Evaluating Congestion Control for Interactive Real-time Media

Document Type Expired Internet-Draft (rmcat WG)
Last updated 2019-05-09 (latest revision 2018-11-05)
Replaces draft-singh-rmcat-cc-eval
Stream IETF
Intended RFC status Informational
Expired & archived
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Stream WG state In WG Last Call (wg milestones: Aug 2013 - Adopt first WG draft..., Dec 2018 - Submit requirements ... )
Document shepherd Martin Stiemerling
IESG IESG state Expired
Consensus Boilerplate Unknown
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Responsible AD (None)
Send notices to Martin Stiemerling <mls.ietf@gmail.com>

This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at


The Real-time Transport Protocol (RTP) is used to transmit media in telephony and video conferencing applications. This document describes the guidelines to evaluate new congestion control algorithms for interactive point-to-point real-time media.


Varun Singh (varun@callstats.io)
Joerg Ott (ott@in.tum.de)
Stefan Holmer (holmer@google.com)

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