RTCWEB                                                       E. Rescorla
Internet-Draft                                                RTFM, Inc.
Intended status:  Standards Track                          July 16, 2012
Expires:  January 17, 2013

                      RTCWEB Security Architecture


   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for enabling real-time
   communications within user-agents using web technologies (e.g
   JavaScript).  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 model.



Status of this Memo

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   Internet-Drafts are working documents of the Internet Engineering
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   Internet-Drafts are draft documents valid for a maximum of six months

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   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 January 17, 2013.

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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 . . . . . . . . . . . . . . . . . . . . . .  9
     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  . . . . . . . . . . . . . . . . . 14
     5.6.  Web-Based Peer Authentication  . . . . . . . . . . . . . . 15
       5.6.1.  Trust Relationships: IdPs, APs, and RPs  . . . . . . . 16
       5.6.2.  Overview of Operation  . . . . . . . . . . . . . . . . 17
       5.6.3.  Items for Standardization  . . . . . . . . . . . . . . 19
       5.6.4.  Binding Identity Assertions to JSEP Offer/Answer
               Transactions . . . . . . . . . . . . . . . . . . . . . 19  Input to Assertion Generation Process  . . . . . . 19  Carrying Identity Assertions . . . . . . . . . . . 20
       5.6.5.  IdP Interaction Details  . . . . . . . . . . . . . . . 20  General Message Structure  . . . . . . . . . . . . 20  IdP Proxy Setup  . . . . . . . . . . . . . . . . . 21
     5.7.  Security Considerations  . . . . . . . . . . . . . . . . . 26
       5.7.1.  Communications Security  . . . . . . . . . . . . . . . 26
       5.7.2.  Privacy  . . . . . . . . . . . . . . . . . . . . . . . 27
       5.7.3.  Denial of Service  . . . . . . . . . . . . . . . . . . 27
       5.7.4.  IdP Authentication Mechanism . . . . . . . . . . . . . 28  IdP Well-known URI . . . . . . . . . . . . . . . . 29  Privacy of IdP-generated identities and the
                   hosting site . . . . . . . . . . . . . . . . . . . 29  Security of Third-Party IdPs . . . . . . . . . . . 29  Web Security Feature Interactions  . . . . . . . . 29
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 30
   7.  Changes since -02  . . . . . . . . . . . . . . . . . . . . . . 30
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 30
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 30
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 31
   Appendix A.  Example IdP Bindings to Specific Protocols  . . . . . 32
     A.1.  BrowserID  . . . . . . . . . . . . . . . . . . . . . . . . 32
     A.2.  OAuth  . . . . . . . . . . . . . . . . . . . . . . . . . . 35
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 36

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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

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   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, but in some cases because we are on a topologically
      restricted network, such as behind a firewall).
   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 and 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

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   mean that it is not possible to have any interaction with them, but
   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.

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                               |                |
                               |     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

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   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 [I-D.ietf-rtcweb-jsep]

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

   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 Section 5.6 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 [RFC6455] 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 way so the browser has no specific knowledge of the
   IdP) to verify 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

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   answer the call, though this is most likely not part of the trusted

   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.

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

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   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 exported [RFC5705] and used to key SRTP for
   the media channels.

   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 [RFC6454].
   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 [RFC2818].
   Thus, clients MUST treat HTTP and HTTPS origins as different
   permissions domains.  [Note:  this follows directly from the origin
   security model and is stated here merely for clarity.]

   Many web browsers currently forbid by default any active mixed
   content on HTTPS pages.  I.e., when JS is loaded from an HTTP origin
   onto an HTTPS page, an error is displayed and the content is not

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   executed unless the user overrides the error.  Any browser which
   enforces such a policy will also not permit access to RTCWEB
   functionality from mixed content pages.  It is RECOMMENDED that
   browsers which allow active mixed content nevertheless disable RTCWEB
   functionality in mixed content settings. [[ OPEN ISSUE:  Should this
   be a 2119 MUST?  It's not clear what set of conditions would make
   this OK, other than that browser manufacturers have traditionally
   been permissive here here.]]  Note that it is possible for a page
   which was not mixed content to become mixed content during the
   duration of the call.  Implementations MAY choose to terminate the
   call or display a warning at that point, but it is also permissible
   to ignore this condition.  This is a deliberate implementation
   complexity versus security tradeoff. [[ OPEN ISSUE::  Should we be
   more aggressive about this?]]

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 for HTTPS

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

   Because HTTP origins cannot be securely established against network
   attackers, implementations MUST NOT allow the setting of permanent
   access permissions for HTTP origins.  Implementations MAY also opt to
   refuse all permissions grants for HTTP origins, but it is RECOMMENDED
   that currently they support one-time camera/microphone 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 browser to know what sort of user
      interface experience to provide to the user, including what
      permissions to request from the user and hence that to enforce
      later.  For instance, 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

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      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.

   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
   MUST implement either full ICE or ICE-Lite.

   Browser implementations MUST verify reachability via ICE prior to

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   sending any non-ICE packets to a given destination.  Implementations
   MUST NOT provide the ICE transaction ID to JavaScript during the
   lifetime of the transaction (i.e., during the period when the ICE
   stack would accept a new response for that transaction).  [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.]  The JS MUST NOT be permitted to control
   the local ufrag and password, though it of course knows it.

   While continuing consent is required, that ICE [RFC5245]; Section 10
   keepalives STUN Binding Indications are one-way and therefore not
   sufficient.  The current WG consensus is to use ICE Binding Requests
   for continuing consent freshness.  ICE already requires that
   implementations respond to such requests, so this approach is
   maximally compatible.  A separate document will profile the ICE
   timers to be used [[TODO:  insert REF here when available.]]

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 API requirements in this
   section are intended to mitigate this issue.  Note that these
   requirements are NOT intended to protect the user's IP address from a
   malicious site.  In general, the site will learn at least a user's
   server reflexive address from any HTTP transaction.  Rather, these
   requirements are intended to allow a site to cooperate with the user
   to hide the user's IP address from the other side of the call.
   Hiding the user's IP address from the server requires some sort of
   explicit privacy preserving mechanism on the client (e.g., Torbutton
   [https://www.torproject.org/torbutton/]) and is out of scope for this

   API Requirement:  The API MUST provide a mechanism to allow the JS 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 JS to indicate that only TURN candidates are to be
      used.  This prevents the peer from learning one's IP address at

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   API Requirement:  The API MUST provide a mechanism for the calling
      application to reconfigure an existing call to add non-TURN
      candidates.  Taken together, this and the previous requirement
      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.  They also allow users to completely hide their IP address
      for the duration of the call.  Finally, they allow a mechanism for
      the user to optimize performance by reconfiguring to allow non-
      turn candidates during an active call if the user decides they no
      longer need to hide their IP address

5.5.  Communications Security

   Implementations MUST implement DTLS [RFC4347] and DTLS-SRTP
   [RFC5763][RFC5764].  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, DTLS-SRTP MUST be selected.  Media traffic MUST NOT be sent
   over plain (unencrypted) RTP.

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

   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:  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)

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      *  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 the cryptographic keys were delivered out-of-band (from
         a server) or were generated as a result of a pairwise
      *  If the far endpoint was directly verified, either via a third-
         party verifiable X.509 certificate or via a Web IdP mechanism
         (see Section 5.6) the "security characteristics" MUST include
         the verified information.

      The following properties are more likely to require some "drill-
      down" from the user:

      *  The "security characteristics" MUST indicate the cryptographic
         algorithms in use (For example:  "AES-CBC" or "Null Cipher".)
      *  The "security characteristics" MUST indicate whether PFS is
      *  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.

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

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   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.

   The mechanisms defined in this document do not require the browser to
   implement any particular identity protocol or to support any
   particular IdP.  Instead, this document provides a generic interface
   which any IdP can implement.  Thus, new IdPs and protocols can be
   introduced without change to either the browser or the calling
   service.  This avoids the need to make a commitment to any particular
   identity protocol, although browsers may opt to directly implement
   some identity protocols in order to provide superior performance or
   UI properties.

5.6.1.  Trust Relationships: IdPs, APs, and RPs

   Any federated identity protocol has three major participants:

   Authenticating Party (AP):  The entity which is trying to establish
      its identity.

   Identity Provider (IdP):  The entity which is vouching for the AP's

   Relying Party (RP):  The entity which is trying to verify the AP's

   The AP and the IdP have an account relationship of some kind:  the AP
   registers with the IdP and is able to subsequently authenticate
   directly to the IdP (e.g., with a password).  This means that the
   browser must somehow know which IdP(s) the user has an account
   relationship with.  This can either be something that the user
   configures into the browser or that is configured at the calling site
   and then provided to the PeerConnection by the calling site.

   At a high level there are two kinds of IdPs:

   Authoritative:    IdPs which have verifiable control of some section
      of the identity space.  For instance, in the realm of e-mail, the
      operator of "example.com" has complete control of the namespace
      ending in "@example.com".  Thus, "alice@example.com" is whoever
      the operator says it is.  Examples of systems with authoritative
      identity providers include DNSSEC, RFC 4474, and Facebook Connect
      (Facebook identities only make sense within the context of the
      Facebook system).

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   Third-Party:    IdPs which don't have control of their section of the
      identity space but instead verify user's identities via some
      unspecified mechanism and then attest to it.  Because the IdP
      doesn't actually control the namespace, RPs need to trust that the
      IdP is correctly verifying AP identities, and there can
      potentially be multiple IdPs attesting to the same section of the
      identity space.  Probably the best-known example of a third-party
      identity provider is SSL certificates, where there are a large
      number of CAs all of whom can attest to any domain name.

   If an AP is authenticating via an authoritative IdP, then the RP does
   not need to explicitly trust the IdP at all:  as long as the RP knows
   how to verify that the IdP indeed made the relevant identity
   assertion (a function provided by the mechanisms in this document),
   then any assertion it makes about an identity for which it is
   authoritative is directly verifiable.

   By contrast, if an AP is authenticating via a third-party IdP, the RP
   needs to explicitly trust that IdP (hence the need for an explicit
   trust anchor list in PKI-based SSL/TLS clients).  The list of
   trustable IdPs needs to be configured directly into the browser,
   either by the user or potentially by the browser manufacturer.  This
   is a significant advantage of authoritative IdPs and implies that if
   third-party IdPs are to be supported, the potential number needs to
   be fairly small.

5.6.2.  Overview of Operation

   In order to provide security without trusting the calling site, the
   PeerConnection component of the browser must interact directly with
   the IdP.  The details of the mechanism are described in the W3C API
   specification, but the general idea is that the PeerConnection
   component downloads JS from a specific location on the IdP dictated
   by the IdP domain name.  That JS (the "IdP proxy") runs in an
   isolated security context within the browser and the PeerConnection
   talks to it via a secure message passing channel.

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         |  https://calling-site.example.com  |
         |                                    |
         |                                    |
         |                                    |
         |         Calling JS Code            |
         |                ^                   |
         |                | API Calls         |
         |                v                   |
         |         PeerConnection             |
         |                ^                   |
         |                | postMessage()     |
         |                v                   |
         |    +-------------------------+     |     +---------------+
         |    | https://idp.example.org |     |     |               |
         |    |                         |<--------->|   Identity    |
         |    |        IdP JS           |     |     |   Provider    |
         |    |                         |     |     |               |
         |    +-------------------------+     |     +---------------+
         |                                    |

   When the PeerConnection object wants to interact with the IdP, the
   sequence of events is as follows:

   1.  The browser (the PeerConnection component) instantiates an IdP
       proxy with its source at the IdP.  This allows the IdP to load
       whatever JS is necessary into the proxy, which runs in the IdP's
       security context.
   2.  If the user is not already logged in, the IdP does whatever is
       required to log them in, such as soliciting a username and
   3.  Once the user is logged in, the IdP proxy notifies the browser
       that it is ready.
   4.  The browser and the IdP proxy communicate via a standardized
       series of messages delivered via postMessage.  For instance, the
       browser might request the IdP proxy to sign or verify a given
       identity assertion.

   This approach allows us to decouple the browser from any particular
   identity provider; the browser need only know how to load the IdP's
   JavaScript--which is deterministic from the IdP's identity--and the
   generic protocol for requesting and verifying assertions.  The IdP
   provides whatever logic is necessary to bridge the generic protocol
   to the IdP's specific requirements.  Thus, a single browser can
   support any number of identity protocols, including being forward
   compatible with IdPs which did not exist at the time the browser was

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5.6.3.  Items for Standardization

   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 and a mechanism for
      carrying assertions in JSEP messages.  Section 5.6.4
   o  The interface to the IdP.  Section 5.6.5 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.

   The first two items are defined in this document.  The final one is
   defined in the companion W3C WebRTC API specification [TODO:REF]

5.6.4.  Binding Identity Assertions to JSEP Offer/Answer Transactions  Input to Assertion Generation Process

   As discussed above, an identity assertion binds the user's identity
   (as asserted by the IdP) to the JSEP offer/exchange transaction and
   specifically to the media.  In order to achieve this, the
   PeerConnection must provide the DTLS-SRTP fingerprint to be bound to
   the identity.  This is provided in a JSON structure for
   extensibility, as shown below:

      "fingerprint" :

   The "algorithm" and digest values correspond directly to the
   algorithm and digest in the a=fingerprint line of the SDP.

   Note:  this structure does not need to be interpreted by the IdP or
   the IdP proxy.  It is consumed solely by the RP's browser.  The IdP
   merely treats it as an opaque value to be attested to.  Thus, new
   parameters can be added to the assertion without modifying the IdP.

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   Once an IdP has generated an assertion, the JSEP message.  This is
   done by adding a new a-line to the SDP, of the form a=identity.  The
   sole contents of this value are a base-64-encoded version of the
   identity assertion.  For example:

    o=- 1181923068 1181923196 IN IP4 ua1.example.com
    c=IN IP4 ua1.example.com
    a=fingerprint: SHA-1 \
    a=identity: \
      ImlkcCI6eyJkb21haW4iOiAiZXhhbXBsZS5vcmciLCAicHJvdG9jb2wiOiAiYm9n \
      dXMifSwiYXNzZXJ0aW9uIjpcIntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5v \
      cmdcIixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIs \
    t=0 0
    m=audio 6056 RTP/AVP 0
    a=pcfg:1 t=1

   Each identity attribute should be paired (and attests to) with an
   a=fingerprint attribute and therefore can exist either at the session
   or media level.  Multiple identity attributes may appear at either
   level, though implementations are discouraged from doing this unless
   they have a clear idea of what security claim they intend to be

5.6.5.  IdP Interaction Details  General Message Structure

   Messages between the PeerConnection object and the IdP proxy are
   formatted using JSON [RFC4627].  For instance, the PeerConnection
   would request a signature with the following "SIGN" message:

                   "id": "1",

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   All messages MUST contain a "type" field which indicates the general
   meaning of the message.

   All requests from the PeerConnection object MUST contain an "id"
   field which MUST be unique for that PeerConnection object.  Any
   responses from the IdP proxy MUST contain the same id in response,
   which allows the PeerConnection to correlate requests and responses.

   All requests from the PeerConnection object MUST contain an "origin"
   field containing the origin of the JS which initiated the PC (i.e.,
   the URL of the calling site).  This origin value can be used by the
   IdP to make access control decisions.  For instance, an IdP might
   only issue identity assertions for certain calling services in the
   same way that some IdPs require that relying Web sites have an API
   key before learning user identity.

   Any message-specific data is carried in a "message" field.  Depending
   on the message type, this may either be a string or a richer JSON
   object.  Errors

   If an error occurs, the IdP sends a message of type "ERROR".  The
   message MAY have an "error" field containing freeform text data which
   containing additional information about what happened.  For instance:

                     "error":"Signature verification failed"

                          Figure 3: Example error  IdP Proxy Setup

   In order to perform an identity transaction, the PeerConnection must
   first create an IdP proxy.  While the details of this are specified
   in the W3C API document, from the perspective of this specification,
   however, the relevant facts are:

   o  The JS runs in the IdP's security context with the base page
      retrieved from the URL specified in Section
   o  The usual browser sandbox isolation mechanisms MUST be enforced
      with respect to the IdP proxy.
   o  JS running in the IdP proxy MUST be able to send and receive
      messages to the PeerConnection and the PC and IdP proxy are able
      to verify the source and destination of these messages.

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   Initially the IdP proxy is in an unready state; the IdP JS must be
   loaded and there may be several round trips to the IdP server, for
   instance to log the user in.  When the IdP proxy is ready to receive
   commands, it delivers a "ready" message.  As this message is
   unsolicited, it simply contains:

                   { "type":"READY" }

   [[ OPEN ISSUE:  if the W3C half of this converges on WebIntents, then
   the READY message will not be necessary.]]

   Once the PeerConnection object receives the ready message, it can
   send commands to the IdP proxy.  Determining the IdP URI

   Each IdP proxy instance is associated with two values:

   domain name:  The IdP's domain name
   protocol:  The specific IdP protocol which the IdP is using.  This is
      a completely IdP-specific string, but allows an IdP to implement
      two protocols in parallel.  This value may be the empty string.

   Each IdP MUST serve its initial entry page (i.e., the one loaded by
   the IdP proxy) from the well-known URI specified in "/.well-known/
   idp-proxy/<protocol>" on the IdP's web site.  This URI MUST be loaded
   via HTTPS [RFC2818].  For example, for the IdP "identity.example.com"
   and the protocol "example", the URL would be:

     https://example.com/.well-known/idp-proxy/example  Authenticating Party

   How an AP determines the appropriate IdP domain is out of scope of
   this specification.  In general, however, the AP has some actual
   account relationship with the IdP, as this identity is what the IdP
   is attesting to.  Thus, the AP somehow supplies the IdP information
   to the browser.  Some potential mechanisms include:

   o  Provided by the user directly.
   o  Selected from some set of IdPs known to the calling site.  E.g., a
      button that shows "Authenticate via Facebook Connect"  Relying Party

   Unlike the AP, the RP need not have any particular relationship with
   the IdP.  Rather, it needs to be able to process whatever assertion
   is provided by the AP.  As the assertion contains the IdP's identity,

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   the URI can be constructed directly from the assertion, and thus the
   RP can directly verify the technical validity of the assertion with
   no user interaction.  Authoritative assertions need only be
   verifiable.  Third-party assertions also MUST be verified against
   local policy, as described in Section  Requesting Assertions

   In order to request an assertion, the PeerConnection sends a "SIGN"
   message.  Aside from the mandatory fields, this message has a
   "message" field containing a string.  The contents of this string are
   defined above, but are opaque from the perspective of the IdP.

   A successful response to a "SIGN" message contains a message field
   which is a JS dictionary dictionary consisting of two fields:

   idp:  A dictionary containing the domain name of the provider and the
      protocol string
   assertion:  An opaque field containing the assertion itself.  This is
      only interpretable by the idp or its proxy.

   Figure 4 shows an example transaction, with the message "abcde..."
   (remember, the messages are opaque at this layer) being signed and
   bound to identity "ekr@example.org".  In this case, the message has
   presumably been digitally signed/MACed in some way that the IdP can
   later verify it, but this is an implementation detail and out of
   scope of this document.  Line breaks are inserted solely for

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       PeerConnection -> IdP proxy:

       IdPProxy -> PeerConnection:
           "message": {
               "domain": "example.org"
               "protocol": "bogus"

                    Figure 4: Example assertion request  Verifying Assertions

   In order to verify an assertion, an RP sends a "VERIFY" message to
   the IdP proxy containing the assertion supplied by the AP in the
   "message" field.

   The IdP proxy verifies the assertion.  Depending on the identity
   protocol, this may require one or more round trips to the IdP.  For
   instance, an OAuth-based protocol will likely require using the IdP
   as an oracle, whereas with BrowserID the IdP proxy can likely verify
   the signature on the assertion without contacting the IdP, provided
   that it has cached the IdP's public key.

   Regardless of the mechanism, if verification succeeds, a successful
   response from the IdP proxy MUST contain a message field consisting
   of a dictionary/hash with the following fields:

   identity  The identity of the AP from the IdP's perspective.  Details
      of this are provided in Section

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   contents  The original unmodified string provided by the AP in the
      original SIGN request.

   Figure 5 shows an example transaction.  Line breaks are inserted
   solely for readability.

         PeerConnection -> IdP Proxy:

         IdP Proxy -> PeerConnection:
            "message": {
              "identity" : {
                "name" : "bob@example.org",
                "displayname" : "Bob"

                  Figure 5: Example verification request  Identity Formats

   Identities passed from the IdP proxy to the PeerConnection are
   structured as JSON dictionaries with one mandatory field:  "name".
   This field MUST consist of an RFC822-formatted string representing
   the user's identity. [[ OPEN ISSUE:  Would it be better to have a
   typed field? ]] The PeerConnection API MUST check this string as

   1.  If the RHS of the string is equal to the domain name of the IdP
       proxy, then the assertion is valid, as the IdP is authoritative
       for this domain.
   2.  If the RHS of the string is not equal to the domain name of the
       IdP proxy, then the PeerConnection object MUST reject the
       assertion unless (a) the IdP domain is listed as an acceptable

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       third-party IdP and (b) local policy is configured to trust this
       IdP domain for the RHS of the identity string.

   Sites which have identities that do not fit into the RFC822 style
   (for instance, Facebook ids are simple numeric values) SHOULD convert
   them to this form by appending their IdP domain (e.g.,
   12345@identity.facebook.com), thus ensuring that they are
   authoritative for the identity.

   The IdP proxy MAY also include a "displayname" field which contains a
   more user-friendly identity assertion.  Browsers SHOULD take care in
   the UI to distinguish the "name" assertion which is verifiable
   directly from the "displayname" which cannot be verified and thus
   relies on trust in the IdP.  In future, we may define other fields to
   allow the IdP to provide more information to the browser.  [[OPEN
   ISSUE:  Should this field exist?  Is it confusing? ]]

5.7.  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.

5.7.1.  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

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   service can strip off any such identity assertions, though it cannot
   forge new ones.  Note that all of the third-party security mechanisms
   available (whether X.509 certificates or a third-party IdP) rely on
   the security of the third party--this is of course also true of your
   connection to the Web site itself.  Users who wish to assure
   themselves of security against a malicious identity provider can only
   do so by verifying peer credentials directly, e.g., by checking the
   peer's fingerprint against a value delivered out of band.

5.7.2.  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
   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. Combined RTCWEB/Tor
   implementations SHOULD arrange to route the media as well as the
   signaling through Tor. [Currently this will produce very suboptimal

5.7.3.  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

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   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.  Either media-level (RTCP) mechanisms must be used or the
   implementation must deny responses entirely, thus termnating the

   Yet another attack, suggested by Magnus Westerlund, is for the
   attacker to cross-connect offers and answers as follows.  It induces
   the victim to make a call and then uses its control of other users
   browsers to get them to attempt a call to someone.  It then
   translates their offers into apparent answers to the victim, which
   looks like large-scale parallel forking.  The victim still responds
   to ICE responses and now the browsers all try to send media to the
   victim.  Implementations can defend themselves from this attack by
   only responding to ICE Binding Requests for a limited number of
   remote ufrags (this is the reason for the requirement that the JS not
   be able to control the ufrag and password).

   Note that attacks based on confusing one end or the other about
   consent are possible 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.

5.7.4.  IdP Authentication Mechanism

   This mechanism relies for its security on the IdP and on the
   PeerConnection correctly enforcing the security invariants described
   above.  At a high level, the IdP is attesting that the user
   identified in the assertion wishes to be associated with the
   assertion.  Thus, it must not be possible for arbitrary third parties
   to get assertions tied to a user or to produce assertions that RPs
   will accept.

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   As described in Section the IdP proxy HTML/JS landing page
   is located at a well-known URI based on the IdP's domain name.  This
   requirement prevents an attacker who can write some resources at the
   IdP (e.g., on one's Facebook wall) from being able to impersonate the
   IdP.  Privacy of IdP-generated identities and the hosting site

   Depending on the structure of the IdP's assertions, the calling site
   may learn the user's identity from the perspective of the IdP.  In
   many cases this is not an issue because the user is authenticating to
   the site via the IdP in any case, for instance when the user has
   logged in with Facebook Connect and is then authenticating their call
   with a Facebook identity.  However, in other case, the user may not
   have already revealed their identity to the site.  In general, IdPs
   SHOULD either verify that the user is willing to have their identity
   revealed to the site (e.g., through the usual IdP permissions dialog)
   or arrange that the identity information is only available to known
   RPs (e.g., social graph adjacencies) but not to the calling site.
   The "origin" field of the signature request can be used to check that
   the user has agreed to disclose their identity to the calling site;
   because it is supplied by the PeerConnection it can be trusted to be
   correct.  Security of Third-Party IdPs

   As discussed above, each third-party IdP represents a new universal
   trust point and therefore the number of these IdPs needs to be quite
   limited.  Most IdPs, even those which issue unqualified identities
   such as Facebook, can be recast as authoritative IdPs (e.g.,
   123456@facebook.com).  However, in such cases, the user interface
   implications are not entirely desirable.  One intermediate approach
   is to have special (potentially user configurable) UI for large
   authoritative IdPs, thus allowing the user to instantly grasp that
   the call is being authenticated by Facebook, Google, etc.  Web Security Feature Interactions

   A number of optional Web security features have the potential to
   cause issues for this mechanism, as discussed below.  Popup Blocking

   If the user is not already logged into the IdP, the IdP proxy may
   need to pop up a top level window in order to prompt the user for
   their authentication information (it is bad practice to do this in an

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   IFRAME inside the window because then users have no way to determine
   the destination for their password).  If the user's browser is
   configured to prevent popups, this may fail (depending on the exact
   algorithm that the popup blocker uses to suppress popups).  It may be
   necessary to provide a standardized mechanism to allow the IdP proxy
   to request popping of a login window.  Note that care must be taken
   here to avoid PeerConnection becoming a general escape hatch from
   popup blocking.  One possibility would be to only allow popups when
   the user has explicitly registered a given IdP as one of theirs (this
   is only relevant at the AP side in any case).  This is what
   WebIntents does, and the problem would go away if WebIntents is used.  Third Party Cookies

   Some browsers allow users to block third party cookies (cookies
   associated with origins other than the top level page) for privacy
   reasons.  Any IdP which uses cookies to persist logins will be broken
   by third-party cookie blocking.  One option is to accept this as a
   limitation; another is to have the PeerConnection object disable
   third-party cookie blocking for the IdP proxy.

6.  Acknowledgements

   Bernard Aboba, Harald Alvestrand, Dan Druta, Cullen Jennings, Hadriel
   Kaplan, Matthew Kaufman, Jim McEachem, Martin Thomson, Magnus

7.  Changes since -02

   The following changes have been made since the -02 draft.

   o  Forbid persistent HTTP permissions.
   o  Clarified the text in S 5.4 to clearly refer to requirements on
      the API to provide functionality to the site.
   o  Fold in the IETF portion of draft-rescorla-rtcweb-generic-idp
   o  Retarget the continuing consent section to assume Binding Requests
   o  Editorial improvements

8.  References

8.1.  Normative References

              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-03 (work in progress),

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              June 2012.

              Perumal, M., Wing, D., and H. Kaplan, "STUN Usage for
              Consent Freshness and Session Liveness",
              draft-muthu-behave-consent-freshness-01 (work in
              progress), July 2012.

   [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.

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

   [RFC4627]  Crockford, D., "The application/json Media Type for
              JavaScript Object Notation (JSON)", RFC 4627, July 2006.

   [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.

   [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.

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              December 2011.

8.2.  Informative References

              Uberti, J. and C. Jennings, "Javascript Session
              Establishment Protocol", draft-ietf-rtcweb-jsep-01 (work
              in progress), June 2012.

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

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

   [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.

   [RFC5705]  Rescorla, E., "Keying Material Exporters for Transport
              Layer Security (TLS)", RFC 5705, March 2010.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, December 2011.

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

Appendix A.  Example IdP Bindings to Specific Protocols

   This section provides some examples of how the mechanisms described
   in this document could be used with existing authentication protocols
   such as BrowserID or OAuth.  Note that this does not require browser-
   level support for either protocol.  Rather, the protocols can be fit
   into the generic framework.  (Though BrowserID in particular works
   better with some client side support).

A.1.  BrowserID

   BrowserID [https://browserid.org/] is a technology which allows a
   user with a verified email address to generate an assertion
   (authenticated by their identity provider) attesting to their
   identity (phrased as an email address).  The way that this is used in
   practice is that the relying party embeds JS in their site which
   talks to the BrowserID code (either hosted on a trusted intermediary
   or embedded in the browser).  That code generates the assertion which
   is passed back to the relying party for verification.  The assertion
   can be verified directly or with a Web service provided by the
   identity provider.  It's relatively easy to extend this functionality
   to authenticate RTCWEB calls, as shown below.

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   +----------------------+                     +----------------------+
   |                      |                     |                      |
   |    Alice's Browser   |                     |     Bob's Browser    |
   |                      | OFFER ------------> |                      |
   |   Calling JS Code    |                     |    Calling JS Code   |
   |          ^           |                     |          ^           |
   |          |           |                     |          |           |
   |          v           |                     |          v           |
   |    PeerConnection    |                     |    PeerConnection    |
   |       |      ^       |                     |       |      ^       |
   | Finger|      |Signed |                     |Signed |      |       |
   | print |      |Finger |                     |Finger |      |"Alice"|
   |       |      |print  |                     |print  |      |       |
   |       v      |       |                     |       v      |       |
   |   +--------------+   |                     |   +---------------+  |
   |   |  IdP Proxy   |   |                     |   |  IdP Proxy    |  |
   |   |     to       |   |                     |   |     to        |  |
   |   |  BrowserID   |   |                     |   |  BrowserID    |  |
   |   |  Signer      |   |                     |   |  Verifier     |  |
   |   +--------------+   |                     |   +---------------+  |
   |           ^          |                     |          ^           |
   +-----------|----------+                     +----------|-----------+
               |                                           |
               | Get certificate                           |
               v                                           | Check
   +----------------------+                                | certificate
   |                      |                                |
   |       Identity       |/-------------------------------+
   |       Provider       |
   |                      |

   The way this mechanism works is as follows.  On Alice's side, Alice
   goes to initiate a call.

   1.  The calling JS instantiates a PeerConnection and tells it that it
       is interested in having it authenticated via BrowserID (i.e., it
       provides "browserid.org" as the IdP name.)
   2.  The PeerConnection instantiates the BrowserID signer in the IdP
   3.  The BrowserID signer contacts Alice's identity provider,
       authenticating as Alice (likely via a cookie).
   4.  The identity provider returns a short-term certificate attesting
       to Alice's identity and her short-term public key.
   5.  The Browser-ID code signs the fingerprint and returns the signed
       assertion + certificate to the PeerConnection.

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   6.  The PeerConnection returns the signed information to the calling
       JS code.
   7.  The signed assertion gets sent over the wire to Bob's browser
       (via the signaling service) as part of the call setup.

   Obviously, the format of the signed assertion varies depending on
   what signaling style the WG ultimately adopts.  However, for
   concreteness, if something like ROAP were adopted, then the entire
   message might look like:

        "seq": 1
      o=- 2890844526 2890842807 IN IP4\n
      s= \n
      c=IN IP4\n
      t=2873397496 2873404696\n
      m=audio 49170 RTP/AVP 0\n
      a=fingerprint: SHA-1 \
            "idp":{     // Standardized
            "assertion":   // Contents are browserid-specific
              "\"assertion\": {
                \"digest\":\"<hash of the contents from the browser>\",
                \"audience\": \"[TBD]\"
                \"valid-until\": 1308859352261,
               \"certificate\": {
                 \"email\": \"rescorla@example.org\",
                 \"public-key\": \"<ekrs-public-key>\",
                 \"valid-until\": 1308860561861,
               }" // certificate is signed by example.org

   Note that while the IdP here is specified as "browserid.org", the
   actual certificate is signed by example.org.  This is because
   BrowserID is a combined authoritative/third-party system in which
   browserid.org delegates the right to be authoritative (what BrowserID
   calls primary) to individual domains.

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   On Bob's side, he receives the signed assertion as part of the call
   setup message and a similar procedure happens to verify it.

   1.  The calling JS instantiates a PeerConnection and provides it the
       relevant signaling information, including the signed assertion.
   2.  The PeerConnection instantiates the IdP proxy which examines the
       IdP name and brings up the BrowserID verification code.
   3.  The BrowserID verifier contacts the identity provider to verify
       the certificate and then uses the key to verify the signed
   4.  Alice's verified identity is returned to the PeerConnection (it
       already has the fingerprint).
   5.  At this point, Bob's browser can display a trusted UI indication
       that Alice is on the other end of the call.

   When Bob returns his answer, he follows the converse procedure, which
   provides Alice with a signed assertion of Bob's identity and keying

A.2.  OAuth

   While OAuth is not directly designed for user-to-user authentication,
   with a little lateral thinking it can be made to serve.  We use the
   following mapping of OAuth concepts to RTCWEB concepts:

              | OAuth                | RTCWEB               |
              | Client               | Relying party        |
              | Resource owner       | Authenticating party |
              | Authorization server | Identity service     |
              | Resource server      | Identity service     |

                                  Table 1

   The idea here is that when Alice wants to authenticate to Bob (i.e.,
   for Bob to be aware that she is calling).  In order to do this, she
   allows Bob to see a resource on the identity provider that is bound
   to the call, her identity, and her public key.  Then Bob retrieves
   the resource from the identity provider, thus verifying the binding
   between Alice and the call.

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           Alice                       IdP                       Bob
           Call-Id, Fingerprint  ------->
           <------------------- Auth Code
           Auth Code ---------------------------------------------->
                                        <----- Get Token + Auth Code
                                        Token --------------------->
                                        <------------- Get call-info
                                        Call-Id, Fingerprint ------>

   This is a modified version of a common OAuth flow, but omits the
   redirects required to have the client point the resource owner to the
   IdP, which is acting as both the resource server and the
   authorization server, since Alice already has a handle to the IdP.

   Above, we have referred to "Alice", but really what we mean is the
   PeerConnection.  Specifically, the PeerConnection will instantiate an
   IFRAME with JS from the IdP and will use that IFRAME to communicate
   with the IdP, authenticating with Alice's identity (e.g., cookie).
   Similarly, Bob's PeerConnection instantiates an IFRAME to talk to the

Author's Address

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

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

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