WebRTC Security Architecture
RFC 8827
Document | Type | RFC - Proposed Standard (January 2021; No errata) | |
---|---|---|---|
Author | Eric Rescorla | ||
Last updated | 2021-01-18 | ||
Stream | IETF | ||
Formats | plain text html xml pdf htmlized bibtex | ||
Reviews | |||
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) | |
Action Holders |
(None)
|
||
Consensus Boilerplate | Yes | ||
Telechat date | |||
Responsible AD | Adam Roach | ||
Send notices to | Sean Turner <sean@sn3rd.com> | ||
IANA | IANA review state | Version Changed - Review Needed | |
IANA action state | RFC-Ed-Ack |
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. +----------------+Show full document text