Major problems of traffic control in ATM networks include how to decide whether a network accepts a new call or not in real time and how to select the best set of Dual Leaky Bucket (DLB) parameter values. To solve these problems, it is necessary to determine the amount of network bandwidth required by the call. In this paper, we present an analysis based on bounding technique to derive an upper bound on bandwidth requirement when the call is characterized by a set of DLB parameters. Consequently, a new definition of the upper bound on bandwidth requirement and simple formulae used for computing the upper bound have been obtained. To clarify the advantages of the derived upper bound, we demonstrate its two applications, one to select the best set of DLB parameter values from candidates for minimizing the amount of bandwidth to be allocated to the call and the other to establish a Connection Admission Control (CAC) scheme. The upper bound-based CAC scheme is fast enough to process in real time due to its simplicity and provides a significant improvement of network utilization compared to the peak rate-based CAC scheme.

In this paper we present two traffic control approaches, a circuit emulation traffic control (CETC) and an adaptive priority traffic control (APTC) for supporting voice services in ATM networks. Most voice services can be handled as CBR traffic, this causes a lot of wasted bandwidth. Sending voice through VBR (variable bit rate) may be a better alternative, because it allows the network to allocate voice bandwidth on demand. In CETC, the service discipline guarantees the quality of service (QOS) for voice circuits. Through mathematical analysis, we show that CETC features an adequate performance in delay-jitter. Moreover, it is feasible in implementation. We also present an APTC approach which uses a dynamic buffer allocation scheme to adjust the buffer size based on the real traffic need, as well as employs an adaptive priority queuing technique to handle various delay requirements for VBR voice traffic. It provides an adequate QOS for voice circuits in addition to improving the multiplexing gain. Simulation results show that voice traffic get satisfied delay performance using our approaches. It may fulfill the emerging needs of voice service over ATM networks.

In this paper, we propose and evaluate the performance of Wireless ATM MAC layer protocol to support efficiently various ATM traffics, such as CBR, VBR, ABR and UBR, in wireless ATM network environments for reverse and forward link. The proposed MAC protocol could extend efficiently the service discipline of ATM traffics from wired network to wireless ATM network environments. Thus, available bandwidth, which is remained except the bandwidth for CBR and VBR traffics, could be effectively allocated to ABR and UBR traffics. Especially, in view of reverse link, two-phase scheduling algorithm supports successfully variable characteristics of VBR traffic. And, in view of forward link, 'Wireless Dynamic Weighted Earliest Deadline First' scheduling algorithm minimizes the mean cell delay and required buffer size. Simulation results show that proposed method provides effective performance in wireless ATM environments.

A decision-directed carrier phase recovery scheme for high-speed satellite communications is proposed. Since the estimation is performed in complex domain from the baseband signal, the scheme has fast acquisition performance, unlike the conventional PLL. This merit makes it applicable for various wireless systems such as wireless local area networks (LANs), wireless asynchronous transfer modes (ATMs) and local multipoint distribution systems (LMDSs) that need high-speed burst signal communications. Also, this scheme can be implemented easily because low pass filters (LPFs) are utilized in filtering the estimates in order to suppress the noise within the carrier recovery loop. Moreover it does not require any divider or voltage-controlled oscillator (VCO). The performance is analyzed through analytical methods and simulation.

This paper studies the application of a media synchronization mechanism to the interleaved transmission of video and audio specified by the H.223 Annex in PHS. The media synchronization problem due to network delay jitters in the interleaved transmission has not been discussed in either the Annex or any related standards. The slide control scheme, which has been proposed by the authors, is applied to live media. We also propose a QOS control scheme to control both quality of the media synchronization and that of the transmission delay. Through simulation we confirm the effectiveness of the slide control scheme and the QOS control scheme in the interleaved transmission.

The software requirements specification process consists of three steps; requirements capture and analysis, requirements definition and specification, and requirements validation. At the beginning of the second step which this paper focuses on, there have been several types of massive documents generated in the first step. Since the developers and the clients/users of the new software system may not have common knowledge in the field which the system deals with, it is difficult for the developers to produce correct requirements specification by using these documents. There has been few research work to solve this problem. The authors have developed a support tool to produce correct requirements specification by arranging and restructuring those documents into clearly understandable forms. In the second step, the developers must specify the functions and their constraints of the new system from those documents. Analyzing the developers' real activities for designing the support tool, the authors propose a model of this step as the following four activities. To specify the functions of the new system, the developers must collect the sentences which may suggest the functions scattering those documents. To define the details of each function, the developers must gather the paragraphs including the descriptions of the functions. To verify the correctness of each function, the developers must survey all related documents. To perform above activities successfully, the developers must manage various versions of those documents correctly. According to these four types of activities, the authors propose the effective ways to support the developers by arranging those documents. This paper shows algorithms based on this model by using the structures of the documents and keywords which may suggest the functions or constraints. To examine the feasibility of their proposal, the authors implemented a prototype tool. Their tool extracts complete information scattering those documents. The effectiveness of their proposal is demonstrated by their experiments.

This paper reports on an Intelligent CASE tool, applicable in a structured programming phase, or from detailed design to coding. This is automation of the bottom level in the hierarchical design process of detailed design and coding, where the largest man-hours are consumed. The main idea is that human designers use a CASE tool for the initial design of a software system, and the design knowledge is automatically acquired from the structured charts and stored in the knowledge base. The acquired design knowledge may be reused in designs. By reusing it, a similar software system may be designed automatically. It has been shown that knowledge acquired in this way has a Logarithmic Learning Effect. Based on this, a quantitative evaluation of productivity is made. By accumulating design experiences (e. g. 10 times), more than 80% of the detailing designs are performed automatically, and productivity increases by up to 4 times. This tool features universality, an essentially zero start-up cost for automatic design, and a substantial increase in software productivity after enough experiences have been accumulated. This paper proposes a new basic idea and its implementation, a quantitative evaluation applying techniques from Industrial Engineering, which proves the effectiveness of the proposed system.

Conventional approach for frequency estimation usually assume a single tone without data modulation. In many applications such an assumption, realized by using either a separate pilot beacon or synchronization preamble is not feasible. This paper deals with frequency estimation of phase-modulated carriers in the absence of timing information and known data pattern. We introduce new frequency estimators that are based on the generalized maximum likelihood principle. The communication channels under consideration include both additive white Gaussian noise (AWGN) channels and correlated Rician fading channels. For the latter class, we distinguish between the case when the fading (amplitude) process is tracked and that when it is not tracked.

Analysis of frequency-dependent lossy transmission lines is very important for designing the high-speed VLSI, MCM and PCB. The frequency-dependent parameters are always obtained as tabulated data. In this paper, a new curve fitting technique of the tabulated data for the moment matching technique in the interconnect analysis is presented. This method based on Chebyshev interpolation enhances the efficiency of the moment matching technique.

One of unique features of CDMA slotted ALOHA (CDMA S-ALOHA) is that user must synchronize his transmission to given slot. Thus orthogonal sequence as spreading sequence would achieve ideal throughput if each of packets accomplish perfect synchronization. In the presence of any ambiguity in synchronizations, however, quasi-synchronous (QS) sequences suit well with CDMA S-ALOHA system. In this paper, we introduce new QS-sequences obtained from the orthogonal Gold sequences and discuss their performance when applying to CDMA S-ALOHA systems. As a result, withstanding to access timing error, good performance is ensured with this sequence under the environment of AWGN, MAI (multiple access interference) and frequency non-selective fading, that is, micro or pico cellular systems and indoor wireless LANs.

DQRUMA (Distributed-Queueing Request Update Multiple Access) protocol has been proposed as an access protocol for the wireless ATM Local Area Networks. DQRUMA protocol is useful to transmit fixed-length packets (e. g. ATM cells). However, it cannot be applied to multimedia environment because it does not include any access control policy for multimedia traffic. In the paper, we propose a slot assignment scheme of DQRUMA protocol in wireless ATM LAN which supports integrated multimedia traffic with different service requirements. In this scheme we can allocate network resources according to the service requirements of each medium because the base station assigns Transmit-Permission flexibly according to the features of each medium.

In a shared-memory multiprocessor, shared data are usually accessed in a critical section that is protected by a lock variable. Therefore, the order of accesses by multiple processors to the shared data corresponds to the order of acquiring the ownership of the lock variable. This paper presents a selective write-update protocol, where data modified in a critical section are stored in a write cache and, at a synchronization point, they are transferred only to the processor that will execute the critical section following the current processor. By using QOLB synchronization primitives, the next processor can be determined at the execution time. We prove that the selective write-update protocol ensures data coherency of parallel programs that comply with release consistency, and evaluate the performance of the protocol by analytical modeling and program-driven simulation. The simulation results show that our protocol can reduce the number of coherence misses in a critical section while avoiding the multicast of write-update requests on an interconnection network. In addition, we observe that synchronization latency can be decreased by reducing both the execution time of a critical section and the number of write-update requests. From the simulation results, it is shown that our protocol provides better performance than a write-invalidate protocol and a write-update protocol as the number of processors increases.

As the current Internet becomes popular in information access, demands for real-time display and playback of continuous media are ever increasing. The applications include real-time audio/video clips embedded in WWW, electronic commerce, and video-on-demand. In this paper, we present a new control protocol R3CP for real-time applications that transmit stored MPEG video stream over a lossy and best-effort based network environment like the Internet. Several control mechanisms are used: a) packet framing based on the meta data; b) adaptive queue-length based rate control scheme; c) data preloading; and d) look-ahead pre-retransmission for lost packet recovery. Different from many adaptive rate control schemes proposed in the past, the proposed flow control is to ensure continuous, periodic playback of video frames by keeping the receiver buffer queue length at a target value to minimize the probability that player finds an empty buffer. Contrary to the widespread belief that "Retransmission of lost packets is unnecessary for real-time applications," we show the effective use of combining look-ahead pre-retransmission control with proper data preloading and adaptive rate control scheme to improve the real-time playback performance. The performance of the proposed protocol is studied via simulation using actual video traces and actual delay traces collected from the Internet. The simulation results show that R3CP can significantly improve frame playback performance especially for transmission paths with poor packet delivery condition.

We consider a finite-capacity single-server queue with constant service and vacation times, which is seen in the time division multiple access (TDMA) scheme. First we derive the probability that j customers remain in the queue when a test customer arrives. Using this probability we then evaluate the probability that the test customer who arrives during the vacation or service time has to wait in the queue for longer than a given time. From these results, we obtain the waiting time distribution for the customer arriving at an arbitrary time. We also show a practical application to wireless TDMA communications systems.

This letter proposes a new retransmission control scheme for slow-frequency-hopped communication systems, in which the number of (re)transmitted packets is adaptively decreased in a certain period. Performance of the proposed scheme is analyzed and compared with that of the conventional scheme in terms of the normalized throughput and the 98% packet transmission delay. The numerical results show the superiority of the proposed scheme.

In this paper, we discuss the initial acquisition of the code division multiplexed DS/SS down-link signals at a user terminal of multiple LEO mobile satellite communication systems. In LEO systems, a receiver generally receives signals from plural satellites for soft hand-off and for satellite diversity as a countermeasure to shadowing. In this situation, the signal from each satellite becomes the interference to the signals from other satellites. In addition to this inter-satellite interference, we have to consider the intra-satellite interference from user channels to a pilot channel because of the loss of orthogonality of channels at initial acquisition stage especially under frequency offsets due to Doppler effect. Thus in this paper, we analytically evaluate the performance of an initial acquisition scheme, taking the intra/inter-satellite interference under Doppler shift into account.

The maximum order complexity (MOC) of a sequence is a very natural generalization of the well-known linear complexity (LC) by allowing nonlinear feedback functions for the feedback shift register which generates a given sequence. It is expected that MOC is effective to reduce such an instability of LC as an extreme increase caused by the minimum changes of a periodic sequence, i. e. , one-symbol substitution, one-symbol insertion or one-symbol deletion per each period. In this paper we will give the bounds (lower and upper bounds) of MOC for the minimum changes of an m-sequence over GF(q) with period qn-1, which shows that MOC is much more natural than LC as a measure for the randomness of sequences in this case.

A homeotropic liquid crystal display utilizing a liquid crystal with positive dielectric anisotropy, 13. 3" XGA TFT-LCD, has been fabricated. The rubbing-free device, appears black in the absence of electric field. When an electric field generated by interdigital electrodes is applied, a bend deformation of molecular director to the direction of the field occurs and thus the cell transmits light, showing brightness uniformity in all directions owing to the dual domainlike director configuration. With an addition of negative-birefringent film, this device shows excellent viewing angle characteristics.

Packet scheduling is one of the key mechanisms that will be employed in the network nodes (routers and switches) for supporting multiple quality of services. In this paper we propose a new packet scheduling algorithm called Urgency-based Round Robin (URR) which computes an index for flows in order to keep track of instantaneous bursts. Basically the index is employed as a measure of the time-dependent service necessity for each flow thus making it possible to detect those flows which might be in need of momentary service. Also, we propose a novel weight allocation scheme to be used together with the scheduler with the aim of preventing network underutilization. Our algorithm can be considered as a version of Weighted Round Robin (WRR) with improved delay characteristics. We show analytically that URR has the desired capability of upper-bounding unfairness. We also show, by simulation, that URR can improve delay performance even under extremely bursty traffic conditions without bandwidth overprovisioning. We also give simulation results for network traffic which exhibits long range dependency (self-similarity) and show that URR is again more effective than a plain round robin multiplexer.

Backfire quadrifilar helical antennas combined with parasitic loops are investigated in detail, focusing on clarifying the function of parasitic loops. First, the basic property is examined for the case of one parasitic loop, and it is found that the loop behaves as a director when the circumferential length of the loop is nearly 0. 9λ, and a reflector when the circumferential length of the loop is nearly 1. 2λ provided the distance between the parasitic loop and the top plane of helical antennas is approximately 0. 1λ, where λ is the wavelength. Next, the function of the parasitic loop is investigated by comparing the current distributions on the helices and the loop with those on a monofilar helix with a ground plane. It is found that the function of the parasitic loop is quite different from that of the ground plane. Then, the case of two parasitic loops is examined, and it is shown that the use of two parasitic loops is very effective and simple measures to control the radiation pattern and gain of the backfire quadrifilar helical antennas. Finally, for this type of antennas with two parasitic loops, an example of structural parameters suited to the use in satellite communications is presented.