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 ...

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...

Layered transmission has been proposed as a solution to video multicast over the Internet. Since the number of layers is limited, there are usually mismatches between the coarse-grained layer subscription rates and the heterogeneous and dynamic rate requirements from the receivers. Existing studies suggest that such mismatches can be reduced by the use of sender or/and receiver adaptations. In this paper, we present a formal study on the problem of sender adaptation, or dynamic layer rate allocation on the sender's side. Specifically, we study the optimization criteria for layer rate allocation, and propose a metric called Application-aware Fairness Index. This metric takes into consideration 1) the non-linear relation between the perceived video quality and the delivered bandwidth, and 2) the degree of fairness in terms of the receiver's satisfaction with heterogeneous bandwidth requirements. We then formulate the rate allocation into an optimization problem with the objective of maximizing the expected fairness index for all receivers in a session, and derive an efficient and scalable solution using dynamic programming. We further demonstrate that such an optimal sender rate allocation can be seamlessly integrated into an end-to-end adaptation protocol called HALM (Hybrid Adaptation Layered Multicast). This protocol takes advantage of the recent development in layered video coding, and is suitable for the current best-effort Internet.

Information systems and distributed applications for the Internet show a growing demand for real-time audiovisual services. Support for those services within the Internet protocol layer will be widely available in the near future. Nevertheless, the heterogeneous structure of the Internet remains to be a great obstacle for establishing real-time video services. Scalable video codecs, generating bit-streams decodable at different rates, have been proposed to address the heterogeneity problem. In this paper we present a scalable codec based on a spatio-temporal resolution pyramid combined with lattice vector quantization for efficient compression. With this approach additionallower bitrate layers can be transmitted at the same overall bitrate as needed by an H.263 single-layer transmission. The complexity of the codec is sufficiently low to allow software-only implementations of Internet video services. This is demonstrated on the example of an implemented video server application. I. I...

: The explosive growth of the Internet and the intranets attracted a great deal of attention to the implementations and performances of networked multimedia services, which involve the transport of real-time multimedia streams over non-guaranteed quality of service (QoS) networks. In this paper, we discuss some issues related to the transport of MPEG-2 streams over such networks by means of the most recent transport protocols compliant with the Real-Time Transport Protocol (RTP) [1] defined by the Internet Engineering Task Force (IETF). MPEG-2 encoded audio and video transmission is important for several applications including high quality video-on-demand as a part of information-on-demand and high quality video conferencing using the existing network infrastructures. 1. INTRODUCTION Multimedia transport protocols play an important role in the effective delivery and networking of multimedia data. In recent years, several application and transport protocols have been developed. In this...

This paper is a detailed survey of the four related protocols. Other Reports on Recent Advances in Networking Back to Raj Jain's Home Page Table of the Contents Multimedia Networking: Goals and Challenges The real-time challenge m Multimedia over Internet m The solution m l RSVP Development m How does RSVP work m RSVP features m RSVP interfaces m l RTP Development m How does RTP work m RTP fixed header fields m RTCP m RTP features m RTP Implementation Resources m l RTSP Development m RTSP operations and methods m l Multimedia Over IP: RSVP, RTP, RTCP, RTSP http://www.cis.ohio-state.edu/~jain/cis788-97/ip_multimedia/index.htm (1 of 24) [2/7/2000 12:37:09 PM] RTSP features m RTSP i

This is a survey of the state of the art in delivering IP services over ATM networks, as it stands in the second quarter of 1997. It also includes a look at the alternatives to that set of technologies. The technology and the choices are changing "on the fly", and have evolved significantly during the course of this project. Moreover, the issues are not exclusively technical, but in many respects reflect the great schism in the data communications world: connection-oriented versus connectionless networks. We have tried to present the technical issues and solutions along with an unbiased overview of the more "philosophical" issues. We indicate how we think the technology and the installed base of equipment is going to develop over the next few years, in order to give a picture of the future of ATM in data networking. Contents 1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 2 Supported Services . . . . . . . . . . . . . . . . . . . . . . . . . 9 3 The Standar...

The extreme heterogeneity of the Internet and the demanding nature of video traffic has created many challenges for delivery of video over the Internet. This paper reviews the previous work in the areas of video coding, internetworking and video delivery over the internetworks. Recent developments in these areas including object-based video coding, content-based scalability, multicasting and active networks are reviewed. A novel approach for video delivery is proposed that incorporates intelligence into visual data, and employs active networks for transport. Data and methods packed together, use the resources offered by the active network routers, in order to efficiently make their way to their destinations. Objectives of this research are outlined and a methodology is presented.

Multicast Multilayer Videoconferencing: Enhancement of a Multilayer Codec and Implementation of the Receiver Driven Layered Multicast Protocol. (December 1997) Ralph Akram Gholmieh, B.S., Saint-Joseph University at Beirut Chair of Advisory Committee: Dr. Pierce E. Cantrell Videoconferencing is an important topic on the current Internet. The heterogeneity of the network, its access-to-all approach and the varying amount of available bandwidth present many challenges to researchers. Most of the available videoconferencing implementations send video and audio in a monolithic stream with an adaptive rate. In one-to-many sessions, the available end-to-end bandwidth between the source and a receiver might sharply differ from one end-user to another. It is clear that one rate cannot satisfy all the online users because that would force the source to use a least common denominator strategy. Earlier work of graduate students at the TAMU Multimedia and Networking Laboratory has focused on ...