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Network Working Group                                     P. Jones (Ed.)
Internet Draft                                                 N. Ismail
Intended status: Informational                                 D. Benham
Expires: September 7, 2015                                    N. Buckles
                                                           Cisco Systems
                                                             J. Mattsson
                                                                Y. Cheng
                                                               R. Barnes
                                                           March 7, 2015

   Requirements for Private Media in a Switched Conferencing Environment


   This document specifies the requirements for ensuring the privacy and
   integrity of real-time media flows between two or more endpoints
   communicating in a switched conferencing environment.  This document
   also provides a high-level overview of switched conferencing in order
   to establish a common understanding of the goals and objectives of
   this work.

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 September 7, 2015.

Copyright Notice

   Copyright (c) 2014 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

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   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
   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. Terminology....................................................3
   4. Background.....................................................4
   5. Motivation for Private Media in Switched Conferencing..........5
      5.1. Switched Conferencing in Cloud Services...................5
      5.2. Private Media Security through Switching..................7
   6. Private Media Trust Model......................................8
      6.1. Trusted Elements..........................................9
      6.2. Untrusted Elements.......................................10
   7. Goals and Non-Goals...........................................11
      7.1. Goals....................................................11
         7.1.1. Ensure End-To-End Confidentiality...................11
         7.1.2. Ensure End-To-End Source Authentication of Media....11
         7.1.3. Provide a More Efficient Service than "Full-Mesh"...11
         7.1.4. Support Cloud-Based Conferencing....................12
         7.1.5. Limiting a User's Access to Content.................12
         7.1.6. Compatibility with the WebRTC Security Architecture.12
      7.2. Non-Goals................................................13
         7.2.1. Securing the Endpoints..............................13
         7.2.2. Concealing that Communication Occurs................13
         7.2.3. Individual Media Source Authentication..............13
         7.2.4. Support for Multicast in Switched Conferencing......14
   8. Requirements..................................................14
   9. IANA Considerations...........................................15
   10. Security Considerations......................................15
   11. References...................................................16
      11.1. Normative References....................................16
      11.2. Informative References..................................16
   12. Acknowledgments..............................................16
   13. Contributors.................................................17
   Authors' Addresses...............................................18

1. Introduction

   Users of multimedia communication products and services have privacy
   expectations that are largely satisfied with the use of SRTP
   [RFC3711] and related technologies when communicating point-to-point
   over the Internet.  When communicating in a conferencing environment
   with two or more participants, though, it is necessary for an
   endpoint to share the SRTP master key and salt with the conference

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   server so that it can authenticate and decrypt received RTP and RTCP
   packets.  The conference server also needs the master key and salt in
   order to transmit media packets it receives to other participants in
   the conference.  The need for conferencing servers to have the master
   key is a security risk for users.

   Within a corporate or other isolated environment where conferencing
   servers are tightly controlled, this security risk can be effectively
   managed.  However, managing this risk is becoming increasing
   difficult as conferencing resources are being deployed in networks
   that are less trusted, including virtualized conferencing servers
   deployed in cloud environments.

   There are also public voice and video conferencing service providers
   in which users must place full trust in order to use those services,
   as it is necessary for an endpoint to share the SRTP master key with
   those conferencing servers.  This exposes corporations, for example,
   to a higher risk of being subjected to corporate espionage.  While it
   is not the intent of this draft to suggest that any existing service
   provider would permit or condone any illicit use of its service, the
   fact is that security threats can come from external sources and
   remain undiscovered for long periods of time.

   It is possible to ensure communication privacy within the context of
   a switched conferencing environment with limited changes in the
   security mechanisms used today.  This document discusses this
   possibility in more detail and presents a set of requirements for
   meeting this objective.

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. Terminology

   Adversary - An unauthorized entity that may attempt to compromise the
   performance of a conference server through various means, including,
   but not limited to, the transmission of bogus media packets or
   attempt to gain access to the plaintext of the media.

   Media content - the portion of the RTP (i.e., the encrypted RTP
   payload) or other packet containing the actual audio, video, or other
   multimedia information that is considered confidential and is subject
   to end-to-end encryption.  This does not include, for example, RTP
   headers, RTP header extensions, or RTCP packets.

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   Switching conference server - A conference server that does not
   decrypt RTP media flows or perform processing on the media payload,
   but instead simply forwards the received media from a sender to the
   other participants in a multimedia conference.  A switching
   conference server may modify some RTP headers.

4. Background

   Traditional multimedia conferencing servers would mix, transcode,
   transrate, and/or recompose media flows from one or more conference
   participants, sending out a different audio and video flow to each
   participant.  For audio, this might entail mixing some number of
   input flows that appear to contain audio intended to be heard by the
   other participants, with each participant receiving a flow that does
   not contain that participant's own audio.  For video, the conference
   server may elect to send only video showing the current active
   speaker, a tiled composition of all participants or the most recent
   active speakers, a video flow with the active speaker presented
   prominently with other participants presented as thumbnail images, or
   some other composite arrangement.  It is also common for audio or
   video to be transcoded.  A typical traditional conferencing server is
   depicted in Figure 1.

            +---+ --{A}--> |                   | <--{C}-- +---+
            | A |          | Media Composition |          | C |
            +---+ <-{BCD}- |                   | -{ABD}-> +---+
                           |    Transcoders    |
            +---+ --{B}--> |    Transraters    | <--{D}-- +---+
            | B |          |                   |          | D |
            +---+ <-{ACD}- |   Decrypt/Encrypt | -{ABC}-> +---+

                 Figure 1 - Traditional Conferencing Server

   Traditional conference servers require a significant amount of
   processing power, which in turn translates into a high cost for
   conferencing hardware manufacturers.  Significantly, too, it is very
   difficult to deploy these servers in a cloud environment due to the
   high processing demands, as the specialized hardware found in the
   traditional voice and video conferencing server does not exist in a
   cloud environment.

   To enable the traditional conferencing server to perform its job, the
   server establishes an SRTP session with each of the conference
   participants so that it can get the keys required to decrypt and
   encrypt media flows from and to each participant.  This means that
   the conference server is necessarily a fully trusted entity in the
   communication path.  Anytime these servers are deployed in a network
   that is not tightly controlled, it increases the risk that an
   attacker might gain access to cryptographic key material, thus

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   allowing the attacker to be able to see and listen to ongoing
   conferences.  In some instances, depending on how the hardware is
   designed and how keys and certificates are managed, it might be
   possible for an attacker to see and listen to previously recorded
   conferences or future conferences.

   The Secure Real-time Transport Protocol (SRTP) [RFC3711] is a profile
   of RTP, which can provide confidentiality, message authentication,
   and replay protection to the RTP traffic and to the RTP Control
   Protocol (RTCP).  Encryption of header extension in SRTP [RFC6904]
   provides a mechanism extending the mechanisms of [RFC3711], to
   selectively encrypt RTP header extensions in SRTP.  [RFC3711] and
   [RFC6904] solves end-to-end use cases between two endpoints, and does
   not consider use cases where a sender delivers media to a receiver
   via a cloud-based conferencing service.

5. Motivation for Private Media in Switched Conferencing

5.1. Switched Conferencing in Cloud Services

   There is a trend in the industry for enterprises to use cloud
   services to host multi-party conferences and meet-me services, either
   exclusively or to meet peak loads on-demand.  At the same time, there
   is shift toward using light-weight, cost-effective switching
   conference servers in cloud services that do not necessarily need to
   mix audio or composite/transcode video.  Also fueling the use of such
   light-weight conference servers is the desire to fully exploit
   virtualized computing resources and dynamic scalability potential
   available in cloud computing environments.

   The increased use of cloud services has exposed a problem.  There are
   two different trust domains from a media perspective: endpoints and
   other devices in a trusted domain, and conference servers controlled
   by the cloud service in an untrusted domain.  Other examples of
   conference devices spread across trusted and untrusted domains are
   likely, but the cloud service trend is triggering the urgency to
   address the need to allow for lightweight media conference while
   enabling media privacy at the same time.

   With a switching conference server, each participant transmits media
   to the server as it would with a traditional conferencing server.
   However, the switching conference server merely forwards media to the
   other participants in the conference (where the other participant may
   be associated with a cascaded conference server or an endpoint on the
   same server), leaving composition to the receiving endpoint.  Since
   some endpoints may have a limited amount of bandwidth, each endpoint
   might negotiate with the switching conference server to receive only
   a subset of the available media flows.  Each transmitting endpoint
   might also send multiple media flows of varying frame sizes and/or
   frame rates (e.g., simulcast or scalability layers), so that the
   server can select the streams most appropriate for each receiver's

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   bandwidth and capabilities.  This allows, for example, an endpoint to
   receive and display higher quality video for the active speaker and
   thumbnails for other participants.  It is also worth noting that, for
   switched media to work successfully, each endpoint in the conference
   must support the media formats transmitted by all other entities in
   the conference.  More modern endpoints support multiple codecs and
   formats, making this commercially practical.

   Figure 2 depicts an example of a switching conference server wherein
   each participant is receiving the media flows transmitted by each of
   the other participants in the conference.

            +---+ --{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 2 - Switching Conference Server

   Note - The use of multiple arrows directed toward each endpoint is
   not intended to suggest the use of separate RTP sessions.

   By using methods such as those described in [RFC6464], it is possible
   for the switching conference server to transmit the appropriate audio
   and video flows to conference participants without having knowledge
   of the contents of the encrypted media.  The examples that follow
   help to illustrate this point.

   In the Figure 3 below, endpoints A, B and D receive the video streams
   from endpoint C, the currently active speaker, which is receiving
   video from endpoint A, the previous active speaker.  Later when
   endpoint B becomes the active speaker (Figure 4), endpoints A, C and
   D will start to receive video from B, while endpoint B continues to
   receive video from endpoint C.  Finally in Figure 5, endpoint A
   becomes the active speaker.

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

                Figure 3 - Endpoint "C" is the Active Speaker

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

                Figure 4 - Endpoint "B" is the Active Speaker

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

                Figure 5 - Endpoint "A" is the Active Speaker

   Switched conferencing can also enable conferences to scale to include
   many more simultaneous participants than would be possible with a
   traditional conferencing server.  Like traditional conferencing
   servers, switching conference servers can also be cascaded or
   interconnected in a meshed topology to increase the size of the
   conference without putting undue burden on any particular server.

5.2. Private Media Security through Switching

   A traditional conferencing server, or MCU, establishes an SRTP
   session with each participating endpoint separately, and needs
   to decrypt packets containing media presented to other endpoints.  By
   using a switching conference server, it is possible to keep the media
   encryption keys private to the endpoints such that the conference
   server does not have access to the keys used for media encryption.

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   The switching conference server just forwards media received to each
   of the other participants in the conference.

   This provides for a significantly improved security model, as one
   can, for example, utilize conferencing resources in the cloud that do
   not necessarily have to be trusted.  That said, there may be
   situations where the switching conference server needs to modify the
   RTP packet received from an endpoint, such as by adding or removing
   an RTP header extension, modifying the payload type value, etc.  It
   would be the responsibility of the switching conference server to
   ensure that media of the expected type and containing the correct
   information is received by a recipient.

   Thus, there is a need to utilize an end-to-end encryption and
   authentication key (or pair of keys) and a hop-by-hop encryption and
   authentication key (or pair of keys).  The purpose for the hop-by-hop
   encryption key is to optionally encrypt RTP header extensions.  The
   current SRTP specification and related specifications do not define
   use of a dual-key approach presently.  However, such an approach is
   possible and would result in ensuring the privacy of media while also
   enabling the more scalable switched conferencing model.

   The assumption is that no changes are made to SRTCP, i.e. SRTCP is
   protected hop-by-hop with a single security context.

   This dual-key model does necessitate a change in the way that keys
   are managed.  However, the topic of key management is outside the
   scope of this requirements document.  However, high-level assumptions
   like if the end-to-end contexts use a group key as SRTP master key or
   if individual SRTP master keys (that may be derived/negotiated from
   another group key) is likely to influence the solution derived from
   this document.

6. Private Media Trust Model

   The architecture suggested in this specification enables switching
   conference servers to be hosted in domains in which the network
   elements may have low trust, or where the trustworthiness is
   uncertain.  This does not mean that the service provider is
   untrusted; it simply means that high trust is not required.  This has
   the benefit of protecting the endpoints in the case of external
   attacks against the conference server.

   In this specification, certain elements are considered trusted and
   others are considered untrusted.  Trust in the context of this
   specification means that the element can be in possession of the
   media encryption key(s) for a past, current, or potentially future
   conference (or portion thereof) used to protect media content.

   There are very few elements that need to be trusted.  However, it is
   also recognized that in certain deployment models, some elements that

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   are classified as untrusted might be placed into the trusted domain
   and considered trusted.  This specification is not intended to
   prevent such deployment models, but it does not rely upon them.

   Each of the elements discussed below has a direct or indirect
   relationship with each other.  The following diagram depicts the
   trust relationships described in the following sub-sections and the
   media or signaling interfaces that exist between them, showing the
   trusted elements on the left and untrusted elements on the right.
   Note that this is a logical diagram and functional elements may be
   co-located or further divided into multiple separate physical
   entities.  Note that it is not necessary that every interface exist
   between all elements, such as both an interface from the endpoint and
   call processing function to a key management function, though both
   are possible options.

                       v             |                              |
                 +----------+        |       +-----------------+    |
                 | Endpoint |--------------> | Call Processing |    |
                 +----------+        |       +-----------------+    |
                    ^                |         ^     ^              |
      Trusted       |                |         |     |       +------+
      Elements      |                |         |     |       |
                    |  +-----------------------+     |       |
                    |  |             |               v       v
                    |  |             |     +----------------------+
                    |  |  +--------------> | Switching Conference |
                    |  |  |          |     |       Server         |
                    v  v  v          |     +----------------------+
              +----------------+     |
              | Key Management |     |            Untrusted
              |    Function    |     |            Elements
              +----------------+     |

          Figure 6 - Relationship of Trusted and Untrusted Elements

6.1. Trusted Elements

   The endpoint is considered a trusted element, as it will be sourcing
   media flows transmitted to other conference participants and will be
   receiving media for rendering for the human user.  While it is
   possible for an endpoint to be compromised and perform in unexpected
   ways, such as transmitting a decrypted copy of media content to an
   adversary, such security issues and defenses are outside the scope of
   this document.

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   The other trusted element is a key management function (KMF).  This
   function is responsible for providing cryptographic keys to the
   endpoints for encrypting and authenticating media content.  The KMF
   is also responsible for providing cryptographic keys to the
   conferencing resources to enable authentication of media packets
   received by a conference participant.  Interaction between the KMF
   and untrusted call processing functions may be necessary to ensure
   conference participants are delivered the appropriate keys or are
   directed to the appropriate conference server.  It is expected that
   the KMF will be tightly controlled and managed to prevent
   exploitation by an adversary, as any kind of security compromise of
   the KMF puts the security of all conferences at risk.

6.2. Untrusted Elements

   The call processing function is responsible for such things as
   authenticating the user, signing messages, and processing call
   signaling messages.  This element is responsible for ensuring the
   integrity, and optionally the confidentiality, of call signaling
   messages between itself, the endpoint, and other network elements.
   However, it is considered an untrusted element for the purposes of
   this specification, as it cannot be trusted to have access to or be
   able to gain access to cryptographic key material that provides
   privacy and integrity of media packets.

   There might be several independent call processing functions within
   an enterprise, service provider network, or the Internet that are
   classified as untrusted.  Any signaling information that passes
   through these untrusted entities is subject to inspection by that
   element and might be altered by an adversary.

   Likewise, there may be certain deployment models where the call
   processing function is considered trusted.  In such cases, trusted
   call processing functions MUST take responsibility for ensuring the
   integrity of received messages before delivering those to the
   endpoint.  How signaling message integrity is ensured is outside the
   scope of this document, but might use such methods as defined in

   The final element is the switching conference server, which is
   responsible for forwarding encrypted media packets and conference
   control information to endpoints in the conference.  It is also
   responsible for conveying secured signaling between the endpoints and
   the key management function, acquiring per-hop authentication keys
   from the KMF, and performing per-hop authentication operations for
   media packets.  This function might also aggregate conference control
   information and initiate various conference control requests.
   Forwarding of media packets requires that the switching conference
   server have access to RTP headers or header extensions and
   potentially modify those message elements, but the actual media
   content MUST not be decipherable by the switching conference server.

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   Further, the switching conference server does not have the ability to
   determine whether an endpoint is authorized to have access to media
   encryption keys.  Merely joining a conference MUST NOT be interpreted
   as having authority.  Media encryption keys are conveyed to the
   endpoint by the KMF in such a way as to prevent the switching
   conference server from having access to those keys.

   It is assumed that an adversary might have access to the switching
   conference server and have the ability to read any of the contents
   that pass through.  For this reason, it is untrusted to have access
   to the media encryption keys.

   As with the call processing functions, it is appreciated that there
   may be some deployments wherein the switching conference server is
   trusted.  However, for the purposes of this specification, the
   switching conference server is considered untrusted so that we can
   ensure to develop a solution that will work even in the more hostile

7. Goals and Non-Goals

7.1. Goals

7.1.1. Ensure End-To-End Confidentiality

   The content of the communication and all media needs to be
   confidential within the group of entities explicitly invited into the
   conference.  An external monitoring adversary should not be able to
   deduce the human-to-human communication that actually occurred from
   capturing the media packets.

   At the same time, it is necessary to allow switching media servers to
   manipulate certain RTP header fields like the payload type value.

7.1.2. Ensure End-To-End Source Authentication of Media

   In a conference system with multiple participants it is vital that
   the media content presented to any of the human participants is from
   the stated participant, and not an adversary that attempts to inject
   misleading content.  Nor should an adversary be able to fool the
   system into becoming a trusted party in the conference.  Only
   explicitly invited parties shall be able to contribute content.

7.1.3. Provide a More Efficient Service than "Full-Mesh"

   A multi-party conference that has the goals of confidentiality and
   source authentication can be established as a "full mesh" (i.e., each
   participating endpoint directly addresses each of the other
   participants).  However, this has a significant issue with the amount
   of consumed resources in both the uplink and the downlink from each

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   A switched conferencing model would yield the efficiencies desired.

7.1.4. Support Cloud-Based Conferencing

   To achieve cost-effective and scalable conferencing, it must be
   possible to run the conference server instances in a cloud-based
   virtualized environment.

   From a security standpoint, this is a significant issue since the
   virtualized server instance and the underlying hardware and software
   upon which it runs might not be secure from an adversary.

7.1.5. Limiting a User's Access to Content

   Since an invited user will be provided with the content protection
   keys, the user can decrypt content from time periods before and after
   the user joined the conference.  However, this is not always
   desirable.  It should be possible to re-key the content protection
   keys every time a user joins or leaves the conference so each
   particular set of conference participants uses a unique key.

   This also changes the trust level required on the conference roster
   handling at any point and how to keep that accurate and secured.

   It should be noted that timely completion of the re-keying operations
   become an obstacle in system design and operation.  Thus, it is a
   goal to allow for this possibility when it is deemed essential, but
   it should not be a requirement on a system to re-key each time the
   participant list changes.

7.1.6. Compatibility with the WebRTC Security Architecture

   It is a goal of this work to ensure compatibility with the WebRTC
   security architecture as described in [I.D-rtcweb-security-arch].  As
   an example, local resources that are considered a part of the trusted
   computing base (TCB), such as keying material derived using DTLS-
   SRTP, will remain within the TCB and not exposed to untrusted

   The browser is reliant on an external calling service to convey
   signaling information that may open the door for a man-in-the-middle
   attack, such as the conveyance of certificate fingerprints over the
   interface between the browser and the calling service.  However, as
   described in [I.D-rtcweb-security-arch], the browser may utilize
   additional services, such as a trusted identify provider, to mitigate
   such risks.

   Having said the foregoing, this document does not aim to define
   requirements for end-to-end security for the WebRTC data channel.

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7.2. Non-Goals

7.2.1. Securing the Endpoints

   The security of a communication session requires that the endpoints
   are not compromised and that the users are trustworthy.  If not,
   credentials and decrypted content may be shared with third parties.
   However, this is hard to prevent through system design.  Thus, it
   should be assumed that the endpoint is secure and the user is
   trustworthy; how to achieve this is out of scope this document.

7.2.2. Concealing that Communication Occurs

   A non-goal is to attempt to prevent a pervasive monitoring adversary
   from knowing that the communication session has occurred.  The reason
   for excluding this as a goal is that it is extremely difficult to
   achieve, as a pervasive monitoring adversary can be expected to be
   able to have knowledge of all IP flows that enter or exit local ISPs,
   across links that straddle nation borders or internet exchange
   points.  To hide the fact communication occurred, the flows required
   to achieve the communication session need to be highly difficult to
   correlate between different legs of the communication.

   At this stage this is deemed too difficult to attempt and will need
   to be a subject for further study.  Existing attempts include The
   Onion Router (TOR), against which it has been claimed to be possible
   to monitor, at least partially, by an adversary with sufficient

   Also of consideration is that trying to conceal the fact that
   communication occurred actually makes it more difficult for network
   administrators to effectively manage and troubleshoot issues with
   conference calls.

7.2.3. Individual Media Source Authentication

   Although the participants in the conference are authenticated, it is
   not a goal to provide source authentication of the media at the
   individual user level, instead being satisfied with being able to
   authenticate media as coming from an invited conference participant
   or not.

   There exist solutions that can provide individual media source
   authentication (e.g., TESLA).  However, they impact the performance
   or security properties they provide.  Thus, further study is required
   to determine impact and resulting security properties if desired to
   have individual source authentication.

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7.2.4. Support for Multicast in Switched Conferencing

   Multicast traffic is, by design, transmitted to every participant in
   a conference.  The focus of this document is only on centralized
   unicast conferencing that utilizes a switched conferencing

8. Requirements

   The following are the security solution requirements for switched
   conferencing that enable end-to-end media privacy between all
   conference participants.

   Note that while some switching media servers might be fully trusted
   entities, the intent of this solution and purpose for these private
   media (PM) requirements is to address those servers that are not
   fully trusted.

   PM-01:  Switching conference server MUST be able to switch the media
           between participants in a conference without having access to
           unencrypted media content.

   PM-02:  Solution MUST maintain all current SRTP security goals,
           namely the ability to provide for end-to-end confidentiality,
           provide for hop-by-hop replay protection, and ensure hop-by-
           hop and end-to-end message integrity. {Editor's Note:
           Question asked, "Does this include third parties?"  Jonathan
           Lennox to suggest ways to make this more concrete.}

   PM-03:  Solution MUST extend replay protection to cover each hop in
           the media path, both ensuring that any received packet is
           destined for the recipient and not a duplicate.

   PM-04:  Keys used for end-to-end encryption and authentication of RTP
           payloads and other information deemed unsuitable for access
           by the switching conference server MUST NOT be generated by
           or accessible to any component that is not in the fully
           trusted domain.

   PM-05:  The switching conference server MUST be capable of making
           changes to the RTP header and, optionally, the RTP header

   PM-06:  The SRTP cryptographic context, which is identified in part
           by an SSRC, contains transform-independent parameters used by
           the sending endpoint, including the RTP packet sequence
           number and rollover counter (ROC), required for packet
           decryption and authentication that, along with the value of
           the SSRC, MUST be protected end-to-end.

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   PM-07:  The switching conference server, or any entity that is not
           fully trusted, MUST NOT be involved in the user or device
           authentication for the purpose of media key distribution.

   PM-08:  The switching conference server MUST be able to switch an
           already active SRTP stream to a new receiver, while
           guaranteeing the timely synchronization between the SRTP
           context of the transmitter and its current and new receivers.

   PM-09:  It MUST be possible for the switching conference server to
           determine if a received media packet was transmitted by a
           conference participant in possession of the end-to-end media
           encryption keys and hop-by-hop authentication keys.

   PM-10:  It MUST be possible for a conference to be optionally re-
           keyed as desired, such as each time a participant joins or
           leaves the conference. {Editor's note:  Who is allowed to
           know who leaves and joins?  Do you trust the conference
           server to tell you reliably?}

   PM-11:  Any solution satisfying this requirements specification MUST
           provide for a means through which WebRTC-compliant endpoints
           can participate in a switched conference using private media
           as outlined herein.

   PM-12:  All RTP senders, including the switching conference server,
           MUST adhere to all congestion control requirements that are
           required by the RTP profile and topology in use, including
           RTP circuit breakers [I.D-ietf-avtcore-rtp-circuit-breakers].
           Since the switching conference server is unable to perform
           transcoding or transrating that requires access to the
           unencrypted media, its reaction to congestion signals is
           often limited to dropping packets that would otherwise be
           forwarded in the absence of congestion, and signaling
           congestion to the RTP source.  This is similar to the
           congestion control behavior of the Media Switching Mixer and
           Selective Forwarding Middlebox/Unit in [I.D-ietf-avtcore-rtp-

9. IANA Considerations

   There are no IANA considerations for this document.

10. Security Considerations


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11. References

11.1. Normative References

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

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

   [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.

               E. Rescorla, "WebRTC Security Architecture", Work in
               Progress, July 2014.

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

               Westerlund, M., and S. Wenger, "RTP Topologies", Work in
               Progress, March 2015.

               Perkins, C. S., and V. Singh, "Multimedia Congestion
               Control: Circuit Breakers for Unicast RTP Sessions", Work
               in Progress, March 2015.

11.2. Informative References

   [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.

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

12. Acknowledgments

   The authors would like to thank Marcello Caramma, Matthew Miller,
   Christian Oien, Magnus Westerlund, Cullen Jennings, Christer
   Holmberg, Bo Burman, Jonathan Lennox, Suhas Nandakumar, Dan Wing,
   Roni Even, and Mo Zanaty for their invaluable input.

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13. Contributors


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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
   Email: paulej@packetizer.com

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

   Email: nermeen@cisco.com

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

   Email: dbenham@cisco.com

   Nathan Buckles
   Cisco Systems, Inc.
   170 W Tasman Dr.
   San Jose

   Email: nbuckles@cisco.com

   John Mattsson
   Ericsson AB
   SE-164 80 Stockholm

   Phone: +46 10 71 43 501
   Email: john.mattsson@ericsson.com

   Yi Cheng
   SE-164 80 Stockholm

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   Phone: +46 10 71 17 589
   Email: yi.cheng@ericsson.com

   Richard Barnes
   331 E Evelyn Ave.
   Mountain View

   Email: rlb@ipv.sx

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