Security Considerations for WebRTC
draft-ietf-rtcweb-security-08
| The information below is for an old version of the document | |||||
|---|---|---|---|---|---|
| Document | Type | Expired Internet-Draft (rtcweb WG) | |||
| Last updated | 2016-04-08 (latest revision 2015-02-26) | ||||
| Replaces | draft-rescorla-rtcweb-security | ||||
| Stream | IETF | ||||
| Intended RFC status | Proposed Standard | ||||
| Formats |
Expired & archived
pdf
htmlized
bibtex
|
||||
| Reviews | |||||
| Additional URLs |
|
||||
| Stream | WG state | Waiting for WG Chair Go-Ahead | |||
| Document shepherd | Sean Turner | ||||
| Shepherd write-up | Show (last changed 2015-03-19) | ||||
| IESG | IESG state | Expired (IESG: Dead) | |||
| Consensus Boilerplate | Unknown | ||||
| Telechat date | |||||
| Responsible AD | Alissa Cooper | ||||
| Send notices to | (None) | ||||
https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-08.txt
Abstract
The Real-Time Communications on the Web (RTCWEB) working group is tasked with standardizing protocols for real-time communications between Web browsers, generally called "WebRTC". 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 soft phones) WebRTC communications are directly controlled by a Web server, which poses new security challenges. For instance, a Web browser might expose a JavaScript API which allows a server to place a video call. Unrestricted access to such an API would allow any site which a user visited to "bug" a user's computer, capturing any activity which passed in front of their camera. This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)