While today's computer networks support only best-effort service, future packet-switching integrated-services networks will have to support real-time communication services that allow clients to transport information with performance guarantees expressed in terms of delay, delay jitter, throughput, and loss rate. An important issue in providing guaranteed performance service is the choice of the packet service discipline at switching nodes. In this paper, we survey several service disciplines that are proposed in the literature to provide per-connection end-to-end performance guarantees in packet-switching networks. We describe their mechanisms, their similarities and differences and the performance guarantees they can provide. Various issues and tradeoffs in designing service disciplines for guaranteed performance service are discussed, and a general framework for studying and comparing these disciplines are presented. >

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Service disciplines for guaranteed performance service in packet-switching networks

Techniques to implement MBMS services in a wireless communication system. In one aspect, a method is provided for processing data for transmission to a plurality of terminals. Frames of information bits (which may have variable rates) are provided to a buffer implementing a matrix. The matrix is padded with padding bits based on a particular padding scheme to support variable frame rates. The frames are then coded based on a particular block code to provide parity bits. The frame of information bits and the parity bits are then transmitted to the terminals. In another aspect, a method is provided for controlling the transmit power of a data transmission to a plurality of terminals. In accordance with the method, TPC streams are received from the terminals and processed to obtain a stream of joint power control commands used to adjust the transmit power of the data transmission.

Multi-media broadcast and multicast service (MBMS) in a wireless communication system

A communication system providing telephony communication across combined circuit switched and packet switched networks, such as a telephone network and the Internet, which are connectable to terminals, such as telephones and computers, for selective communication therebetween. The communication system includes an authorization and account control object in the packet switched network, multiple gateways between the circuit switched and packet switched networks providing controlled connectivity between those networks, and an information retrieval object in the packet switched network, wherein the authorization and account control object maintains a substantially real time record of user accounts and usage, and the information and retrieval object provides substantially real time retrieval of selected information from the authorization and account control object. The retrieval object provides on demand to terminals which provide authentication for access to an identified account information regarding that account. The information regarding the account is substantially real time information including information with respect to communications in progress, which are chargeable to the account which has been authenticated. The authorization and account control object is preferably a unitary logical object having distributed instances thereof handling multitudinous accounts of widely separated terminals. The retrieval object provides isolation of the authorization and account control object permitting simultaneous multitasking by the authorization and account control object and the retrieval object respectively.

Network session management

Sensor networks are meant for sensing and disseminating information about the environment they sense. The criticality of a sensed phenomenon determines its importance to the end user. Hence, data dissemination in a sensor network should be information aware. Such information awareness is essential firstly to disseminate critical information more reliably and secondly to consume network resources proportional to the criticality of information. In this paper, we describe a protocol called RelnForM to deliver packets at desired reliability at a proportionate communication cost. RelnForm sends multiple copies of each packet along multiple paths from source to sink, such that data is delivered at the desired reliability. It uses the concept of dynamic packet state in context of sensor networks, to control the number of paths required for the desired reliability, and does so using only local knowledge of channel error rates and topology. We show that for uniform unit disk graphs, the number of edge-disjoint paths between nodes is equal to the average node degree with very high probability. RelnForm utilizes this property in its randomized forwarding mechanism which results in use of all possible paths and efficient load balancing.

ReInForM: reliable information forwarding using multiple paths in sensor networks

The theory of large deviations provides a simple unified basis for statistical mechanics, information theory and queueing theory. The objective of this paper is to use large deviation theory and the Laplace method of integration to provide an simple intuitive overview of the theory of effective bandwidth for high-speed digital networks, especially ATM networks. This includes (1) identification of the appropriate energy function, entropy function and effective bandwidth function of a source, (2) the calculus of the effective bandwidth functions, (3) bandwidth allocation and buffer management, (4) traffic descriptors, and (5) envelope processes and conjugate processes for fast simulation and bounds. >

Effective bandwidth in high-speed digital networks

Apparatus and methods for multicasting blocks of data to a plurality of receivers, the blocks including a first block and a second block, the first block comprising k1≧1 data packets, the transmission of the first block comprising an initial transmission of the k1 data packets and h1≧repair packets and one or more subsequent transmissions of additional repair packets in response to repair requests, wherein any k1 of the data packets and repair packets provide sufficient information to recover the k1 data packets; and multicasting a second block comprising k2≧1 data packets, the transmission of the second block comprising an initial transmission of the k2 data packets and h2≧repair packets, wherein any k2 of the data packets and the repair packets of the second block provide sufficient information to recover the k2 data packets of the second block and either or both k2 and h2 differ from k1 and h1, respectively. The invention provides multicast data transmission with adaptive proactive forward error correction (FEC) in combination with automatic repair requesting (ARQ), and reduces response feedback implosion in multicast over digital packet networks.

Multicast with proactive forward error correction

Quality-of-service (QOS) requirements for applications in high-speed networks are typically specified on an end-to-end basis. Mapping this end-to-end requirement to nodal requirements facilitates providing QOS guarantees and simplifies connection admission. In this paper, we evaluate strategies for local allocation of the end-to-end QOS. A QOS allocation policy is said to perform better than another when the maximum network load that it can support is greater. A major contribution of this work is the development of a nodal metric that predicts the relative performance of QOS allocation policies in a network setting. Computation of the nodal metric and direct evaluation of allocation policy performance for two simple network models yield valuable insight into the choice of allocation policies. It is found that with the packet loss probability as the QOS metric, there is little difference in the performance of allocation policies in the regime of applications with low loss requirements. From a practical viewpoint, this suggests that a simple allocation policy may be adopted in this scenario with only a small decrease in carried load with respect to an optimal policy. ----------------- This research was supported in part by the National Research Foundation under grant NCR-9116183 and the Defense Advanced Projects Research Agency under contract NAG2-595. ------------------------------------------------------

Local Allocation of End-to-End Quality-of-Service in High-Speed Networks

The authors examine the performance of two different approaches for handling the loss and/or corruption of messages as they are transmitted between two end users in a high-speed network. In the link-by-link approach, two adjacent nodes in an end-to-end path locally detect and recover from message loss or corruption along their joining link. In the end-to-end approach, recovery is done solely on the basis of a single end-to-end protocol. The authors develop analytic performance models, validated with simulation, for comparing the performance of these two approaches. The authors find that for the range of network parameters of practical interest, an end-to-end approach towards error control is superior to a link-by-link approach, even under assumptions that would overly favor the link-by-link approach, while at the same time requiring fewer network resources (e.g. buffers, computation time) than the link-by-link approach. The performance differences arise primarily from the increased buffer requirements of the link-by-link approach. >

Performance comparison of error control schemes in high-speed computer communication networks

Measurement and analysis of the network behavior are crucial to understanding the Internet performance and designing appropriate control mechanisms for better performance. End hosts and their applications, however, have a limited capability in accessing and acquiring information about the network behavior. To them, end-to-end measurement of the network behavior is usually the only available information. This thesis focuses on two fundamental measures of network performance: end-to-end packet delay and loss. 
First, we address the issue of accuracy in end-to-end delay. Raw delay measurements in the Internet contain impairments due to unsynchronized clocks between two measuring hosts. We propose a linear programming (LP) based algorithm to estimate and remove clock skew in delay measurements. We compare the LP-based algorithm with three other algorithms, and show that the LP-based algorithm is robust, and performs well over actual delay measurements and in simulation. 
Next, we consider the problem of adaptively adjusting the playout delay at the receiver of packet audio applications. We first present efficient algorithms that compute a bound on the achievable performance of any playout delay adjustment algorithm, and a new adaptive playout delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. 
We also look at the correlation between end-to-end delay and loss. We quantify the correlation as sample mean delay conditioned on loss and loss conditioned on delay, and analyze the measurements based on them. The results indicate that it is likely that the packet delay would increase in the near future if a packet loss is detected. 
Recent developments from the MINC (Multicast-based Inference of Network-internal Characteristics) project have shown that we can attain asymptotically converging estimates of link loss using Maximum Likelihood Estimators (MLE) from end-to-end multicast measurements. We validate the effectiveness of the MLE algorithm by showing that the inferred link loss estimates are close to the actual link loss inside the network. Also we study the performance of MLE estimates given a limited number of packets and under a wide range of loss rates and tree topologies. 
We conclude this dissertation with a discussion for future research.

Don Towsley

Papers

Multicast with proactive forward error correction

Local Allocation of End-to-End Quality-of-Service in High-Speed Networks

Performance comparison of error control schemes in high-speed computer communication networks

Performance comparison of error control schemes in high-speed computer communication networks

Measurement and analysis of end-to-end delay and loss in the internet