RFC 2354 (rfc2354) - Page 2 of 12

Options for Repair of Streaming Media

RFC 2354         Options for Repair of Streaming Media         June 1998


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 [18] 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
   the probability that a receiver has something to decode.  A receiver
   is assumed to have a `quality' ranking of the differing encodings,
   and so is capable of choosing the `best' unit for playout, given
   multiple options.

   A session is defined as interactive if the end-to-end delay is less
   then 250ms, including media coding and decoding, network transit and
   host buffering.

3  Network Loss Characteristics

   If it is desired to repair a media stream subject to packet loss, it
   is useful to have some knowledge of the loss characteristics which
   are likely to be encountered.  A number of studies have been
   conducted on the loss characteristics of the Mbone [2, 8, 21] and
   although the results vary somewhat, the broad conclusion is clear:
   in a large conference it is inevitable that some receivers will
   experience packet loss.  Packet traces taken by Handley [8] show a
   session in which most receivers experience loss in the range 2-5%,
   with a somewhat smaller number seeing significantly higher loss
   rates.  Other studies have presented broadly similar results.

   It has also been shown that the vast majority of losses are of single
   packets.  Burst losses of two or more packets are around an order of
   magnitude less frequent than single packet loss, although they do
   occur more often than would be expected from a purely random process.
   Longer burst losses (of the order of tens of packets) occur
   infrequently.  These results are consistent with a network where
   small amounts of transient congestion cause the majority of packet
   loss.  In a few cases, a network link is found to be severely



Perkins & Hodson             Informational