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
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
Consensus Boilerplate Yes
Telechat date
Responsible AD Adam Roach
Send notices to Sean Turner <>
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


   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

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
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
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   include Simplified BSD License text as described in Section 4.e of
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   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  Dedicated Calling Services  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  Key Continuity  Short Authentication Strings  Third-Party Identity  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
   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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