RTCWEB                                                       E. Rescorla
Internet-Draft                                                RTFM, Inc.
Intended status: Standards Track                            June 8, 2016
Expires: December 10, 2016

                      WebRTC 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 (commonly
   called "WebRTC").  This document defines the security architecture
   for WebRTC.

Status of This Memo

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   This Internet-Draft will expire on December 10, 2016.

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   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   5
   3.  Trust Model . . . . . . . . . . . . . . . . . . . . . . . . .   5
     3.1.  Authenticated Entities  . . . . . . . . . . . . . . . . .   5
     3.2.  Unauthenticated Entities  . . . . . . . . . . . . . . . .   6
   4.  Overview  . . . . . . . . . . . . . . . . . . . . . . . . . .   6
     4.1.  Initial Signaling . . . . . . . . . . . . . . . . . . . .   8
     4.2.  Media Consent Verification  . . . . . . . . . . . . . . .  10
     4.3.  DTLS Handshake  . . . . . . . . . . . . . . . . . . . . .  11
     4.4.  Communications and Consent Freshness  . . . . . . . . . .  11
   5.  Detailed Technical Description  . . . . . . . . . . . . . . .  12
     5.1.  Origin and Web Security Issues  . . . . . . . . . . . . .  12
     5.2.  Device Permissions Model  . . . . . . . . . . . . . . . .  12
     5.3.  Communications Consent  . . . . . . . . . . . . . . . . .  15
     5.4.  IP Location Privacy . . . . . . . . . . . . . . . . . . .  15
     5.5.  Communications Security . . . . . . . . . . . . . . . . .  16
     5.6.  Web-Based Peer Authentication . . . . . . . . . . . . . .  18
       5.6.1.  Trust Relationships: IdPs, APs, and RPs . . . . . . .  19
       5.6.2.  Overview of Operation . . . . . . . . . . . . . . . .  20
       5.6.3.  Items for Standardization . . . . . . . . . . . . . .  22
       5.6.4.  Binding Identity Assertions to JSEP Offer/Answer
               Transactions  . . . . . . . . . . . . . . . . . . . .  22  Carrying Identity Assertions  . . . . . . . . . .  23  a=identity Attribute  . . . . . . . . . . . . . .  23
       5.6.5.  Determining the IdP URI . . . . . . . . . . . . . . .  24  Authenticating Party  . . . . . . . . . . . . . .  25  Relying Party . . . . . . . . . . . . . . . . . .  25
       5.6.6.  Requesting Assertions . . . . . . . . . . . . . . . .  25
       5.6.7.  Managing User Login . . . . . . . . . . . . . . . . .  27
     5.7.  Verifying Assertions  . . . . . . . . . . . . . . . . . .  27
       5.7.1.  Identity Formats  . . . . . . . . . . . . . . . . . .  28
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  28
     6.1.  Communications Security . . . . . . . . . . . . . . . . .  29
     6.2.  Privacy . . . . . . . . . . . . . . . . . . . . . . . . .  30

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     6.3.  Denial of Service . . . . . . . . . . . . . . . . . . . .  30
     6.4.  IdP Authentication Mechanism  . . . . . . . . . . . . . .  31
       6.4.1.  PeerConnection Origin Check . . . . . . . . . . . . .  32
       6.4.2.  IdP Well-known URI  . . . . . . . . . . . . . . . . .  32
       6.4.3.  Privacy of IdP-generated identities and the hosting
               site  . . . . . . . . . . . . . . . . . . . . . . . .  32
       6.4.4.  Security of Third-Party IdPs  . . . . . . . . . . . .  33
       6.4.5.  Web Security Feature Interactions . . . . . . . . . .  33  Popup Blocking  . . . . . . . . . . . . . . . . .  33  Third Party Cookies . . . . . . . . . . . . . . .  33
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  34
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  34
   9.  Changes . . . . . . . . . . . . . . . . . . . . . . . . . . .  34
     9.1.  Changes since -10 . . . . . . . . . . . . . . . . . . . .  34
     9.2.  Changes since -06 . . . . . . . . . . . . . . . . . . . .  34
     9.3.  Changes since -05 . . . . . . . . . . . . . . . . . . . .  35
     9.4.  Changes since -03 . . . . . . . . . . . . . . . . . . . .  35
     9.5.  Changes since -03 . . . . . . . . . . . . . . . . . . . .  35
     9.6.  Changes since -02 . . . . . . . . . . . . . . . . . . . .  35
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  36
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  36
     10.2.  Informative References . . . . . . . . . . . . . . . . .  38
   Appendix A.  Example IdP Bindings to Specific Protocols . . . . .  39
     A.1.  BrowserID . . . . . . . . . . . . . . . . . . . . . . . .  39
     A.2.  OAuth . . . . . . . . . . . . . . . . . . . . . . . . . .  42
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  43

1.  Introduction

   The Real-Time Communications on the Web (WebRTC) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The major use cases for WebRTC 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) WebRTC communications are directly
   controlled by some Web server, via a JavaScript (JS) API as shown in
   Figure 1.

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

                     Figure 1: A simple WebRTC system

   A more complicated system might allow for interdomain calling, as
   shown in Figure 2.  The protocol to be used between the domains is
   not standardized by WebRTC, but given the installed base and the form
   of the WebRTC API is likely to be something SDP-based like SIP.

                    +--------------+             +--------------+
                    |              | SIP,XMPP,...|              |
                    |  Web Server  |<----------->|  Web Server  |
                    |              |             |              |
                    +--------------+             +--------------+
                           ^                             ^
                           |                             |
                     HTTP  |                             |  HTTP
                           |                             |
                           v                             v
                           JS API                    JS API
                     +-----------+                  +-----------+
                     |           |        Media     |           |
                     |  Browser  |<---------------->|  Browser  |
                     |           |                  |           |
                     +-----------+                  +-----------+

                   Figure 2: A multidomain WebRTC 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 WebRTC which addresses the threats and
   requirements described in that document.

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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
   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.

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, and can infer
      authentication from firewall behavior).

   o  Other users: WebRTC 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

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   must first identify other elements before we can determine whether
   and how much to trust them.  Additionally, sometimes we need to
   identify the communicating peer before we know what policies to

   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
   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 WebRTC 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
   figures 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 as OpenID
   or BrowserID) that can be used to demonstrate their identity to other
   parties.  For instance, Alice might have an account with a social
   network which she can then use to authenticate to other web sites
   without explicitly having an account with those sites; this is a
   fairly conventional pattern on the Web. Section 5.6.1 provides an
   overview of Identity Providers and the relevant terminology.  Alice
   and Bob might have relationships with different IdPs as well.

   This separation of identity provision and signaling isn't
   particularly important in "closed world" cases where Alice and Bob

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   are users on the same social network and have identities based on
   that domain (Figure 3) However, there are important settings where
   that is not the case, such as federation (calls from one domain to
   another; Figure 4) 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.

   Note that the servers themselves are also authenticated by an
   external identity service, the SSL/TLS certificate infrastructure
   (not shown).  As is conventional in the Web, all identities are
   ultimately rooted in that system.  For instance, when an IdP makes an
   identity assertion, the Relying Party consuming that assertion is
   able to verify because it is able to connect to the IdP via HTTPS.

                               |                |
                               |     Signaling  |
                               |     Server     |
                               |                |
                                   ^        ^
                                  /          \
                          HTTPS  /            \   HTTPS
                                /              \
                               /                \
                              v                  v
                           JS API              JS API
                     +-----------+            +-----------+
                     |           |    Media   |           |
               Alice |  Browser  |<---------->|  Browser  | Bob
                     |           | (DTLS+SRTP)|           |
                     +-----------+            +-----------+
                           ^      ^--+     +--^     ^
                           |         |     |        |
                           v         |     |        v
                     +-----------+   |     |  +-----------+
                     |           |<--------+  |           |
                     |   IdP1    |   |        |    IdP2   |
                     |           |   +------->|           |
                     +-----------+            +-----------+

                 Figure 3: A call with IdP-based identity

   Figure 4 shows essentially the same calling scenario but with a call
   between two separate domains (i.e., a federated case), as in
   Figure 2.  As mentioned above, the domains communicate by some
   unspecified protocol and providing separate signaling and identity

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   allows for calls to be authenticated regardless of the details of the
   inter-domain protocol.

           +----------------+    Unspecified    +----------------+
           |                |      protocol     |                |
           |    Signaling   |<----------------->|    Signaling   |
           |    Server      |  (SIP, XMPP, ...) |    Server      |
           |                |                   |                |
           +----------------+                   +----------------+
                    ^                                   ^
                    |                                   |
              HTTPS |                                   | HTTPS
                    |                                   |
                    |                                   |
                    v                                   v
                 JS API                               JS API
           +-----------+                             +-----------+
           |           |             Media           |           |
     Alice |  Browser  |<--------------------------->|  Browser  | Bob
           |           |           DTLS+SRTP         |           |
           +-----------+                             +-----------+
                 ^      ^--+                      +--^     ^
                 |         |                      |        |
                 v         |                      |        v
           +-----------+   |                      |  +-----------+
           |           |<-------------------------+  |           |
           |   IdP1    |   |                         |    IdP2   |
           |           |   +------------------------>|           |
           +-----------+                             +-----------+

            Figure 4: A federated call with IdP-based identity

4.1.  Initial Signaling

   For simplicity, assume the topology in Figure 3.  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 (with technologies such as BrowserID, Federated
   Google Login, Facebook Connect, OAuth, OpenID, WebFinger), and is
   often provided as a side effect service of a user's ordinary accounts
   with some service.  In this example, we show Alice and Bob using a
   separate identity service, though the identity service may be the
   same entity as the calling service or there may be no identity
   service at all.

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   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 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 far.

   Once the PeerConnection is created, the calling service JS needs to
   set up some media.  Because this is an audio/video call, it creates a
   MediaStream with two MediaStreamTracks, 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, so the browser prompts Alice for

   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  Interactive Connectivity Establishment (ICE) [RFC5245] candidates

   o  A fingerprint attribute binding the communication to a key pair
      [RFC5763].  Note that this key may simply be ephemerally generated
      for this call or specific to this domain, and Alice may have a
      large number of such keys.

   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].  preferably over TLS
   [RFC5246].  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

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   the browser to display a trusted element in the browser chrome
   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 a PeerConnection is instantiated with the message from
   Alice's side.  Then, a similar process occurs as on Alice's browser:
   Bob's browser prompts him for device permission, 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. This level of security is
   provided merely by having the fingerprint in the message and having
   that message received securely from the signaling server.  Because
   the far end sent an identity assertion along with their message, they
   know that this is verifiable from the IdP as well.  Note that if the
   call is federated, as shown in Figure 4 then Alice is able to verify
   Bob's identity in a way that is not mediated by either her signaling
   server or Bob's.  Rather, she verifies it directly with Bob's IdP.

   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.  I.e., they simply have whatever information their
   calling site claims about the caller/calllee's identity.  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

4.2.  Media Consent Verification

   As described in ([I-D.ietf-rtcweb-security]; Section 4.2) media
   consent verification is provided 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

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   it is not possible for an attacker who is on-path between Alice and
   Bob but not attached to the signaling service to spoof these checks
   because they do not have the ICE credentials.  Bob has the same
   security guarantees with respect to Alice.

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 or
   channels.  This is performed via DTLS [RFC4347] (for the data
   channel) and DTLS-SRTP [RFC5763] keying for SRTP [RFC3711] for the
   media channel and SCTP over DTLS [I-D.ietf-tsvwg-sctp-dtls-encaps]
   for data channels.  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
   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 mounting a man-
   in-the-middle attack on their traffic.  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
   (i.e., that the signaling service cannot mount a passive attack on
   the communications).

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 so that Alice does not continue to send
   large traffic volumes to entities which went abruptly offline.  ICE
   specifies periodic STUN keepalives but only if media is not flowing.
   Because the consent issue is more difficult here, we require WebRTC
   implementations to periodically send keepalives.  As described in
   Section 5.3, these keepalives MUST be based on the consent freshness
   mechanism specified in [I-D.muthu-behave-consent-freshness].  If a

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   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 WebRTC 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.  That is, when JavaScript is loaded from an
   HTTP origin onto an HTTPS page, an error is displayed and the HTTP
   content is not executed unless the user overrides the error.  Any
   browser which enforces such a policy will also not permit access to
   WebRTC functionality from mixed content pages (because they never
   display mixed content).  Browsers which allow active mixed content
   MUST nevertheless disable WebRTC functionality in mixed content

   Note that it is possible for a page which was not mixed content to
   become mixed content during the duration of the call.  The major risk
   here is that the newly arrived insecure JS might redirect media to a
   location controlled by the attacker.  Implementations MUST either
   choose to terminate the call or display a warning at that point.

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.

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   In addition, they SHOULD support requests for access that promise
   that media from this grant will be sent to a single communicating
   peer (obviously there could be other requests for other peers).
   E.g., "Call customerservice@ford.com".  The semantics of this request
   are that the media stream from the camera and microphone will only be
   routed through a connection which has been cryptographically verified
   (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP
   handshake) as being associated with the stated identity.  Note that
   it is unlikely that browsers would have an X.509 certificate, but
   servers might.  Browsers servicing such requests SHOULD clearly
   indicate that identity to the user when asking for permission.  The
   idea behind this type of permissions is that a user might have a
   fairly narrow list of peers he is willing to communicate with, e.g.,
   "my mother" rather than "anyone on Facebook".  Narrow permissions
   grants allow the browser to do that enforcement.

   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 client to know what sort of
      user interface experience to provide to the user, including what
      permissions to request from the user and hence what 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
      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
      device access, 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 when the indication is hidden.  [Note:

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      this may not be necessary in systems that are non-windows-based
      but that have good notifications support, such as phones.]

   [[OPEN ISSUE: This section does not have WG consensus.  Because
   screen/application sharing presents a more significant risk than
   camera and microphone access (see the discussion in
   [I-D.ietf-rtcweb-security] S 4.1.1), we require a higher level of
   user consent.

   o  Browsers MUST not permit permanent screen or application sharing
      permissions to be installed as a response to a JS request for
      permissions.  Instead, they must require some other user action
      such as a permissions setting or an application install experience
      to grant permission to a site.

   o  Browsers MUST provide a separate dialog request for screen/
      application sharing permissions even if the media request is made
      at the same time as camera and microphone.

   o  The browser MUST indicate any windows which are currently being
      shared in some unambiguous way.  Windows which are not visible
      MUST not be shared even if the application is being shared.  If
      the screen is being shared, then that MUST be indicated.


   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 communicating peers.  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.

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5.3.  Communications Consent

   Browser client implementations of WebRTC MUST implement ICE.  Server
   gateway implementations which operate only at public IP addresses
   MUST implement either full ICE or ICE-Lite [RFC5245].

   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 during the
   lifetime of the transaction (i.e., during the period when the ICE
   stack would accept a new response for that transaction).  The JS MUST
   NOT be permitted to control the local ufrag and password, though it
   of course knows it.

   While continuing consent is required, the ICE [RFC5245]; Section 10
   keepalives use STUN Binding Indications which 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; see [I-D.muthu-behave-consent-freshness].

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.
   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 specification.

   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

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      used.  This prevents the peer from learning one's IP address at
      all.  This mechanism MUST also permit suppression of the related
      address field, since that leaks local addresses.

   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

   Note that some enterprises may operate proxies and/or NATs designed
   to hide internal IP addresses from the outside world.  WebRTC
   provides no explicit mechanism to allow this function.  Either such
   enterprises need to proxy the HTTP/HTTPS and modify the SDP and/or
   the JS, or there needs to be browser support to set the "TURN-only"
   policy regardless of the site's preferences.

5.5.  Communications Security

   Implementations MUST implement SRTP [RFC3711].  Implementations MUST
   implement DTLS [RFC4347] and DTLS-SRTP [RFC5763][RFC5764] for SRTP
   keying.  Implementations MUST implement

   All media channels MUST be secured via SRTP.  Media traffic MUST NOT
   be sent over plain (unencrypted) RTP; that is, implementations MUST
   NOT negotiate cipher suites with NULL encryption modes.  DTLS-SRTP
   MUST be offered for every media channel.  WebRTC implementations MUST
   NOT offer SDES or select it if offered.

   All data channels MUST be secured via DTLS.

   All implementations MUST implement DTLS 1.0, with the cipher suite
   TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA with the the P-256 curve
   [FIPS186].  The DTLS-SRTP protection profile
   SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
   Implementations SHOULD implement DTLS 1.2 with the
   TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.
   Implementations MUST favor cipher suites which support PFS over non-
   PFS cipher suites and SHOULD favor AEAD over non-AEAD cipher suites.

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   API Requirement:  The API MUST generate a new authentication key pair
      for every new call by default.  This is intended to allow for

   API Requirement:  The API MUST provide a means to reuse a key pair
      for calls.  This can be used to enable key continuity-based
      authentication, and could be used to amortize key generation

   API Requirement:  Unless the user specifically configures an external
      key pair, different key pairs MUST be used for each origin.  (This
      avoids creating a super-cookie.)

   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.

      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 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.  X.509 identities and Web IdP
         identities have similar semantics and should be displayed in a
         similar way.

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      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".)
         However, if Null ciphers are used, that MUST be presented to
         the user at the top-level UI.

      *  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 identify the endpoint on the other
   side without trusting 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.

   The mechanisms defined in this document do not require the browser to
   implement any particular identity protocol or to support any

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   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 Web application at the
   calling site.  The use case for having this information configured
   into the browser is that the user may "log into" the browser to bind
   it to some identity.  This is becoming common in new browsers.
   However, it should also be possible for the IdP information to simply
   be provided by the calling application.

   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 configure trust in the IdP at all.  The
   identity mechanism can directly verify that the IdP indeed made the
   relevant identity assertion (a function provided by the mechanisms in
   this document), and any assertion it makes about an identity for
   which it is authoritative is directly verifiable.  Note that this
   does not mean that the IdP might not lie, but that is a
   trustworthiness judgement that the user can make at the time he looks
   at the identity.

   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.

   Note that there are two logically separate functions here:

   o  Identity assertion generation.

   o  Identity assertion verification.

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   The same IdP JS "endpoint" is used for both functions but of course a
   given IdP might behave differently and load new JS to perform one
   function or the other.

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

   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.  This allows the IdP to load whatever JS is necessary into
       the proxy.  The resulting code runs in the IdP's security

   2.  The IdP registers an object with the browser that conforms to the
       API defined in [webrtc-api].

   3.  The browser invokes methods on the object registered by the IdP
       proxy to create or verify identity assertions.

   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--the location of which is determined based on the IdP's
   identity--and to call the generic API for requesting and verifying
   identity 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,

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   including being forward compatible with IdPs which did not exist at
   the time the browser was written.

5.6.3.  Items for Standardization

   There are two parts to this work:

   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.  This is specified in
      Section 5.6.4.

   o  The interface to the IdP, which is defined in the companion W3C
      WebRTC API specification [webrtc-api].

   The WebRTC API specification also defines JavaScript interfaces that
   the calling application can use to specify which IdP to use.  That
   API also provides access to the assertion-generation capability and
   the status of the validation process.

5.6.4.  Binding Identity Assertions to JSEP Offer/Answer Transactions

   An identity assertion binds the user's identity (as asserted by the
   IdP) to the SDP 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
   as a JavaScript object (also known as a dictionary or hash) with a
   single "fingerprint" key, as shown below:

       "fingerprint": [ {
         "algorithm": "sha-256",
         "digest": "4A:AD:B9:B1:3F:...:E5:7C:AB"
       }, {
         "algorithm": "sha-1",
         "digest": "74:E9:76:C8:19:...:F4:45:6B"
       } ]

   The "fingerprint" value is an array of objects.  Each object in the
   array contains "algorithm" and "digest" values, which correspond
   directly to the algorithm and digest values in the "a=fingerprint"
   line of the SDP [RFC4572].

   This object is encoded in a JSON [RFC4627] string for passing to the

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   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.  Carrying Identity Assertions

   Once an IdP has generated an assertion, it is attached to the SDP
   message.  This is done by adding a new identity attribute to the SDP.
   The sole contents of this value are a base-64 encoded [RFC4648]
   identity assertion.  For example:

   o=- 1181923068 1181923196 IN IP4 ua1.example.com
   c=IN IP4 ua1.example.com
   a=fingerprint:sha-1 \
   t=0 0
   m=audio 6056 RTP/SAVP 0

   The identity attribute attests to all "a=fingerprint" attributes in
   the session description.  It is therefore a session-level attribute.

   Multiple "a=fingerprint" values can be used to offer alternative
   certificates for a peer.  The "a=identity" attribute MUST include all
   fingerprint values that are included in "a=fingerprint" lines.

   The RP browser MUST verify that the in-use certificate for a DTLS
   connection is in the set of fingerprints returned from the IdP when
   verifying an assertion.  a=identity Attribute

   The identity attribute is session level only.  It contains an
   identity assertion, encoded as a base-64 string [RFC4648].

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   The syntax of this SDP attribute is defined using Augmented BNF

   identity-attribute  = "identity:" identity-assertion
                         [ SP identity-extension
                           *(";" [ SP ] identity-extension) ]
   identity-assertion  = base64
   base64              = 1*(ALPHA / DIGIT / "+" / "/" / "=" )
   identity-extension  = extension-att-name [ "=" extension-att-value ]
   extension-att-name  = token
   extension-att-value = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff)
                         ; byte-string from [RFC4566] omitting ";"

   No extensions are defined for this attribute.

   The identity assertion is a JSON [RFC4627] encoded dictionary that
   contains two values.  The "assertion" attribute contains an opaque
   string that is consumed by the IdP.  The "idp" attribute is a
   dictionary with one or two further values that identify the IdP, as
   described in Section 5.6.5.

5.6.5.  Determining the IdP URI

   In order to ensure that the IdP is under control of the domain owner
   rather than someone who merely has an account on the domain owner's
   server (e.g., in shared hosting scenarios), the IdP JavaScript is
   hosted at a deterministic location based on the IdP's domain name.
   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 opaque IdP-specific string, but allows an IdP to
      implement two protocols in parallel.  This value may be the empty
      string.  If no value for protocol is provided, a value of
      "default" is used.

   Each IdP MUST serve its initial entry page (i.e., the one loaded by
   the IdP proxy) from a well-known URI [RFC5785].  The well-known URI
   for an IdP proxy is formed from the following URI components:

   1.  The scheme, "https:".  An IdP MUST be loaded using HTTPS

   2.  The authority, which is the IdP domain name.  The authority MAY
       contain a non-default port number.  Any port number is removed
       when determining if an asserted identity matches the name of the
       IdP.  The authority MUST NOT include a userinfo sub-component.

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   3.  The path, starting with "/.well-known/idp-proxy/" and appended
       with the IdP protocol.  Note that the separator characters '/'
       (%2F) and '\' (%5C) MUST NOT be permitted in the protocol field,
       lest an attacker be able to direct requests outside of the
       controlled "/.well-known/" prefix.  Query and fragment values MAY
       be used by including '?' or '#' characters.

   For example, for the IdP "identity.example.com" and the protocol
   "example", the URL would be:


   The IdP MAY redirect requests to this URL, but they MUST retain the
   "https" scheme.  This changes the effective origin of the IdP, but
   not the domain of the identities that the IdP is permitted to assert
   and validate.  I.e., the IdP is still regarded as authoritative for
   the original domain.  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,
   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 5.7.1.

5.6.6.  Requesting Assertions

   The input to identity assertion is the JSON-encoded object described
   in Section 5.6.4 that contains the set of certificate fingerprints
   the browser intends to use.  This string is treated as opaque from
   the perspective of the IdP.

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   The browser also identifies the origin that the PeerConnection is run
   in, which allows the IdP to make decisions based on who is requesting
   the assertion.

   An application can optionally provide a user identifier hint when
   specifying an IdP.  This value is a hint that the IdP can use to
   select amongst multiple identities, or to avoid providing assertions
   for unwanted identities.  The "username" is a string that has no
   meaning to any entity other than the IdP, it can contain any data the
   IdP needs in order to correctly generate an assertion.

   An identity assertion that is successfully provided by the IdP
   consists of the following information:

   idp:  The domain name of an IdP and the protocol string.  This MAY
      identify a different IdP or protocol from the one that generated
      the assertion.

   assertion:  An opaque value containing the assertion itself.  This is
      only interpretable by the identified IdP or the IdP code running
      in the client.

   Figure 5 shows an example assertion formatted as JSON.  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 readability.

       "domain": "example.org",
       "protocol": "bogus"
     "assertion": "{\"identity\":\"bob@example.org\",

                        Figure 5: Example assertion

   For use in signaling, the assertion is serialized into JSON,
   base64-encoded [RFC4648], and used as the value of the "a=identity"

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5.6.7.  Managing User Login

   In order to generate an identity assertion, the IdP needs proof of
   the user's identity.  It is common practice to authenticate users
   (using passwords or multi-factor authentication), then use Cookies
   [RFC6265] or HTTP authentication [RFC2617] for subsequent exchanges.

   The IdP proxy is able to access cookies, HTTP authentication or other
   persistent session data because it operates in the security context
   of the IdP origin.  Therefore, if a user is logged in, the IdP could
   have all the information needed to generate an assertion.

   An IdP proxy is unable to generate an assertion if the user is not
   logged in, or the IdP wants to interact with the user to acquire more
   information before generating the assertion.  If the IdP wants to
   interact with the user before generating an assertion, the IdP proxy
   can fail to generate an assertion and instead indicate a URL where
   login should proceed.

   The application can then load the provided URL to enable the user to
   enter credentials.  The communication between the application and the
   IdP is described in [webrtc-api].

5.7.  Verifying Assertions

   The input to identity validation is the assertion string taken from a
   decoded a=identity attribute.

   The IdP proxy verifies the assertion.  Depending on the identity
   protocol, the proxy might contact the IdP server or other servers.
   For instance, an OAuth-based protocol will likely require using the
   IdP as an oracle, whereas with a signature-based scheme might be able
   to verify the assertion without contacting the IdP, provided that it
   has cached the relevant public key.

   Regardless of the mechanism, if verification succeeds, a successful
   response from the IdP proxy consists of the following information:

   identity:  The identity of the AP from the IdP's perspective.
      Details of this are provided in Section 5.7.1.

   contents:  The original unmodified string provided by the AP as input
      to the assertion generation process.

   Figure 6 shows an example response formatted as JSON for illustrative

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     "identity": "bob@example.org",
     "contents": "{\"fingerprint\":[ ... ]}"

                   Figure 6: Example verification result

5.7.1.  Identity Formats

   The identity provided from the IdP to the RP browser MUST consist of
   a string representing the user's identity.  This string is in the
   form "<user>@<domain>", where "user" consists of any character except
   '@', and domain is an internationalized domain name [RFC5890].

   The PeerConnection API MUST check this string as follows:

   1.  If the domain portion 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.  Comparison of domain names is
       done using the label equivalence rule defined in Section
       of [RFC5890].

   2.  If the domain portion of the string is not equal to the domain
       name of the IdP proxy, then the PeerConnection object MUST reject
       the assertion unless:

       1.  the IdP domain is trusted as an acceptable third-party IdP;

       2.  local policy is configured to trust this IdP domain for the
           domain portion of the identity string.

   Sites that have identities that do not fit into the RFC822 style (for
   instance, identifiers that are simple numeric values, or values that
   contain '@' characters) SHOULD convert them to this form by escaping
   illegal characters and appending their IdP domain (e.g.,
   user%40133@identity.example.com), thus ensuring that they are
   authoritative for the identity.

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.

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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.  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.

   In order to protect against malicious content JavaScript, that
   JavaScript MUST NOT be allowed to have direct access to---or perform
   computations with---DTLS keys.  For instance, if content JS were able
   to compute digital signatures, then it would be possible for content
   JS to get an identity assertion for a browser's generated key and
   then use that assertion plus a signature by the key to authenticate a
   call protected under an ephemeral DH key controlled by the content
   JS, thus violating the security guarantees otherwise provided by the
   IdP mechanism.  Note that it is not sufficient merely to deny the
   content JS direct access to the keys, as some have suggested doing
   with the WebCrypto API.  [webcrypto].  The JS must also not be
   allowed to perform operations that would be valid for a DTLS
   endpoint.  By far the safest approach is simply to deny the ability
   to perform any operations that depend on secret information
   associated with the key.  Operations that depend on public
   information, such as exporting the public key are of course safe.

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

   Additionally, any identifier which persists across multiple calls is
   potentially a problem for privacy, especially for anonymous calling
   services.  Such services SHOULD instruct the browser to use separate
   DTLS keys for each call and also to use TURN throughout the call.
   Otherwise, the other side will learn linkable information.
   Additionally, browsers SHOULD implement the privacy-preserving CNAME
   generation mode of [I-D.ietf-avtcore-6222bis].

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 WebRTC at all, WebRTC implementations need
   to be more careful.

   Consider the case of a call center which accepts calls via WebRTC.
   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 WebRTC implementations

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   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 terminating 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).

   [I-D.ietf-rtcweb-rtp-usage] Section 13 documents a number of
   potential RTCP-based DoS attacks and countermeasures.

   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.

6.4.  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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6.4.1.  PeerConnection Origin Check

   Fundamentally, the IdP proxy is just a piece of HTML and JS loaded by
   the browser, so nothing stops a Web attacker from creating their own
   IFRAME, loading the IdP proxy HTML/JS, and requesting a signature.
   In order to prevent this attack, we require that all signatures be
   tied to a specific origin ("rtcweb://...") which cannot be produced
   by content JavaScript.  Thus, while an attacker can instantiate the
   IdP proxy, they cannot send messages from an appropriate origin and
   so cannot create acceptable assertions.  I.e., the assertion request
   must have come from the browser.  This origin check is enforced on
   the relying party side, not on the authenticating party side.  The
   reason for this is to take the burden of knowing which origins are
   valid off of the IdP, thus making this mechanism extensible to other
   applications besides WebRTC.  The IdP simply needs to gather the
   origin information (from the posted message) and attach it to the

   Note that although this origin check is enforced on the RP side and
   not at the IdP, it is absolutely imperative that it be done.  The
   mechanisms in this document rely on the browser enforcing access
   restrictions on the DTLS keys and assertion requests which do not
   come with the right origin may be from content JS rather than from
   browsers, and therefore those access restrictions cannot be assumed.

   Note that this check only asserts that the browser (or some other
   entity with access to the user's authentication data) attests to the
   request and hence to the fingerprint.  It does not demonstrate that
   the browser has access to the associated private key.  However,
   attaching one's identity to a key that the user does not control does
   not appear to provide substantial leverage to an attacker, so a proof
   of possession is omitted for simplicity.

6.4.2.  IdP Well-known URI

   As described in Section 5.6.5 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

6.4.3.  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

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   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

6.4.4.  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.

6.4.5.  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

   The IdP proxy is unable to generate popup windows, dialogs or any
   other form of user interactions.  This prevents the IdP proxy from
   being used to circumvent user interaction.  The "LOGINNEEDED" message
   allows the IdP proxy to inform the calling site of a need for user
   login, providing the information necessary to satisfy this
   requirement without resorting to direct user interaction from the IdP
   proxy itself.  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.

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7.  IANA Considerations

   This specification defines the "identity" SDP attribute per the
   procedures of Section 8.2.4 of [RFC4566].  The required information
   for the registration is included here:

   Contact Name:  Eric Rescorla (ekr@rftm.com)

   Attribute Name:  identity

   Long Form:  identity

   Type of Attribute:  session-level

   Charset Considerations:  This attribute is not subject to the charset

   Purpose:  This attribute carries an identity assertion, binding an
      identity to the transport-level security session.

   Appropriate Values:  See Section of RFCXXXX [[Editor Note:
      This document.

8.  Acknowledgements

   Bernard Aboba, Harald Alvestrand, Richard Barnes, Dan Druta, Cullen
   Jennings, Hadriel Kaplan, Matthew Kaufman, Jim McEachern, Martin
   Thomson, Magnus Westerland.  Matthew Kaufman provided the UI material
   in Section 5.5.

9.  Changes

9.1.  Changes since -10

   Update cipher suite profiles.

   Rework IdP interaction based on implementation experience in Firefox.

9.2.  Changes since -06

   Replaced RTCWEB and RTC-Web with WebRTC, except when referring to the

   Forbade use in mixed content as discussed in Orlando.

   Added a requirement to surface NULL ciphers to the top-level.

   Tried to clarify SRTP versus DTLS-SRTP.

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   Added a section on screen sharing permissions.

   Assorted editorial work.

9.3.  Changes since -05

   The following changes have been made since the -05 draft.

   o  Response to comments from Richard Barnes

   o  More explanation of the IdP security properties and the federation
      use case.

   o  Editorial cleanup.

9.4.  Changes since -03

   Version -04 was a version control mistake.  Please ignore.

   The following changes have been made since the -04 draft.

   o  Move origin check from IdP to RP per discussion in YVR.

   o  Clarified treatment of X.509-level identities.

   o  Editorial cleanup.

9.5.  Changes since -03

9.6.  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  Added some more privacy and linkage text in various places.

   o  Editorial improvements

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10.  References

10.1.  Normative References

   [FIPS186]  National Institute of Standards and Technology (NIST),
              "Digital Signature Standard (DSS)", NIST PUB 186-4 , July

              Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
              (work in progress), July 2013.

              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-26 (work in progress), March

              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-08 (work in progress), February 2015.

              Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
              Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
              dtls-encaps-09 (work in progress), January 2015.

              Perumal, M., Wing, D., R, R., and T. Reddy, "STUN Usage
              for Consent Freshness", draft-muthu-behave-consent-
              freshness-04 (work in progress), July 2013.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818,
              DOI 10.17487/RFC2818, May 2000,

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,

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   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, DOI 10.17487/RFC4347, April 2006,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC4572]  Lennox, J., "Connection-Oriented Media Transport over the
              Transport Layer Security (TLS) Protocol in the Session
              Description Protocol (SDP)", RFC 4572,
              DOI 10.17487/RFC4572, July 2006,

   [RFC4627]  Crockford, D., "The application/json Media Type for
              JavaScript Object Notation (JSON)", RFC 4627,
              DOI 10.17487/RFC4627, July 2006,

   [RFC4648]  Josefsson, S., "The Base16, Base32, and Base64 Data
              Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006,

   [RFC5234]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234,
              DOI 10.17487/RFC5234, January 2008,

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246,
              DOI 10.17487/RFC5246, August 2008,

   [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, DOI 10.17487/RFC5763, May
              2010, <http://www.rfc-editor.org/info/rfc5763>.

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   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,

   [RFC5785]  Nottingham, M. and E. Hammer-Lahav, "Defining Well-Known
              Uniform Resource Identifiers (URIs)", RFC 5785,
              DOI 10.17487/RFC5785, April 2010,

   [RFC5890]  Klensin, J., "Internationalized Domain Names for
              Applications (IDNA): Definitions and Document Framework",
              RFC 5890, DOI 10.17487/RFC5890, August 2010,

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,

              Dahl, Sleevi, , "Web Cryptography API", June 2013.

              Available at http://www.w3.org/TR/WebCryptoAPI/

              Bergkvist, Burnett, Jennings, Narayanan, , "WebRTC 1.0:
              Real-time Communication Between Browsers", October 2011.

              Available at http://dev.w3.org/2011/webrtc/editor/

10.2.  Informative References

              Uberti, J., Jennings, C., and E. Rescorla, "Javascript
              Session Establishment Protocol", draft-ietf-rtcweb-jsep-14
              (work in progress), March 2016.

   [RFC2617]  Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
              Leach, P., Luotonen, A., and L. Stewart, "HTTP
              Authentication: Basic and Digest Access Authentication",
              RFC 2617, DOI 10.17487/RFC2617, June 1999,

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   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,

   [RFC5705]  Rescorla, E., "Keying Material Exporters for Transport
              Layer Security (TLS)", RFC 5705, DOI 10.17487/RFC5705,
              March 2010, <http://www.rfc-editor.org/info/rfc5705>.

   [RFC6265]  Barth, A., "HTTP State Management Mechanism", RFC 6265,
              DOI 10.17487/RFC6265, April 2011,

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, DOI 10.17487/RFC6455, December 2011,

              van Kesteren, A., "XMLHttpRequest Level 2", January 2012.

Appendix A.  Example IdP Bindings to Specific Protocols

   [[TODO: These still need some cleanup.]]

   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 WebRTC 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.

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   5.  The Browser-ID code signs the fingerprint and returns the signed
       assertion + certificate to the PeerConnection.

   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.

   The offer might look something like:

      o=- 2890844526 2890842807 IN IP4\n
      s= \n
      c=IN IP4\n
      t=2873397496 2873404696\n
      a=fingerprint:SHA-1 ...\n
      a=identity [[base-64 encoding of identity assertion:
          "idp":{     // Standardized
          // Assertion contents are browserid-specific
          "assertion": "{
              \"assertion\": {
                \"digest\":\"<hash of the SIGN message>\",
                \"audience\": \"<audience>\"
                \"valid-until\": 1308859352261,
              \"certificate\": {
                \"email\": \"rescorla@example.org\",
                \"public-key\": \"<ekrs-public-key>\",
                \"valid-until\": 1308860561861,
                \"signature\": \"<signature from example.org>\"
              \"content\": \"<content of the SIGN message>\"
      m=audio 49170 RTP/AVP 0\n

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   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.

   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 WebRTC concepts:

              | OAuth                | WebRTC               |
              | 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

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   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.

           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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