Options for Repair of Streaming Media (RFC2354)
Original Publication Date: 1998-Jun-01
Included in the Prior Art Database: 2000-Sep-13
Internet Society Requests For Comment (RFCs)
C. Perkins: AUTHOR [+2]
This document summarizes a range of possible techniques for the repair of continuous media streams subject to packet loss. The techniques discussed include redundant transmission, retransmission, interleaving and forward error correction. The range of applicability of these techniques is noted, together with the protocol requirements and dependencies.
Network Working Group C. Perkins
Request for Comments: 2354 O. Hodson
Category: Informational University College London
Options for Repair of Streaming Media
Status of this Memo
This memo provides information for the Internet community. This memo
does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.
Copyright (C) The Internet Society (1998). All Rights Reserved.
This document summarizes a range of possible techniques for the
repair of continuous media streams subject to packet loss. The
techniques discussed include redundant transmission, retransmission,
interleaving and forward error correction. The range of
applicability of these techniques is noted, together with the
protocol requirements and dependencies.
A number of applications have emerged which use RTP/UDP transport to
deliver continuous media streams. Due to the unreliable nature of
UDP packet delivery, the quality of the received stream will be
adversely affected by packet loss. A number of techniques exist by
which the effects of packet loss may be repaired. These techniques
have a wide range of applicability and require varying degrees of
protocol support. In this document, a number of such techniques are
discussed, and recommendations for their applicability made.
It should be noted that this document is introductory in nature, and
does not attempt to be comprehensive. In particular, we restrict our
discussion to repair techniques which require the involvement of the
sender of a media stream, and do not discuss possibilities for
receiver based repair.
For a more detailed survey, the reader is referred to .
2 Terminology and Protocol Framework
A unit is defined to be a timed interval of media data, typically
derived from the workings of the media coder. A packet comprises one
or more units, encapsulated for transmission over the network. For
example, many audio coders operate on 20ms units, which are typically
combined to produce 40ms or 80ms packets for transmission. The
framework of RTP  is assumed. This implies that packets have a
sequence number and timestamp. The sequence number denotes the order
in which packets are transmitted, and is used to detect losses. The
timestamp is used to determine the playout order of units. Most loss
recovery schemes rely on units being sent out of order, so an
application must use the RTP timestamp to schedule playout.
The use of RTP allows for several different media coders, with a
payload type field being used to distinguish between these at the
receiver. Some loss repair schemes send multiple copies of units, at
different times and possibly with different encodings, to increase