Evaluating Congestion Control for Interactive Real-time Media
draft-ietf-rmcat-eval-criteria-08

The information below is for an old version of the document
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
Formats
Expired & archived
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Reviews
Additional Resources
- Mailing list discussion
Stream WG state In WG Last Call
Document shepherd Martin Stiemerling
IESG IESG state Expired
Consensus Boilerplate Unknown
Telechat date
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This Internet-Draft is no longer active.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
https://www.ietf.org/archive/id/draft-ietf-rmcat-eval-criteria-08.txt

Abstract

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.

Authors

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.)