WebRTC Security Architecture
RFC 8827

• Status
• IESG evaluation record
• IESG writeups
• Email expansions
• History

• Versions
• 00
• 01
• 02
• 03
• 04
• 05
• 06
• 07
• 08
• 09
• 10
• 11
• 12
• 13
• 14
• 15
• 16
• 17
• 18
• 19
• 20



Internet Engineering Task Force (IETF)                       E. Rescorla
Request for Comments: 8827                                       Mozilla
Category: Standards Track                                   January 2021
ISSN: 2070-1721

                      WebRTC Security Architecture

Abstract

   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
   https://www.rfc-editor.org/info/rfc8827.

Copyright Notice

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

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (https://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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

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
           Transactions
       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
               Site
       9.4.4.  Security of Third-Party IdPs
         9.4.4.1.  Confusable Characters
       9.4.5.  Web Security Feature Interactions
         9.4.5.1.  Popup Blocking
         9.4.5.2.  Third Party Cookies
   10. IANA Considerations
   11. References
     11.1.  Normative References
     11.2.  Informative References
   Acknowledgements
   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.

                            +----------------+