Achieving High Throughput for Data Transfer over ATM Networks

File-transfer rates for ftp are often reported to be relatively slow, compared to the raw bandwidth available in emerging gigabit networks. While a major bottleneck is disk I/O, protocol issues impact performance as well. Ftp was developed and optimized for use over the TCP/IP protocol stack of the Internet. However, TCP has been shown to run inefficiently over ATM. In an effort to maximize network throughput, data-transfer protocols can be developed to run over UDP or directly over IP, rather than over TCP. If error-free transmission is required, techniques for achieving reliable transmission can be included as part of the transfer protocol. However, selected image-processing applications can tolerate a low level of errors in images that are transmitted over a network. In this paper we report on experimental work to develop a high-throughput protocol for unreliable data transfer over ATM networks. We attempt to maximize throughput by keeping the communications pipe full, but still keep packet loss under five percent. We use the Bay Area Gigabit Network Testbed as our experimental platform

FEC Performances in Multimedia Streaming

In this paper, the performances of packet-level media-independent FEC schemes are computed in terms of both packet loss ratio and average burst length of multimedia data after error recovery. The set of equations leading to the analytical formulation of both parameters are first given for a renewal error process. Finally, the FEC performances parameters are computed in the case of a Gilbert-model loss process and compared to experimental data

G.729 Error Recovery for Internet Telephony

This memorandum discusses the use of the ITU G.729 CS-ACELP speech coder on the Internet for telephony applications. In particular, the memo explores issues of error resiliency and recovery. In particular, the paper considers two questions. First, given N consecutive frame erasures (due to packet losses), how long does the decoder take to resynchronize its state with the encoder? What is the strength of the resulting error signal, both in terms of objective and subjective measures? The second issue explores which particular factors contribute to the strength of the error signal: the distortion of the speech due to the incorrect concealment of the erasures, or the subsequent distortion due to the loss of state synchronization, even though correct packets are being received. Both objective and subjective measures are used to characterize the importance of each of these two factors

Multicast Routing of Hierarchical Data

The issue of multicast of broadband, real-time data in a heterogeneous environment, in which the data recipients differ in their reception abilities, is considered. Traditional multicast schemes, which are designed to deliver all the source data to all recipients, offer limited performance in such an environment, since they must either force the source to overcompress its signal or restrict the destination population to those who can receive the full signal. We present an approach for resolving this issue by combining hierarchical source coding techniques, which allow recipients to trade off reception bandwidth for signal quality, and sophisticated routing algorithms that deliver to each destination the maximum possible signal quality. The field of hierarchical coding is briefly surveyed and new multicast routing algorithms are presented. The algorithms are compared in terms of network utilization efficiency, lengths of paths, and the required mechanisms for forwarding packets on the resulting paths

Regular Topologies for Gigabit Wide-Area Networks

In general terms, this project aimed at the analysis and design of techniques for very high-speed networking. The formal objectives of the project were to: (1) Identify switch and network technologies for wide-area networks that interconnect a large number of users and can provide individual data paths at gigabit/s rates; (2) Quantitatively evaluate and compare existing and proposed architectures and protocols, identify their strength and growth potentials, and ascertain the compatibility of competing technologies; and (3) Propose new approaches to existing architectures and protocols, and identify opportunities for research to overcome deficiencies and enhance performance. The project was organized into two parts: 1. The design, analysis, and specification of techniques and protocols for very-high-speed network environments. In this part, SRI has focused on several key high-speed networking areas, including Forward Error Control (FEC) for high-speed networks in which data distortion is the result of packet loss, and the distribution of broadband, real-time traffic in multiple user sessions. 2. Congestion Avoidance Testbed Experiment (CATE). This part of the project was done within the framework of the DARTnet experimental T1 national network. The aim of the work was to advance the state of the art in benchmarking DARTnet's performance and traffic control by developing support tools for network experimentation, by designing benchmarks that allow various algorithms to be meaningfully compared, and by investigating new queueing techniques that better satisfy the needs of best-effort and reserved-resource traffic. This document is the final technical report describing the results obtained by SRI under this project. The report consists of three volumes: Volume 1 contains a technical description of the network techniques developed by SRI in the areas of FEC and multicast of real-time traffic. Volume 2 describes the work performed under CATE. Volume 3 contains the source code of all software developed under CATE

Dynamic Time Windows and Generalized Virtual Clocks-Combined Closed-Loop/Open-Loop Mechanisms for Congestion Control of Data Traffic in High Speed Wide Area Networks

This paper presents a set of mechanisms for congestion control of data traffic in high speed wide area networks (HSWANs) along with preliminary performance results. The model of the network assumes reservation of resources based on average requirements. The mechanisms address (a) the different network time constants (short term and medium-term), (b) admission control that allows controlled variance of traffic as a function of medium-term congestion, and (c) prioritized scheduling which is based on a new fairness criterion. This latter criterion is perceived as the appropriate fairness measure for HSWANs. Preliminary performance studies show that the queue length statistics at switching nodes (mean, variance and max) are approximately proportional to the end-point \u27time window\u27 size. Further, * when network utilization approaches unity, the time window mechanism can protect the network from buffer overruns and excessive queueing delays, and * when network utilization level is smaller, the time window may be increased to allow a controlled amount of variance that attempts to simultaneously meet the performance goals of the end-user and that of the network. The prioritized scheduling algorithms proposed and studied in this paper are a generalization of the Virtual Clock algorithm [Zhang 1989]. The study here investigates * necessary and sufficient conditions for accomplishing desired fairness, * simulation and (limited analytical results for expected waiting times, * ability to protect against misbehaving users, and * relationship between end-point admission control (Time-Window) and internal scheduling (\u27Pulse\u27 and Virtual Clock) at the switch

Applications of analysis and synthesis techniques for complex sounds

Master'sMASTER OF SCIENC

Energy-efficient wireless communication

In this chapter we present an energy-efficient highly adaptive network interface architecture and a novel data link layer protocol for wireless networks that provides Quality of Service (QoS) support for diverse traffic types. Due to the dynamic nature of wireless networks, adaptations in bandwidth scheduling and error control are necessary to achieve energy efficiency and an acceptable quality of service. In our approach we apply adaptability through all layers of the protocol stack, and provide feedback to the applications. In this way the applications can adapt the data streams, and the network protocols can adapt the communication parameters

Soft ARQ for layered streaming media

A growing and important class of traffic in the Internet is so-called `streaming media,' in which a server transmits a packetized multimedia signal to a receiver that buffers the packets for playback. This playback buffer, if adequately sized, counteracts the adverse impact of delay jitter and reordering suffered by packets as they traverse the network, and if large enough also allows lost packets to be retransmitted before their playback deadline expires. We call this framework for retransmitting lost streaming-media packets `soft ARQ' since it represents a relaxed form of Automatic Repeat reQuest (ARQ). While state-of-the-art media servers employ such strategies, no work to date has proposed an optimal strategy for delay-constrained retransmissions of streaming media-specifically, one which determines what is the optimal packet to transmit at any given point in time. In this paper, we address this issue and present a framework for streaming media retransmission based on layered media representations, in which a signal is decomposed into a discrete number of layers and each successive layer provides enhanced quality. In our approach, the source chooses between transmitting (1) newer but critical coarse information (e.g., a first approximation of the media signal) and (2) older but less important refinement reformation (e.g., added details) using a decision process that minimizes the expected signal distortion at the receiver. To arrive at the proper mix of these two extreme strategies, we derive an optimal strategy for transmitting layered data over a binary erasure channel with instantaneous feedback. To provide a quantitative performance comparison of different transmission policies, we conduct a Markov-chain analysis, which shows that the best transmission policy is time-invariant and thus does not change as the frames' layers approach their expiration times