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This paper addresses the problem of streaming packetized media

Content distribution networks (CDNs) are a mechanism to deliver content to end users on behalf of origin Web sites. Content distribution offloads work from origin servers by serving some or all of the contents of Web pages. We found an order of magnitude increase in the number and percentage of popular origin sites using CDNs between

Abstract—Due to the explosive growth of the Internet and increasing demand for multimedia information on the web, streaming video over the Internet has received tremendous attention from academia and industry. Transmission of real-time video typically has bandwidth, delay, and loss requirements. However, the current best-effort Internet does not offer any quality of service (QoS) guarantees to streaming video. Furthermore, for video multicast, it is difficult to achieve both efficiency and flexibility. Thus, Internet streaming video poses many challenges. To address these challenges, extensive research has been conducted. This special issue is aimed at dissemination of the contributions in the field of streaming video over the Internet. To introduce this special issue with the necessary background and provide an integral view on this field, we cover six key areas of streaming video. Specifically, we cover video compression, application-layer QoS control, continuous media distribution services, streaming servers, media synchronization mechanisms, and protocols for streaming media. For each area, we address the particular issues and review major approaches and mechanisms. We also discuss the tradeoffs of the approaches and point out future research directions. Index Terms—Application-layer QoS control, continuous media distribution services, Internet, protocol, streaming video,

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...

This paper presents a system which employs the accepted notion of melodic pitch contours to support content-based navigation around a body of multimedia documents including MIDI and digital audio files. The system adopts an open hypermedia model which enables the user to find available links from an arbitrary fragment of a piece of music, based on the content or location of that fragment. The design of the tools, indexed contour database and the fast contour-matching algorithms are discussed. 1.1 Keywords Open hypermedia, content based navigation, branching audio, melodic contours, pitch contours, query by humming 2.

This paper presents a model and theory for streaming layered video. We model the bandwidth available to the streaming application as a stochastic process whose statistical characteristics are unknown a priori. The random bandwidth models short term variations due to congestion control (such as TCP-friendly conformance). We suppose that the video has been encoded into a base and an enhancement layer, and that to decode the enhancement layer the base layer has to be available to the client. We make the natural assumption that the client has abundant local storage and attempts to prefetch as much of the video as possible during playback. At any instant of time, starvation or partial starvation can occur at the client in either of the two layers. During periods of starvation, the client applies video error concealment to hide the loss. We study the dynamic allocation of the available bandwidth to the two layers in order to minimize the impact of client starvation. For the case of an infinitely-long video, we find that the optimal policy takes on a surprisingly simple and static form. For finite-length videos, the optimal policy is a simple static policy when the enhancement layer is deemed at least as important as the base layer. When the base layer is more important, we design a threshold policy heuristic which switches between two static policies. We provide numerical results that compare the performance of no-prefetching, static and threshold policies. I.

The Internet and intranets have been used to deliver continuous media, both stored and live, for a number of years. Most of the attention has focused on providing guaranteed quality of service (RSVP) and end-to-end data transport (RTP), with every application using its own control protocol. In this paper, we describe a control architecture that offers most standard advanced telephony features and integrates stored and conference multimedia. The protocol re-uses much of the “infrastructure ” of HTTP, including its security and proxy mechanisms. The architecture is instantiated by two related, but independent protocols: the Session Initiation Protocol (SIP) for inviting participants to a multimedia session and the Real-Time Stream Protocol (RTSP) to control playback and recording for stored continuous media. 1

Internet telephony must offer the standard telephony services. However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network (PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services. 1. Introduction Internet ...

Abstract. Replicating Web documents at a worldwide scale can help reduce user-perceived latency and wide-area network traffic. This paper presents the design of Globule, a platform that automates all aspects of such replication: serverto-server peering negotiation, creation and destruction of replicas, selection of the most appropriate replication strategies on a per-document basis, consistency management and transparent redirection of clients to replicas. Globule is initially directed to support standard Web documents. However, it can also be applied to stream-oriented documents. To facilitate the transition from a non-replicated server to a replicated one, we designed Globule as a module for the Apache Web server. Therefore, converting Web documents should require no more than compiling a new module into Apache and editing a configuration file. 1