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
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Contributions published or made publicly available before November
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material may not have granted the IETF Trust the right to allow
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Without obtaining an adequate license from the person(s) controlling
the copyright in such materials, this document may not be modified
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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
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.
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