In mobile computing environments, a problem may exist between loss recovery mechanisms employed by the TCP (transmission control protocol) and RLP (radio link protocol). It is because that local retransmissions performed by the RLP could interfere with the TCP end-to-end error recovery when there are long and correlated packet losses due to bursty channel errors. That is, a spurious timeout would occur at the transport layer. In this paper, a new method is proposed to effectively suppress the occurrence of TCP spurious timeouts. In this new method a small number of ACKs (acknowledgements) is buffered at the base station prior to the emergence of every bad state period in the wireless channel, and these ACKs are henceforth released by the base station one at a time to reset the TCP sender's retransmission timer. Comprehensive comparisons between the proposed method and a baseline method are conducted through simulations to show that the improvement in throughput performance can be as large as 22%.

The ranging algorithm allows active stations to measure their distances to the headend for synchronization purpose in Hybrid Fiber Coax (HFC) networks. A practicable mechanism to resolve contention among numerous stations is to randomly delay the transmission of their control messages. Since shorter contention cycle time increases slot throughput, this study develops three mechanisms, fixed random delay, variable random delay, and optimal random delay, to minimize the contention cycle time. Simulation demonstrates that the optimal random delay effectively minimizes the contention cycle time and approaches the theoretical optimum throughput of 0.18 from pure ALOHA. Furthermore, over-estimation reduces the impact on contention cycle time more than under-estimation through sensitivity analysis, and both phenomenon damage slot throughput. Two estimation schemes, maximum likelihood and average likelihood, are thereby presented to estimate the number of active stations for each contention resolution round. Simulation proofs that the proposed estimation schemes are effective even when the estimated number of active stations in initial contention round is inaccurate.

Dual-Banyan is a buffered banyan asynchronous transfer mode (ATM) switch encompassing bifurcated queueing as its buffering strategy. This paper describes an efficient analytical model, based on iterative calculations, for performance evaluation of the Dual-Banyan switch under uniform and non-uniform traffic patterns with much less time than the simulation. The efficiency of the given model is verified through a comparison with simulation results. Another benefit of our model is the possibility to adopt it in any non-uniform incoming traffic. At last, we compare performance of Dual-Banyan switch and Input Buffer Banyan, and show that Dual-Banyan switch has significant better performance levels.

We propose a new analytical model of Transmission Control Protocol (TCP) in wireless environments where transmission errors occur frequently. In our proposed model, we consider the exponential increase of a congestion window and the exponential increase of a timeout back-off. Finally, we have clarified the behavior of TCP mechanisms of different versions and TCP throughput characteristics analytically. From our result, the behavior of TCP mechanisms is different in each implementation version. These differences mean that the required characteristics of wireless links are different in each implementation version. Therefore, our proposed model is a base analysis of designing wireless link mechanisms.

It has been shown that the performances of single-receiver ARQ schemes are largely dependent on the packet-error process, i.e., for dependent packet-error environment, they are under- or over- estimated by analyzing them under the assumption that packet-errors occur at random. While, multi-receiver ARQ's have not been analyzed and evaluated for dependent packet-error process. In this paper, we analyze the throughput efficiency of the fundamental multi-receiver Go-back-N ARQ scheme, which can be implemented very simply, over an unreliable channel modeled by the two-state Markov process. Any receiver erroneously receives a packet with probability inherent to each state. From numerical results, we show that the throughput efficiency of the fundamental multi-receiver Go-Back-N ARQ scheme depends on the number of receivers, round-trip-delay, and the characteristic of the Markov process. Also we show that the throughput efficiency of the fundamental multi-receiver Go-Back-N ARQ scheme for larger decay factor and larger difference between packet error probability at each state is considerably better than that for the random error pattern.

It has been shown that virtual output queuing (VOQ) and a sophisticated scheduling algorithm enable an input-queued switch to achieve 100% throughput for independent arrival process. Several of the scheduling algorithms that have been proposed can be classified as either iterative scheduling algorithms or symmetric crossbar arbitration algorithms. i-OCF (oldest-cell-first) and TSA (two step arbiter) are well-known examples of iterative scheduling algorithms and symmetric crossbar arbitration algorithms, respectively. However, there are drawbacks in using these algorithms. i-OCF takes long time to find completely a conflict-free match between input ports and output ports because it requires multiple iterations. If i-OCF cannot find a conflict-free match completely, the switch throughput falls. TSA has the possibility that it finds a conflict-free match faster than i-OCF because it does not need any iterations. However, TSA suffers from the starvation problem. In this paper, we propose a new scheduling algorithm. It uses two schedulers, which we call scheduler 1 and scheduler 2, in parallel. After cells were transmitted, the information that input port i granted the offer from output port j in scheduler 2 is mapped to scheduler 1 if and only if input port i has at least one cell destined for output port j. If the information is moved, input port i and output port j are matched in scheduler 1 at the beginning of the next time slot. Our proposed algorithm uses one scheduler based on TSA and the other scheduler based on i-OCF. Numerical results show that the proposed scheduling algorithm does not require multiple iterations to find a conflict-free match completely and suffer from the starvation problem for both uniform and bursty traffic.

In this paper we propose a MIN (Multistage Interconnection Network) whose performance in the faulty case degrades as gracefully as possible. We focus on a two-dilated baseline network as a sort of MIN. The link connection pattern in our MIN is determined so that all the available paths established between an input terminal and an output terminal via an identical input of a SE (Switching Element) in some stage will never pass through an identical SE in the next stage. Extra links are useful in improving the performance of the MIN and do not complicate the routing scheme. There is no difference between our MIN and others constructed from a baseline network with regard to numbers of links and cross points in all SEs. The theoretical computation and simulation-based study show that our MIN is superior to others in performance, especially in robustness against concentrated SE faults in an identical stage.

Nowadays, many IP based communication systems are connected by high-speed networks to realize high-speed TCP/IP communication. However, since the mechanism of TCP/IP protocols is based on a best effort service, the quality of the communication may change by the time or the route of the traffic. Therefore, it is important for network providers to investigate the quality of their users' communication. In order to analyze the traffic on an Internet backbone, the realtime analysis is one of the important factors. So far, several tools were developed for the purpose of the traffic measurement. However, none of them can analyze sufficient statistics to evaluate the quality for each end user's communication in realtime. Therefore, we have designed and implemented a performance monitor, which collects the statistics representing the performance such as TCP throughput while capturing the traffic. The statistics are collected by every pair of IP addresses and by every application. The monitor also provides the function to analyze effectively for the statistics records such as sorting and filtering of the records and the graphical user interface to operate the software tool. This paper describes the design and implementation of the performance monitor.

To improve the throughput efficiencies of ARQ protocols over a high random packet-error channel, contiguous multiple copy-transmission (CMCT) strategy for which each packet is (re-)transmitted by sending its multiple copies in contiguous slots has been used so far. However, in burst error environments, all copies may be damaged in an error burst resultting the performance degradation of CMCT. To cope with this situation, we propose, in this paper, a new strategy called intermittent multiple copy-transmission (IMCT) whereby multiple copies are sent at a fixed interval. The throughput efficiency of go-back-N ARQ using CMCT or IMCT is analyzed and considered under a two-state Markov channel model expressing burst error property of a channel. As a result, it is shown that (i) the degree of improvement of throughput efficiency by CMCT or IMCT depends on the degree of error burst and (ii) the proposed IMCT can improve the throughput efficiency of go-back-N ARQ for high and burst error channels.

In this letter, performance evaluation of a system that combines between Code-Division Multiple Access (CDMA) and ALOHA protocol in multimedia networks is presented. In our analysis, we compare the performance between the two basic techniques of ALOHA protocol, i.e., Slotted-ALOHA (S-ALOHA) and Unslotted-ALOHA (U-ALOHA), when combined with CDMA scheme to support voice and data users operating in same CDMA channel. The quality of service (QoS) required for voice and data media is completely taken care of. We obtain the throughput of data media, and the outage probability for voice considering both voice and data offered loads. Throughput performance of S-ALOHA technique is almost twice of that of U-ALOHA. However, we show in this letter that when we combine CDMA with the two basic techniques of ALOHA to accomplish multimedia transmission, both techniques have almost same performance. And, thus, CDMA U-ALOHA can be a good candidate for multimedia networks.

This paper proposes a new satellite communication system that enables high-speed communication in a mobile environment. The system configuration combines a terrestrial mobile network and an existing satellite system, and includes a tracking antenna that was newly developed to receive 30 Mbit/s signals from commercial communication satellites. A prototype system comprising the mobile network, the satellite system and a vehicle in which the tracking antenna is installed was constructed for purposes of evaluation and demonstration. A LAN system was incorporated in the experimental vehicle by using the tracking antenna, a satellite router and a Personal Digital Cellular phone. The validity of the proposed system was verified by the tracking antenna driving tests, system UDP tests and FTP throughput tests in a mobile environment.

In this paper, a slot-averaged SIR model is derived to analyze the performances of 1/N-slotted DS-CDMA packet radio networks. DS-CDMA packet radio networks can be analyzed with channel threshold model, but it is not appropriate for a 1/N-slotted access scheme since the interference level varies slot by slot during a packet transmission time. A packet in a DS-CDMA system will be channel-encoded and interleaved so that an excessive interference of some slots does not result in a corruption of a packet directly. In a 1/N-slotted DS-CDMA packet system, interleaving and channel coding can cope with the excessive interference in some slots. Proposed slot-averaged SIR model reflects this effect and is simple enough to be used in the performance analyses of various packet access control schemes. This paper verifies the slot-averaged SIR model and evaluates the throughputs of 1/N-slotted access schemes using it. This paper also compares the results with those of the conventional channel threshold model. Packet access schemes such as slotted ALOHA, slotted CLSP and slotted CLSP/CC are analyzed. The results show that many different aspects from the results of the previous works can be evaluated and this work will offer more accurate view on the behavior of 1/N-slotted DS-CDMA packet radio networks.

We propose a p-persistent protocol for slow-frequency-hopped slotted random access networks, assuming that all the users know the number of backlog users in a slot. The proposed protocol delays new packet transmission until the number of users with a packet to be retransmitted decreases to the threshold or less. Performance of the proposed protocol is evaluated with a two-dimensional Markov chain for systems with finite population in terms of throughput efficiency and the average transmission delay. Numerical results show that the proposed protocol can achieve better performance by suppressing new packet transmission, compared to the conventional frequency-hopped slotted ALOHA. The optimum value of the threshold is also numerically derived.

We propose the analytical calculation method of average packet success probability of CDMA Slotted ALOHA systems, which derives accurate probability, and that is applicable to the system with any spreading codes and any amplitude distributions. In the method, we consider the bit-to-bit dependence of amplitude of signals, used spreading sequences, relative timing delays, and relative carrier phases. This bit-to-bit dependence is the case that the parameters above mentioned are constant for a slot time. By using the method to obtain the average packet success probability, we derive useful throughput performance of CDMA Slotted ALOHA systems on fading channel, and show that the normalized throughput becomes worse in the case of larger spreading factor.

In this paper, we discuss the access control in multimedia CDMA ALOHA protocol. We introduce a new algorithm for the access control based on Modified Channel Load Sensing Protocol (MCLSP) in an integrated voice and two different classes of data users, high bit rate and low bit rate, exist in a multi-code CDMA Slotted ALOHA system. With our new algorithm, we show that the throughput of high bit rate data users, as well as, the total throughput of the data medium can be optimized and take a maximum value even at high values of offered loads. We also investigate the performance when voice activity detection (VAD) is considered in voice transmission.

In this paper, using a standard propagation model, the performance evaluation of a code-division-multiple-access (CDMA) cellular system with high data rate services for the reverse and forward links is investigated. In the approach, we propose "Equivalent Load" to estimate two cases of the system termed as the static analysis and the dynamic analysis. Performance measures of the static analysis obtained include the system capacity plane, outage probability and throughput. Performance measures of the dynamic analysis obtained include the allowed maximum data rate and the constraint set between the load situation and the data rate. We also estimate the effects of the power control on the system.

For a wireless communication system to work effectively without interference, the electromagnetic environment needs to be controlled. We experimentally and analytically investigated the requirements for controlling the electrical field strength and delay spread so as to achieve the best communication without electromagnetic interference in selected regions for a 2.4-GHz-band wireless LAN system. To control the coverage, partitions were placed around desks in a test environment and covered on the inside with electromagnetic absorbing board from the top of the desks to the top of the partitions; four indoor environments that combined one of two wall-material types and one of two partition heights were used. The transmission loss and delay spread were measured, then calculated using ray tracing to verify the effectiveness of using ray-tracing calculation. The throughput and BER characteristics were measured for the same environments to clarify the requirements for controlling the coverage. We found that covered and uncovered regions could be created by using partitions with absorbing boards and that the delay spread must be less than 15 ns and the received-signal must be stronger than -75 dBm for a region to be covered. We verified that the delay spread can be calculated to within 5 ns and the received-signal level can be calculated to within 5 dB of the measured data by using ray tracing. Therefore, ray tracing can be used to design antenna positions and indoor environments where electromagnetic environments are controlled for 2.4-GHz-band wireless LAN systems.

This paper proposes the iterative quasi-oldest-cell-first (i-QOCF) scheduling algorithm, a new scheduling algorithm for input-queued ATM switches with virtual output queuing (VOQ). In the i-QOCF scheduling algorithm, each input port and each output port maintains its own list. The length of the list can be N, 2 N, ..., B N, where B is the size of the separate queue for an output port at input ports, and N is the number of output ports. The list maintained by an input port contains the identifiers for those output ports to which that input port will send a cell. The list maintained by an output port contains the identifiers for input ports that have a cell destined for that output port. If we use a list whose length is B N, then the identifiers in the list appear in the oldest order, and i-QOCF gives preference to cells that have been waiting for the longest time. If we use a list whose length is less than B N, then the identifiers in the list appear in the quasi-oldest order, and i-QOCF gives preference to cells that have been waiting for the quasi-longest time. We determine the performance of i-QOCF in a comparison with i-OCF in terms of cell delay time. We find that an input-queued ATM switch with i-QOCF and VOQ can achieve 100% throughput for independent arrival processes. Under uniform traffic, 3-QOCF is enough to achieve convergence during one cell time. If we use 3-QOCF, the list length is 3 N, then its cell delay time is almost the same as that of 4-OCF (Oldest-Cell-First).

We propose a shared multi-buffer ATM switch, in which each unicast cell has chances to be read from a shared buffer during three consecutive read cycles and each multicast cell is read from a shared buffer if the shared buffer is not used for read of a unicast cell at the last read cycle. The HOL effect that the unicast cells experience is not augmented by the multicast cells and utilization rate of the output ports is increased because both a unicast cell and a multicast cell have the opportunity to be read for each output port. For a fixed multicast rate, the proposed scheme shows 98.9% throughput even though the offered load reaches 1. We designed the proposed shared multi-buffer ATM switch in 0.6 µm single-poly triple metal CMOS technology. The designed shared multi-buffer ATM switch has 88 ports and operates at 20 MHz, which supports 155.52 Mbps STM-1 source rate for each port.

This paper focuses on flow control in high-speed and large-scale networks. Each node in the network handles its local traffic flow only on the basis of the information it knows. It is preferable, however, that the decision making of each node leads to high performance of the whole network. To this end, the relationship between local decision making and global performance of flow control is the essential object. We propose phenomenological models of flow control of high-speed and large-scale networks, and investigate the stability of these models.