Current literature on input buffer management reveals that, in representative ATM networks under highly bursty traffic conditions, the fuzzy thresholding approach yields lower cell loss rate at the cost of lower throughput. Also, under less bursty traffic, the traditional fixed thresholding approach achieves higher throughput at the expense of higher cell loss rate. The integration of these two properties into practice is termed adaptive dynamic buffer management (ADBM) approach for input buffers and its assessment is the objective of this paper. The argument is that, given that the traffic conditions are constantly changing, to achieve efficiency during actual operation, the network control must dynamically switch, at every ATM switch, under the call processor's control, between the two input buffer management techniques, dictated by the nature of the traffic at the inputs of the corresponding switch. The need to involve the call processor marks the first effort in the literature to dynamically configure input buffer management architectures at the switch fabric level under higher level call processor control. It stems from the fact that the switch fabric operates very fast and cannot engage in complex decision making without incurring stiff penalty. To achieve this goal, the network control needs knowledge of the burstiness of the traffic at the inputs of every ATM switch. The difficulties with this need are two-fold. First, it is not always easy to obtain the traffic model and model parameters for a specific user's call. Second, even where the traffic model and the model parameters are known for a specific user's call, this knowledge is valid only at the source switch where the user interfaces with the network. At all other switches in the network, the cells of the traffic in question interact asynchronously with the cells from other traffic sources and are subject to statistical multiplexing. Thus, to obtain the exact nature of the composite traffic at the inputs of any ATM switch, is a challenge. Conceivably, one may determine the burstiness by counting the number of cells incurred at the inputs of an ATM switch over a defined time interval. The challenge posed by this proposition lies in the very definition of burstiness in that the time interval must approach, in the limit, zero or the resolution of time in the network. To address this challenge, first, a 15-node representative ATM network is modeled in an asynchronous, distributed simulator and, second, simulated on a network of workstations under realistic traffic stimuli. Third, burstiness indices are measured for the synthetic, stochastic traffic at the inputs of every ATM switch as a function of the progress of simulation for different choices of time interval values, ranging from 20,000 timesteps down to 1,000 timesteps. A timestep equals 2.73 µs. Results reveal that consistent burstiness indices are obtained for interval choices between 1,000 and 5,000 timesteps and that a burstiness index of 25, measured at 3,000 timestep interval, constitutes a reasonable and practical threshold value that distinguishes highly bursty traffic that warrants the use of the fuzzy thresholding approach from less bursty traffic that can benefit from the fixed thresholding scheme. A comparative performance analysis of ADBM yields the following. For pure fixed and pure fuzzy thresholding schemes, the throughputs are at 73.88% and 71.53% while the cell drop rates are at 4.31% and 2.44%,respectively. For the ADBM approach, where the input buffer management alternates at each individual ATM switch between the fixed and fuzzy schemes, governed by measured burstiness index threshold of 25 for a 3,000 timestep interval, the throughput is 74.77%, which is higher than even the pure fixed scheme while the cell drop rate is 2.21% that is lower than that of the pure fuzzy scheme. In essence, ADBM successfully integrates the best characteristics of the fuzzy and fixed thresholding schemes.

This paper presents throughput performance along with power profiles in the time and frequency domains over 100 Mbps based on field experiments using the implemented Variable Spreading Factor-Orthogonal Frequency and Code Division Multiplexing (VSF-OFCDM) transceiver with a 100-MHz bandwidth in a real multipath fading channel. We conducted field experiments in which a base station (BS) employs a 120-degree sectored beam antenna with the antenna height of 50 m and a van equipped with a mobile station (MS) is driven at the average speed of 30 km/h along measurement courses that are approximately 800 to 1000 m away from the BS, where most of the locations along the courses are under non-line-of-sight conditions. Field experimental results show that, by applying 16QAM data modulation and Turbo coding with the coding rate of R = 1/2 to a shared data channel together with two-branch antenna diversity reception, throughput over 100 and 200 Mbps is achieved when the average received signal-to-interference plus noise power ratio (SINR) is approximately 6.0 and 14.0 dB, respectively in a broadband channel bandwidth where a large number of paths such as more than 20 are observed. Furthermore, the location probability for achieving throughput over 100 and 200 Mbps becomes approximately 90 and 20% in these measurement courses, which experience a large number of paths, when the transmission power of the BS is 10 W with a 120-degree sectored beam transmission.

It is reported that TCP does not perform well in high-speed wide area networks. Because MulTCP behaves like the aggregate of N TCP flows, MulTCP can be used to achieve throughputs of 1 Gbps or more. However, no performance evaluation of MulTCP in high-speed wide area networks has been published. Computer simulations are used to evaluate the performance of MulTCP. The results clarify that synchronized packet losses greatly impact the performance of MulTCP.

In the recent draft of the 802.11e MAC, the Hybrid Coordinator (HC) gives transmission opportunity (TXOP) for a station to transmit burst frames in TXOP for improving throughput efficiency. In this paper, we express the expected throughput for the Block Ack policy in Polled TXOP as a closed form function of the number of burst MSDUs, the number of fragmented MPDUs, the data payload length, the frame retry count, the wireless channel condition, and the selected PHY mode. Based on our simulation study and analysis, we show that the TXOP and proper Ack policy can enhance system performance, and that there are the Block Ack efficient block size and frame length to transmit the burst frames for each PHY mode. Moreover, we also show that when the Block Ack mechanism is combined with link adaptation, it has better throughput performance.

Since the TCP is the transport protocol for most Internet applications, evaluation of TCP throughput is important. In this paper, we establish a framework of evaluating TCP throughput by simple measurement. TCP throughput is generally measured by sending TCP traffic and monitoring its arrival or using data from captured packets, neither of which suits our proposal because of heavy loads and lack of scalability. While there has been much research into the analytical modeling of TCP behavior, this has not been concerned with the relationship between modeling and measurement. We thus propose a lightweight method for the evaluation of TCP throughput by associating measurement with TCP modeling. Our proposal is free from the defects of conventional methods, since measurement is performed to obtain the input parameters required to calculate TCP throughput. Numerical examples show the proposed framework's effectiveness.

The data throughput of wireless LAN of IEEE 802.11 g in mobile environment is measured according to the distance between AP and NIC, and the moving speed. It shows there is no degradation of the maximum data throughput of 20 Mbps due to the vehicle speed up to 160 km/h.

This paper proposes methods of computing maximum throughput for Petri nets that model workflows including resources, called WF-nets with resources. We first propose the formal definitions of WF-nets with resources and their subclasses: marked graph/state machine WF-nets with conflict-free resources (CF-Res-MG/SMWF-nets). We also show several properties of CF-Res-MG/SMWF-nets. Next we give the methods of computing maximum throughput for CF-Res-MG/SMWF-nets under condition that all tokens have to be processed in the order of First-In First-Out (FIFO). Concretely, for CF-Res-MGWF-nets, on the basis of Ramamoorthy's method of computing cycle time, we give an algorithm of computing maximum throughput under the FIFO condition. For CF-Res-SMWF-nets, there is not any method of computing maximum throughput under the FIFO condition. So we are the first to give an algorithm of computing maximum throughput for CF-Res-SMWF-nets under the FIFO condition. Finally we show an example of computing maximum throughput by using our method.

In this paper, the CSK/SSMA ALOHA system with nonorthogonal sequences which combines the ALOHA system with Code Shift Keying (CSK) using nonorthogonal sequences is proposed. The throughput performance was evaluated by theoretical analysis. Moreover, the throughput performance of the system is compared with those of the DS/SSMA ALOHA and M-ary/SSMA ALOHA systems. It is found that the throughput performance of our system to be better than those of the other two systems.

Since a radio channel is shared among terminals in an ad hoc network, packet collisions are frequent. In case of transmitting packets especially using TCP, data and ACK packets are transmitted in opposite directions on the same radio channel. Therefore, frequent collisions are unavoidable, and this seriously degrades TCP throughput. It is possible to transmit to two or more nodes which adjoin from a certain node simultaneously on the radio channel. To reduce the likelihood of packet collisions when an intermediate node transmits both data and ACK packets, these two types of packet can be combined and transmitted at the same time to increase the efficiency of radio channel utilization. In this paper, we propose a new technique to improve TCP performance by combining data and ACK packets. Our proposed technique is applicable to generic ad hoc networks easily. By means of a simulation using networks with various topologies, we have found that throughput can be improved by up to 60% by applying our proposed technique.

Unlike most of the previous work for smart antennas that covered each area individually (antenna-array design, signal processing and communications algorithms and network throughput), this paper may be considered as a review of comprehensive effort on smart antennas that examines and integrates antenna array design, the development of signal processing algorithms (for angle of arrival estimation and adaptive beamforming), strategies for combating fading, and the impact on the network throughput. In particular, this study considers problems dealing with the impact of the antenna design on the network throughput. In addition, fading channels and tradeoffs between diversity combining and adaptive beamforming are examined as well as channel coding to improve the system performance.

Maximizing the throughput of a network while supporting fairness among nodes is one of the most critical issues in designing wireless networks. In single-hop networks, a lot of schemes have been proposed to satisfy this criterion, and efficient protocols like the IEEE 802.11 and the HiperLAN/2 standards have been established for wireless LAN. In multi-hop wireless networks, however, throughput and fairness have different characteristics from those of single-hop networks. In this paper, the tradeoff between throughput and fairness on multi-hop networks is studied by computer simulation, assuming three node distribution models, namely, normal, constant, and uniform distribution and four different bandwidth (channel) scheduling methods, i.e., first-in first-out buffer based, weighted traffic model based, bandwidth reservation based, and maximum throughput based scheduling. Furthermore, as a realistic model, a hybrid scheme is investigated where partial bandwidth is allocated to the bandwidth reservation based scheduling and the remaining to the maximum throughput based one.

This paper proposes a new scheme that can evaluate the cell throughput performance of wireless local area network (LAN) systems, which use carrier sense multiple access/collision avoidance (CSMA/CA) and multiple transmission bit-rates (multi-rate). We extend the interference model of the conventional scheme in order to deal with interference more accurately in multi-cell environments. Unlike the conventional scheme, the proposed scheme is able to handle multi-rate systems. We use the proposed scheme to evaluate the IEEE 802.11a system and systems whose signal-bandwidth is expanded from that of the IEEE 802.11a system. We find that a system with 35(75) MHz signal-bandwidth achieves about 1.3(1.25) times higher cell throughput than the IEEE 802.11a system. Furthermore, the system with 35(75) MHz signal-bandwidth is also shown to have the potential to achieve up to 1.5(1.8) times higher cell throughput performance than the IEEE 802.11a system if the transmission efficiency on the media access control (MAC) layer is assumed to be ideal. It is concluded that the proposed scheme confirms that the approach to expand the signal bandwidth of the IEEE 802.11a system is effective to improve the cell throughput performance. This result is virtually impossible to derive with the conventional scheme.

In this letter, we present an adaptive hopping technique for a wireless personal area network (WPAN) system employing a frequency hop spread spectrum (FH/SS). Analytical results based on the closed-form solutions for the aggregate throughput show that the proposed hopping algorithm using two defined hopping criteria is more friendly towards all kinds of interferers and gives an enhanced throughput with a moderate computational complexity.

We propose a new dependable system called PREGMA (Platform for Reliable Environment based on a General-purpose Machine Architecture). PREGMA aims to meet two requirements -- fault tolerance and low cost -- for Internet services. It can provide fault tolerance, so we can avoid system failure and prevent data corruption, even if faults occur. That is, it masks the faults by running multiple replicated servers, each possessing its own data, in a loosely synchronized manner and delivering the majority vote as output to clients. Moreover, PREGMA is composed of COTS (Commercial Off-The-Shelf) components without modification, which makes it possible to offer the services at a low cost. We investigated two approaches for achieving redundancy of the Coordinator, which is the core of PREGMA: using the primary backup method and the active replication method. We evaluated the effectiveness of PREGMA in terms of throughput overhead, data integrity and recovery time. The results for a prototype show that PREGMA using the Coordinator with the primary backup method outperforms that with the active replication method and has throughput only 3% lower than a non-redundant system. The results also show that, in the event of failure, the recovery time is only less than one second and no data corruption occurs.

This paper presents a novel data placement and retrieval scheme for video-on-demand systems. In particular, a zone-based data placement scheme is employed to reduce the average seek time of the disk array storage system and thus increase the disk access bandwidth to allow the video server provide more services of video programs concurrently. Furthermore, due to the inherent nature of video access, a popular program always requires more accesses and therefore occupies more disk I/O bandwidth as request for serving such program increases. A new retrieval strategy is proposed to maintain a single copy of each video program disregarding the popularity of the video programs, and to achieve maximum I/O throughput of the video server.

This paper presents a new modulation scheme for Very-High Speed Digital Subscriber Lines (VDSL) modem, featuring a Multi-Code Multi-Carrier Code Division Multiple Access (MC2-CDMA) modulation. The system takes advantage from both the CDMA modulation and the Multi-Carrier transmission, and furthermore the channel throughput is increased adopting a multi-code approach. Starting from an overview of this novel scheme, encompassing the transmitter, channel and receiver description, a brief review of the equalization techniques is also considered and a proper bit-loading algorithm is derived to find out the achievable overall channel rate. The aim of this paper, besides introducing this novel scheme, is to demonstrate its suitability for a VDSL environment, where the achievable channel rate represents a real challenge. By means of a further optimisation, a general improvement of the system performance with respect to the standardized Discrete Multi Tone (DMT) modulation is also demonstrated.

A new bandwidth-efficient asynchronous multicarrier DS CDMA scheme is proposed for the uplink. In this new scheme, each user employs a set of FIR filters whose impulse responses are a mutually orthogonal (MO) complementary set of sequences. The intentional inter-symbol interference (ISI) and multiple access interference (MAI) are eliminated by the properties of these sequences. We also propose applying this new scheme in a DS CDMA packet network in which slotted ALOHA or pure ALOHA protocol is used. Packet throughput figures are obtained for the new ALOHA/bandwidth-efficient asynchronous MC DS CDMA packet network. Numerical results are given for both slotted and pure ALOHA cases. With the same bandwidth and number of simultaneous users, the throughput is compared favorably to similar figures for single-carrier DS CDMA with random spreading sequences.

This paper compares by computer simulation the achievable throughput performance employing fast packet scheduling algorithms focusing on the throughput of each user in High Speed Downlink Packet Access (HSDPA). Three packet scheduling algorithms are employed: the Maximum carrier-to-interference power ratio (CIR), Proportional Fairness (PF), and Round Robin (RR) methods. The simulation results elucidate that although the Maximum CIR method achieves an aggregated user throughput within a cell higher than that using the PF and RR methods, the PF method is advantageous because it enhances the user throughput for a large number of users with a lower received signal-to-interference power ratio (SIR), who are located outside the normalized distance of 0.6-0.7 from a cell site (this corresponds to the area probability of 50-60% within the cell) compared to the Maximum CIR method. It is also shown that when the PF method is employed, the probability of user throughput of greater than 2 Mbps in the vicinity of the cell site becomes approximately 45% (5%) for L = 1-path (2-path) fading channel, while it is almost 80% (50%) when using the Maximum CIR method. Finally, we show that the average user throughput in a 2-path Rayleigh fading channel is reduced by approximately 30% compared to that in a 1-path channel due to severe multipath interference (MPI) and that the average user throughput is strongly affected by the total traffic produced within a cell, which is directly dependent on the number of users within a cell and the data size per packet call.

We propose a new adaptive FEC scheme combined with ELN (Explicit Loss Notification) that was proposed for improving TCP performance in wireless cellular networks. In our method, transmission errors on the wireless link are measured at the packet level and the error status is notified the TCP sender with ELN. According to this information, an appropriate FEC code is determined in order to maximize the TCP performance. We first compare the TCP performance using Snoop Protocol, ELN and the fixed FEC, through which we find the appropriate FEC code against given BER (bit error ratio). We then show how the adaptive FEC can be realized using our solution, and also examine the appropriate observation period of measuring BER enough for the fading speed on the noisy wireless link. We finally demonstrate that our method can achieve better performance than the conventional fixed FEC by using the Gilbert model as a wireless error model.

In this letter, we describe an adaptive hybrid SR ARQ scheme using punctured TCM and code combining. Numerical results show that the proposed scheme yields better throughput efficiency than the scheme using TCM at the values of Es/No smaller than 9 dB.