LSR (Label Switching Router)s in MPLS (Multiprotocol Label Switching) networks map arriving IP flows into some labels on Layer 2 switching fabric and establish LSP (Label Switching Path)s. By using LSPs, LSRs not only transmit IP datagrams fast by cut-through mechanism, but also solve traffic engineering issue to optimize the delay of some IP datagram flows. So far, we have analyzed the performance of LSR focusing only on the maximum number of LSPs which can be set on Layer 2. In this paper, we will also consider the bandwidth allocated to each LSP and analyze the IP datagram transmission delay and the cut-through rate of LSR. We suppose the label mapping method as the data-driven scheme in the analytical model, so that the physical bandwidth of LSR is shared by both the default LSP for hop-by-hop transmission and the cut-through LSPs. Thus, we will investigate the impact of the bandwidth allocation among these LSPs on the performance.

In this paper we propose a close-loop queueing model of MPLS switch under different label-setup and release policies, supporting both traffic-driven and topology-driven connection setup procedures. This model can emulate the behavior of TCP under the MPLS switch when the maximum window size is sustained and the packet loss rate is negligible. From the proposed flow-based MPLS switch model, one can clearly observe the competition of multiple IP flow for limited number of labels, and how the label-setup policy and the label-release policy affect the system performance. We find that Norton's theorem can be applied to solve this sophisticated queueing model. Therefore, with very limited computational complexity with respect to the number of IP flows or labels, the proposed mathematical model and the approximation of label competition can be used to obtain the desired performance metrics, such as the throughput, the label-setup rate, and the channel utilization. Finally, the trade-off among performance metrics can be observed as well.

Wireless LANs have been used for realizing fully-distributed users in a multimedia environment that has the ability to provide real-time bursty traffic (such as voice or video) and data traffic. In this paper, we present a new realistic and detailed system model and a new effective analysis for the performance of wireless LANs which support multimedia communication with non-persistent carrier sense multiple access with collision avoidance (CSMA/CA) protocol. In this CSMA/CA model, a user with a packet ready to transmit initially sends some pulse signals with random intervals within a collision avoidance period before transmitting the packet to verify a clear channel. The system model consists of a finite number of users to efficiently share a common channel. Each user can be a source of both voice traffic and data traffic. The time axis is slotted, and a frame has a large number of slots and includes two parts: the collision avoidance period and the packet transmission period. A discrete-time Markov process is used to model the system operation. The number of slots in a frame can be arbitrary, dependent on the chosen lengths of the collision avoidance period and packet transmission period. Numerical results are shown in terms of channel utilization and average packet delay for different packet generation rates. They indicate that the network performance can be improved by adequate choice of ratios between the collision avoidance period and transmission period, and the pulse transmission probability.

To satisfy huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes two different procedures of multi-channel multiple access schemes with the slotted ALOHA operation for both data and voice traffic and presents an exact analysis to numerically evaluate the performance of the systems. In scheme I, there is no limitation on access between data transmissions and voice transmissions, i.e., all channels can be accessed by all transmissions. In scheme II, a channel reservation policy is applied, where a number of channels are used exclusively for voice packets while the remaining channels are used for both data packets and voice packets. We call the system using scheme I "Non-reservation system" and call the system using scheme II "Reservation system. " Performance characteristics we obtained include loss probability for voice traffic, average packet delay for data traffic and channel utilization for both traffic. The performance of the schemes and the effects of the design parameters are numerically evaluated and compared to a wide-bandwidth conventional single-channel slotted ALOHA system with single data traffic. The analysis presented in this paper will be not only useful for the performance evaluation and the optimum design of multi-channel multi-traffic systems in wireless environments, but also applicable to evaluate other performance measures in priority networks, cellular mobile radio networks or multi-hop wireless networks.

A new carrier based dynamic channel assignment for FDMA/TDMA cellular systems, called borrowing with directional carrier locking strategy, is proposed in this paper. When a call arrives at a cell and finds all voice channels busy, a carrier which consists of multiple voice channels can be borrowed from its neighboring cells for carrying the new call if such borrowing will not violate the cochannel interference constraint. Two analytical models, cell group decoupling analysis and phantom cell analysis, are constructed for evaluating the performance of the proposed strategy. Using cell group decoupling (CGD) analysis, a cell is decoupled together with its neigbors from the rest of the network for finding its call blocking probability. Unlike conventional approaches, decoupling enables the analysis to be confined to a local/small problem size and thus efficient solution can be found. For a planar cellular system with three-cell channel reuse pattern, using CGD analysis involves solving of seven-dimenional Markov chains. It becomes less efficient as the number of carriers assigned to each cell increases. To tackle this, we adopt the phantom cell analysis which can simplify the seven-dimensional Markov chain to two three-dimentional Markov chains. Using phantom cell analysis for finding the call blocking probability of a cell, two phantom cells are used to represent its six neighbors. Based on extensive numerical results, we show that the proposed strategy is very efficient in sharing resources among base stations. For low to medium traffic loads and small number of voice channels per carrier, we show that both analytical models provide accurate prediction on the system call blocking probability.

Fourier analysis of sinusoidal and/or quasi-periodic signals in additive noise has been used in various fields. So far, many analysis algorithms including the well-known DFT have been developed. In particular, many adaptive algorithms have been proposed to handle non-stationary signals whose discrete Fourier coefficients (DFCs) are time-varying. Notch Fourier Transform (NFT) and Constrained Notch Fourier Transform(CNFT) proposed by Tadokoro et al. and Kilani et al., respectively, are two of them, which are implemented by filter banks and estimate the DFCs via simple sliding algorithms of their own. This paper presents, for the first time, statistical performance analyses of the NFT and the CNFT. Estimation biases and mean square errors (MSEs) of their sliding algorithms will be derived in closed form. As a result, it is revealed that both algorithms are unbiased, and their estimation MSEs are related to the signal frequencies, the additive noise variance and orders of comb filters used in their filter banks. Extensive simulations are performed to confirm the analytical findings.

A closed-loop queueing model of flow-based label switches, supporting label reservation protocols of different label-setup and release policies, is presented. This model can emulate the behavior of TCP under the label switch when the maximum window size has been achieved and the packet loss rate is negligible. The label-setup policy is that the IP controller does not start to set up a label until the accumulated packets of the same flow in the switch buffer have exceeded a triggering threshold. Meanwhile, the reserved bandwidth is released when the flow is detected idle and the label-release timer has expired. This policy can achieve higher channel utilization with minimal label processing overhead in spite of suffering from certain delay penalty. To avoid unnecessary TCP timeout or large packet delay under such policy, we also introduce a label-setup timer. Norton's theorem is applied to obtain approximate solutions of this queueing model. Although the analytical method is an approximate one, the simulation results show that the accuracy is high and this model can clearly illustrate how the label-setup and the lable-release timer affect the system performance. Besides, one can observe the trade-off between the throughput and the channel utilization.

In this paper, a token-controlled network with exhaustive service strategy is analyzed. The mean and variance of service time of a station, and the mean token rotation time on the network are derived under the condition that the buffer capacity of each station is individually finite. For analysis, an extended stochastic Petri-net model of a station is presented. Then, by analyzing the model, the mean service time of a station and the mean token rotation time are derived, as functions of the given network parameters such as the total number of stations on the network, the arrival rate of frames, the transmission rate of frames, and the buffer capacity. The variance of service time of a station is also derived. By examining derived results, it is shown that they exactly describe the actual operations of the network. In addition, computer simulations with sufficient confidence intervals help to validate the results.

This paper evaluates the performance of a reliable multicast protocol for bulk-data transfer over unreliable networks via IP-multicast. Bulk-data type reliable multicast appears promising for commercial publishing and large-scale data replication. The proposed reliable multicast transport protocol (RMTP) provides high-performance due to the use of IP multicast while also providing confirmed and error free transfer by end-to-end controls. The protocol includes a multi-round selective repeat scheme dedicated for bulk-data multicast applications. This paper examines the multicast retransmission procedures in RMTP through analysis and tests on an implemented system and clarifies the basic performance behavior of the protocol. Evaluations are conducted with regard to retransmission redundancy, transfer time, and packet processing load with various error conditions and number of receivers. Against the response concentration problem seen in end-to-end communication, the backoff time algorithm is applied to the protocol; the limits it places on system scalability are clarified.

The optimistic analytical results for performance analysis of buffered banyan networks are mainly due to certain independence assumptions used for simplifying analysis. To capture more effects of cell correlation, a refined analytical model for both single-buffered and multiple buffered banyan networks is proposed in this paper. When cell output contention occurs at a 2 2 switch element, two contention resolution schemes are used. One is based on randomly choosing the winning cell and another is to give priority to the cell which has been delayed in the current buffer for at least one stage cycle. The switch throughput, cell transfer delay and cell delay deviation for single-buffered banyan networks with and without using priority scheme are derived. Then the model is generalized to multiple buffered banyan networks where analytical expressions for throughput and delay are obtained. We show that using the priority scheme the cell delay deviation is reduced and the influence on throughput performance is insignificant. The results obtained from our analytical model are compared with the simulations and good agreement is observed. Comparisons with some proposed analytical models in the literature reveal that our model is more accurate and powerful in predicting the performance of buffered banyan networks.

Adaptive estimation of nonstationary sinusoidal signals or quasi-periodic signals in additive noise is of essential importance in many diverse engineering fields, such as communications, biomedical engineering, power systems, pitch detection in transcription and so forth. So far, Kalman filtering based techniques, recursive least square (RLS), simplified RLS (SRLS) and LMS algorithms, for examples, have been developed for this purpose. This work presents in detail a performance analysis for the SRLS algorithm proposed recently in the literature, which is used to estimate an enhanced sinusoid. Its dynamic and tracking properties, noise and lag misadjustments are developed and discussed. It is found that the SRLS estimator is biased, and its misadjustments are functions of not only the noise variance but also, unpleasantly, of the signal parameters. Simulations demonstrate the validity of the analysis. Application of the SRLS to a real-life piano sound is also given to peek at its effectiveness.

Many network protocols for routing messages have been proposed for mobile computing environments. In this paper, we consider the query processing strategy which operates over these network protocols. To begin with, we introduce five fundamental location update methods based on ideas extracted from the representative network protocols. They are the single broadcast notification (SBN), the double broadcast notification (WBN), the single default notification (SDN), the double default notification (WDN), and the no notification (NN). As a network protocol, each method is strong in performance in some system enrivonment, but weak in others. In practical situations, where various kinds of applications are used for various purposes, however, it is required to use a single method. We therefore propose an adaptive query processing strategy where these five location update methods can be dynamically selected. Moreover, we analyze the performance of this adaptive query processing strategy via the Markov chain. We also use the statistical approach to estimate the traffic of individual hosts. Finally, we show the efficiency of our proposed strategy over a wide area of system environments.

Multipath interconnection networks can support higher bandwidth than those of nonblocking networks by passing multiple packets to the same output simultaneously and these packets are buffered in the output buffer. The delay-throughput performance of the output buffer in multipath networks is closely related to output traffic distribution, packet arrival process at each output link connected to a given output buffer. The output traffic distributions are different according to the various input traffic patterns. Focusing on nonuniform output traffic distributions, this paper develops a new, general analytic model of the output buffer in multipath networks, which enables us to investigate the delay-throughput performance of the output buffer under various input traffic patterns. This paper also introduces Multipath Crossbar network as a representative multipath network which is the base architecture of our analysis. It is shown that the output buffer performances such as packet loss probability and delay improve as nonuniformity of the output traffic distribution becomes larger.

The problem of test generation for VLSI circuits computationally requires prohibitive costs. Parallel processing on a multiprocessor system is one of available methods in order to speedup the process for such time-consuming problems. In this paper, we analyze the performance of parallel test generation for combinational circuits. We present two types of parallel test generation systems in which the communication methods are different; vector broadcasting (VB) and fault broadcasting (FB) systems, and analyze the number of generated test vectors, the costs of test vector generation, fault simulation and communication, and the speedup of these parallel test generation systems, where the two types of communication factors; the communication cut-off factor and the communication period, are applied. We also present experimental results on the VB and FB systems implemented on a network of workstations using ISCAS'85 and ISCAS'89 benchmark circuits. The analytical and experimental results show that the total number of test vectors generated in the VB system is the same as that in the FB system, the speedup of the FB system is larger than that of the VB, and it is effective in reducing the communication cost to switch broadcasted data from vectors to faults.

This paper first presents the performance analysis of the NACF algorithm. The results show the possibility of the degradation in the convergence speed. To improve the convergence speed, the bias term is introduced into the NACF algorithm and its efficiency is investigated through the computer simulations.

Efficient retrieval of nested objects is an important issue in advanced database systems. So far, a number of indexing methods for nested objects have been proposed. However, they do not consider retrieval of nested objects based on the set comparison operators such as and . Previouly, we proposed four set access facilities for nested objects and compared their performance in terms of retrieval cost, storage cost, and update cost. In this paper, we extend the study and present refined algorithms and cost formulas applicable to more generalized situations. Our cost models and analysis not only contribute to the study of set-valued retrieval but also to cost estimation of various indexing methods for nested objects in general.

In this paper, we evaluate the performance of handoff schemes in microcellular personal communication systems (PCS) which cater to both pedestrian and vehicular users. Various performance parameters, including blocking of new calls,channel utilization, handoff blocking and call termination probabilities for each user type are evaluated. We study different queuing disciplines for handoff calls and their impact on system performance. We also study the tradeoff in handoff blocking and call termination probabilities between user types as the handoff traffic carried by the system from each user type is varied.

In this paper, we present an analysis of a high-speed slotted ring with a single packet buffer at each station. Assuming that distances between stations affect the network performance only through the sum of themselves (this will be called the "lumpability assumption"), we introduce a model system called the lumped model in which stations are aggregated at a single point on the ring with their relative positions preserved. At the instant when each slot visits the aggregated point of the lumped model, we build a Markov chain by recording the system state of buffers and slots. From the steady state probabilities of the Markov chain, we obtain the mean waiting time and the blocking probability of each station. It will be shown analytically and by simulation that the analysis based on the lumped model yields accurate results for various network conditions.

With rapid increase of information requirements from various application areas, there has been much research on the efficient information retrieval. A signature is an abstraction of information, and has been applied in many proposals of information retrieval systems. In this paper we evaluate the performance of various signature-based information retrieval methods and provide guidelines for the most effective usage to a given operational environment. We derive analytic performance evaluation models of these access methods based on retrieval time, storage overhead and insertion time. The relationships between various performance parameters are thoroughly investigated. We also perform simulation experiments by using wide range of parameter values and show that the performance experiments agree with those analytic models.