In recent years, the "Internet Multicast Backbone," or MBone, has risen from a small, research curiosity to a largescale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. In this paper, we describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet) . Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a "real" application---the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone.

In just a few years the "Internet Multicast Backbone", or MBone, has risen from a small, research curiosity to a large scale and widely used communications infrastructure. A driving force behind this growth was our development of multipoint audio, video, and shared whiteboard conferencing applications that are now used daily by the large and growing MBone community. Because these real-time media are transmitted at a uniform rate to all the receivers in the network, the source must either run below the bottleneck rate or overload portions of the multicast distribution tree. In this dissertation, we propose a solution to this problem by moving the burden of rate-adaptation from the source to the receivers with a scheme we call Receiver-driven Layered Multicast, or RLM. In RLM, a source distr...

In this paper we analyze the packet loss process in a single server queueing system with a finite buffer capacity. The model we use addresses the packet loss probabilities for packets within a block of consecutive sequence of packets. In contrast to other work which used an independence assumption to compute the loss probabilities of packets within a block, we present an analytical approach that yields efficient recursions for the computation of the distribution of the number of lost packets within a block of packets of fixed or variable size for several arrival models and a several number of sessions. Numerical examples are provided to compare the distribution obtained from our analysis with the distribution obtained by using the independence assumption. The results give insight to the following areas related to high-speed networks: (i) Forward error correction schemes become less efficient due to the bursty nature of the packet loss processes. (ii) Real time traffic such as voice and...

In this paper, we examine the transport and storage of video compressed with a variable bit rate (VBR). We focus primarily on networked video, although we also briefly consider other applications of VBR video, including satellite transmission (channel sharing), playback of stored video, and wireless transport. Packet video research requires careful integration between the network and the video systems; however, a major stumbling block has resulted because commonly used terms are often interpreted differently by the video and networking communities.

The Multimedia Multicast Channel is a dissemination-oriented communication abstraction providing a service analogous to that of a cable television broadcast channel. A source transmits multimedia information such as video and audio streams onto a channel, and a varying number of receivers "tune in" to the channel to receive a selected set of the streams. To support heterogeneity, each receiver may tailor the selected streams to meet individual needs through the use of filters. The design encourages a very loose coupling between the source and the receivers, promoting open-loop control for the underlying network protocols. KEY WORDS: multicast, multimedia, dissemination, computer networks. This research has been supported in part by grants from DEC, IBM, NCR, NSF, TRW, and UC MICRO. The views expressed are those of the authors, and not necessarily those of the supporters. 1 Introduction The Multimedia Multicast Channel (MMC) is a programming abstraction which supports dissemination...

Weinvestigate the interaction of congestion control with the partitioning of source information into components of varying importance for variable bit rate packet voice and packet video. High priority transport for the more important signal components results in substantially increased objective service quality. Using a Markovchain voice source model with simple PCM speech encoding and a priority queue, simulation results show a signal-to-noise ratio improvementof45dBwithtwo priorities over an unprioritized system. Performance is sensitive to the fraction of traffic placed in eachpriority, and the optimal partition depends on network loss conditions. When this partition is optimized dynamically, quality degrades gracefully over a wide range of load values. Results with DCT encoded speech and video samples show similar behavior. Variations are investigated such as further partition of low priority information into multiple priorities. A simulation with delay added to represe...

Distribution of continuous media traffic such as digital audio and video over packet-switching networks has become increasingly feasible due to a number of technology trends leading to powerful desktop computers and high-speed integrated services networks. Protocols supporting the transmission of continuous media are already available. In these protocols, transmission errors due to packet loss are generally not recovered. Instead existing protocol designs focus on preventive error control techniques that reduce the impact of losses by adding redundancy, e.g., forward error correction, or by preventing loss of important data, e.g., channel coding. The goal of this study is to show that retransmission of continuous media data often is, contrary to conventional wisdom, a viable option in most packet-switching networks. If timely retransmission can be performed with a high probability of success, a retransmission-based approach to error control is attractive because it imposes little overh...

different encoding formats

The wavelet transform has become a cutting-edge technology in image compression research. This article explains what wavelets are and provides a practical, nuts-andbolts tutorial on wavelet-based compression that will help readers to understand and experiment with this important new technology. Keywords: image coding, signal compression, wavelet transform, image transforms 1 Introduction The advent of multimedia computing has lead to an increased demand for digital images. The storage and manipulation of these images in their raw form is very expensive; for example, a standard 35mm photograph digitized at 12 ¯m per pixel requires about 18 MBytes of storage and one second of NTSC-quality color video requires almost 23 MBytes of storage. To make widespread use of digital imagery practical, some form of data compression must be used. Digital images can be compressed by eliminating redundant information. There are three types of redundancy that can be exploited by image compression system...

This paper gives an introduction to the issues involved in asynchronous video transfers. Brief overviews of video coding, rate control, multiplexing, as well as delay, error and loss control are given. Section 1. Background A collection of the world's telecommunication operators and their equipment suppliers are currently standardizing the asynchronous transfer mode (ATM) within the International Telecommunication Union (ITU) as a basis for broadband integrated services digital networks. In parallel with the ITU, the ATM Forum is furthering the development process by including computer communication manufacturers, as well as end--users in the standardization process. Although the representations give different perspectives on the proposed network solutions, the overall goal of building networks capable of transferring multimedia efficiently is shared between the ITU and the ATM Forum. In addition to the work on ATM, new developments for internet protocols lean towards supporting full--fledged multimedia communications. The simple internet protocol plus (RFC 1710) has been selected by the Internet Engineering Task Force as a basis for the next generation internet protocol (version 6, often denoted IPng; see RFC 1752). It features connection--oriented services, denoted flows, and signaling by means of the resource reservation protocol (RSVP) [103]. An implied assumption in the development of both ATM and IPng is that network users will get access to copious amounts of transmission capacity at affordable prices, even in the local loop. This capacity is needed to lower the latency of bulk data transfers and to enable audio and video communications. The provision of the latter has received considerable attention among researchers and is the topic of this paper. Asynchronous ...