Security Considerations for WebRTC
RFC 8826
| Document | Type | RFC - Proposed Standard (January 2021; No errata) | |
|---|---|---|---|
| Author | Eric Rescorla | ||
| Last updated | 2021-01-18 | ||
| Replaces | draft-rescorla-rtcweb-security | ||
| Stream | Internent Engineering Task Force (IETF) | ||
| Formats | plain text html xml pdf htmlized (tools) htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Sean Turner | ||
| Shepherd write-up | Show (last changed 2018-01-04) | ||
| IESG | IESG state | RFC 8826 (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 | IANA OK - No Actions Needed | |
| IANA action state | No IANA Actions | ||
Internet Engineering Task Force (IETF) E. Rescorla
Request for Comments: 8826 Mozilla
Category: Standards Track January 2021
ISSN: 2070-1721
Security Considerations for WebRTC
Abstract
WebRTC is a protocol suite for use with real-time applications that
can be deployed in browsers -- "real-time communication on the Web".
This document defines the WebRTC threat model and analyzes the
security threats of WebRTC in that model.
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/rfc8826.
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
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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. The Browser Threat Model
3.1. Access to Local Resources
3.2. Same-Origin Policy
3.3. Bypassing SOP: CORS, WebSockets, and Consent to Communicate
4. Security for WebRTC Applications
4.1. Access to Local Devices
4.1.1. Threats from Screen Sharing
4.1.2. Calling Scenarios and User Expectations
4.1.2.1. Dedicated Calling Services
4.1.2.2. Calling the Site You're On
4.1.3. Origin-Based Security
4.1.4. Security Properties of the Calling Page
4.2. Communications Consent Verification
4.2.1. ICE
4.2.2. Masking
4.2.3. Backward Compatibility
4.2.4. IP Location Privacy
4.3. Communications Security
4.3.1. Protecting Against Retrospective Compromise
4.3.2. Protecting Against During-Call Attack
4.3.2.1. Key Continuity
4.3.2.2. Short Authentication Strings
4.3.2.3. Third-Party Identity
4.3.2.4. Page Access to Media
4.3.3. Malicious Peers
4.4. Privacy Considerations
4.4.1. Correlation of Anonymous Calls
4.4.2. Browser Fingerprinting
5. Security Considerations
6. IANA Considerations
7. References
7.1. Normative References
7.2. Informative References
Acknowledgements
Author's Address
1. Introduction
The Real-Time Communications on the Web (RTCWEB) Working Group has
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. A simple
case is shown below.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTPS / \ HTTPS
or / \ or
WebSockets / \ WebSockets
v v
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