RTP Payload for Interleaved Audio
draft-ietf-avt-rtp-interleave-00
| Document | Type |
Expired Internet-Draft
(avt WG)
Expired & archived
|
|
|---|---|---|---|
| Author | Orion Hodson | ||
| Last updated | 2002-05-07 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | (None) | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
| Document shepherd | (None) | ||
| IESG | IESG state | Expired | |
| 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This Internet-Draft is no longer active.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
This document describes a payload format for use with the Real-time Transport Protocol (RTP) version 2 for interleaving encoded audio data. It is intended for use in audio streaming delay tolerant applications operating over best-effort packet networks. The goal of interleaving is to disperse burst losses into a series of shorter losses. The total amount of audio lost is not changed by interleaving, but the individual loss events are shorter and easier to conceal at the receiver.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)