Showing papers in "IEEE Journal on Selected Areas in Communications in 1991"
IBM1
TL;DR: The authors propose a computationally simple approximate expression to provide a unified metric to represent the effective bandwidth used by connections and the corresponding effective load of network links, which can then be used for efficient bandwidth management, routing, and call control procedures aimed at optimizing network usage.
Abstract: The authors propose a computationally simple approximate expression for the equivalent capacity or bandwidth requirement of both individual and multiplexed connections, based on their statistical characteristics and the desired grade-of-service (GOS). The purpose of such an expression is to provide a unified metric to represent the effective bandwidth used by connections and the corresponding effective load of network links. These link metrics can then be used for efficient bandwidth management, routing, and call control procedures aimed at optimizing network usage. While the methodology proposed can provide an exact approach to the computation of the equivalent capacity, the associated complexity makes it infeasible for real-time network traffic control applications. Hence, an approximation is required. The validity of the approximation developed is verified by comparison to both exact computations and simulation results. >
1,442 citations
TL;DR: Autonet as mentioned in this paper is a self-configuring local area network composed of switches interconnected by 100 Mb/s, full-duplex, point-to-point links, which uses cut-through to achieve a packet forwarding latency as low as 2 ms/switch.
Abstract: Autonet is a self-configuring local area network composed of switches interconnected by 100 Mb/s, full-duplex, point-to-point links. The switches contain 12 ports that are internally connected by a full crossbar. Switches use cut-through to achieve a packet forwarding latency as low as 2 ms/switch. Any switch port can be cabled to any other switch port or to a host network controller. A processor in each switch monitors the network's physical configuration. A distributed algorithm running on the switch processor computes the routes packets are to follow and fills in the packet forwarding table in each switch. With Autonet, distinct paths through the set of network links can carry packets in parallel, allowing many pairs of hosts to communicate simultaneously at full link bandwidth. A 30-switch network with more than 100 hosts has been the service network for Digital's Systems Research Center since February 1990. >
646 citations
TL;DR: The authors present the architecture of a general-purpose broadband-ISDN (B-IS DN) switch chip and its novel feature: the weighted round-robin cell (packet) multiplexing algorithm and its implementation in hardware, thus proving the feasibility of the architecture.
Abstract: The authors present the architecture of a general-purpose broadband-ISDN (B-ISDN) switch chip and, in particular, its novel feature: the weighted round-robin cell (packet) multiplexing algorithm and its implementation in hardware. The flow control and buffer management strategies that allow the chip to operate at top performance under congestion are given, and the reason why this multiplexing scheme should be used under those circumstances is explained. The chip architecture and how the key choices were made are discussed. The statistical performance of the switch is analyzed. The critical parts of the chip have been laid out and simulated, thus proving the feasibility of the architecture. Chip sizes of four to ten links with link throughput of 0.5 to 1 Gb/s and with about 1000 virtual circuits per switch have been realized. The results of simulations of the chip are presented. >
561 citations
TL;DR: A discrete multitone (DMT) transceiver design for high bit rate digital subscriber line (HDSL) access is presented and analyzed and is an excellent candidate for HDSL implementation.
Abstract: A discrete multitone (DMT) transceiver design for high bit rate digital subscriber line (HDSL) access is presented and analyzed. The DMT transmitter and receiver structure and algorithms are detailed, and the computational requirements of DMT for HDSL are estimated. At a sampling rate of 640 kHz, using an appropriate combination of a short finite-impulse-response (FIR) equalizer and a length-512 DMT system, 1.6 Mb/s data transmission is possible within the carrier serving area (CSA) at an error rate of 10/sup -7/ on a single twisted pair. A significant performance margin can be achieved when two coordinated twisted pairs are used to deliver a total data rate of 1.6 Mb/s. In terms of a performance-per-computation figure of merit, the DMT system is an excellent candidate for HDSL implementation. >
549 citations
TL;DR: Basic design objectives and requirements for a policing or usage parameter control function are described, which serve as a basis for the comparison of some of the mechanisms proposed so far, namely the leaky bucket, the jumping window, the triggered jumpingwindow, the moving window, and the exponentially weighted moving average mechanisms.
Abstract: Asynchronous transfer mode (ATM) networks, as proposed by CCITT as the solution for the future broadband ISDN, will provide high flexibility with respect to the varying bandwidth requirements of the different services. They will also support variable bit rates within a connection. The packetized information transfer, without flow control between the user and the network, in combination with the asynchronous multiplexing principle, results in a need to control the individual cell stream during the entire duration of the calls to ensure an acceptable quality of service for all coexisting calls sharing the same network resources. This kind of control will be provided by introducing a policing or usage parameter control function. Basic design objectives and requirements for such a function are described. These requirements serve as a basis for the comparison of some of the mechanisms proposed so far, namely the leaky bucket, the jumping window, the triggered jumping window, the moving window, and the exponentially weighted moving average mechanisms. >
423 citations
TL;DR: The results suggest that the transmission capacity not used by the small window session will be approximately fairly divided among the large window sessions, and the worst-case performance of round-robin scheduling with windows is shown to approach limits that are perfectly fair in the max-min sense.
Abstract: The author studies a simple strategy, proposed independently by E.L. Hahne and R.G. Gallager (1986) and M.G.H. Katevenis (1987), for fairly allocating link capacity in a point-to-point packet network with virtual circuit routing. Each link offers its packet transmission slots to its user sessions by polling them in round-robin order. In addition, window flow control is used to prevent excessive packet queues at the network nodes. As the window size increases, the session throughput rates are shown to approach limits that are perfectly fair in the max-min sense. If each session has periodic input (perhaps with jitter) or has such heavy demand that packets are always waiting to enter the network, then a finite window size suffices to produce perfectly fair throughput rates. The results suggest that the transmission capacity not used by the small window session will be approximately fairly divided among the large window sessions. The focus is on the worst-case performance of round-robin scheduling with windows. >
337 citations
TL;DR: The authors examine the phenomenon of congestion to better understand the congestion management techniques that will be needed in high-speed, cell-based networks.
Abstract: The authors examine the phenomenon of congestion to better understand the congestion management techniques that will be needed in high-speed, cell-based networks. The first step of this study is to use high time-resolution local area network (LAN) traffic data to explore the nature of LAN traffic variability. Then the data are used for a trace-driven simulation of a connectionless service that provides LAN interconnection. The simulation allows one to characterize what congestion might look like in a high-speed, cell-based network. When realistic data are applied to simple models of LAN interconnection, it is observed that during periods, congestion persists and losses can be significant; congestion losses cannot be avoided by modest increases in buffer capacity; consequences of misengineering can be serious; and, fortunately, most congested periods are preceded by signs of impending danger. >
330 citations
TL;DR: The author proposes to characterize the burstiness of packet-arrival processes with indexes of dispersion for intervals and counts, which are relatively straightforward and can estimate and convey much more information than simpler indexes that are often used to describe burstiness quantitatively.
Abstract: The author proposes to characterize the burstiness of packet-arrival processes with indexes of dispersion for intervals and counts. These indexes, which are functions of the variance of intervals and counts, are relatively straightforward and can estimate and convey much more information than simpler indexes (such as the coefficient of variation) that are often used to describe burstiness quantitatively. The author defines and evaluates the indexes of dispersion for some of the simple analytical models that are frequently used to represent highly variable processes. The indexes for a number of measured point processes that were generated by workstations communicating to file servers over a local area network are also estimated. It is shown that nonstationary components in the measured packet-arrival data distort the shape of the indexes, and ways to handle nonstationary data are proposed. To show how to incorporate measures of variability into analytical models and to offer an example of how to model the measured packet-arrival processes, the author describes a fitting procedure based on the index of dispersion for counts for the Markov-modulated Poisson process. >
257 citations
TL;DR: It is concluded that while the leaky bucket can easily control the peak rate, difficulties arise in controlling the mean bit rate and the burst duration (because of the poor selectivity).
Abstract: The performance of the leaky bucket, a device for controlling the source traffic parameters of an ATM network is analyzed. The most critical situation, i.e., an on/off bursty source, is assumed. Moreover, the delay jitter introduced by a possible multiplexing of sources inside the customer premises network is not considered. The leaky bucket can be analyzed as a G/D/1/N queue with finite waiting room N and a suitable arrival process. An alternative approach, the fluid flow approximation, in which the bit flow is considered as a continuous variable, is presented. By comparison with the exact model, it is shown that the accuracy of the fluid flow approach is sufficient for practical purposes. An explicit formula for the cell loss probability, in case of exponential distributions for both the burst duration and the inactivity period of the source, is derived. On the basis of that formula, some numerical evaluations are presented with the aim of assessing the effectiveness of the leaky bucket mechanism. It is concluded that while the leaky bucket can easily control the peak rate, difficulties arise in controlling the mean bit rate (because of the long time required) and the burst duration (because of the poor selectivity). >
250 citations
TL;DR: It is specified that the provision of a synchronization function be performed within a packet switched network, and, accordingly, a two-level communication architecture is presented.
Abstract: Protocols to provide synchronization of data elements with arbitrary temporal relationships of both stream and non-stream broadband traffic types are proposed. It is specified that the provision of a synchronization function be performed within a packet switched network, and, accordingly, a two-level communication architecture is presented. The lower level, called the network synchronization protocol (NSP), provides the ability to establish and maintain individual connections with specified synchronization characteristics. The upper level, the application synchronization protocol (ASP), supports an integrated synchronization service for multimedia applications. The ASP identifies the temporal relationships among an application's data objects and manages the synchronization of arriving data for playout. The proposed NSP and ASP are mapped to the session and application layers of the open-systems-interconnection (OSI) reference model, respectively. >
247 citations
TL;DR: A comparative performance study is given, indicating the excellent performance characteristics of a simple buffer management scheme called partial buffer sharing, and the introduction of a second bearer capability provides a 10/sup -6/ cell loss rate instead of 10/Sup -10/.
Abstract: Various space priority mechanisms and their detailed performance evaluation are described. A comparative performance study is given, indicating the excellent performance characteristics of a simple buffer management scheme called partial buffer sharing. The introduction of a second bearer capability provides a 10/sup -6/ cell loss rate instead of 10/sup -10/. >
TL;DR: The authors present a high-performance self-routing packet switch architecture that can support a wide range of services having diverse performance objectives and traffic characteristics and achieves high performance by utilizing both internal and output queuing techniques within a single architecture.
Abstract: The authors present a high-performance self-routing packet switch architecture, called Sunshine, that can support a wide range of services having diverse performance objectives and traffic characteristics. Sunshine is based on Batcher-banyan networks and achieves high performance by utilizing both internal and output queuing techniques within a single architecture. This queuing strategy results in an extremely robust and efficient architecture suitable for a wide range of services. An enhanced architecture allowing the bandwidth from an arbitrary set of transmission links to be aggregated into trunk groups to create high bandwidth pipes is also presented. Trunk groups appear as a single logical port on the switch and can be used to increase the efficiency of the switch in an extremely bursty environment or to increase the access bandwidth for selected high-bandwidth terminations. Simulation results are presented. >
TL;DR: An efficient real-time scheduling algorithm is introduced that substantially increases the schedulable region without incurring prohibitive complexity costs and is compared with the ones generated by the static priority scheduling algorithm and a variant of the minimum laxity threshold algorithm.
Abstract: Whether or not the introduction of traffic classes improves upon the performance of ATM networks is discussed within the framework provided by a class of networks that guarantees quality of service. To provide a meaningful comparison the authors define the concept of a schedulable region, a region in the space of loads for which the quality of service is guaranteed. The authors show the dependence of the schedulable region on the scheduling algorithm employed, quality of service parameters, and traffic statistics. An efficient real-time scheduling algorithm is introduced that substantially increases the schedulable region without incurring prohibitive complexity costs. The schedulable region associated with this algorithm is compared with the ones generated by the static priority scheduling algorithm and a variant of the minimum laxity threshold algorithm. The size and shape of the schedulable region is explored by means of simulations. >
TL;DR: Reduced base model construction methods for stochastic activity networks are discussed and examples which illustrate the method and demonstrate its effectiveness in reducing the size of a state space are presented.
Abstract: Reduced base model construction methods for stochastic activity networks are discussed. The basic definitions concerning stochastic networks are reviewed and the types of variables used in the construction process are defined. These variables can be used to estimate both transient and steady-state system characteristics. The construction operations used and theorems stating the validity of the method are presented. A procedure for generating the reduced base model stochastic process for a given stochastic activity network and performance variable is presented. Some examples which illustrate the method and demonstrate its effectiveness in reducing the size of a state space are presented. >
TL;DR: A congestion management strategy for integrated services packet networks that is robust with regard to transmission speed and network size is proposed and is further developed to incorporate several frame sizes into the strategy, thereby providing flexibility in meeting throughput and delay requirements of different applications.
Abstract: A congestion management strategy for integrated services packet networks that is robust with regard to transmission speed and network size is proposed. The strategy supports several classes of services with zero loss and different delay bounds as well as services without stringent loss and delay guarantees. Loss-free and bounded-delay transmission is accomplished by means of an admission policy which ensure smoothness of the traffic at the network edge, and by a service discipline called stop-and-go queuing, which maintains the traffic smoothness throughout the network. Both the admission policy and the stop-and-go queuing are based on a time framing concept described elsewhere by the author (IEEE Trans. Commun., vol.39, Dec.1991). This concept is further developed to incorporate several frame sizes into the strategy, thereby providing flexibility in meeting throughput and delay requirements of different applications. Stop-and-go queueing is realizable with minor modification to a first-in first-out (FIFO) queueing structure. >
TL;DR: The authors present techniques for breaking the bottleneck caused by delays in the feedback loop, which can severely limit the maximum data rate of a detector, operating with nonlinear cancellation (NLC).
Abstract: The authors present techniques for breaking the bottleneck caused by delays in the feedback loop, which can severely limit the maximum data rate of a detector, operating with nonlinear cancellation (NLC). The authors first show how to simplify the loop to avoid high-speed switching of analog signals by using multiple decision elements, each with a different threshold level. The authors then show how to use lookahead computation to increase the delay permissible in the feedback loop. The authors describe nonlinear cancellation, and outline some of the problems of implementing NLC at high data rates. Techniques for overcoming these problems are developed and are illustrated with the example of a simple one-tap NLC. The authors describe how to generalize the techniques to more complex NLCs and present some more examples. >
TL;DR: The performance of an asynchronous transfer mode multiplexer whose input consists of the superposition of a multiplicity of homogeneous on-off sources modeled by a two-state Markovian process is studied and a new matching procedure that leads to accurate results compared to simulation is developed.
Abstract: The performance of an asynchronous transfer mode (ATM) multiplexer whose input consists of the superposition of a multiplicity of homogeneous on-off sources modeled by a two-state Markovian process is studied. The approach is based on the approximation of the actual input process by means of a suitably chosen two-state Markov modulated Poisson process (MMPP), as a simple and effective choice for the representation of superposition arrival streams. To evaluate the cell loss performance, a new matching procedure that leads to accurate results compared to simulation is developed. The application limits of the proposed method are also discussed. The outstanding physical meaning of this procedure permits a deep insight into the multiplexer performance behavior as the source parameters and the multiplexer buffer size are varied. >
TL;DR: A tutorial on the physical environment in which high bit rate digital subscriber line (HDSL) transceivers will have to evolve and succeed is presented, with special attention given to the most damaging impairments that are encountered in subscriber lines.
Abstract: The author presents a tutorial on the physical environment in which high bit rate digital subscriber line (HDSL) transceivers will have to evolve and succeed. Special attention is given to the most damaging impairments that are encountered in subscriber lines, such as propagation loss, linear distortion, crosstalk, bridged taps, and impulse noise. Somewhat less important impairments, such as change of gauge, temperature variation, and thermal noise, are also briefly described. The author concludes with a discussion of the capacity of a twisted-pair channel in a crosstalk-dominated environment. >
TL;DR: The authors study the performance of a multichannel modulation method for asymmetric digital subscriber lines (ADSLs) and very high-speed digital subscriber Lines (VHDSLs) to find out whether data rates in excess of 100 Mb/s can be transmitted reliably.
Abstract: The authors study the performance of a multichannel modulation method for asymmetric digital subscriber lines (ADSLs) and very high-speed digital subscriber lines (VHDSLs). In the ADSL case, over all unloaded North American subscriber lines in the test set, a unidirectional 1.536 Mb/s data rate service from the end office to the customer premises is possible on a single twisted pair at an error rate of 10/sup -7/ with at least a 6 dB margin used coded multichannel modulation with sufficient transmit power. In the VHDSL case, data rates in excess of 100 Mb/s can be transmitted reliably, at an error rate of 10/sup -7/, using uncoded multichannel modulation on a single twisted pair over a distance >
TL;DR: In this paper, the authors analyzed the burst component of the considered superposition queuing process in a variable-bit-rate constant-output-rate (COR) fluid system.
Abstract: When variable-bit-rate sources are multiplexed in an asynchronous transfer mode (ATM) network, there arise queues with a particular form of correlated arrival process. Such queues are analyzed by exploiting a result expressing the distribution of work in system of the G/G/1 queue originally derived by V.E. Benes (1963). A simple alternative demonstration of this result is analyzed and extended to the case of fluid input systems. The result is applied first to a queue where the arrival process is a superposition of periodic sources (the Sigma D/sub i//D/1 queue), and then to a variable-input-rate constant-output-rate fluid system. The latter is shown to model the so-called burst component of the considered superposition queuing process. The difference between this and the real queue, the cell component, can be evaluated by means of the results obtained for the Sigma D/sub i//D/1 queue. The relative importance of these two components is explored with reference to the particular case of a superposition of on/off sources. >
TL;DR: Various priority queueing strategies characterized mainly by different degrees of resource sharing and a general system model for performance evaluation are introduced and performance comparisons and design tradeoffs are addressed.
Abstract: The problem of furnishing an asynchronous transfer mode (ATM) based broadband-ISDN (B-ISDN) with two bearer services supporting different grades of transfer quality is addressed. The focus is on priority bandwidth and buffer management in the ATM communications nodes (switches, multiplexers or concentrators, and expanders) in the context of a multichannel network architecture. Detailed queueing analyses and simulations and results are provided to evaluate the differentiation between traffic classes that can be achieved by different strategies. The implementation complexity of the different schemes is discussed. Various priority queueing strategies characterized mainly by different degrees of resource sharing and a general system model for performance evaluation are introduced. Performance comparisons and design tradeoffs are addressed. >
TL;DR: The authors present dynamic call admission control using the distribution of the number of cells arriving during the fixed interval, which tolerates loose bandwidth enforcement and loose policing control, and dispenses with modeling of the arrival processes.
Abstract: The authors present dynamic call admission control using the distribution of the number of cells arriving during the fixed interval. This distribution is estimated from the measured number of cells arriving at the output buffer during the fixed interval and traffic parameters specified by users. Call acceptance is decided on the basis of online evaluation of the upper bound of cell loss probability, derived from the estimated distribution of the number of calls arriving. QOS (quality of service) standards can be guaranteed using this control when there is no estimation error. The control mechanism is effective when the number of call classes is large. It tolerates loose bandwidth enforcement and loose policing control, and dispenses with modeling of the arrival processes. Numerical examples demonstrate the effectiveness of this control, and implementation is also discussed. >
TL;DR: A method of characterizing video codec sources in asynchronous transfer mode (ATM) networks as an autoregressive moving average (ARMA) process is described and it is shown that the multiplexed stream of video cells is also an ARMA process.
Abstract: A method of characterizing video codec sources in asynchronous transfer mode (ATM) networks as an autoregressive moving average (ARMA) process is described. Measurements of long-term mean and the autocorrelation function of cell interarrival times allow the parameter estimation of the ARMA model. The video source is then described by ARMA model. Furthermore, it is shown that the multiplexed stream of video cells is also an ARMA process. Such a cell stream is then applied to a model of a queuing system to obtain performance measures of the system. Perturbation analysis is then performed on the functional behavior of the queuing system by appropriate perturbation of the model parameters to determine cell waiting time sensitivity due to slight variations of the input process. >
TL;DR: The effect of transmission delay on speech quality in telecommunications is described, with human factors such as conversational mode and the talker's knowledge of the cause of delay taken into account.
Abstract: The effect of transmission delay on speech quality in telecommunications is described, with human factors such as conversational mode and the talker's knowledge of the cause of delay taken into account. Objective quality estimation methods for delay effects are proposed, and these methods are applied in an actual communications network. In connection with delay perception in a telephone conversation, the assumption was verified that a talker expects a particular response time from a partner, and that delay that is outside this expectation time window is noticed. Taking this information into account, a subjective conversational experiment is controlled by six kinds of tasks by varying the temporal characteristics. Thus, a subjective assessment of delay effects is obtained by laboratory tests in relation to the detectability threshold, opinion rating, and conversational efficiency. Objective quality measures for each test were defined as a linear combination of temporal parameters that correspond closely to subjective qualities. >
TL;DR: A modification to the basic structure of the tandem banyan switching fabric is proposed, which decreases the hardware complexity of the switch while maintaining its performance.
Abstract: The authors propose a new space-division fast packet switch architecture based on banyan interconnection networks, called the tandem banyan switching fabric (TBSF). It consists of placing banyan networks in tandem, offering multiple paths from each input to each output, thus overcoming in a very simple way the effect of conflicts among packets (to which banyan networks are prone) and achieving output buffering. From a hardware implementation perspective, this architecture is simple in that it consists of several instances of only two VLSI chips, one implementing the banyan network and the other implementing the output buffer function. The basic structure and operation of the tandem banyan switching fabric are described, and its performance is discussed. The authors propose a modification to the basic structure which decreases the hardware complexity of the switch while maintaining its performance. An implementation of the banyan network using a high-performance BiCMOS sea-of-gates on 0.8- mu m technology is reported. >
TL;DR: A multicast function and a 600 Mb/s link switch architecture, which are suitable for ATM network systems connecting various media, and an expansion method using the 32*32 switching board to achieve large-scale switching systems such as 256*256 or 1024*1024 switches are discussed.
Abstract: A set of 0.8 mu m CMOS VLSIs developed for shared buffer switches in asynchronous transfer mode (ATM) switching systems is described. A 32*32 unit switch consists of eight buffer memory VLSIs, two memory control VLSIs, and two commercially available first in first out (FIFO) memory LSIs. Using the VLSIs, the switch can be mounted on a printed board. To provide excellent traffic characteristics not only under random traffic conditions but also under burst traffic conditions, this switch has a 2-Mb shared buffer memory, the largest reported to date. which can save 4096 cells among 32 output ports. This switch has a priority control function to meet the different cell loss rate requirements and switching delay requirements of different service classes. A multicast function and a 600 Mb/s link switch architecture, which are suitable for ATM network systems connecting various media, and an expansion method using the 32*32 switching board to achieve large-scale switching systems such as 256*256 or 1024*1024 switches are discussed. >
TL;DR: The main conclusion is that both the MMPP model and the fluid flow approximation can provide accurate loss predictions for parameter ranges of practical interest.
Abstract: Three different approximation techniques are examined. The performance models studied differ primarily in the manner in which the superposition of the voice sources (i.e., the arrival process) is modeled. The first approach models the superimposed voice sources as a renewal process, and performance calculations are based only on the first two moments of the renewal process. The second approach is based on modeling the superimposed voice sources as a Markov modulated Poisson process (MMPP). The choice of parameters for the MMPP attempts to capture aspects of the arrival process in a more intuitive manner than previously proposed approaches for determining the MMPP parameters and is shown to compute loss more accurately. Finally, a fluid flow approximation for computing packet loss is evaluated. For all three approaches, a unifying example, the case of multiplexing voice sources over a T1-rate link is considered. The main conclusion is that both the MMPP model and the fluid flow approximation can provide accurate loss predictions for parameter ranges of practical interest. >
TL;DR: A new cell loss recovery method to be applied to virtual paths (VP's) of ATM networks and its advantages over conventional methods on virtual circuits (VC's) include reduced coding/decoding delays and facility sharing.
Abstract: A cell loss recovery method to be applied to the virtual paths (VPs) of asynchronous transfer mode (ATM) networks is introduced. The method uses forward error correction (FEC). Its advantages over conventional methods on virtual circuits (VCs) include reduced coding/decoding delays and facility sharing. In addition to consecutive cell loss compensation, another feature of this method is that cell loss compensation facilities are only required at VP terminating nodes. When a cell is discarded due to buffer overflow, successive cells also tend to be discarded. Performance estimations show an outstanding reduction in cell loss rate. >
TL;DR: A signaling protocol for user applications that invokes multimedia services provided by the EXPANSE research prototype, a broadband ISDN testbed, and a generic framework that flexibly accommodates a wide variety of services involving multiple parties with heterogeneous terminals is described.
Abstract: The author introduces a signaling protocol for user applications that invokes multimedia services provided by the EXPANSE research prototype, a broadband ISDN testbed A generic framework that flexibly accommodates a wide variety of services involving multiple parties with heterogeneous terminals is described The author describes the generic services supported, introduces the EXPANSE call model, discusses syntax and message flow procedures, and relates EXPANSE signaling to the ISDN signaling control part (ISCP) and long-term broadband ISDN signaling work currently under study by CCITT >
TL;DR: Numerical examples show how the traffic characteristics of the designated traffic stream are affected by the total traffic load at the switching node, the number of other bursty traffic sources, and the degree of burstiness of the traffic source itself.
Abstract: Admission control in asynchronous transfer mode (ATM) networks is considered. An ATM switch is modeled as a discrete-time single-server queue at which the following three different kinds of arrival processes are allowed to join together: arrivals of cells with a general interarrival time distribution; Bernoulli arrivals of cells in batches; and interrupted Poisson processes. An exact analysis is given to derive the waiting-time distributions and interdeparture-time distributions for arriving cells subject to admission control in ATM networks. The model is extended to approximately obtain the end-to-end delay distributions for the designated traffic stream. Such an analysis is important for voice or coded video because they require a playout mechanism at the destination. Since the analysis is approximate, the simulation results needed to assess its accuracy are provided. Numerical examples show how the traffic characteristics of the designated traffic stream are affected by the total traffic load at the switching node, the number of other bursty traffic sources, and the degree of burstiness of the traffic source itself. >