Evaluating Congestion Control for Interactive Real-time Media

Document Type Replaced Internet-Draft (rmcat WG)
Authors Varun Singh  , Joerg Ott 
Last updated 2013-10-20
Replaced by RFC 8868
Stream Internet Engineering Task Force (IETF)
Intended RFC status (None)
Expired & archived
pdf htmlized bibtex
Stream WG state Candidate for WG Adoption
Document shepherd No shepherd assigned
IESG IESG state Replaced by draft-ietf-rmcat-eval-criteria
Consensus Boilerplate Unknown
Telechat date
Responsible AD (None)
Send notices to (None)

This Internet-Draft is no longer active.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@comnet.tkk.fi)
Joerg Ott (jo@comnet.tkk.fi)

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