Evaluating Congestion Control for Interactive Real-time Media
draft-singh-rmcat-cc-eval-04

Document Type Replaced Internet-Draft (rmcat WG)
Last updated 2013-10-20
Replaced by draft-ietf-rmcat-eval-criteria
Stream IETF
Intended RFC status (None)
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Stream WG state Candidate for WG Adoption
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IESG IESG state Replaced by draft-ietf-rmcat-eval-criteria
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This Internet-Draft is no longer active.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-singh-rmcat-cc-eval-04.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@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.)