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Network Working Group                                   R. Copeland, Ed.
Internet-Draft                  Institut Mines Telecom-Telecom Sud Paris
Intended status: Informational                                  K. Corre
Expires: March 30, 2017                                      Orange Labs
                                                               I. Friese
                                                     Deutsche Telekom AG
                                                          S. El Jaouhari
                                 Institut Mines Telecom-Telecom Bretagne
                                                      September 26, 2016

Requirements for Trust and Privacy in WebRTC Peer-to-peer Authentication


   This document studies the relationships of WebRTC communication users
   with their web Calling Services (CS) and their Identity Providers
   (IdPs), in order to identify requirements for IdP based peer-to-peer
   authentication.  This study focuses in particular on issues of
   privacy, security and trust that are raised by the introduction of
   the IdP into the WebRTC call model, and by a different browser-based
   calling paradigm, compared with Mobile networks or traditional VoIP
   systems.  The document lists privacy and trust scenarios for WebRTC
   authentication for individuals as well as organizations.  This
   contribution is proposed to the RTCWEB working group.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
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   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on March 30, 2017.

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Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Call Context Aspects  . . . . . . . . . . . . . . . . . . . .   6
     3.1.  Existing Protocols and Drafts . . . . . . . . . . . . . .   6
     3.2.  WebRTC Architecture Components  . . . . . . . . . . . . .   7
     3.3.  WebRTC Call Models  . . . . . . . . . . . . . . . . . . .   7
       3.3.1.  Single-CS Call Model  . . . . . . . . . . . . . . . .   7
       3.3.2.  Dual-CS Call Model  . . . . . . . . . . . . . . . . .   7
     3.4.  Service Types and their Trust Models  . . . . . . . . . .   9
     3.5.  Destination Categories  . . . . . . . . . . . . . . . . .  10
   4.  Architecture  Vulnerabilities . . . . . . . . . . . . . . . .  10
     4.1.  Untrusted Parties . . . . . . . . . . . . . . . . . . . .  10
     4.2.  Network Messages Across Multiple Parties  . . . . . . . .  10
     4.3.  Confidentiality of Call Logs  . . . . . . . . . . . . . .  11
     4.4.  Dependency of Identifiers . . . . . . . . . . . . . . . .  12
     4.5.  IdP Selection Issues  . . . . . . . . . . . . . . . . . .  12
   5.  Identity Privacy  . . . . . . . . . . . . . . . . . . . . . .  12
     5.1.  Desirable and Undesirable Identity Privacy  . . . . . . .  13
     5.2.  Undetectable Calling  . . . . . . . . . . . . . . . . . .  13
     5.3.  Pseudonymous Calling  . . . . . . . . . . . . . . . . . .  13
     5.4.  Unlinkable Calling  . . . . . . . . . . . . . . . . . . .  14
     5.5.  Potential Methods of Identity Protection  . . . . . . . .  14
       5.5.1.  Sensitive User Information  . . . . . . . . . . . . .  14
       5.5.2.  Proposed Surrogated identities with Pseudonyms  . . .  14
   6.  Trust Relationships . . . . . . . . . . . . . . . . . . . . .  15
     6.1.  Three-Way Trust: User-CSP-IdP . . . . . . . . . . . . . .  15
     6.2.  Choice Indicates Trust  . . . . . . . . . . . . . . . . .  16
     6.3.  User trust in Calling Services  . . . . . . . . . . . . .  16
     6.4.  User trust in Identity Providers  . . . . . . . . . . . .  17
     6.5.  Communication Service trust in Identity Providers . . . .  17
     6.6.  Trusted Identities for non-Browser Interworking . . . . .  18

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   7.  Use Cases for Privacy Requirements  . . . . . . . . . . . . .  18
     7.1.  Anonymous Caller Connecting to Call-Centers . . . . . . .  18
     7.2.  Call Center Worker's Privacy  . . . . . . . . . . . . . .  19
     7.3.  Online Gaming Calling by Pseudonyms . . . . . . . . . . .  19
     7.4.  Hosted Enterprise WebRTC Conferencing Service . . . . . .  20
     7.5.  Variable Trust modes for Employee's Calls . . . . . . . .  20
     7.6.  Employee using untrusted WebRTC service . . . . . . . . .  20
     7.7.  WebRTC service Interworking with SIP users  . . . . . . .  21
   8.  Requirements Summary  . . . . . . . . . . . . . . . . . . . .  21
     8.1.  Anonymity . . . . . . . . . . . . . . . . . . . . . . . .  21
     8.2.  Unlinkability . . . . . . . . . . . . . . . . . . . . . .  22
     8.3.  Independent IdP . . . . . . . . . . . . . . . . . . . . .  22
     8.4.  User Information Confidentiality  . . . . . . . . . . . .  23
     8.5.  Calling Services  . . . . . . . . . . . . . . . . . . . .  23
     8.6.  Usability and Notification  . . . . . . . . . . . . . . .  23
   9.  Conclusions . . . . . . . . . . . . . . . . . . . . . . . . .  24
   10. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  24
   11. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  25
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  25
     12.1.  Normative references . . . . . . . . . . . . . . . . . .  25
     12.2.  Informative references . . . . . . . . . . . . . . . . .  26
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  27

1.  Introduction

   This study provides requirements for supporting identity privacy in
   peer-to-peer (P2P) WebRTC calling services.  While the WebRTC
   specification aims to manage the media flow, it does not include
   procedures for call initiation and privacy setting.  However, WebRTC
   architecture supports peer authentication that is performed
   separately from the calling service, decoupling user authentication
   from the granting of service resources.  The authentication is
   performed by a third party Identity Provider (IdP), while service
   resources are granted by each Calling Services Provider (CSP).

   WebRTC calling creates a different paradigm from 'traditional' VoIP
   services or Telecom, because it is browser-based.  All that is needed
   for a website to support WebRTC calling is to download a client to
   the device to access the user-agent JavaScript APIs.  This simplicity
   will encourage many websites to add calling facilities to their
   'shop-window'.  In turn, such easy click-to-link services will entice
   occasional website visitors to initiate 'opportunistic' calls, often
   using unknown, maybe untrusted calling services, giving little
   thought to the risk of leaking personal information.  Hence, this
   paradigm increases risks of abuse of user data, identity theft and
   commercial exploitation.  This necessitates establishing appropriate
   privacy protection, even without user explicit input of preferences.
   It is possible to achieve this by attaching privacy rules to the

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   IdP's maintained user identity, so that the IdP will support
   anonymity where required.

   An independent identity should prevent the undesirable lock-in effect
   of 'service-bound' identities and allow for a single identity, with
   its linked pseudonyms identifiers (with same credentials), to be used
   instead of numerous identifiers and separate passwords.  Users should
   be able to specify particular privacy rules that are applied during
   authentication across all services, or for a particular service type,
   since the privacy rules are attached to the identifier, not to the

   The current peer authentication procedure provides some flexibility
   for the choice of IdP, but does not allow users to determine privacy
   requirements for different circumstances.  There are no means of
   evaluating trust models and required privacy between calling parties,
   their CSs and IdPs.  The call model (single or dual CS model) affects
   the level of trust in the other parties.  Users may trust their
   chosen IdPs and CS, but the same cannot be assumed for CS and IdPs of
   their communication partners.  The service type and the means of
   activating it also influence the trust level, e.g. a social media
   contact may be more trusted than a call to an unknown website.
   Similarly, the type of destination (e.g. public organizations versus
   unfamiliar websites) also impacts the trust level.  Such destinations
   have their own privacy requirements that need to be negotiated, e.g.
   callers' traceability may be mandated in order to avoid nuisance
   calls, but at the same time, unlinkability is required for employees
   when they respond to public enquiries.

   The accumulated record of calls is a precious commodity in the
   monetized web space, but many users wish to exercise better control
   of what is divulged.  To solve this, it is argued that by using
   context-based privacy to obscure certain details or to present
   surrogate identities, the calling service logs will contain only
   filtered information, in accordance with users' wishes.

   Hence, this study proposes requirements for user-controlled, multi-
   purpose, identity, with service-independent authentication by IdPs,
   which is protected not only by preferences and negotiated privacy
   rules, but also by detected context-based privacy settings.

2.  Terminology

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

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   Privacy and data minimization terms, such as Anonymity,
   Unlinkability, Undetectability, and Pseudonymity, follow the
   definitions by Pfitzmann et al.  [TerminologyForPrivacy], but refer
   to inter-party interactions, not user-to-server, where the 'sender'
   (Caller) and the Destination (called party) are separate entities
   from their own services and endpoint clients.  Table 1 contains
   further terms that convey a particular meaning or an extension
   meaning in this document.

                             Terms Definitions

   | Term           | Description                                      |
   | Called-party,  | A target destination, as nominated by the        |
   | Destination or | caller, who is an individual called party, an    |
   | Callee         | organization representative or an intelligent    |
   |                | object, with a WebRTC valid identity.            |
   | Caller or      | An individual, an organization representative or |
   | Sender         | an intelligent object with a valid identity who  |
   |                | is initiating a WebRTC call or session by using  |
   |                | a WebRTC-enabled browser.                        |
   | Calling        | The WebRTC based service that is used to connect |
   | Service (CS)   | calling parties.  This service may provide       |
   |                | calling features and group calling management,   |
   |                | as well as users' preferences and group privacy  |
   |                | policies.  A CS is assumed to be server-side     |
   |                | software from a CS Provider, with clients        |
   |                | downloaded to the users' endpoints.              |
   | IdP (Identity  | IdP (Identity Provider) creates, maintains, and  |
   | provider)      | manages identity information for entities        |
   |                | (users, services, or systems) and provides       |
   |                | authentication to other service providers.  It   |
   |                | is a trusted third party that can be relied upon |
   |                | by users and servers when they establish a       |
   |                | dialog that must be authenticated.               |
   | IdP Proxy      | A client (user agent) constructed by each IdP    |
   |                | for its own operation, and is downloaded to      |
   |                | users' endpoints when they wish to verify a      |
   |                | given 'assertion' for a user.                    |
   | Unlinkability  | While the definition in [TerminologyForPrivacy]  |
   | or             | refers to unlinkability of subject to a message  |
   | Untraceability | or a particular attribute, in this document it   |
   |                | is defined as the inability to link calling      |
   |                | parties to their routable address for calling    |
   |                | back.                                            |
   | Privacy Rules  | Rules containing parameters for service delivery |
   |                | that are compiled from preferences of the        |

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   |                | involved individuals, groups, or institutions,   |
   |                | to determine when, how, and to what extent       |
   |                | information is divulged.                         |
   | Surrogated     | Identities that identify a real person uniquely  |
   | Identities     | through their association with a universal       |
   |                | unique surrogate key (a database indexing key    |
   |                | that is system-generated as an artificial        |
   |                | primary key value, independent of any            |
   |                | attribute).  Such identities need not have their |
   |                | own credentials, and can use either meaningful   |
   |                | pseudonyms or meaningless (anonymous)            |
   |                | identifiers.                                     |

                                  Table 1

3.  Call Context Aspects

   This section describes the dependency of privacy settings on the call
   model (single/dual CS), service type and destination category, in the
   light of the P2P authentication procedure.

3.1.  Existing Protocols and Drafts

   WebRTC calls connect two browsers that are exchanging media and data,
   as presented in the "WebRTC Overview" [I-D.ietf-rtcweb-overview].
   Prior to the media connection, the calling parties may authenticate
   each other, in a possibly reciprocal peer-to-peer authentication
   process, as described in "WebRTC Security Architecture"

   The RFC "Privacy Considerations for Internet Protocols" [RFC6973]
   offers guidance for privacy considerations in Internet protocols.
   This RFC identifies privacy threats and threat mitigation solutions
   which includes data minimization.

   The Internet draft "Security Considerations"
   [I-D.ietf-rtcweb-security]  for WebRTC identifies two threats to
   privacy in WebRTC.  These are the correlation of anonymous calls
   through persistent identifiers such as DTLS certificates, or the risk
   of browser fingerprinting through the WebRTC API itself.

   Examples of WebRTC binding to specific identity protocols are given
   in "WebRTC Security Architecture", such as OAuth 2.0 [RFC6749] (or
   OpenID Connect [OIDC] ).  However, these protocols do not address
   privacy and identity management for peer-to-peer authentication.

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3.2.  WebRTC Architecture Components

   Current WebRTC communications rely on standard browser APIs
   (getUserMedia) that execute at both endpoints to enable media
   streaming, and WebRTC APIs (RTCPeerConnection and RTCDatachannel)
   that execute at the endpoints to manage the flow.  The endpoints also
   execute a CS client, which drives the initial call setup.

   To enable decoupling of the CS from the identity authentication
   process, WebRTC security architecture [I-D.ietf-rtcweb-security-arch]
   proposes that the WebRTC PeerConnection component interacts directly
   with the IdP, to initiate and manage the authentication process.  It
   negotiates Session Description Protocol (SDP) with the other party by
   sending an SDP offer and accepting, rejecting or offering a counter

   Both parties set up their own PeerConnection instances and download
   an IdP proxy from their own IdP.  Each IdP authenticates its user's
   and returns an identity assertion containing the identity and session
   key fingerprint as claims.  When a party receives an SDP offer or
   answer containing an identity assertion, that party also downloads
   the IdP proxy from the other party's IdP.  This proxy is then used to
   verify the received identity assertion.  Each IdP Proxy only connects
   to its own IdP server.  This architecture is presented in Figure 1.

3.3.  WebRTC Call Models

3.3.1.  Single-CS Call Model

   In a single-CS model, which is the dominant model in the current
   Internet Voice services, only one calling service is involved, where
   both the caller and the called party are registered to the same
   service.  The service manages the subscribing users' privacy policy
   via a set of options, which are then reconciled for the call.  These
   services utilize 'service-bound' identities, where users must log on
   with the service-specific credentials.  By contrast, the WebRTC
   Single-CS call model permits users to select their own identities,
   and perform user-to-user mutual authentication, even though both
   users are logged on to the same service.

3.3.2.  Dual-CS Call Model

   While early implementations of WebRTC calling between individuals go
   no further than the single-CS model, it is expected that the dual-CS
   model will become more common if WebRTC services are adopted in
   business, especially for small-to-medium enterprises.  In dual-CS
   model, the calling parties log on to two different CSs, with their
   own sets of privacy rules and security policies, so CSs must discover

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   the respective privacy/security requirements and negotiate an
   acceptable set of rules for both parties per session.

   Figure 1 shows the calling parties (A and B) using different services
   (CS-A and CS-B) and different IdPs (IdP-1 and IdP-2).  It shows the
   mutual P2P authentication that is performed by each side
   symmetrically.  A gets its own identity assertion from IdP-1 and
   verifies B via a downloaded IdP-2 proxy.  B gets its own assertion
   from IdP-2 and verifies A by the downloaded IdP-1 proxy.  Each proxy
   runs in its own sandbox to protect it from interference.  The
   mechanism for interaction between calling Services can be, for
   example, SIP or XMPP.

                    +----------+           +----------+
                    | Caller's |Unspecified| Called   |
    +----------+    | Service  | Protocol  | Service  |    +----------+
    | IdP-1    |    |  CS-A    |<--------->|  CS-B    |    | IdP-2    |
    |          |    | (caller) |(SIP,      | (callee) |    |          |
    +-v------^-+    +-----^----+  XMPP,...)+-------^--+    +--^-----v-+
      |      |           /                          \         |     |
      |      |          /                            \        |     |
      |      |         /                              \       |     |
      | +----|----------------+             +-----------------|---+ |
      | |    |                |    Media    |                 |   | |
      | |    | Browser A      |<----------->|    Browser B    |   | |
      | | +-------+ +-------+ |             | +-------+ +-------+ | |
      | | | IdP-1 | | IdP-2 | |             | | IdP-1 | | IdP-2 | | |
      | | | Proxy | | Proxy | |             | | Proxy | | Proxy | | |
      | | |sandbox| |sandbox| |             | |sandbox| |sandbox| | |
      | | +-------+ +---^---+ |             | +----^--+ +-------+ | |
      | +---------------|-----+             +------|--------------+ |
      |                 |                          |                |
      |                 |                          |                |
      +-----------------|--------------------------+                |
       B verifies A     |                                           |
                                                  A Verifies B

                      Figure 1: Dual CS Calling Model

   Calling parties may connect to unknown parties who are using
   unfamiliar services across the globe, and are authenticated by IdPs
   that the caller may not know or even be aware of.  In such cases,
   many users would prefer to prevent unnecessary data disclosure.

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3.4.  Service Types and their Trust Models

   In order to understand the privacy requirements for WebRTC
   architecture, it is important to classify calling services by the
   manner of initiating the sessions, choosing destinations (called
   parties), and consulting the other party CS.  The following is a
   classification of web calling service types that may have varying
   privacy requirements, due to different trust models:

   a.  Contact-List-Service: The service enables users to maintain their
       contact list, and make or receive calls from them.  This service
       type is used by social media, VoIP OTT, and internal enterprise
       call servers.

   b.  Click-to-Link Service: Browsing users activate this service type
       when they click on a website link to talk to anyone at that
       destination, not to an individual person.  The 'linked' party in
       the visited website uses its own CS in a single-CS call model.
       This service type is common for shopping websites and customers'
       enquiries, which today are mostly chat services only.  The caller
       may not trust such services, and users often wish for anonymity
       to be preserved.  On the other hand, unlinkability is often
       required for the responding individuals at the website.

   c.  Negotiated Service: This type of service, which is common for
       business-to-business, allows both calling parties to have their
       own privacy rules.  In a single-CS call model, the service
       reconciles the differences, but in a dual-CS call model, the
       calling services should conduct a dialogue resulting in
       negotiated privacy rules per call.

   d.  Interworking Service: WebRTC calling services should enable
       connecting to other types of services, using temporary or
       'interworking' special identities.  This service type is used,
       for example, for interworking with SIP servers and telephony.
       Although users cannot authenticate each other, the interworking
       depends on one-sided IdP authentication.

   e.  Conferencing Multi-Party Service: Services that connect several
       parties in one call need different mechanisms to mix the media,
       such as a bridge, router/mixer or a matrix, but they also need to
       reconcile privacy needs of several parties.  This is often a
       single-CS service between subscribing participants, but it may
       also support multiple calling services, where each call leg is
       managed by the caller's own CS.  Peer authentication could still
       be performed between the conferencing shared identity and each

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   These service types influence the required privacy, since they are
   correlated to particular trust models, e.g. contact-List calling
   permits full information exchange, but Clink-to-Link sets the
   strictest privacy level.

3.5.  Destination Categories

   The destination category influences the level of security and privacy
   that the calling parties require.  The destination may be an
   untrusted web site in click-to-link service type that users prefer to
   connect while preserving anonymity and/or unlinkability, but such a
   destination may also be a completely trusted, often used website.
   Businesses often wish to apply different confidentiality rules for
   internal or external calls, i.e. distinguishing between classes of
   destinations.  Hence, privacy rules should be determined per
   destination category, by: the addressing mode (click-to-link or
   contact list); the domain (government etc.); or previously logged
   calls.  In the absence of explicit user preference, context-based
   privacy level can be set by the CS according to the destination
   category.  The IdP can also perform such a service, based on the call
   log and the target domain names.

4.  Architecture Vulnerabilities

   This section considers the new paradigm that webRTC creates, which
   gives rise to different trust models between the parties and other

4.1.  Untrusted Parties

   The decoupling of the authentication from the service logic, both in
   networked functions and in business entities, increases
   vulnerability, but also the variety of trust models that are needed
   to support permissive and intuitive web calling patterns.  This
   environment encourages users to take more risks and launch
   interactions without careful considerations of the information that
   may leak to various parties.  In additions, not only a casually
   invoked service may receive user data, but also the IdP and the CS of
   the other party, who may not be trusted.

4.2.  Network Messages Across Multiple Parties

   WebRTC architecture relies on the IdP to perform the authentication
   independently from the call initiation process.  This entails further
   network based interactions and more parties gaining knowledge of
   them.  The additional network messaging between more parties increase
   the risk of Man-in-the-Middle (MITM) attacks.  The number of involved
   parties in person-to-person calls rises in peer authentication,

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   depending on the call model.  A single-CS call model with service-
   bound identification involves only one CS with two parties.  In the
   dual-CS call model with independent IdPs, there are two calling
   parties, two service providers and two IdPs, i.e. double the number
   of involved parties.  This increases the risk of leaking information
   and abuse of call history, as well as MitM attacks.

4.3.  Confidentiality of Call Logs

   Currently, both IdPs and calling services acquire user knowledge,
   which is a) contextual; b) historical; and c) subscription-based.
   This information can be static (user personal details, additional
   email identities) or dynamic (calling patterns, frequent
   destinations, preferred services/websites, and more).  Every time a
   new call is set up, the exchanged information can provide means of
   tracking user activity or linking back to the parties.  The
   accumulation of such information reveals trends, habits, preferences
   and behavior patterns that are highly valuable to traders and
   marketeers.  Exploiting this data is often the only monetization
   methods available to web service providers, but it is a cause for
   concern for users who regard it as compromised privacy.

   The IdP gains knowledge of personal details (to enhance user
   profile), and associated identities (to enhance identity resilience),
   which users may be reluctant to share with numerous websites.  In
   addition, the IdP can log authentication requests of every CS using
   its identity service, while the CS only has visibility of calls made
   to and from that service.  Since users' calling activities are now
   spread over a number of web communications services, an IdP will have
   wider perspective on the user's intelligence.

   IdP is not able to know much about the involved CS, as the Peer-
   Connection method interfaces between the endpoint's client and the
   IdP Proxy directly, not via a CSP server.  On the other hand, the CS
   has better understanding of the call context in the preamble before
   initiating a call request.

   As the IdP Proxies are deployed on both calling parties'devices, the
   IdPs can log identity verification requests for incoming as well as
   outgoing calls.  It may be possible for an IdP to track call history
   for a particular destination user who is using another IdP, even if
   an pseudonymous identity is given, and perhaps eventually link the
   records with the real identity.  However, this risk only arises with
   habitual pseudonymous calling to the same destination.

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4.4.  Dependency of Identifiers

   The aim of providing completely independent and portable identities
   is not easy to achieve.  While the IdP-generated identity (with an
   IdP domain) is not related to any specific calling service, i.e.
   portable between services, it is still dependent on the IdP.  If
   users wish to keep their well-known published identifiers when moving
   to another IdP, they need identities that ensure both CS and IdP
   independence.  This requires users to use a global user identifier,
   such as a Universally Unique Identifier (UUID) [RFC4122] that would
   acts as a unifying key for all identities.  This globally unique
   identifier could link several identifiers, both IdP-generated and CS-

4.5.  IdP Selection Issues

   In WebRTC, each CS client in the device is responsible for setting up
   the authentication requests for its own party.  The CS client decides
   what form of authentication to apply, i.e. peer authentication,
   server-side Single Sign-On, or service-specific authentication.  This
   means that the CS controls the selection process, and may restrict
   the choices of IdP to choose from, or even prevent an IdP to be

   Current WebRTC specifications define two options for the CS to select
   an IdP for an identity assertion request:

   o  If the setIdentityProvider() method has been called by the CS, the
      provided IdP will be used.

   o  If the setIdentityProvider() method has NOT been called, the
      browser may use a pre-configured IdP.

   Pre-configuring an IdP via the browser means that yet another party -
   the browser vendor - is a stakeholder in the WebRTC call initiation.
   It is argued that currently, users do not have sufficient control on
   the selection of the IdP with these facilities.

5.  Identity Privacy

   This section sheds a new light on the privacy requirements for
   undetectable, pseudonymous, and unlinkable callers that arise from
   the webRTC peer calling.  Although these terms do exist, the
   associated privacy requirements have not been previously identified.

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5.1.  Desirable and Undesirable Identity Privacy

   Many websites may be content to receive enquiries from anonymous
   callers, because this may generate impulse-buying, so they only
   request user details before completing a transaction.  Such websites
   also require some privacy rules themselves, to protect specific
   personnel serving at a call center.  Web calling encourages
   opportunistic calling by users who are merely visiting the websites,
   where users identities are 'incognito', i.e. their status is
   'undetectable' or 'unobservable'.  In certain circumstances, calls
   from undetectable identities should always be supported, e.g. calls
   to emergency services that are passed through without any
   authentication.  While undetectable status is passive, in other cases
   callers may specifically wish to withhold their personal details for
   a variety of legitimate reasons, e.g. to avoid revealing interests in
   sensitive material or avoid personal embarrassment.  In such cases,
   users can choose to use assumed names (pseudonyms).  Similarly, there
   are good reasons to support the requirement of unlinkability that
   prevents tracing back previous calls, e.g. to avoid traders chasing

   The contrasting requirements to prevent anonymity should also be
   considered, in order to prevent abuse, e.g. for nuisance calls or
   malicious disruption of service.  The solution to block all callers
   who are not on the personal contact list may suffice for individual
   users, but this is too restrictive for a business.

   Therefore, since privacy protection is both desirable and undesirable
   depending on context and point of view, privacy rules need finer
   granularity, so that they can be applied judiciously, according to
   context and circumstances.  Hence, the requirements are for WebRTC
   authentication to support different states of user privacy and
   anonymity: Undetectable (undisclosed identity), Pseudonymous (false
   or fictitious identity) or unlinkable (untraceable address/path).

5.2.  Undetectable Calling

   Undetectable calling may be initiated without logging to any CS,
   while the user is unknown.  When a call is made without logging to
   the CS, a call request may be processed without any authentication.

5.3.  Pseudonymous Calling

   Pseudonymous calling means that the username identifier is replaced
   with an assumed name that hides the user's identity.  The
   authentication can be performed by an IdP who is aware of the
   pseudonym owner.  Hence, the other parties can be reassured that the
   identity is verified.  Such authentication may not be sufficient for

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   monetized transaction and non-repudiation, but is considered
   acceptable for web calling.

5.4.  Unlinkable Calling

   Unlinkability prevents other parties from calling back, or from
   tracing the user's cyber activities, such as visited websites and
   calling patterns.  Unlinkable callers seek to hide the originating
   website or redirected services.  Unlinkability is also both desirable
   and undesirable, depending on the context.  Preventing linkability is
   often needed to protect individual employees who respond to enquiries
   from the public.  Conversely, linkability is highly desirable by
   emergency and health services, to locate incapacitated callers in

5.5.  Potential Methods of Identity Protection

5.5.1.  Sensitive User Information

   User information that may be subject to privacy includes:

   o  Forename and last names, which are often incorporated in the
      username part of the identifier.

   o  Domain name of associated organization, which often incorporates
      the organization name in the @domain part.

   o  Message path and IP address, which are revealed in the SDP
      (Session Description Protocol).

5.5.2.  Proposed Surrogated identities with Pseudonyms

   Using pseudonyms avoids undesirable disclosure of the identity and/or
   incidental private information.  However, pseudonyms should still be
   associated by a common key to the real user.  This is achieved by a
   fully independent identifier that acts as a 'surrogate' key, i.e. an
   indexing key that is not based on any meaningful personal details, as
   described in [SurrogateKeys] for indexing data.  Such surrogate keys
   provide stability, because they are not affected by users changing
   circumstances (e.g. married names) and personal attributes (e.g.
   changing employer domain name).

   The surrogate key for identities is a Global User ID (GUID) that
   uniquely identifies a particular user.  GUIDs can be generated by a
   number of known algorithms.  They may inject user-specific variable
   attributes as 'salt' to the otherwise random number generator, to
   ensure uniqueness.  It is proposed that this GUID/surrogate key will
   link the IdP-based identity, service-bound identities and pseudonym

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   identities, at the discretion of the user.  All the linked identities
   (i.e. the 'surrogate identities') can share the same credentials that
   the IdP has verified.  Service-bound identities from a variety of
   calling services that do have their own credentials (usually just
   passwords) can also link to the surrogate key, thus benefiting from
   deeper user verification and from the SSO effect that such
   arrangement brings.

6.  Trust Relationships

   This section analyzes discernible trust models which are proposed as
   the basis of setting up appropriate privacy levels.

6.1.  Three-Way Trust: User-CSP-IdP

   The current WebRTC security architecture only assumes that users
   trust their CSP, or that an IdP is used in P2P authentication if the
   CS is untrusted.  Trust in the IdP is only considered regarding its
   verified, https origin.  In this model, the browser constitutes the
   trust anchor.  However, this simple trust model does not describe
   trust in the privacy context, nor the difference between a user's own
   IdP and the other IdP.

   Users need to trust other involved actors, i.e. CSs and IdPs, to
   manage their privacy and provide solid identity claims.  The CS in
   turn may need to trust both IdPs regarding user authentication and
   identity claims.  However, IdPs do not require particular trust
   relations with the CS, as they merely provide a service, without risk
   to themselves.  Figure 2 and subsequent sections details this trust

        +------------------->   CSP   |
        |                   +---------+
    +---------+              |
    |  Alice  |              |
    +---------+              +-------+
     |  |                    |       |
     |  |            +-------v-+  +--v------+
     |  +------------>  IdP A  |  |  IdP B  |
     |               +---------+  +-^-------+
     |                              |

     Figure 2: Trust relationships between communication setup actors

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6.2.  Choice Indicates Trust

   The purpose of establishing trust is to make a decision in situations
   where an action with an inherent risk depends on informed judgement.
   Explicit choice can be interpreted as a measure of trust.  Users'
   chosen IdPs are more trusted than mandated IdPs imposed by the
   communication services, though enterprise mandated IdPs are trusted
   by virtue of the enterprise selection.  Similarly, calling services
   that are casually invoked by click-to-Link in a visited website are
   less trustworthy than those which the user has registered to.

   Trust is also assumed towards a call-party that is known to the user,
   but there is no implied trust level for the calling services and IdPs
   of that party.  Trust may be associated with the type of the call
   destination, which can be categorized as:

   o  Fully trusted parties (government, public organization, known

   o  Inner-circle of a social network or family members

   o  Uncertain (not-in-contact-list, no information)

   o  Untrusted (known as unreliable).

6.3.  User trust in Calling Services

   Traditionally, CSP had access to full user profile information and
   accumulated call history, but user habits now favor using different
   services according to context, so the CS has only their own usage
   records.  While some CS may enjoy greater trust, in other cases,
   users do not wish to share or even store their call history.  These
   preferences are usually agreed when users register with the CS, but
   it is up to the CS to respect them.

   Since the CS manages the call signaling, it is well placed to
   intercept the peer-to-peer media stream that otherwise is deemed
   private.  Even if the caller's CS can be trusted not to do so, the CS
   of the other party is not so well trusted.  Trust in the CS also
   varies between single-CS and dual-CS Call Model.  In a single-CS call
   model both parties use the same service to communicate, however this
   does not guarantee that they have the same trust models and the same
   privacy requirements.  In a dual-CS call model, the other party's CS
   merits even less trust, as it may not even be known (depending on
   user-interface implementations).  Hence, variable precautions and
   privacy negotiations are necessary according to the context and the
   involved parties.  In cases where the CS is untrusted, enforcing
   authentication by an independent IdP ensures that the exchange of

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   media key is on a third-party path (the identity path) between the
   authenticated users.  Therefore, the CS, who is not in control of
   this path, cannot mount a MitM attack.  However, the CS, not the
   user, determines whether an IdP is to be used, and users have no
   means of ensuring that they are protected by the IdP authentication.

6.4.  User trust in Identity Providers

   The IdP, more than the CS, is the custodian of user intelligence;
   hence it must have trust relationship with users that subscribe to
   its services.  It is assumed that such services include storing and
   linking service-bound identities, to allow for flexible means of
   authentication of related identifiers.  Using an IdP makes it easier
   for users to control privacy, since a single agreement with their
   chosen IdP is simpler than managing numerous web services, some of
   which are use only rarely.

   Users may have more than one IdP, perhaps different IdPs for special
   purposes, e.g. most commonly, a separate IdP from an employer.  They
   may choose an IdP that they do not fully trust for private activities
   that they wish to keep separate.  In such cases, the user can limit
   what personal details are disclosed to the IdP, but the IdP will
   still know of any authentication request to this identity.

   Trusting the IdP of the other party is a more difficult issue, since
   this IdP may not be even known.  The P2P authentication procedure
   ensures that the IdP origin is not circumvented, but there are no
   ways of assessing the strength and veracity of the origin statement.

   Currently, called users have no way of controlling the downloading of
   an 'alien' IdP Proxy (of the other party) to their device, since this
   is performed automatically by the CS client at the behest of the
   caller.  Hence, both IdP proxies are subject to the same sandbox
   restrictions, although they have different trust models.

6.5.  Communication Service trust in Identity Providers

   The CS may rely on a third-party IdP to authenticate users when they
   log in, and link the given identity with its internal user account.
   In such cases, the CSP must trust the IdP regarding the
   authentication strength and the validity of the provided profile

   In most cases, the CSP provides a set of preferred IdPs for users to
   choose from, through SSO implementations on the website and usage of
   the setIdentityProvider function.  However, users could also select
   an IdP, e.g. with OIDC discovery.  In dual-CS call model, the CS
   could receive a SDP message containing an assertion from an unknown

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   IdP.  The verification of the assertion could be performed using the
   browser's default IdP, with the CS only receiving a confirmation that
   the identity is authenticated.  In these cases, the CS has only low
   trust level in the IdP, while IdPs that have been vetted by the CS
   are higly trusted.

6.6.  Trusted Identities for non-Browser Interworking

   WebRTC browser-based calling services may need to communicate with
   users on non-browser services, including users of existing SIP
   servers.  The interworking should not be only at the level of
   signaling and applications, but also at the authentication stage.
   For a mobile network, as specified in 3GPP TS 23.228, mutual
   authentication is not possible, but the WebRTC identities, which have
   been authenticated by an IdP or a CS, are linked to the allocated SIP
   identities.  The 3GPP WebRTC-SIP Client at the device enables it to
   contact the local network proxy.

7.  Use Cases for Privacy Requirements

   While [I-D.ietf-rtcweb-use-cases-and-requirements] provides use cases
   for webRTC media, in this study, use cases demonstrate the calling
   context scenarios that require different privacy settings, which
   enhance the examples in [I-D.cazeaux-rtcweb-oauth-identity].

7.1.  Anonymous Caller Connecting to Call-Centers

   Alice is surfing on websites of several insurance or healthcare
   companies and wants to discuss matters of some sensitivity.  She
   clicks on links within these websites, in order to talk to their
   experts.  Alice is concerned with her privacy and prefers to remain
   anonymous, especially towards her employer.  Some websites treat her
   identity as undetectable, since she has not logged in to the service,
   but they allow such callers to visit.  For websites that require
   authentication, she will use a pseudonym and authenticate to her
   personal IdP, to avoid her employer's IdP becoming aware of which
   websites she calls.  This use case demonstrates a single-CS call
   model with the 'Link-to-Call' Service type.  The privacy requirements
   demonstrates undetectability, pseudonymity and unlinkability for the
   caller.  The alternative for Alice is to create unrelated identities
   for each website, but this is much more laborious.  Using an
   independent IdP with surrogate pseudonyms, Alice can rely on the same
   credentials, while reassuring the destination websites that she is
   properly authenticated and is not making nuisance calls.

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7.2.  Call Center Worker's Privacy

   Bob is a member of a company's product support group, working in a
   customer support center.  The company presents clickable icons on the
   website that connect visitors to the right expert.  Bob answers
   Alice's call, but when Alice calls again, she cannot contact Bob
   directly, and her call is answered by another group member.  This use
   case represents single-CS Click-to-Link service model from the called
   destination point of view.  Once Alice indicates that she would like
   to talk, the called destination invokes its own calling service, so
   the roles are reversed: the destination is the caller and Alice
   becomes the called party.  This enables the destination group members
   (Bob) to remain unlinkable vis-a-vis Alice.  On the other hand, Alice
   is an undetectable identity, since she has not logged into the
   service, so her call request uses the browser default IdP to retrieve
   a surrogate pseudonym identity, but she is still traceable in terms
   of IP address and path.  Hence, this example also shows that
   unlinkability is not necessarily attached to all other anonymity
   states, e.g. detectability.

7.3.  Online Gaming Calling by Pseudonyms

   Alice is playing poker on a gaming website.  Alice is a customer,
   with an account which the gaming business administrator has verified
   (e.g. via a credit card).  Alice wishes to communicate with other
   players through a voice channel provided by the gaming facility.  She
   is registered on the gaming site under her chosen pseudonym, which is
   all she wants to reveal to the other players.  The calling service
   verifies the users and their accounts through a server-side IdP
   validation, so the identities with their pseudonyms are strictly
   service-bound.  This use case demonstrates single-CS /Contact-List
   call model, where the calls are placed between two registered and
   logged-on users of the same service.  The model is the same as
   traditional OTT VoIP, where the CS manages the users' identities.
   Callers and called-parties implicitly rely on the calling service to
   provide anonymity, where the anonymity set is determined by the
   number of players.

   It should be possible for the gaming CS to permit P2P authentication
   and independent IdPs, but the gaming host may still require
   authorization of user accounts (if money changes hands).  The IdP
   authentication benefits the CS, since the IdP's identification is
   coupled with deeper user verification.

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7.4.  Hosted Enterprise WebRTC Conferencing Service

   Alice is working for a corporation that provides her with a
   comprehensive web-based communication suite of internal and external
   conferencing, which is hosted by a Service Provider.  The
   conferencing Service Provider uses the mandatory corporate IdP to
   authenticate the employees.  Alice calls Bob, who works for a
   partnering company, and is logged on to his own company's CS.  Alice
   uses Bob's identity from his own CS, which is recorded in her contact
   list.  Although Alice's company does not insist on confidentiality,
   Bob's company does, so Bob's calling service demands that all the
   conferencing participants use the security level that matches Bob's.
   This use case demonstrates dual-CS negotiated call model, where the
   parties have their own preferences determined by their respective
   organizations.  The service providers need to agree on a common set
   of privacy rules.  Although an IdP is mandated by Alice's
   organization, external calling parties may not have the same IdP,
   hence external callers should be authenticated in the mutual peer-to-
   peer authentication process.

7.5.  Variable Trust modes for Employee's Calls

   Employees can have different requirements of privacy depending on
   type of calls and types of destinations.  Alice is a Sales
   representative calling Bob (a potential customer), to conduct a
   consumer survey and she wishes to remain untraceable (unlinkable) and
   unrecognised (pseudonymous).  However, when she calls her colleague,
   Charlie, to discuss invoices, she would like Charlie to call her
   back.  Thus, Alice needs unlinkability when calling Bob, but full
   linkability when calling Charlie.  This use case shows privacy
   decision by context, according to destination type.  Rules may be
   defined per class of destinations (e.g. internal-colleague, external-
   corporate, or external-personal).  The privacy rules may be executed
   by the IdP, but other rules may be executed by the CS, hence
   discrepancies may occur, for example, when the destination-based
   privacy rule conflicts with corporate policies for this customer.
   Hence, this example also shows the need for privacy policy
   negotiation and reconciliation.

7.6.  Employee using untrusted WebRTC service

   Alice is an employee making use of a WebRTC service that she
   considers to be untrusted, in order to communicate some important
   messages to Bob, while she is out of the office.  Alice registers to
   an untrusted CS with her corporate identity.  Bob can 'discover'
   Alice's address when he logs into the same untrusted service, because
   her corporate identity was linked with the CS service-bound identity,
   when she registered.  This use case is an example of single-CS call

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   model, using an untrusted calling service in combination with a
   trusted IdP.  Mutual peer authentication can take place, with each
   party authenticating the CS based identity via surrogate or explicit
   corporate identities.  Alice wants to be sure that her trusted
   corporate IdP is used, in order to minimize risks of an MitM attack
   by the CS, and ensure that the media flow is confidential.
   Currently, the decision to use an IdP, or a particular user-chosen
   IdP, is in the hands of the CS, so the possible attacker is
   responsible for setting the protection!  Hence, it is very important
   that users gain the power to proactively protect their communication
   by opting to use IdP authentication.

7.7.  WebRTC service Interworking with SIP users

   Alice uses a social media website to connect to her friends, and her
   contact list includes people with mobile numbers only.  Alice can
   initiate and receive web calls from her mobile, to connect with
   mobile phones.  Users receiving calls from Alice will see Alice's
   phone number displayed, or her IdP identity.  Alice can also call
   Bob's mobile number from her laptop using a social media service.  To
   connect to Bob, Alice is authenticated by her social media service
   provider (her CS), who also provides her with a SIP identity that is
   linked to her other identities.  Her SIP identity [RFC3261] is
   authenticated by the mobile service provider, who had provided a pool
   of SIP identities to the social media calling service.  This use case
   demonstrates dual-CS Call Model with the interworking service type,
   using server-side authentication.

8.  Requirements Summary

   This section lists the new requirements, as discussed above.  These
   requirements call for greater user's autonomy, greater transparency,
   and greater variety of trust models that affect the level of divulged

8.1.  Anonymity

   1.  It should be possible to set different anonymity rules by
       standard service types, call models and destination categories.

   2.  Personal information must not leak via identity assertions.

   3.  IdP should facilitate pseudonimity via surrogate linked

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8.2.  Unlinkability

   1.  It should be possible to set different unlinkability rules by
       standard service types, call models and destination categories.

   2.  Callers should be able to request and enforce unlinkability with
       respect of a called party, separately from other anonymity

   3.  Called destinations should be able to refuse unlinkability
       requests (e.g. to avoid nuisance calls), while respecting

   4.  Unlinkability (e.g. via "origin" in the SDP) should be subject to
       pre-defined policy, whether that policy is CS-based or IdP-based.
       Currently, such policies are not transparent to users.

   5.  Non-disclosure of organization domains is a type of
       unlinkability, as well as anonymity.

8.3.  Independent IdP

   1.  Users should be able to choose an IdP independently from any
       calling service, though some services will still mandate or
       restrict the choice of IdP.  In particular, authentication by a
       trusted IdP must be an option for users who activate untrusted

   2.  IdPs must not lock-in users through non-portable identifiers.

   3.  Users should be able to create and link surrogate identities and
       pseudonyms to a globally unique identifier that is portable
       between IdPs.

   4.  Users should be able to associate service-bound identities with
       their independent identity (albeit with distinctive assertion
       tokens), thus achieving Single-Sign-On.

   5.  Browsers should allow users to set their chosen default IdPs and
       log in on browser start-up.  Currently browsers select their own
       factory-set IdPs.

   6.  User-chosen IdPs should be able to prompt users to log in.
       Currently, IdP proxies cannot open a dialogue with the user.

   7.  Users should be able to set IdP-based privacy rules for untrusted

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8.4.  User Information Confidentiality

   1.  Linked identifiers and supporting personal verification data must
       be subject to users' privacy preference.

   2.  Session setup messages, as well as identity assertions, should be
       protected to prevent tampering and eavesdropping.

   3.  Users' affiliations with organizations should be subject to
       privacy preferences.  Currently, corporate requirements are not

8.5.  Calling Services

   1.  Users should be able to set privacy rules for untrusted CS or
       destinations websites acting as calling services, regardless of
       the service own parameters.

   2.  CS should be able to determine privacy parameters per
       organizations, for data confidentiality and anonymity.

   3.  Despite users' choice of IdP, calling services should not be
       precluded from mandating their choice of IdPs, or offering a
       preferred IdP list.

   4.  There must be a transparent method of resolving conflicting
       privacy requirements arising from the respective CS options.

   5.  The original website that redirects to a calling service should
       not be named as 'origin', if users wish to avoid divulging it.

   6.  To distinguish between withheld identity and undetected identity,
       the "origin" field should only provide a status indicator.

8.6.  Usability and Notification

   1.  Users should be informed how their privacy will be handled by the
       calling service, and which identity and IdP are used.

   2.  It should be possible to request unlinkability and pseudonimity
       for a shared group identity, but allow members to maintain
       separate identities with personal privacy preferences.

   3.  The IdP must notify users (or CS clients) if it is not possible
       to support undetectable, pseudonymous or unlinkable calling.
       Currently, there are no IdP notifications to the user.

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   4.  Users should be notified if a default IdP is assigned, and if
       other than their chosen IdP is assigned.

9.  Conclusions

   The web calling paradigm is transformed by browser-based calling
   facilities that are easily added to shop windows websites.  However,
   this encourages opportunistic calling with increased risks from
   untrusted parties.  The spread of such calling services means users
   have to maintain numerous identities/passwords, while established
   services lock-in users to the service-bound identity.  Users need to
   manage their own structured identities, independently of any service.
   They also need to control their privacy preferences, and vary such
   preferences for high-risk connections.  A solution is needed not only
   to allow users to control their privacy requirements according to
   web-calling context, but also to protect destination websites from
   abuse of anonymity.

10.  Contributors

   This document is the result of a very fruitful work within the
   reThink project, with also the following contributors:

   I.  Tariq Javed
   Institut Mines Telecom-Telecom Sud Paris
   9 rue C.Fourier
   Evry 91011
   Email: ibrahim_tariq.javed@telecom-sudparis.eu

   N.  Crespi
   Institut Mines Telecom-Telecom Sud Paris
   9 rue C.Fourier
   Evry 91011
   Email: noel.crespi@mines-telecom.fr

   A.  Bouabdallah
   Institut Mines Telecom-Telecom Bretagne
   2 rue de la Chataigneraie
   Cesson Sevigne 35576
   Email: ahmed.bouabdallah@telecom-bretagne.eu

   S.  Becot
   Orange Labs
   4, rue du Clos Courtel
   Cesson Sevigne 35510

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   Email: simon.becot@orange.com

   J.-M.  Crom
   Orange Labs
   4, rue du Clos Courtel
   Cesson Sevigne 35510
   Email: jeanmichel.crom@orange.com

11.  Acknowledgements

   This document was inspired by a draft authored by Victoria Beltran,
   Orange (Email: vicbelma@gmail.com); Emmanuel Bertin, Orange (Email:
   emmanuel.bertin@orange.com); Stephane Cazeaux, Orange (Email:

   Thank you to Ricardo Chaves (INESC-ID) who provided us with his
   interesting feedback and comments.

   This work has received funding from the European Union's Horizon 2020
   research and innovation program under grant agreement No 645342,
   project reTHINK.

12.  References

12.1.  Normative references

              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-15
              (work in progress), January 2016.

              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-12 (work in progress), June 2016.

              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements", draft-
              ietf-rtcweb-use-cases-and-requirements-16 (work in
              progress), January 2015.

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

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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,

   [RFC4122]  Leach, P., Mealling, M., and R. Salz, "A Universally
              Unique IDentifier (UUID) URN Namespace", RFC 4122,
              DOI 10.17487/RFC4122, July 2005,

   [RFC4949]  Shirey, R., "Internet Security Glossary, Version 2",
              FYI 36, RFC 4949, DOI 10.17487/RFC4949, August 2007,

   [RFC6749]  Hardt, D., Ed., "The OAuth 2.0 Authorization Framework",
              RFC 6749, DOI 10.17487/RFC6749, October 2012,

   [RFC6973]  Cooper, A., Tschofenig, H., Aboba, B., Peterson, J.,
              Morris, J., Hansen, M., and R. Smith, "Privacy
              Considerations for Internet Protocols", RFC 6973,
              DOI 10.17487/RFC6973, July 2013,

12.2.  Informative references

              Beltran, V., Bertin, E., and S. Cazeaux, "Additional Use-
              cases and Requirements for WebRTC Identity Architecture",
              draft-cazeaux-rtcweb-oauth-identity-00 (work in progress),
              March 2015.

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

              Sayre, J. and H. Story, "The WebID Protocol & Browsers",
              May 2011, <https://www.w3.org/2011/identity-

   [OIDC]     "OpenID Connect Core 1.0 incorporating errata set 1",

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              Carter, B., "Intelligent Versus Surrogate Keys", 1997,

              Pfitzmann, A., Hansen, M., and H. Tschofenig , "A
              terminology for privacy by data minimization: Anonymity,
              Unlinkability, Undetectability,
              Unobservability,Pseudonymity, and Identity Management -
              V0.34", August 2010.

   [WebRTC]   Bergkvist, A., Burnett, D., Jennings, C., and A.
              Narayanan, "WebRTC 1.0: Real-time Communication Between
              Browsers. World Wide Web Consortium WD WD-webrtc-
              20120821.", August 2012.

Authors' Addresses

   Rebecca Copeland (editor)
   Institut Mines Telecom-Telecom Sud Paris
   9 rue C.Fourier
   Evry  91011

   Email: rebecca.copeland@coreviewpoint.com

   Kevin Corre
   Orange Labs
   4, rue du Clos Courtel
   Cesson Sevigne  35510

   Email: kevin1.corre@orange.com

   Ingo Friese
   Deutsche Telekom AG
   Winterfeldtstr. 21
   Berlin  10781

   Email: Ingo.Friese@telekom.de

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   Saad El Jaouhari
   Institut Mines Telecom-Telecom Bretagne
   2 rue de la Chataigneraie
   Cesson Sevigne  35576

   Email: saad.eljaouhari@telecom-bretagne.eu

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