Evaluating Congestion Control for Interactive Real-time Media
draft-singh-rmcat-cc-eval-04
This document is an Internet-Draft (I-D).
Anyone may submit an I-D to the IETF.
This I-D is not endorsed by the IETF and has no formal standing in the
IETF standards process.
| Document | Type |
Replaced Internet-Draft
(rmcat WG)
Expired & archived
|
|
|---|---|---|---|
| Authors | Varun Singh , Joerg Ott | ||
| Last updated | 2013-10-20 | ||
| Replaced by | draft-ietf-rmcat-eval-criteria | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | (None) | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | Candidate for WG Adoption | |
| Document shepherd | (None) | ||
| IESG | IESG state | Replaced by draft-ietf-rmcat-eval-criteria | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
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
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)