WebRTC Security Architecture
RFC 8827

Document Type RFC - Proposed Standard (January 2021; No errata)
Author Eric Rescorla 
Last updated 2021-01-18
Stream IETF
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Stream WG state Submitted to IESG for Publication
Document shepherd Sean Turner
Shepherd write-up Show (last changed 2018-03-10)
IESG IESG state RFC 8827 (Proposed Standard)
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Consensus Boilerplate Yes
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Responsible AD Adam Roach
Send notices to Sean Turner <sean@sn3rd.com>
IANA IANA review state Version Changed - Review Needed
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Internet Engineering Task Force (IETF)                       E. Rescorla
Request for Comments: 8827                                       Mozilla
Category: Standards Track                                   January 2021
ISSN: 2070-1721

                      WebRTC Security Architecture


   This document defines the security architecture for WebRTC, a
   protocol suite intended for use with real-time applications that can
   be deployed in browsers -- "real-time communication on the Web".

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2021 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   than English.

Table of Contents

   1.  Introduction
   2.  Terminology
   3.  Trust Model
     3.1.  Authenticated Entities
     3.2.  Unauthenticated Entities
   4.  Overview
     4.1.  Initial Signaling
     4.2.  Media Consent Verification
     4.3.  DTLS Handshake
     4.4.  Communications and Consent Freshness
   5.  SDP Identity Attribute
     5.1.  Offer/Answer Considerations
       5.1.1.  Generating the Initial SDP Offer
       5.1.2.  Generating an SDP Answer
       5.1.3.  Processing an SDP Offer or Answer
       5.1.4.  Modifying the Session
   6.  Detailed Technical Description
     6.1.  Origin and Web Security Issues
     6.2.  Device Permissions Model
     6.3.  Communications Consent
     6.4.  IP Location Privacy
     6.5.  Communications Security
   7.  Web-Based Peer Authentication
     7.1.  Trust Relationships: IdPs, APs, and RPs
     7.2.  Overview of Operation
     7.3.  Items for Standardization
     7.4.  Binding Identity Assertions to JSEP Offer/Answer
       7.4.1.  Carrying Identity Assertions
     7.5.  Determining the IdP URI
       7.5.1.  Authenticating Party
       7.5.2.  Relying Party
     7.6.  Requesting Assertions
     7.7.  Managing User Login
   8.  Verifying Assertions
     8.1.  Identity Formats
   9.  Security Considerations
     9.1.  Communications Security
     9.2.  Privacy
     9.3.  Denial of Service
     9.4.  IdP Authentication Mechanism
       9.4.1.  PeerConnection Origin Check
       9.4.2.  IdP Well-Known URI
       9.4.3.  Privacy of IdP-Generated Identities and the Hosting
       9.4.4.  Security of Third-Party IdPs  Confusable Characters
       9.4.5.  Web Security Feature Interactions  Popup Blocking  Third Party Cookies
   10. IANA Considerations
   11. References
     11.1.  Normative References
     11.2.  Informative References
   Author's Address

1.  Introduction

   The Real-Time Communications on the Web (RTCWEB) Working Group
   standardized protocols for real-time communications between Web
   browsers, generally called "WebRTC" [RFC8825].  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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