RTCWEB                                                       E. Rescorla
Internet-Draft                                                RTFM, Inc.
Intended status:  Standards Track                       January 22, 2012
Expires:  July 25, 2012

                      RTCWEB Security Architecture


   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The major use cases for RTCWEB technology are
   real-time audio and/or video calls, Web conferencing, and direct data
   transfer.  Unlike most conventional real-time systems (e.g., SIP-
   based soft phones) RTCWEB communications are directly controlled by
   some Web server, which poses new security challenges.  For instance,
   a Web browser might expose a JavaScript API which allows a server to
   place a video call.  Unrestricted access to such an API would allow
   any site which a user visited to "bug" a user's computer, capturing
   any activity which passed in front of their camera.  [I-D.ietf-
   rtcweb-security] defines the RTCWEB threat model.  This document
   defines an architecture which provides security within that threat



Status of this Memo

   This Internet-Draft is submitted 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

Rescorla                  Expires July 25, 2012                 [Page 1]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   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 July 25, 2012.

Copyright Notice

   Copyright (c) 2012 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
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
   modifications of such material outside the IETF Standards Process.
   Without obtaining an adequate license from the person(s) controlling
   the copyright in such materials, this document may not be modified
   outside the IETF Standards Process, and derivative works of it may
   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.

Rescorla                  Expires July 25, 2012                 [Page 2]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Trust Model  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Authenticated Entities . . . . . . . . . . . . . . . . . .  5
     3.2.  Unauthenticated Entities . . . . . . . . . . . . . . . . .  5
   4.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  6
     4.1.  Initial Signaling  . . . . . . . . . . . . . . . . . . . .  7
     4.2.  Media Consent Verification . . . . . . . . . . . . . . . .  9
     4.3.  DTLS Handshake . . . . . . . . . . . . . . . . . . . . . . 10
     4.4.  Communications and Consent Freshness . . . . . . . . . . . 10
   5.  Detailed Technical Description . . . . . . . . . . . . . . . . 10
     5.1.  Origin and Web Security Issues . . . . . . . . . . . . . . 10
     5.2.  Device Permissions Model . . . . . . . . . . . . . . . . . 11
     5.3.  Communications Consent . . . . . . . . . . . . . . . . . . 12
     5.4.  IP Location Privacy  . . . . . . . . . . . . . . . . . . . 13
     5.5.  Communications Security  . . . . . . . . . . . . . . . . . 13
     5.6.  Web-Based Peer Authentication  . . . . . . . . . . . . . . 15
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 16
     6.1.  Communications Security  . . . . . . . . . . . . . . . . . 16
     6.2.  Privacy  . . . . . . . . . . . . . . . . . . . . . . . . . 16
     6.3.  Denial of Service  . . . . . . . . . . . . . . . . . . . . 17
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 18
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 18
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 19

Rescorla                  Expires July 25, 2012                 [Page 3]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

1.  Introduction

   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The major use cases for RTCWEB technology are
   real-time audio and/or video calls, Web conferencing, and direct data
   transfer.  Unlike most conventional real-time systems, (e.g., SIP-
   based[RFC3261] soft phones) RTCWEB communications are directly
   controlled by some Web server, as shown in Figure 1.

                               |                |
                               |   Web Server   |
                               |                |
                                   ^        ^
                                  /          \
                          HTTP   /            \   HTTP
                                /              \
                               /                \
                              v                  v
                           JS API              JS API
                     +-----------+            +-----------+
                     |           |    Media   |           |
                     |  Browser  |<---------->|  Browser  |
                     |           |            |           |
                     +-----------+            +-----------+

                     Figure 1: A simple RTCWEB system

   This system presents a number of new security challenges, which are
   analyzed in [I-D.ietf-rtcweb-security].  This document describes a
   security architecture for RTCWEB which addresses the threats and
   requirements described in that document.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Trust Model

   The basic assumption of this architecture is that network resources
   exist in a hierarchy of trust, rooted in the browser, which serves as
   the user's TRUSTED COMPUTING BASE (TCB).  Any security property which

Rescorla                  Expires July 25, 2012                 [Page 4]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   the user wishes to have enforced must be ultimately guaranteed by the
   browser (or transitively by some property the browser verifies).
   Conversely, if the browser is compromised, then no security
   guarantees are possible.  Note that there are cases (e.g., Internet
   kiosks) where the user can't really trust the browser that much.  In
   these cases, the level of security provided is limited by how much
   they trust the browser.

   Optimally, we would not rely on trust in any entities other than the
   browser.  However, this is unfortunately not possible if we wish to
   have a functional system.  Other network elements fall into two
   categories:  those which can be authenticated by the browser and thus
   are partly trusted--though to the minimum extent necessary--and those
   which cannot be authenticated and thus are untrusted.  This is a
   natural extension of the end-to-end principle.

3.1.  Authenticated Entities

   There are two major classes of authenticated entities in the system:

   o  Calling services:  Web sites whose origin we can verify (optimally
      via HTTPS).
   o  Other users:  RTCWEB peers whose origin we can verify
      cryptographically (optimally via DTLS-SRTP).

   Note that merely being authenticated does not make these entities
   trusted.  For instance, just because we can verify that
   https://www.evil.org/ is owned by Dr. Evil does not mean that we can
   trust Dr. Evil to access our camera an microphone.  However, it gives
   the user an opportunity to determine whether he wishes to trust Dr.
   Evil or not; after all, if he desires to contact Dr. Evil (perhaps to
   arrange for ransom payment), it's safe to temporarily give him access
   to the camera and microphone for the purpose of the call, but he
   doesn't want Dr. Evil to be able to access his camera and microphone
   other than during the call.  The point here is that we must first
   identify other elements before we can determine whether and how much
   to trust them.

   It's also worth noting that there are settings where authentication
   is non-cryptographic, such as other machines behind a firewall.
   Naturally, the level of trust one can have in identities verified in
   this way depends on how strong the topology enforcement is.

3.2.  Unauthenticated Entities

   Other than the above entities, we are not generally able to identify
   other network elements, thus we cannot trust them.  This does not
   mean that it is not possible to have any interaction with them, but

Rescorla                  Expires July 25, 2012                 [Page 5]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   it means that we must assume that they will behave maliciously and
   design a system which is secure even if they do so.

4.  Overview

   This section describes a typical RTCWeb session and shows how the
   various security elements interact and what guarantees are provided
   to the user.  The example in this section is a "best case" scenario
   in which we provide the maximal amount of user authentication and
   media privacy with the minimal level of trust in the calling service.
   Simpler versions with lower levels of security are also possible and
   are noted in the text where applicable.  It's also important to
   recognize the tension between security (or performance) and privacy.
   The example shown here is aimed towards settings where we are more
   concerned about secure calling than about privacy, but as we shall
   see, there are settings where one might wish to make different
   tradeoffs--this architecture is still compatible with those settings.

   For the purposes of this example, we assume the topology shown in the
   figure below.  This topology is derived from the topology shown in
   Figure 1, but separates Alice and Bob's identities from the process
   of signaling.  Specifically, Alice and Bob have relationships with
   some Identity Provider (IdP) that supports a protocol such OpenID or
   BrowserID) that can be used to attest to their identity.  This
   separation isn't particularly important in "closed world" cases where
   Alice and Bob are users on the same social network and have
   identities based on that network.  However, there are important
   settings where that is not the case, such as federation (calls from
   one network to another) and calling on untrusted sites, such as where
   two users who have a relationship via a given social network want to
   call each other on another, untrusted, site, such as a poker site.

Rescorla                  Expires July 25, 2012                 [Page 6]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

                               |                |
                               |     Signaling  |
                               |     Server     |
                               |                |
                                   ^        ^
                                  /          \
                          HTTPS  /            \   HTTPS
                                /              \
                               /                \
                              v                  v
                           JS API              JS API
                     +-----------+            +-----------+
                     |           |    Media   |           |
               Alice |  Browser  |<---------->|  Browser  | Bob
                     |           | (DTLS-SRTP)|           |
                     +-----------+            +-----------+
                           ^      ^--+     +--^     ^
                           |         |     |        |
                           v         |     |         v
                     +-----------+   |     |  +-----------+
                     |           |<--------+  |           |
                     |   IdP     |   |        |    IdP    |
                     |           |   +------->|           |
                     +-----------+            +-----------+

                 Figure 2: A call with IdP-based identity

4.1.  Initial Signaling

   Alice and Bob are both users of a common calling service; they both
   have approved the calling service to make calls (we defer the
   discussion of device access permissions till later).  They are both
   connected to the calling service via HTTPS and so know the origin
   with some level of confidence.  They also have accounts with some
   identity provider.  This sort of identity service is becoming
   increasingly common in the Web environment in technologies such
   (BrowserID, Federated Google Login, Facebook Connect, OAuth, OpenID,
   WebFinger), and is often provided as a side effect service of your
   ordinary accounts with some service.  In this example, we show Alice
   and Bob using a separate identity service, though they may actually
   be using the same identity service as calling service or have no
   identity service at all.

   Alice is logged onto the calling service and decides to call Bob. She
   can see from the calling service that he is online and the calling
   service presents a JS UI in the form of a button next to Bob's name

Rescorla                  Expires July 25, 2012                 [Page 7]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   which says "Call".  Alice clicks the button, which initiates a JS
   callback that instantiates a PeerConnection object.  This does not
   require a security check:  JS from any origin is allowed to get this

   Once the PeerConnection is created, the calling service JS needs to
   set up some media.  Because this is an audio/video call, it creates
   two MediaStreams, one connected to an audio input and one connected
   to a video input.  At this point the first security check is
   required:  untrusted origins are not allowed to access the camera and
   microphone.  In this case, because Alice is a long-term user of the
   calling service, she has made a permissions grant (i.e., a setting in
   the browser) to allow the calling service to access her camera and
   microphone any time it wants.  The browser checks this setting when
   the camera and microphone requests are made and thus allows them.

   In the current W3C API, once some streams have been added, Alice's
   browser + JS generates a signaling message The format of this data is
   currently undefined.  It may be a complete message as defined by ROAP
   [I-D.jennings-rtcweb-signaling] or separate media description and
   transport messages as defined in JSEP [REF] or may be assembled
   piecemeal by the JS.  In either case, it will contain:

   o  Media channel information
   o  ICE candidates
   o  A fingerprint attribute binding the communication to Alice's
      public key [RFC5763]

   [Note that the above implies that this information should appear in
   JSEP's transport-level messages.]  Prior to sending out the signaling
   message, the PeerConnection code contacts the identity service and
   obtains an assertion binding Alice's identity to her fingerprint.
   The exact details depend on the identity service (though as discussed
   in [I-D.rescorla-rtcweb-generic-idp] PeerConnection can be agnostic
   to them), but for now it's easiest to think of as a BrowserID
   assertion.  The assertion may bind other information to the identity
   besides the fingerprint, but at minimum it needs to bind the

   This message is sent to the signaling server, e.g., by XMLHttpRequest
   [XmlHttpRequest] or by WebSockets
   [I-D.ietf-hybi-thewebsocketprotocol].  The signaling server processes
   the message from Alice's browser, determines that this is a call to
   Bob and sends a signaling message to Bob's browser (again, the format
   is currently undefined).  The JS on Bob's browser processes it, and
   alerts Bob to the incoming call and to Alice's identity.  In this
   case, Alice has provided an identity assertion and so Bob's browser
   contacts Alice's identity provider (again, this is done in a generic

Rescorla                  Expires July 25, 2012                 [Page 8]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   way so the browser has no specific knowledge of the IdP) to verity
   the assertion.  This allows the browser to display a trusted element
   indicating that a call is coming in from Alice.  If Alice is in Bob's
   address book, then this interface might also include her real name, a
   picture, etc.  The calling site will also provide some user interface
   element (e.g., a button) to allow Bob to answer the call, though this
   is most likely not part of the trusted UI.

   If Bob agrees [I am ignoring early media for now], a PeerConnection
   is instantiated with the message from Alice's side.  Then, a similar
   process occurs as on Alice's browser:  Bob's browser verifies that
   the calling service is approved, the media streams are created, and a
   return signaling message containing media information, ICE
   candidates, and a fingerprint is sent back to Alice via the signaling
   service.  If Bob has a relationship with an IdP, the message will
   also come with an identity assertion.

   At this point, Alice and Bob each know that the other party wants to
   have a secure call with them.  Based purely on the interface provided
   by the signaling server, they know that the signaling server claims
   that the call is from Alice to Bob. Because the far end sent an
   identity assertion along with their message, they know that this is
   verifiable from the IdP as well.  Of course, the call works perfectly
   well if either Alice or Bob doesn't have a relationship with an IdP;
   they just get a lower level of assurance.  Moreover, Alice might wish
   to make an anonymous call through an anonymous calling site, in which
   case she would of course just not provide any identity assertion and
   the calling site would mask her identity from Bob.

4.2.  Media Consent Verification

   As described in ([I-D.ietf-rtcweb-security]; Section 4.2) This
   proposal specifies that media consent verification be performed via
   ICE.  Thus, Alice and Bob perform ICE checks with each other.  At the
   completion of these checks, they are ready to send non-ICE data.

   At this point, Alice knows that (a) Bob (assuming he is verified via
   his IdP) or someone else who the signaling service is claiming is Bob
   is willing to exchange traffic with her and (b) that either Bob is at
   the IP address which she has verified via ICE or there is an attacker
   who is on-path to that IP address detouring the traffic.  Note that
   it is not possible for an attacker who is on-path but not attached to
   the signaling service to spoof these checks because they do not have
   the ICE credentials.  Bob's security guarantees with respect to Alice
   are the converse of this.

Rescorla                  Expires July 25, 2012                 [Page 9]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

4.3.  DTLS Handshake

   Once the ICE checks have completed [more specifically, once some ICE
   checks have completed], Alice and Bob can set up a secure channel.
   This is performed via DTLS [RFC4347] (for the data channel) and DTLS-
   SRTP [RFC5763] for the media channel.  Specifically, Alice and Bob
   perform a DTLS handshake on every channel which has been established
   by ICE.  The total number of channels depends on the amount of
   muxing; in the most likely case we are using both RTP/RTCP mux and
   muxing multiple media streams on the same channel, in which case
   there is only one DTLS handshake.  Once the DTLS handshake has
   completed, the keys are extracted and used to key SRTP for the media

   At this point, Alice and Bob know that they share a set of secure
   data and/or media channels with keys which are not known to any
   third-party attacker.  If Alice and Bob authenticated via their IdPs,
   then they also know that the signaling service is not attacking them.
   Even if they do not use an IdP, as long as they have minimal trust in
   the signaling service not to perform a man-in-the-middle attack, they
   know that their communications are secure against the signaling
   service as well.

4.4.  Communications and Consent Freshness

   From a security perspective, everything from here on in is a little
   anticlimactic:  Alice and Bob exchange data protected by the keys
   negotiated by DTLS.  Because of the security guarantees discussed in
   the previous sections, they know that the communications are
   encrypted and authenticated.

   The one remaining security property we need to establish is "consent
   freshness", i.e., allowing Alice to verify that Bob is still prepared
   to receive her communications.  ICE specifies periodic STUN
   keepalizes but only if media is not flowing.  Because the consent
   issue is more difficult here, we require RTCWeb implementations to
   periodically send keepalives.  If a keepalive fails and no new ICE
   channels can be established, then the session is terminated.

5.  Detailed Technical Description

5.1.  Origin and Web Security Issues

   The basic unit of permissions for RTCWEB is the origin
   [I-D.abarth-origin].  Because the security of the origin depends on
   being able to authenticate content from that origin, the origin can
   only be securely established if data is transferred over HTTPS

Rescorla                  Expires July 25, 2012                [Page 10]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   [RFC2818].  Thus, clients MUST treat HTTP and HTTPS origins as
   different permissions domains and SHOULD NOT permit access to any
   RTCWEB functionality from scripts fetched over non-secure (HTTP)
   origins.  If an HTTPS origin contains mixed active content
   (regardless of whether it is present on the specific page attempting
   to access RTCWEB functionality), any access MUST be treated as if it
   came from the HTTP origin.  For instance, if a
   https://www.example.com/example.html loads
   https://www.example.com/example.js and
   http://www.example.org/jquery.js, any attempt by example.js to access
   RTCWeb functionality MUST be treated as if it came from
   http://www.example.com/.  Note that many browsers already track mixed
   content and either forbid it by default or display a warning. [[ OPEN
   ISSUE:  This seems to be wrong, but I'm not sure what's right yet. ]]

5.2.  Device Permissions Model

   Implementations MUST obtain explicit user consent prior to providing
   access to the camera and/or microphone.  Implementations MUST at
   minimum support the following two permissions models:

   o  Requests for one-time camera/microphone access.
   o  Requests for permanent access.

   In addition, they SHOULD support requests for access to a single
   communicating peer.  E.g., "Call customerservice@ford.com".  Browsers
   servicing such requests SHOULD clearly indicate that identity to the
   user when asking for permission.

   API Requirement:  The API MUST provide a mechanism for the requesting
      JS to indicate which of these forms of permissions it is
      requesting.  This allows the client to know what sort of user
      interface experience to provide.  In particular, browsers might
      display a non-invasive door hanger ("some features of this site
      may not work..." when asking for long-term permissions) but a more
      invasive UI ("here is your own video") for single-call
      permissions.  The API MAY grant weaker permissions than the JS
      asked for if the user chooses to authorize only those permissions,
      but if it intends to grant stronger ones it SHOULD display the
      appropriate UI for those permissions and MUST clearly indicate
      what permissions are being requested.

   API Requirement:  The API MUST provide a mechanism for the requesting
      JS to relinquish the ability to see or modify the media (e.g., via
      MediaStream.record()).  Combined with secure authentication of the
      communicating peer, this allows a user to be sure that the calling
      site is not accessing or modifying their conversion.

Rescorla                  Expires July 25, 2012                [Page 11]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   UI Requirement:  The UI MUST clearly indicate when the user's camera
      and microphone are in use.  This indication MUST NOT be
      suppressable by the JS and MUST clearly indicate how to terminate
      a call, and provide a UI means to immediately stop camera/
      microphone input without the JS being able to prevent it.

   UI Requirement:  If the UI indication of camera/microphone use are
      displayed in the browser such that minimizing the browser window
      would hide the indication, or the JS creating an overlapping
      window would hide the indication, then the browser SHOULD stop
      camera and microphone input.  [Note:  this may not be necessary in
      systems that are non-windows-based but that have good
      notifications support, such as phones.]

   Clients MAY permit the formation of data channels without any direct
   user approval.  Because sites can always tunnel data through the
   server, further restrictions on the data channel do not provide any
   additional security. (though see Section 5.3 for a related issue).

   Implementations which support some form of direct user authentication
   SHOULD also provide a policy by which a user can authorize calls only
   to specific counterparties.  Specifically, the implementation SHOULD
   provide the following interfaces/controls:

   o  Allow future calls to this verified user.
   o  Allow future calls to any verified user who is in my system
      address book (this only works with address book integration, of

   Implementations SHOULD also provide a different user interface
   indication when calls are in progress to users whose identities are
   directly verifiable.  Section 5.5 provides more on this.

5.3.  Communications Consent

   Browser client implementations of RTCWEB MUST implement ICE.  Server
   gateway implementations which operate only at public IP addresses may
   implement ICE-Lite.

   Browser implementations MUST verify reachability via ICE prior to
   sending any non-ICE packets to a given destination.  Implementations
   MUST NOT provide the ICE transaction ID to JavaScript.  [Note:  this
   document takes no position on the split between ICE in JS and ICE in
   the browser.  The above text is written the way it is for editorial
   convenience and will be modified appropriately if the WG decides on
   ICE in the JS.]

   Implementations MUST send keepalives no less frequently than every 30

Rescorla                  Expires July 25, 2012                [Page 12]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   seconds regardless of whether traffic is flowing or not.  If a
   keepalive fails then the implementation MUST either attempt to find a
   new valid path via ICE or terminate media for that ICE component.
   Note that ICE [RFC5245]; Section 10 keepalives use STUN Binding
   Indications which are one-way and therefore not sufficient.  We will
   need to define a new mechanism for this.  [PROPOSED SOLUTION:
   Replace STUN Binding Indications with STUN Binding Requests and
   require that a failed transaction causes the results above.]

5.4.  IP Location Privacy

   A side effect of the default ICE behavior is that the peer learns
   one's IP address, which leaks large amounts of location information,
   especially for mobile devices.  This has negative privacy
   consequences in some circumstances.  The following two API
   requirements are intended to mitigate this issue:

   API Requirement:  The API MUST provide a mechanism to suppress ICE
      negotiation (though perhaps to allow candidate gathering) until
      the user has decided to answer the call [note:  determining when
      the call has been answered is a question for the JS.]  This
      enables a user to prevent a peer from learning their IP address if
      they elect not to answer a call and also from learning whether the
      user is online.

   API Requirement:  The API MUST provide a mechanism for the calling
      application to indicate that only TURN candidates are to be used.
      This prevents the peer from learning one's IP address at all.  The
      API MUST provide a mechanism for the calling application to
      reconfigure an existing call to add non-TURN candidates.  Taken
      together, these requirements allow ICE negotiation to start
      immediately on incoming call notification, thus reducing post-dial
      delay, but also to avoid disclosing the user's IP address until
      they have decided to answer.

5.5.  Communications Security

   Implementations MUST implement DTLS and DTLS-SRTP.  All data channels
   MUST be secured via DTLS.  DTLS-SRTP MUST be offered for every media
   channel and MUST be the default; i.e., if an implementation receives
   an offer for DTLS-SRTP and SDES and/or plain RTP, DTLS-SRTP MUST be

   [OPEN ISSUE:  What should the settings be here?  MUST?]
   Implementations MAY support SDES and RTP for media traffic for
   backward compatibility purposes.

Rescorla                  Expires July 25, 2012                [Page 13]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   API Requirement:  The API MUST provide a mechanism to indicate that a
      fresh DTLS key pair is to be generated for a specific call.  This
      is intended to allow for unlinkability.  Note that there are also
      settings where it is attractive to use the same keying material
      repeatedly, especially those with key continuity-based

   API Requirement:  The API MUST provide a mechanism to indicate that a
      fresh DTLS key pair is to be generated for a specific call.  This
      is intended to allow for unlinkability.

   API Requirement:  When DTLS-SRTP is used, the API MUST NOT permit the
      JS to obtain the negotiated keying material.  This requirement
      preserves the end-to-end security of the media.

   UI Requirements:    A user-oriented client MUST provide an
      "inspector" interface which allows the user to determine the
      security characteristics of the media. [largely derived from
      The following properties SHOULD be displayed "up-front" in the
      browser chrome, i.e., without requiring the user to ask for them:

      *  A client MUST provide a user interface through which a user may
         determine the security characteristics for currently-displayed
         audio and video stream(s)
      *  A client MUST provide a user interface through which a user may
         determine the security characteristics for transmissions of
         their microphone audio and camera video.
      *  The "security characteristics" MUST include an indication as to
         whether or not the transmission is cryptographically protected
         and whether that protection is based on a key that was
         delivered out-of-band (from a server) or was generated as a
         result of a pairwise negotiation.
      *  If the far endpoint was directly verified (see Section 5.6) the
         "security characteristics" MUST include the verified
      The following properties are more likely to require some "drill-
      down" from the user:

      *  If the transmission is cryptographically protected, the The
         algorithms in use (For example:  "AES-CBC" or "Null Cipher".)
      *  If the transmission is cryptographically protected, the
         "security characteristics" MUST indicate whether PFS is
      *  If the transmission is cryptographically protected via an end-
         to-end mechanism the "security characteristics" MUST include
         some mechanism to allow an out-of-band verification of the
         peer, such as a certificate fingerprint or an SAS.

Rescorla                  Expires July 25, 2012                [Page 14]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

5.6.  Web-Based Peer Authentication

   In a number of cases, it is desirable for the endpoint (i.e., the
   browser) to be able to directly identity the endpoint on the other
   side without trusting only the signaling service to which they are
   connected.  For instance, users may be making a call via a federated
   system where they wish to get direct authentication of the other
   side.  Alternately, they may be making a call on a site which they
   minimally trust (such as a poker site) but to someone who has an
   identity on a site they do trust (such as a social network.)

   Recently, a number of Web-based identity technologies (OAuth,
   BrowserID, Facebook Connect), etc. have been developed.  While the
   details vary, what these technologies share is that they have a Web-
   based (i.e., HTTP/HTTPS identity provider) which attests to your
   identity.  For instance, if I have an account at example.org, I could
   use the example.org identity provider to prove to others that I was
   alice@example.org.  The development of these technologies allows us
   to separate calling from identity provision:  I could call you on
   Poker Galaxy but identify myself as alice@example.org.

   Whatever the underlying technology, the general principle is that the
   party which is being authenticated is NOT the signaling site but
   rather the user (and their browser).  Similarly, the relying party is
   the browser and not the signaling site.  Thus, the browser MUST
   securely generate the input to the IdP assertion process and MUST
   securely display the results of the verification process to the user
   in a way which cannot be imitated by the calling site.

   In order to make this work, we must standardize the following items:

   o  The precise information from the signaling message that must be
      cryptographically bound to the user's identity.  At minimum this
      MUST be the fingerprint, but we may choose to add other
      information as the signaling protocol firms up.  This will be
      defined in a future version of this document.
   o  The interface to the IdP.  [I-D.rescorla-rtcweb-generic-idp]
      specifies a specific protocol mechanism which allows the use of
      any identity protocol without requiring specific further protocol
      support in the browser.
   o  The JavaScript interfaces which the calling application can use to
      specify the IdP to use to generate assertions and to discover what
      assertions were received.  These interfaces should be defined in
      the W3C document.

Rescorla                  Expires July 25, 2012                [Page 15]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

6.  Security Considerations

   Much of the security analysis of this problem is contained in
   [I-D.ietf-rtcweb-security] or in the discussion of the particular
   issues above.  In order to avoid repetition, this section focuses on
   (a) residual threats that are not addressed by this document and (b)
   threats produced by failure/misbehavior of one of the components in
   the system.

6.1.  Communications Security

   While this document favors DTLS-SRTP, it permits a variety of
   communications security mechanisms and thus the level of
   communications security actually provided varies considerably.  Any
   pair of implementations which have multiple security mechanisms in
   common are subject to being downgraded to the weakest of those common
   mechanisms by any attacker who can modify the signaling traffic.  If
   communications are over HTTP, this means any on-path attacker.  If
   communications are over HTTPS, this means the signaling server.
   Implementations which wish to avoid downgrade attack should only
   offer the strongest available mechanism, which is DTLS/DTLS-SRTP.
   Note that the implication of this choice will be that interop to non-
   DTLS-SRTP devices will need to happen through gateways.

   Even if only DTLS/DTLS-SRTP are used, the signaling server can
   potentially mount a man-in-the-middle attack unless implementations
   have some mechanism for independently verifying keys.  The UI
   requirements in Section 5.5 are designed to provide such a mechanism
   for motivated/security conscious users, but are not suitable for
   general use.  The identity service mechanisms in Section 5.6 are more
   suitable for general use.  Note, however, that a malicious signaling
   service can strip off any such identity assertions, though it cannot
   forge new ones.

6.2.  Privacy

   The requirements in this document are intended to allow:

   o  Users to participate in calls without revealing their location.
   o  Potential callees to avoid revealing their location and even
      presence status prior to agreeing to answer a call.

   However, these privacy protections come at a performance cost in
   terms of using TURN relays and, in the latter case, delaying ICE.
   Sites SHOULD make users aware of these tradeoffs.

   Note that the protections provided here assume a non-malicious
   calling service.  As the calling service always knows the users

Rescorla                  Expires July 25, 2012                [Page 16]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   status and (absent the use of a technology like Tor) their IP
   address, they can violate the users privacy at will.  Users who wish
   privacy against the calling sites they are using must use separate
   privacy enhancing technologies such as Tor.

6.3.  Denial of Service

   The consent mechanisms described in this document are intended to
   mitigate denial of service attacks in which an attacker uses clients
   to send large amounts of traffic to a victim without the consent of
   the victim.  While these mechanisms are sufficient to protect victims
   who have not implemented RTCWEB at all, RTCWEB implementations need
   to be more careful.

   Consider the case of a call center which accepts calls via RTCWeb.
   An attacker proxies the call center's front-end and arranges for
   multiple clients to initiate calls to the call center.  Note that
   this requires user consent in many cases but because the data channel
   does not need consent, he can use that directly.  Since ICE will
   complete, browsers can then be induced to send large amounts of data
   to the victim call center if it supports the data channel at all.
   Preventing this attack requires that automated RTCWEB
   implemementations implement sensible flow control and have the
   ability to triage out (i.e., stop responding to ICE probes on) calls
   which are behaving badly, and especially to be prepared to remotely
   throttle the data channel in the absence of plausible audio and video
   (which the attacker cannot control).

   Another related attack is for the signaling service to swap the ICE
   candidates for the audio and video streams, thus forcing a browser to
   send video to the sink that the other victim expects will contain
   audio (perhaps it is only expecting audio!) potentially causing
   overload.  Muxing multiple media flows over a single transport makes
   it harder to individually suppress a single flow by denying ICE
   keepalives.  Media-level (RTCP) mechanisms must be used in this case.

   Note that attacks based on confusing one end or the other about
   consent are possible primarily even in the face of the third-party
   identity mechanism as long as major parts of the signaling messages
   are not signed.  On the other hand, signing the entire message
   severely restricts the capabilities of the calling application, so
   there are difficult tradeoffs here.

7.  Acknowledgements

   Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan,
   Matthew Kaufman, Magnus Westerland.

Rescorla                  Expires July 25, 2012                [Page 17]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

8.  References

8.1.  Normative References

              Barth, A., "The Web Origin Concept",
              draft-abarth-origin-09 (work in progress), November 2010.

              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-01 (work in progress),
              October 2011.

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

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.

8.2.  Informative References

              Fette, I. and A. Melnikov, "The WebSocket protocol",
              draft-ietf-hybi-thewebsocketprotocol-17 (work in
              progress), September 2011.

              Jennings, C., Rosenberg, J., Uberti, J., and R. Jesup,
              "RTCWeb Offer/Answer Protocol (ROAP)",
              draft-jennings-rtcweb-signaling-01 (work in progress),
              October 2011.

              Kaufman, M., "Client Security User Interface Requirements
              for RTCWEB", draft-kaufman-rtcweb-security-ui-00 (work in
              progress), June 2011.

              Rescorla, E., "RTCWeb Generic Identity Provider
              Interface", draft-rescorla-rtcweb-generic-idp-00 (work in
              progress), January 2012.

   [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.

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, April 2006.

Rescorla                  Expires July 25, 2012                [Page 18]

Internet-Draft              RTCWEB Sec. Arch.               January 2012

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [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.

              van Kesteren, A., "XMLHttpRequest Level 2".

Author's Address

   Eric Rescorla
   RTFM, Inc.
   2064 Edgewood Drive
   Palo Alto, CA  94303

   Phone:  +1 650 678 2350
   Email:  ekr@rtfm.com

Rescorla                  Expires July 25, 2012                [Page 19]