RTP Payload for Interleaved Audio

Document Type Expired Internet-Draft (avt WG)
Author Orion Hodson 
Last updated 2002-05-07
Stream Internet Engineering Task Force (IETF)
Intended RFC status (None)
Expired & archived
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Stream WG state WG Document
Document shepherd No shepherd assigned
IESG IESG state Expired
Consensus Boilerplate Unknown
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This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at


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.


Orion Hodson (o.hodson@cs.ucl.ac.uk)

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)