State of the art, real-time, rate-adaptive, multimedia applications adjust their transmission rate to match the available network capacity. Unfortunately, this source-based rate-adaptation performs poorly in a heterogeneous multicast environment because there is no single target rate --- the conflicting bandwidth requirements of all receivers cannot be simultaneously satisfied with one transmission rate. If the burden of rate-adaption is moved from the source to the receivers, heterogeneity is accommodated. One approach to receiver-driven adaptation is to combine a layered source coding algorithm with a layered transmission system. By selectively forwarding subsets of layers at constrained network links, each user receives the best quality signal that the network can deliver. We and others have proposed that selective-forwarding be carried out using multiple IP-Multicast groups where each receiver specifies its level of subscription by joining a subset of the groups. In this paper, we ...

We describe a mechanism for scalable control of multicast continuous media streams. The mechanism uses a novel probing mechanism to solicit feedback information in a scalable manner and to estimate the number of receivers. In addition, it separates the congestion signal from the congestion control algorithm, so as to cope with heterogeneous networks. This mechanism has been implemented in the IVS videoconference system using options within RTP to elicit information about the quality of the video delivered to the receivers. The H.261 coder of IVS then uses this information to adjust its output rate, the goal being to maximize the perceptual quality of the image received at the destinations while minimizing the bandwidth used by the video transmission. We find that our prototype control mechanism is well suited to the Internet environment. Furthermore, it prevents video sources from creating congestion in the Internet. Experiments are underway to investigate how the scalable probing mech...

This paper describes the INRIA Videoconferencing System (IVS), a low bandwidth tool for real-time video between workstations on the Internet using UDP datagrams and the IP multicast extension. The video coder-decoder (codec) is a software implementation of the UIT-T recommendation H.261 originally developed for the Integrated Services Digital Network (ISDN). Our focus in this paper is on adapting this codec for the Internet environment. We propose a packetization scheme, an error control scheme and an output rate control scheme that adapts the image coding process based on network conditions. This work shows that it is possible to maintain videoconferences with reasonable quality across packet-switched networks without requiring special support from the network such as resource reservation or admission control.

Variable-bit-rate compressed video can exhibit significant, multiple-time-scale bit rate variability. In this paper we consider the transmission of stored video from a server to a client across a network, and explore how the client buffer space can be used most effectively toward reducing the variability of the transmitted bit rate. Two basic results are presented. First, we show how to achieve the greatest possible reduction in rate variability when sending stored video to a client with given buffer size. We formally establish the optimality of our approach and illustrate its performance over a set of long MPEG-1 encoded video traces. Second, we evaluate the impact of optimal smoothing on the network resources needed for video transport, under two network service models: Deterministic Guaranteed service [1, 31] and Renegotiated CBR (RCBR) service [9]. Under both models the impact of optimal smoothing is dramatic. 1 Introduction A broad range of applications is enabled by the capac...

..................................................................... ....... i Acknowledgements ............................................................... ii 1. Introduction .................................................................... 1 2. Quality of Service Terminology, Principles and Concepts ................. 17 2.1 Terminology.......................................................... .......................17 2.2 Qos Principles........................................................... ...................18 2.2.1 Integration Principle .................................................................19 2.2.2 Separation Principle .................................................................19 2.2.3 Transparency Principle............................................................ ..19 2.2.4 Asynchronous Resource Management Principle .................................20 2.2.5 Performance Principle............................................................ ...20 2.3 QoS S...

Over the past several years there has been a considerable amount of research within the field of quality of service (QoS) support for distributed multimedia systems. To date, most of the work has been within the context of individual architectural layers such as the distributed system platform, operating system, transport subsystem and network. Much less progress has been made in addressing the issue of overall end-to-end support for multimedia communications. In recognition of this, a number of research teams have proposed the development of QoS architectures which incorporate quality of service configurable interfaces and quality of service driven control and management mechanisms across all architectural layers. This paper examines the state-of-the-art in the development of QoS architectures. The approach taken is to present QoS terminology and a generalised QoS framework for understanding and discussing quality of service in the context of distributed multimedia systems. Following this, we evaluate a number of QoS architectures that have emerged in the literature. 1.

We introduce a point to point real-time video transmission scheme over the Internet combining a low-delay TCP-friendly transport protocol in conjunction with a novel compression method that is error resilient and bandwidth-scalable. Compressed video is packetized into individually decodable packets of equal expected visual importance. Consequently, relatively constant video quality can be achieved at the receiver under lossy conditions. Furthermore, the packets can be truncated to instantaneously meet the time varying bandwidth imposed by a TCP-friendly transport protocol. As a result, adaptive flows that are friendly to other Internet traffic are produced. Actual Internet experiments together with simulations are used to evaluate the performance of the compression, transport, and the combined schemes.

Variable bit-rate (VBR) compressed video traffic is expected to be a significant component of the traffic mix in integrated services networks. This traffic is hard to manage because it has strict delay and loss requirements while simultaneously exhibiting burstiness at multiple time scales. We show that burstiness over long time scales, in conjunction with resource reservation using one-shot traffic descriptors, can substantially degrade the loss rate, end-to-end delay, and statistical multiplexing gain of a connection. We use large-deviation theory to model the performance of multiple time-scale traffic and to motivate the design of renegotiated constant bit rate (RCBR) service. Sources using

Layered Video Multicast with Retransmissions (LVMR) is a system for distributing video using layered coding over the Internet. The two key contributions of the system are: (1) improving the quality of reception within each layer by retransmitting lost packets given an upper bound on recovery time and applying an adaptive playback point scheme to help achieve more successful retransmission, and (2) adapting to network congestion and heterogeneity using hierarchical rate control mechanism. This paper concentrates on the rate control aspects of LVMR. In contrast to the existing sender-based and receiver-based rate control in which the entire information about network congestion is either available at the sender (in sender-based approach) or replicated at the receivers (in receiver-based approach), the hierarchical rate control mechanism distributes the information between the sender, receivers, and some agents in the network in such a way that each entity maintains only the info...

We describe a mechanism for dynamic adjustment of the bandwidth requirements of multimedia applications. The sending application uses RTP receiver reports to compute packet loss and jitter. Based on these metrics the congestion state seen by the receivers is determined and the bandwidth is adjusted by a linear regulator with dead zone. The suggested mechanism has been implemented and controls the bandwidth of the vic video conferencing system. Currently we are evaluating the proposed algorithms by simulations and experiments on the Internet and on our local ATM network. 1 Introduction Multimedia applications pose unique challenges for network control and management: they offer high data rates, stringent real-time contraints, long connection durations and relatively inflexible demands on bandwidth. Due to the long connection duration, the standard cycle of QoS negotiation purely at the beginning of a session leads either to high call rejection probability at busy times or unnece...