exchange...................................................9
      8.1. Session Signaling.........................................9
      8.2. Negotiating SRTP Protection Profiles and Key Exchange....11
         8.2.1. Endpoint and KMF....................................11
         8.2.2. Switching Conference Server and KMF.................13
   9. Changing Media Forwarded and EKT Field........................14
   10. IANA Considerations..........................................14
   11. Security Considerations......................................14
   12. References...................................................15
      12.1. Normative References....................................15
      12.2. Informative References..................................16
   13. Acknowledgments..............................................16
   Authors' Addresses...............................................17

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 on the conference
   server, as might be the case with a traditional multipoint control
   unit (MCU).  Instead, media flows transmitted by conference
   participants are simply forwarded by the switching conference server
   to each of the other participants, selectively forwarding flows based
   on voice activity detection or other criteria.  In some instances,
   the switching conference server may make limited modifications to RTP
   [RFC3550] headers, for example, but the actual media content (e.g.,
   voice or video data) is unaltered.

   An advantage of switched conferencing is that conference servers can
   be deployed on general-purpose computing hardware, as there is no
   need for the specialized hardware required to manipulate media flows
   that one finds on a traditional hardware MCU.  This, in turn, means
   that it is possible to deploy switching conference servers in
   virtualized environments, including private and public clouds.

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   However, deploying conference resource in a cloud environment may
   introduce a higher security risk.  Whereas traditional conference
   servers were usually deployed in private networks that were protected
   from public access by firewalls, 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, most
   significantly being ports where the administrator can log in to make
   configuration changes, install software updates, 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-avtcore-private-media-reqts].
   Attempting to meet those requirements, this document defines a
   solution framework wherein privacy is ensured by making it impossible
   for a switching conference server 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 conference server 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 conference server 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 conference

   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",
   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

3. Private Media Trust Model

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

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.  This framework specifies the re-use of

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   several technologies, including SRTP [RFC3711], EKT [I.D-draft-ietf-
   avtcore-srtp-ekt], and DTLS-SRTP [RFC5764].

4.1. Switching Conference Server Behavior

   Before going into the specifics of how media privacy is ensured,
   first consider Figure 1 below depicting the behavior of a switching
   conferencing server forwarding media between participants.

            +---+ --{A}--> |                    | <-{C}--- +---+
            | A | <-{B}--- |Switching Conference| --{A}--> | C |
            |   | <-{C}--- |       Server       | --{B}--> |   |
            +---+ <-{D}--- |                    | --{D}--> +---+
                           |       Packet       |
            +---+ --{B}--> |   Authentication   | <-{D}--- +---+
            | B | <-{A}--- |                    | --{A}--> | D |
            |   | <-{C}--- |                    | --{B}--> |   |
            +---+ <-{D}--- |   Media Privacy    | --{C}--> +---+

                   Figure 1 - Switching Conference Server

   In the above figure, each of the participating endpoints sends media
   to the switching conference server where each flow is protected using
   the security mechanisms that will be discussed later.  Each endpoint
   then receives media from each of the other participants in the
   conference.  Importantly, the switching conference server is unable
   to decrypt the media content or modify media content without being
   detected by receiving endpoints.

   The framework does not require, however, for each of the participant
   flows to be transmitted to every other endpoint in the conference.
   In many situations, the switching conference server will transmit
   only a subset of the media flows to each participant.  This might be
   to restrict the bandwidth usage, provide a primary video flow and
   thumbnail flows to single-screen video endpoints, etc.

   The following two diagrams and corresponding explanatory text are for
   illustrative purposes to describe one possible operational mode.

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            +---+ --{A}--> |                    | <--{C}-- +---+
            | A |          |Switching Conference|          | C |*
            +---+ <-{C}--- |       Server       | ---{A}-> +---+
                           |                    |
            +---+ --{B}--> |                    | <--{D}-- +---+
            | B |          |                    |          | D |
            +---+ <-{C}--- |                    | ---{C}-> +---+

                Figure 2 - Endpoint "C" is the Active Speaker

   As depicted in Figure 2, each of the endpoints in the conference is
   receiving a single flow.  In particular, all but one endpoints are
   receiving media flows from endpoint "C", the current active speaker.
   Endpoint "C" is receiving media from endpoint "A", the former active

            +---+ --{A}--> |                    | <--{C}-- +---+
            | A |          |Switching Conference|          | C |
            +---+ <-{B}--- |       Server       | ---{B}-> +---+
                           |                    |
            +---+ --{B}--> |                    | <--{D}-- +---+
           *| B |          |                    |          | D |
            +---+ <-{C}--- |                    | ---{B}-> +---+

                Figure 3 - Endpoint "B" is the Active Speaker

   When the active speaker transitions, so do the video flows.  As
   depicted in Figure 3, the active speaker transitions from "C" to "B".
   Now, each of the endpoints receives a copy of the media flows from
   "B", while "B" receives the media flow from "C", the former active

   How many flows and what type of flows a switching conference server
   transmits to a receiving endpoint are outside the scope of this

4.2. End-to-End Media Privacy

   To ensure the confidentiality of RTP media packets, endpoints utilize
   EKT keys known to conference participants to encrypt the media
   content of the RTP packet (i.e., the RTP payload) using authenticated
   encryption.  These keys may change from time-to-time for various
   reasons, such as when a new conference participant joins a conference
   or when a conference participant leaves a conference.  When it is
   decided that a conference is to be re-keyed is outside the scope of
   this document, but it is important that an unstrusted switching
   conference server is never given access to those keys.

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   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.

4.3. Hop-by-Hop Operations

   To ensure the integrity of transmitted media packets, this framework
   requires that every packet be authenticated.  While media is both
   encrypted and authenticated end-to-end, RTP packets are also
   authenticated hop-by-hop.  The authentication key used for hop-by-hop
   authentication is derived from the SRTP master key shared only on the
   respective hop.  If conference servers are cascaded, then there will
   also be SRTP master keys and derived authentication keys 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.

   It is expected that the conference servers 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
   switching media server 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
   extensions as per [RFC6904].  This will give the switching conference
   server 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 conference server
   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 conference server, it will be detected.  Likewise, the
   endpoint can detect replayed packets originally sent by the media
   server.  Packets received by an endpoint that were originally sent to
   a different endpoint will fail to pass authentication checks.

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

       * Header extensions are optionally Encrypted Hop-by-Hop

                    Figure 4 - Private Media SRTP Packet

   The rollover counter value is shown and transmitted as plaintext.
   This is necessary since a switching conference server may not
   transmit media from one "silent" participant to another participant
   in the conference for a long period of time.  When media from that
   "silent" participant is later sent to that other participant, the
   receiving participant 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.

   The EKT field shown in Figure 4 is the "Full EKT Field".  The "Short
   EKT Field" may also be present in its place.

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.  However, this framework extends the

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   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 SRTP master key, and as outlined
   in Section 3.2.1, other associated values that relate to the master
   key (e.g., master salt and key length values).  For AES-CCM, there
   will also be an associated "Tag_Size_Flag" value (see [I.D-draft-

   The existing parameters in the SRTP cryptographic context are used
   for hop-by-hop operations, including 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

   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) derived
   from the EKT Key(s) by the endpoints communicating in a switched
   conferencing environment.

   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 default 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

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   data used with those algorithms specified in Section 9.2 is redefined
   as follows:

     Associated Data: The version V (2 bits), padding flag P (1 bit),
                      the sequence number (16 bits), timestamp (32
                      bits), and SSRC (32 bits).

   The authentication tag for the end-to-end encrypted payload
   immediately follows the encrypted payload in the packet format.

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

   Only a part of the RTP packet is authenticated with the above
   definition of "Associated Data" since packets are authenticated hop-
   by-hop and there is a desire to allow switching conference servers to
   make changes to certain parts of the RTP header. For these reasons,
   there is 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 conference server to make changes to certain
   fields that would otherwise be protected.

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 encryptions,
   SRTCP authentication, and optional SRTCP encryption.  Likewise, the
   switching conference server needs a hop-by-hop key when communicating
   with an endpoint or cascaded conference server.  The challenge is in
   securely exchanging these keys to the appropriate entities.

   To facilitate key exchange, we utilize DTLS-SRTP and procedures
   defined in EKT.  We will elaborate on this further in the following

8.1. Session Signaling

   The session signaling protocol is not significant to this
   specification, since the call processing functions are untrusted.
   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, we will assume SIP is
   used and SDP [RFC4566] conveys the fingerprint information as per

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   The endpoint ("User Agent" in SIP terminology) will send an INVITE
   message containing SDP for the media session along with fingerprints.
   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

   For this example, we will assume 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, including the switching conference server.  It's important
   to note, however, that this does not lend to the security of media,
   as the call processing function is not trusted.

   The Key Management Function (KMF) needs to receive information about
   the call.  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 switching conference
   server and the KMF (see Error! Reference source not found.).
   Regardless, 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.  The endpoint
   to KMF information exchange is outside the scope of this document.

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

   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.

   [Editor's Note: How would an endpoint that is outside an enterprise
   domain (e.g., an associate at another company) be able to interact
   with the enterprise KMF?  It might be necessary to have a trusted
   call processing entity that signs messages that the foreign endpoint
   can validate so that it knows that it can trust the certificate
   fingerprint of the KMF.]

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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 conference server.  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 conference servers.

   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 conference server does not
   terminate the DTLS signaling.  Rather, DTLS packets received by the
   conference server are forwarded to the KMF and vice versa.  The
   figure below depicts this.

          +-----+    Server / KMF +-----------------------------+
          |     |    Interface    | Switching Conference Server |
          |     |<--------------->|                             |
          |     |                 |                             |
          | KMF |<--------------->|<-------------+ (Tunnels     |
          |     |    DTLS-        |              v  DTLS-SRTP)  |
          +-----+    SRTP         +-----------------------------+
                     Tunnel                      ^
                                                 | DTLS-SRTP
                                            | Endpoint |

                    Figure 5 - 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 deriving SRTP keys and encrypt and
   authenticate the media content in SRTP packets.  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". [Editor's note: we are proposing
   changes to the EKT draft that will include the ROC separated from the
   EKT Ciphertext.  Additionally, we need a mechanism to negotiate SRTP
   Protection Profiles for the end-to-end encryption/authentication.
   This might be an extension to EKT, a new extension, or even an
   application-layer exchange over the DTLS connection to the KMF.]

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   This framework also calls for the extension of 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].

   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.  [Editor's
   note: We could use DTLS-SRTP only to negotiate the SRTP Protection
   Profiles and then introduce a new extension to allow the KMF to send
   out the hop-by-hop key and salt to both the endpoint and conference
   server.  Open to alternative suggestions from the workgroup.]

   During the lifetime of the conference, conference participants may
   come and go.  During those events, the KMF will send a new EKT
   message to clients providing a new EKT key to use from that point

   If a new participant does not support the same SRTP Protection
   Profile in use by the conference, 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 we
   select a small number of protection profiles that must be implemented
   by all endpoints.

   Summary of what we need to realize this framework:

      - Endpoint must securely convey its certificate information to
        the KMF and negotiate a participant identifier (e.g., a UUID
        securely conveyed, but need not be encrypted) before a
        connection to the conference server is attempted.

      - A means through EKT or another extension to negotiate the SRTP
        security profiles for end-to-end encryption/authentication

      - A means through EKT or another extension of sending the
        participant identifier (the participant identifier could
        implicitly identify the conference)

      - 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

      - A means of conveying per-hop SRTP master key and salt
        information to the switching conference server

   To help in understanding better the sequence of messages, consider
   the following figure:

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   Endpoint                    KMF                      Server
      |                         |                          |
      | External Signaling      |                          |
      | To exchange Cert and    |                          |
      | and participant ID      |                          |
      | ----------------------> |                          |
      |                         |                          |
      | DTLS connection         |                          |
      | -------------------------------------------------> | \
      |                         | <======================= | /
      |                         |              DTLS Tunnel |
      |                         |                          |
      | For brevity, we use === 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 6 - 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

8.2.2. Switching Conference Server and KMF

   [Editor's Note: there must be an interface between the switching
   conference server and the KMF so that cipher suites and key
   information can be conveyed for each participant in each conference
   for hop-by-hop operations.  This interface is out of scope for this

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9. Changing Media Forwarded and EKT Field

   Endpoints transmit media to the switching conference server as they
   would in a traditional conference, 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.  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 conference server 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.

   A switching conference server may forward a single audio and video
   flow to a receiver, or it may forward multiple flows.  The number of
   media flows very much depends on the capabilities of the receiving
   device.  How the number of media flows to forward is determined or
   negotiated is outside the scope of this document.

   To aid in determining when to transition the active speaker's audio
   or video, endpoints MUST implement [RFC6464] in order to provide a
   hint to the switching media server as to which endpoint should be
   designated as the one of the active speaker(s).

10. IANA Considerations

   There are no IANA considerations for this document.

11. Security Considerations


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12. References

12.1. 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.

   [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.

               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

               McGrew, D. and K. Igoe, "AES-GCM and AES-CCM
               Authenticated Encryption in Secure RTP (SRTP)", Work in
               Progress, July 2014.

   [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.

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12.2. Informative References

               Jones, P. et al., "Requirements for Private Media in a
               Switched Conferencing Environment", Work in Progress,
               March 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.

   [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.

13. Acknowledgments

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

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Authors' Addresses

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

   Phone: +1 919 476 2048

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


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


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