The paper considers the problem of rendering post production applications in a distributed environment. The aim is to limit the concessions that conventional, in-house methods force graphics operators to make to the quality of the end products to get them on time and within budget. A reference architecture for distributed rendering is proposed that takes advantage of the parallel properties of the application to reduce the job turnaround time. Performance issues that stimulate the industrial sector to use the environment are discussed through a simplified evaluation framework. The authors also examine how to serve multiple customers in a metacomputing environment employing limited resources. User strategies for submitting jobs over a priority-based service system that minimize the associated rendering cost are defined. Furthermore, a pricing scheme is implemented that succeeds in driving the user actions to an operating point in which certain provider objectives are met and, at the same time, discourages the users from utilizing a congested system, in which meeting of deadlines cannot be assured. Results and directions are not only restricted to the post production area, but apply over a wide range of remote operations.

A queued TDM system with a general arrival process is analyzed. Each call consists of a geometrically distributed number of cells. The interarrival times between successive calls are generally distributed independent random variables. Call applications, which cannot be immediately satisfied, are queued in a finite capacity buffer, and are forwarded according to a FIFO discipline. Waiting times, queue lengths, and throughput are the performance measures considered in this paper.>

The paper models and evaluates key design issues for the Adaptation Layer in ATM networks. The role and efficient design of the adaptation layer is crucial for future B-ISDN based on ATM. We concentrate on packet-mode adaptation services analyzing and simulating relevant protocols employed for narrowband ISDN signaling and packet data. Processing and transmission resources are modeled. In particular we investigate the performance of software based solution resident in microprocessors and interfaces to the ATM world via a generic interface component. This set-up which is currently under development is found adequate to handle narrowband ISDN signaling and data traffic. It offers a cost effective and readily available solution for the transition phase towards fully customized broadband components.

The problem of resource allocation for future integrated broadband communication networks (IBCNs) is addressed. It mainly involves resource allocation at the connection level. The resource allocation problem is decomposed into the following interdependent tasks: given that a network can accommodate the bandwidth demand of a call request, determine a route for the corresponding asynchronous transfer mode (ATM) virtual connection; and allocate bandwidth, i.e. links inside the trunks of the chosen route, to this connection according to predefined limits on bandwidth use by various service calls. Various link allocation schemes combined with routing algorithms are examined. Their performance in terms of service call blocking is evaluated using a software package developed, for that purpose. It is shown that the traditional complete sharing (CS) and complete partitioning (CP) policies are not adequate for IBCNs. Movable boundary (MB) policies are more flexible and present near-optimal performance when access of broadband service to narrowband service resources is allowed and suitable routing algorithms are dynamically applied.>

Algorithms for the control of the retransmission procedure in random multiple access schemes are needed to ensure stability of the system operation under high traffic conditions. Optimal retransmission control policies cannot be applied in practice since they are based on global information about the system state. In the case of the most well-known implementation of the CSMA/CD protocol, ANSI/IEEE Std 802.3-1985, a heuristic approach, known as the "truncated binary exponential backoff" algorithm, is employed. Such schemes are not amenable to theoretical analysis because of the state space. explosion problem of the underlying Markovian model. In this paper, a new retransmission control algorithm, suitable for CSMA/CD protocols, based on information acquired by monitoring the channel transmissions is proposed. A Markovian model is presented and the algorithm is analyzed in detail. The performance of the algorithm is evaluated, several numerical and simulation results are presented and comparisons to other protocols are made.

Delay, queue length, and throughput are the main performance characteristics of a data transmission system with ARQ (automaticrepeat-request) error control. Various protocols which have been proposed use, as a part or as a whole, the basic selective-repeat ARQ protocol. Their performance analysis has, however, been restricted to the throughput characteristics. An easily applicable method to analyze this basic protocol has not appeared in the literature so far. In this paper, two different methods for the queue length and delay analysis are presented. The system is modeled as a discrete time queue with infinite buffer storage. Transmission errors are considered to be independent, and block arrivals may follow an arbitrary interarrival time distribution. The first method uses an exact Markov state model, on which the theory of absorbing and ergodic Markov chains is applied, and leads to a computational algorithm. The second method, which is based on a specific assumption, uses a substantially simpler stochastic model and results in equations which are easily solved by means of iterative computation. In the case of geometrically distributed interarrival times, simple analytical formulas are extracted. The results are compared to the exact ones (that is, those obtained by the first method or by extended simulation runs) and surprising agreement is observed.

In this paper, we propose a new queueing model appropriate for the analysis of a buffered CSMA/CD protocol. We assume that each user has a finite buffer capacity. The system, in general, can be modeled as a multidimensional semi-Markov chain. The enormous state space for the complete description is reduced by appropriate exploitation of the features of the system, e.g., fairness towards all users, dependence of the probability of successful transmission only on the total number of busy users, etc. Using a further assumption we construct a two-dimensional semi-Markov chain, examining the state of the system immediately after the successful transmission of packets. The obtained solution is extremely accurate and exhibits excellent agreement with simulation results, which corroborates the correctness of our method.

A queueing model appropriate for the analysis of multipleaccess schemes with finite user buffer capacity is proposed. This model can be applied to several slotted random multiple-access schemes, such as the URN, ALOHA, random TDMA channels, etc. The examined system, in general, can be modeled as a multidimensional Markov chain. The enormous state space for its complete description makes numerical analysis intractable. Nevertheless, symmetry properties of the system can be exploited so that the state space will be reduced. In particular, the queue length at one station and the number of busy stations are used to characterize the behavior of the system. In this way, a two-dimensional Markov chain can be constructed and an approximate solution is obtained which is in excellent agreement with simulation results.

The purpose of this paper is to give reduced complexity algorithms in order to compute linear models in situations of non-stationnary signals. The case of impulse response identification is examined as well as the case of pure AR-modeling. Both problems lead to a linear system which must be solved to give the unknown AR-model or the unknown impulse response. Furthermore the linear systems that occur in both cases have an identical left hand side member characterized by a symmetric matrix having the supplementary property of being the product of two non-symmetric Toeplitz matrices. The "near to Toeplitz" structure of this matrix permits the fast solution of the system with specially adapted algorithms to this situation. Many properties of the near to Toeplitz matrices are outlined as the key formulas for the derivation of all fast schemes. Order, time and order-time recursive methods are derived using a unified theoretical framework. The methodology used here is similar to that used by Morf et al. The algorithms presented are new and can be converted to efficient and automated computer packages.

This paper considers the use of time compression multiplexing (TCM) for telephone loop communications. The analysis treats TCM transmission degradations in general, and applies the results to telephone speech transmission. To gain an understanding of the nature of TCM degradations, transmission through a linear, time invariant network and demultiplexing is considered. The channel signals are then assumed to be wide-sense stationary random processes, and expressions for the spectral properties of the TCM distortion and interchannel crosstalk are determined. To accurately evaluate the performance of a TCM system, a practical and economical equalizer is proposed. It is designed to meet speech transmission requirements for the telephone loop plant. A worst case application is chosen to evaluate the system performance. The evaluation of the TCM transmission performance utilizes both the random signal model and a computer simulation using digitized speech. The random signal analysis shows that the spectra of the distortion and crosstalk are significant for human speech. Results of the digitized speech simulation demonstrate that the subjective effect of the signal distortion is not severe, but the crosstalk is somewhat intelligible and more severe than expected.