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STUN Usage for Consent Freshness

Document Type Replaced Internet-Draft (rtcweb WG)
Expired & archived
Authors Muthu Arul Perumal , Dan Wing , Ram R , Tirumaleswar Reddy.K
Last updated 2013-09-23 (Latest revision 2013-07-15)
Replaced by draft-ietf-rtcweb-stun-consent-freshness
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status (None)
Additional resources Mailing list discussion
Stream WG state Adopted by a WG
Other - see Comment Log
Document shepherd (None)
IESG IESG state Replaced by draft-ietf-rtcweb-stun-consent-freshness
Consensus boilerplate Unknown
Telechat date (None)
Responsible AD (None)
Send notices to (None)

This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:


Verification of peer consent before sending traffic is necessary in WebRTC deployments to ensure that a malicious JavaScript cannot use the browser as a platform for launching attacks. A related problem is session liveness. WebRTC application may want to detect connection failure and take appropriate action. This document describes how a WebRTC browser can verify peer consent to continue sending traffic and detect connection failure.


Muthu Arul Perumal
Dan Wing
Ram R
Tirumaleswar Reddy.K

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)