Advances in microsensor and radio technology will enable small but smart sensors to be deployed for a wide range of environmental monitoring applications. The low per-node cost will allow these wireless networks of sensors and actuators to be densely distributed. The nodes in these dense networks will coordinate to perform the distributed sensing and actuation tasks. Moreover, as described in this paper, the nodes can also coordinate to exploit the redundancy provided by high density so as to extend overall system lifetime. The large number of nodes deployed in these systems will preclude manual configuration, and the environmental dynamics will preclude design-time preconfiguration. Therefore, nodes will have to self-configure to establish a topology that provides communication under stringent energy constraints. ASCENT builds on the notion that, as density increases, only a subset of the nodes are necessary to establish a routing forwarding backbone. In ASCENT, each node assesses its connectivity and adapts its participation in the multihop network topology based on the measured operating region. This paper motivates and describes the ASCENT algorithm and presents analysis, simulation, and experimental measurements. We show that the system achieves linear increase in energy savings as a function of the density and the convergence time required in case of node failures while still providing adequate connectivity.

The architecture of World Wide Web (WWW) browsers and servers support full file transfer for document retrieval. TCP is used for data transfers by Web browsers and their associated Hypertext Transfer Protocol (HTTP) servers. Full file transfer and TCP are unsuitable for continuous media, such as real time audio and video. In order for the WWW to support continuous media, we require the transmission of video and audio on-demand and in real time, as well as new protocols for real time data. We extend the architecture of the WWW to encompass the dynamic, real time information space of video and audio. Our WWW browser Vosaic, short for Video Mosaic, incorporates real time video and audio into standard hypertext pages and which are displayed in place. Video and audio transfers occur in real time; there is no file retrieval latency. The video and audio result in compelling Web pages. Real time video and audio data can be effectively served over the present day Internet with the proper transm...

This paper examines the effectiveness of combining multiple description coding (MDC) and multiple path transport (MPT) for video and image transmission in a multihop mobile radio network. The video and image information is encoded nonhierarchically into multiple descriptions with the following objectives. The received picture quality should be acceptable, even if only one description is received and every additional received description contributes to enhanced picture quality. Typical applications will need a higher bandwidth/higher reliability connection than that provided by a single link in current mobile networks. For supporting these applications, a mobile node may need to set up and use multiple paths to the desired destination, either simply because of the lack of raw bandwidth on a single channel or because of its poor error characteristics, which reduce its effective throughput. In the context of this work, the principal reasons for considering such an architecture are providing high bandwidth and more robust end-to-end connections. We describe a protocol architecture that addresses this need and, with the help of simulations, we demonstrate the feasibility of this system and compare the performance of the MDC-MPT scheme to a system using layered coding and asymmetrical paths for the base and enhancement layers.

Real-time streaming of audio-visual content over Internet Protocol (IP) based networks has enabled a wide range of multimedia applications. An Internet streaming solution has to provide real-time delivery and presentation of a continuous media content while compensating for the lack of Quality-of-Service (QoS) guarantees over the Internet. Due to the variation and unpredictability of bandwidth and other performance parameters (e.g. packet loss rate) over IP networks, in general, most of the proposed streaming solutions are based on some type of a data loss handling method and a layered video coding scheme. In this paper, we describe a real-time streaming solution suitable for non-delay-sensitive video applications such as video-on-demand and live TV viewing. The main aspects of our proposed streaming solution are: 1. An MPEG-4 based scalable video coding method using both a prediction-based base layer and a "ne-granular enhancement layer; 2. An integrated transport-decoder bu!er model with priority re-transmission for the recovery of lost packets, and continuous decoding and presentation of video. In addition to describing the above two aspects of our system, we also give an overview of a recent activity within MPEG-4 video on the development of a "ne-granular-scalability coding tool for streaming applications. Results for the performance of our scalable video coding scheme and the re-transmission mechanism are also presented. The latter results are based on actual testing conducted over Internet sessions used for streaming MPEG-4 video in realtime.

. We describe and evaluate the implementation of a streaming media system called Yima, which consists of a scalable continuous media server and client components. We report on the real-life experiences that we gained from streaming near NTSC quality video and audio to residential locations within a metropolitan area. We investigated the feasibility of such streaming services with current broadband technology. We describe our experimental setup and the results, which indicate that streaming applications, such as video-on-demand, are not only technically feasible but also may be economically viable in the near future. 1

This research identifies the problems encountered in transmitting voice over the Internet and proposes approaches to solve these problems. The current Internet is not very suitable for transmitting real-time data because its underlying protocols and switches were only engineered to transmit non-real time data. The problems posed by voice over the Internet are studied by conducting a series of experiments. These problems caused by high loss, large delay, and jitter will seriously affect the transmission quality. A good design thus needs to combine different components that solve these problems together. Silence removal and compression is used to reduce bandwidth usage. Jitter buffers are used to smooth the burstiness in the received stream caused by the network. To conceal loss, we investigate existing methods and propose two new nonredundant reconstruction algorithms based on a simple but effective average reconstruction scheme. One is to apply adaptive filtering on top of average reco...

This document describes Real-Time Transport Protocol (RTP) payload formats for carrying each of MPEG-4 Audio and MPEG-4 Visual bitstreams without using MPEG-4 Systems. For the purpose of directly mapping MPEG-4 Audio/Visual bitstreams onto RTP packets, it provides specifications for the use of RTP header fields and also specifies fragmentation rules. It also provides specifications for Multipurpose Internet Mail Extensions (MIME) type registrations and the use of Session Description Protocol (SDP).

This paper motivates the need for Multiple Path Transport #MPT# of video and image information in a multihop mobile radio network. The video and image information is encoded non-hierarchically into multiple descriptions #MDC# with the following objectives. The received picture quality should be acceptable even if only one description is received and every additionally received description contributes to enhanced picture quality.Typical applications will need a higher bandwidth#higher reliability connection than that provided by current mobile networks. For supporting these applications a mobile node may need to set up and use multiple paths to the desired destination, either simply because of the lackofraw bandwidth on a single channel or because of its poor error characteristics, which reduces its e#ective throughput. In the context of this work, the principal reasons for considering suchanarchitecture are providing high bandwidth and more robust end-to-end connections. We describe a protocol architecture that addresses this need and, with the help of simulations, we demonstrate the feasibilityofthis system and compare the performance of the MDC-MPT scheme to that of a hierarchical coding scheme.

Abstract: Presently, digital continuous media (CM) are well established as an integral part of many applications. In recent years, a considerable amount of research has focused on the efficient retrieval of such media. Scant attention has been paid to servers that can record such streams in real time. However, more and more devices produce direct digital output streams. Hence, the need arises to capture and store these streams with an efficient data stream recorder that can handle both recording and playback of many streams simultaneously and provide a central repository for all data. We propose a design for a large scale data stream recorder. Our goal was to introduce a unified architecture that integrates multi-stream recording and retrieval in a coherent manner. The discussion raises practical issues such as support for multizone disk drives, variable bit rate media, and disk drives that have a different write than read bandwidth. We provide initial solutions for some issues while others will need to be investigated further. 1

In this paper we focus on self-contention – contention between packets of the same transport layer connection along the path from source to destination. We observe that selfcontention plays an important role in degrading TCP performance in multi-hop wireless networks and that the use of the popular IEEE 802.11 MAC protocol exacerbates selfcontention. We propose and study two MAC-layer approaches to alleviate self-contention. The first approach, called quickexchange (QE), is designed with the intent of reducing the effects of inter-flow self-contention (e.g. between packets of the same connection traveling in opposite directions). The design of our second mechanism, called fast-forward (FF), is geared towards decreasing intra-flow self-contention (e.g. between packets of the same connection traveling in the same direction). We simulate and study our proposed schemes and observe that quick-exchange consistently improves network aggregate goodput (by as much as 20 % in string topologies, 15 % in random static scenarios, and 10 % in random mobile scenarios). In contrast to our expectations, fast-forward causes sporadic and often negative effects on goodput for TCP connections. Upon investigation we find that while the MAC is, in some respect, operating more efficiently, as demonstrated by improved UDP throughput; interactions with TCPs congestion control mechanism cause the goodput to degrade. We analyze various effects that cause the respective behaviors with QE and FF in detail.