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) | ||
| Formats |
Expired & archived
plain text
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html
bibtex
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| 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) | ||
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.)