We report a flexible and lightweight wearable microstrip antenna that can be sewn into clothing and hats. This antenna is composed of felt and a conductive woven fabric. Experimental results clearly show that this antenna operates normally as a conventional microstrip antenna, and is practical and feasible for personal satellite communications.

This paper presents a new automatic-frequency control (AFC) configuration capable of removing wide range frequency offsets (up to about 0.625 fs, where fs is signal symbol rate). The new configuration consists of an AFC that removes frequency offsets between 0.125 fs and another AFC that detects the frequency offset range coarsely between 0.625 fs. This paper describes the principle of the new AFC configuration. The proposed AFC configuration employs four correlators to enhance the acquisition range. It also adopts the reverse modulation scheme to decrease the acquisition time. The performance of the new AFC configuration is confirmed via computer simulations. It is shown that the proposed configuration can accommodate wide range frequency offsets as well as reduce the acquisition time.

Fading in mobile satellite communications severely degrades the performance of data transmission. It is commonly modeled with non-frequency selective Rayleigh fading. For this type of channel, a new structure for a bit-interleaved coded modulation (BICM) scheme is presented and evaluated to determine its effectiveness compared to previously proposed schemes. This scheme is referred to as rate-compatible punctured BICM (RCP-BICM), in that its BICM encoder is able to yield a wide range of data rates by using a punctured convolutional code obtained by periodically perforating parity bits from the output of a low-rate-1/2 systematic convolutional code. A trellis-coded modulation (TCM) scheme and a turbo TCM (TTCM) scheme are discussed and evaluated for comparison with the RCP-BICM scheme. Simulation results demonstrate that the RCP-BICM scheme with hard-decision iterative decoding is superior to the TCM scheme by 3 dB at a bit error rate (BER) of 10^-5 over an Rayleigh fading channel, and comes at a BER of 10^-5 within 1 dB of the TCM scheme over an additive white Gaussian noise (AWGN) channel.

A prototype modem with unequal error protection (UEP) capability was developed using multiple block coded modulation (MBCM). Benefited from its unique structure, MBCM can be used to provide UEP straightforward. We propose a new method to increase the robustness of the carrier recovery process by taking advantage of the MBCM code structure. We also use a frame format to facilitate the synchronization operation. This modem was developed in preparation for the mobile satellite communication experiments using the Engineering Test Satellite of VIII-type (ETS-VIII). In addition of MBCM, some other types of modulation schemes have been implemented in the same modem to enable a range of communication experiments to be performed. The modem can operate at variable data rates. The results of laboratory measurements agreed well with computer simulation results. Typical link budgets based on the parameters of ETS-VIII are also presented.

A Generalized Symbol-rate-increased (GSRI) Pragmatic Trellis coded Type-I Hybrid ARQ based on a Selective-Repeat (SR) ARQ with multicopy (MC) retransmission (SR+MC scheme) for high speed mobile satellite communication system is analyzed. The SR+MC ARQ is a suitable scheme for mobile satellite systems and further improvement of the throughput performance can be expected by an additional combination of an error control coding. In this paper, we investigate the performance of the SR+MC scheme employing GSRI Pragmatic TCM. GSRI TC-MPSK can arbitrarily set the bandwidth expansion ratio keeping higher coding gain than conventional TCM scheme. Also Pragmatic TCM has an advantage in that the modulation level can be easily changeable. By changing the modulation level and the bandwidth expansion ratio, this scheme can optimize the performance according to the channel conditions. Numerical and simulation results show that the GSRI Trellis Coded Type-I Hybrid ARQ presents better performance than conventional Pragmatic Trellis Coded Type-I Hybrid ARQ.

High speed core networks with optical fibers have spread widely, but it is still difficult to access the core networks from many rural areas. Synchronous CDMA systems with GEO satellite links are attractive to solve this problem, since they have wide service areas and are suitable for packet-based networks due to their statistically multiplexing effects. Additionally, the synchronous CDMA systems have more effective frequency utilization and power efficiency than asynchronous ones. In the synchronous CDMA systems, transmitted signals from fixed earth stations are required to achieve synchronization with each other. The broadband systems require extremely precise timing control as their bit rates increase. In this paper, we propose a synchronization method for a synchronous CDMA communication system using a GEO satellite and verify the feasibility of Gigachip rate synchronous CDMA systems.

The advanced satellite broadcasting system in the 21 GHz band or higher frequency bands is expected to be suitable for use in high quality multimedia services in the future. To establish this system, rain attenuation mitigation is very important and the time diversity system has been proposed as an appropriate technology for this purpose. This paper shows principle of time diversity as an attenuation mitigation technology and also shows the effect of time diversity. We also propose a method for predicting time diversity gain as a function of the rain attenuation, cumulative time percentage, and time delay of two data contents or broadcasts.

Low Earth Orbit (LEO) satellite constellations have been proposed in recent years to provide broadband network access. This research focuses on Walker Delta type constellation. Walker Delta has overlapping ascending and descending orbits. Although Inter Satellite Links (ISLs) can be utilized between satellites orbiting in the same direction, ISLs cannot be utilized between satellites orbiting in opposite directions. As a result, a Walker Delta Constellation with ISLs has two locally separate overlapping meshes, an ascending and a descending mesh. To reach from one local mesh to the other, the traffic has to pass through the highest latitude intra-plane ISLs. Therefore the propagation delay between terminals connected to different meshes is greater than between terminals connected to the same mesh. Due to characteristic handover of LEO satellites, terminals can connect to a satellite in the other mesh during communication, causing drastic variation of propagation delay which results in degradation of communication quality. These issues can be solved by continuously connecting the communication terminals to the same mesh. In this paper, a satellite selection method for Walker Delta Constellations with double mesh coverage is proposed. It employs geographical location information of the communicating terminals, to connect them to the same mesh. In addition, the proposed method selects the mesh that minimize propagation delay for that communication session. It is shown through simulation that the proposed method is effective in reducing delay and jitter for a connection while improving overall communication quality of the network.

For a LEO constellation system, it is important to optimize the orbit parameters to maximize the quality of communication service. At the Next-generation LEO System (NeLS) Research Center, the LEO parameters were evaluated for a mobile satellite communication system. A 2π constellation was selected to maintain a stable inter-satellite link (ISL). An elevation angle above 20 degrees was required for a service area of 70 degree of latitude. The performance of optical ISL terminals has since improved as a result of key technological developments at the NeLS Research Center. As a consequence, the constellation parameters for ISL have become more flexible. Furthermore, the ability of ground station to access two satellites has improved communication quality. In this paper, we address the optimum constellation parameters for dual-satellite coverage. An equation for determining the optimum inclination angle was derived from the constellation parameters. Moreover, by using the new constellation parameters, we found that the satellite network consists of a bi-directional Manhattan Street Network (MSN), and the ISL network structure was improved.

This paper discusses the orbital motion and elevation properties of a quasi-zenith satellite system using circular orbits. The satellites are deployed on inclined geosynchronous orbits with identical sub-satellite loci on earth. The satellites trace the locus at even intervals. This satellite system can provide mobile satellite communications and navigation services at very high elevations to middle-latitude regions. In general, the orbital parameters of the satellite system are determined by numerical simulation to maximize the minimum elevation angle in areas where satellite services are to be provided. However, an understanding of the properties of the orbit and consequent elevation properties are important for efficient constellation design. This paper formulates the orbital motion of inclined geosynchronous circular orbits, including the relative motion to the rotating earth. Although elliptical orbit constellations are also possible and can gain higher elevation, only circular orbits, which can be accurately formulated without using an analytically unsolvable Kepler's equation, are discussed in this paper. Elevation properties are evaluated using the geocentric angle between the sub-satellite point and an arbitrary point in the intended service area. This angle is a typical parameter that can be derived as a single-valued function of the elevation at a specific point. Optimum orbital parameters for an intended service area can be easily estimated without numerical simulation using the results of the evaluation described in this paper. These results can also be used to infer whether a circular-orbit constellation is applicable to an intended service area.

In this paper, the design of a mobile satellite Internet access (MSIA) system and a mobile broadband satellite access system, called Mobile Broadband Interactive Satellite Multimedia Access Technology System (MoBISAT) are presented. MSIA system provides Internet service, broadcasting, and digital A/V service in both fixed and mobile environments using Ku-band geostationary earth orbit (GEO) satellite. A Ku-band two-way active phased array antenna installed on top of the transportation vehicles can enable the transmission of signals to satellite as well as signal tracking and reception. The forward link and return link are a high speed Time Division Multiplex (TDM) and TDMA transmission media, respectively, both of which carry signaling and user traffic. The MoBISAT, which is a next generation mobile broadband satellite access system, provides both Ku-band satellite TV and Ka-band high-speed Internet to the passengers and crews for land, maritime, and air vehicles. This paper addresses the main technological solutions adopted for the implementation and test results for the MSIA system and the main design features of the MoBISAT system.

Electronic mail (email) is one of the most popular services using Internet Protocol (IP). Emails are now sent to and from portable phones in addition to laptop computers. The volume of emails is increasing because of the recent addition of attached electronic files. In addition, short message delivery services (SMS) that send email header information and the essence of email contents are also commonly used. Global short message delivery services via satellites are considered attractive for meeting the demand of increased short message delivery services. To establish a communications link for indoor subscribers, it is preferable to concentrate high transmission power into a spot-beam even using a non-geostationary satellite orbit (non-GSO) satellite system with low Earth orbits. The idea can be realized by a kind of TDMA scheme where only a single spot-beam per satellite is fired at a timeslot in an efficiently scheduled manner. This paper proposes a global short message delivery scheme and a scheduling algorithm that enables each satellite to concentrate its power into a spot-beam illuminating the addressed area in a prescribed timeslot, so that a subscriber in the area can efficiently receive a message addressed to the subscriber. Since the algorithm must guarantee that the spot-beam assignment causes no interference with other spot-beams from adjacent satellites, numerical evaluations are given for a 6-hour period MEO system, as an example, to demonstrate the efficacy and performance.

Cutoff rate of m-ary PPM is derived for an avalanche photodiode (APD) based receiver. The cutoff rate is compared among m=8-256 under the condition of the same total signal energy in 256 slots, where the effects of the timing jitter and atmospheric turbulence are included. Numerical results are shown for the illustrative examples of the cutoff rate.

A high-attenuation waveguide filter using electromagnetic bandgap (EBG) substrates is introduced. With a simple design modification on the EBG covers, the waveguide filter produced an almost full Ku-band rejection bandwidth showing better than 20 dB input-to-output isolation from 12.3 to 17.2 GHz.

A vertically connected wireless link (VCWL) using the 60-GHz band has been proposed for reliable and economical transmission of various satellite media to individual building units. This paper describes a prototype of such a VCWL that employs a self-heterodyne scheme. The CNR performance of the prototype was evaluated in a real environment. The results showed that signals transmissions of the required quality could be delivered to the units of a five-story apartment. For the placement of multiple transmitters in close proximity, the prototype required 12 dB of CIR.

Diagonal algebraic space time (DAST) block codes was proved to achieve the full transmit diversity over a quasi-static fading channel and to maintain 1 symbol/s/Hz. When the number of transmit antennas employed is larger than 2, DAST codes outperform the codes from orthogonal design with the equivalent spectral efficiency. However, due to the limitation on the signal constellation with complex integer points, no good 3bits/symbol DAST block code was given previously. In this paper, we propose a general form of 8-star-PSK constellations with integer points and present some theoretical results on the performance of the equivalent 8-star-PSK modulations. By using our proposed 8-star-PSKs, we present a searching algorithm to construct DAST codes with 3 bits per symbol under some criteria and investigate their performances over flat Rayleigh fading channels. It is shown that (5,2) 8-star-PSK scheme has a comparable performance to conventional 8PSK over additive white Gaussian noise (AWGN) channel and the corresponding DSAT codes constructed can achieve significant performance gain over flat Rayleigh fading channel.

We propose a class of multiple input multiple output (MIMO) systems using the transmit diversity pre-combining (TDPC) scheme, assuming perfect channel state information (CSI) at a transmitter. In such a class of systems, each transmitter antenna is weighted independently according to the channel condition so that the Euclidean distance between the closest symbols at a receiver could be maximized under a total transmit power constraint. We also introduce an effective space-time trellis code (STTC) for the proposed class of MIMO systems. From computer simulations, we see that incorporating both the TDPC scheme and the proposed STTC outperforms the MIMO system using only the TDPC scheme and the conventional STTC-MIMO system.

The characteristics of the spreading sequence significantly affect the signal-to-interference power ratio (SIR) of the received signal in direct sequence code division multiple access (DS-CDMA) system. In this paper, we analyze the receiver performance of the forward link of a DS-CDMA system in terms of the SIR and bit error rate (BER) when pseudo noise (PN) codes and concatenated orthogonal/PN (OPN) codes are used as the spreading sequence. The use of OPN spreading codes can cancel out the intra-cell interference signals with equal path delay, but the use of PN spreading codes cannot, significantly degrading the performance. As a result, the BER performance of the OPN spreading system is better than that of the PN spreading system. The use of OPN spreading sequences can provide the system capacity at least two times larger than the use of PN spreading sequences in the single-cell environment even when the channel has a large number of multipaths. The two spreading systems also show significant difference in the user capacity even in a multi-cell environment.

A simple millimeter-wave quasi-maximal-ratio-combin-ing antenna diversity system based on the millimeter-wave self-heterodyne transmission technique is described. The millimeter-wave self-heterodyne transmission technique is useful for developing millimeter-wave systems with enhanced characteristics in regard to system miniaturization, development and fabrication cost, and the frequency stability of the signal transmission. We also show that applying this technique with an antenna diversity receiver configuration can easily solve a problem peculiar to millimeter-wave systems--the fact that the transmission link always requires a line-of-sight path--without requiring hardware designed with millimeter-scale precision. In this paper, we theoretically analyze the operating principle of a combining antenna diversity system based on the millimeter-wave self-heterodyne transmission technique. We further prove that we can obtain a diversity gain in accordance with that of a maximal-ratio combining diversity system without resorting to any complicated control of the received signal envelope and phase. Our experiments using the simplest two-branch diversity structure have validated the operating principle derived in our theoretical analysis. Our results show that a received CNR improvement of 3 dB is obtained as a diversity gain. We also demonstrate that circuit precision corresponding to the wavelength of the intermediate frequency, rather than to the millimeter wavelength, is sufficient to obtain the diversity effect when we control the signal phase or delay in combining the received signals.

We have developed a code-division-multiplexing (CDM) transmission scheme for future cellular communication systems, which uses cyclic shifted-and-extended (CSE) codes generated from an M-sequence to enable seamless communication in highly mobile environments. Because the correlation characteristics of CSE codes are determined by the M-sequence, the cross-correlation values are accumulated as a result of combining transmitted signals with opposite polarities in parallel channels. The accumulated cross-correlation values significantly degrade transmission performance, especially with multi-level modulation schemes such as quadrature amplitude modulation (QAM). We thus propose a cancellation technique to eliminate the accumulated cross-correlation values. We have evaluated the transmission performance of the CDM transmission scheme with the proposed technique by computer simulation. The new scheme enables high-quality data transmission in fast-fading channels.

Multi-Input Multi-Output (MIMO) systems, which utilize multiple antennas at both the receiver and transmitter, promise very high data rates in a rich scattering environment. It was proven in literature that with optimal power allocation, MIMO eigenmode transmission system (EMTS) is optimal because MIMO capacity is maximized. However, the performance of MIMO EMTS is very sensitive to the accuracy of channel state information and thus it is of practical importance to analyze its performance when channel state information is corrupted under realistic system and propagation conditions. In this paper, we lower bound the mutual information of MIMO EMTS with imperfect channel estimation and delayed quantized feedback in a spatially correlated continuous fading channel. Our results showed that this lower bound is tight and can serve as a comprehensive guide to the actual performance of MIMO EMTS under practical operating conditions.

This paper addresses the utility-based radio resource allocation problem in DS-CDMA systems carrying multimedia traffic. The proposed scheme, aiming at achieving optimal resource allocation, considers the joint power and data rate allocation. To avoid high computational complexity of nonlinear optimization, we reformulate the radio resource allocation problem as a market model, where resource is regarded as a commodity. Since the market model satisfies the incentive-compatible constraint, the optimal resource allocation can be obtained at the market equilibrium in a distributed manner. According to whether to allocate a minimal transmission data rate to each user, two algorithms, UCA and FCA, are proposed. UCA emphasize on maximizing system overall utilities, while FCA guarantees fairness to users. Simulation results show that the proposed radio resource allocation scheme and algorithms are flexible and efficient for multimedia DS-CDMA systems.

Smart or adaptive antennas promise to provide significant space-time communications against fading in wireless communication systems. In this paper, we propose multiple-input multiple-output (MIMO) beamforming for frequency-selective fading channels to maximize the Signal-to-Noise and Interference Ratio (SINR) based on an iterative update algorithm of transmit and receive weight vectors with prior knowledge of the channel state information (CSI) at both the transmitter and receiver. We derive the necessary conditions for an optimum weight vector solution and propose an iterative weight update algorithm for an optimal SINR reception. The Maximum Signal-to-Noise (MSN) method, where noise includes the additive gaussian noise and interference signals, is used as a criterion. The proposed MIMO with M N arrays allows the cancellation of M + N - 2 delayed channels. Computer simulations are presented to verify our analysis. The results show that significant improvements in performance are possible in wireless communication systems.

This paper presents a multipath interference canceller (MPIC) configuration based on multipath interference (MPI) replica generation per transmit antenna (called PTA-MPIC). This configuration is associated with Space Time Transmit Diversity (STTD) for the common control physical channel (CCPCH), which takes advantage of tentative decision data after STTD decoding, and with closed-loop type phase control (PC) transmit diversity for the dedicated physical channel (DPCH) employing tentative decision data after diversity combining, in the W-CDMA forward link. This paper also proposes transmitter carrier phase verification, i.e., an antenna verification method used in PC transmit diversity, that utilizes the dedicated pilot symbols in a DPCH after the PTA-MPIC removes the MPI components. The one-stage PTA-MPIC removes the MPI from the common pilot channel (CPICH), the CCPCH, and the synchronization channel (SCH). The simulation results show that this canceller reduces the required average transmit E_b/N₀ of the DPCH at the average BER of 10^-3 by approximately 3.0 dB compared to that using a MF-based Rake receiver (the transmit power ratio of each common channel to DPCH is R_CPICH/DPCH = 3 dB, R_CCPCH/DPCH = 5 dB, and R_SCH/DPCH = 3 dB, with TPC and without antenna diversity reception at the user equipment). Furthermore, it is shown that in the two-stage PTA-MPIC with MPI suppression for all channels associated with PC transmit diversity, the required average transmit E_b/N₀ employing the proposed antenna verification is reduced by approximately 0.3 dB, 0.5 dB, and 1.2 dB compared to that using the conventional antenna verification when the transmission power ratio of the interfering DPCH to the desired DPCH is R_Int/Des = 0 dB, 3 dB, and 6 dB for ten DPCHs. This is because the number of detection errors of the transmitted carrier phase in the second antenna due to feedback information bit decoding error is reduced.

Call Admission Control (CAC) is a very important issue in CDMA systems to guarantee a required quality of service (QoS) and to increase system capacity. In this paper, we proposed and analyzed the CAC scheme using multiple criterions (MCAC), which can provide a quicker processing time and better performance. One is based on the number of active users with the minimum/maximum threshold by considering the spillover ratio, and the other is based on the signal to interference ratio (SIR). If active users are lower/higher than the minimum/maximum number of users threshold (N_min )/(N_max ), we accept/reject the new call without any other considerations based on the first criterion. And if the number of active users is between the N_min and N_max, we consider the current SIR to guarantee QoS based on the second criterion. Then the system accepts the new call when the SIR satisfies the system requirements, otherwise, the call is rejected. The multiple criterions scheme is investigated and its performance is compared with the number of user based CAC and power based CAC.

In this paper, the design of a QoS scheduling scheme for the Internet traffic is proposed by considering the correlation property of the arriving traffic. The basic concept of the Weighted Fair Queuing (WFQ) is adopted in the proposed scheme, however, the correlation property of the traffic stream is applied as the heuristic to adjust the share weight factors of each traffic type dynamically. The Auto Regressive Integrated Moving Average (ARIMA) model is applied in this paper to characterize the correlation property of the Internet traffic. And the share weight factors are derived from the parameters of the AR part and MA part. Experimental simulations are performed to illustrate the effectiveness of the proposed scheme. In addition to comparing the performance of each service types, we also define a fair play parameter (FPP) to examine the fairness index among various traffic streams of the proposed scheme. The experimental results indicate that the proposed scheme demonstrates a quite good performance in scheduling the integrated services and the fairness among service classes can also be achieved, especially when the link capacity is limited.

The issue of guaranteeing Quality of Services (QoS) in a network has emerged in recent years. The Proportional Delay Differentiated Model has been presented to provide the predictable and controllable queueing delay differentiation for different classes of connections. However, most related works have focused on providing this model for a wired network. This study proposes a novel scheduler to provide proportional delay differentiation in a wireless network that includes a multi-state link. This scheduler, Look-ahead Waiting-Time Priority (LWTP), offers proportional delay differentiation and a low queueing delay, by adapting to the location-dependent capacity of the wireless link and solving the head-of-line (HOL) blocking problem. The simulation results demonstrate that the LWTP scheduler actually achieves delay ratios much closer to the target delay proportion between classes and yields smaller queueing delays than past schedulers.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

In this paper we propose an MPLS switch model with a GPS-based scheduler to provide guaranteed QoS in the Internet. Important applications of this model include IP VPN (MPLS-VPN) services and MPLS networks. The considered scenarios include cases with sufficient labels or with limited number of labels under competing traffic streams. Due to tractability issues in direct analysis, we use a worst case equivalent system model to obtain the theoretical upper bounds of delays and backlogs of the original system model. Simulation results are used to verify the analytical results. We can obtain the worst case delay bounds and the maximum backlogs. Furthermore, the trade-off among the number of labels, the worst delay bounds, and the maximum backlogs can be observed. With our model, one can determine the number of required labels according to the allowable delay and the available buffer size.

A round-ended wide straight slot cut in the broad wall of a rectangular waveguide is analyzed by the Method of Moments (MoM) using numerical eigenmode basis functions derived by the edge-based finite element method (FEM), referred to as MoM/FEM. The frequency characteristics of the calculated transmission coefficients are compared with the measured ones, and good agreement is observed for a wide variety of antenna parameters. For simpler analysis that does not use MoM/FEM, an equivalent rectangular slot approximation for a round-ended slot is discussed. The resonant frequencies of empirically introduced "equal-area" and "equal-perimeter" slots are compared with those of round-ended slots for a wide variety of parameters such as slot width, wall thickness and dielectric constant inside the waveguide. Based upon MoM/FEM, which can be a reliable reference, it is found that the equal-area slot always gives a better approximation of the order of 1% over that of the equal-perimeter one which is of the order of 5%. For higher accuracy, a new rectangular slot approximation of a round-ended slot is proposed to be a linear combination of equal-area and equal perimeter approximation. The error is around 0.25% for a wide variety of parameters such as slot width-to-length ratio, wall thickness and dielectric constant of the filling material inside the waveguide.

Common-mode excitation caused by an imperfect ground plane on a printed circuit board (PCB) has been conventionally explained with the 'current driven' scheme, in which the common-mode current is driven by the ground voltage across the unintentional inductance of the ground plane. We have developed an alternative method for estimating common-mode excitation that is driven by the difference of the common-mode voltages for two connected transmission lines. A parameter called current division factor (CDF) that represents the degree of imbalance of a transmission line explains the common-mode voltage. In this paper, we calculate the CDF with two-dimensional (2-D) static electric field analysis by using the boundary element method (BEM) for asymmetric transmission lines with an arbitrary cross-section. The proposed 2-D method requires less time than three-dimensional simulations. The EMI increase due to a signal line being close to the edge of the ground pattern was evaluated through CDF calculation. The estimated increase agreed well--within 2 dB--with the measured one.

The current paper presents the design, analysis, and implementation of a low-profile resonant dc-to-dc converter that utilizes a coreless printed circuit board transformer as a substitute for the conventional magnetic core-based transformer. A prototype series resonant converter, fabricated in a 40 mm80 mm area with a 4 mm thickness while achieving the maximum efficiency of 85% at a 58 W output power, is used as an example to address the theoretical and practical issues involved in the design, analysis, and implementation of a PCB transformer-based low-profile dc-to-dc converter.

By modelling the quantization error as additive white noise in the transform domain, Wiener filter is used to reduce quantization noise for DCT coded images in DCT domain. Instead of deriving the spectrum of the transform coefficient, a DPCM loop is used to whiten the quantized DCT coefficients. The DPCM loop predicts the mean for each coefficient. By subtracting the mean, the quantized DCT coefficient is converted into the sum of prediction error and quantization noise. After the DPCM loop, the prediction error can be assumed uncorrelated to make the design of the subsequent Wiener filter easy. The Wiener filter is applied to remove the quantization noise to restore the prediction error. The original coefficient is reconstructed by adding the DPCM predicted mean with the restored prediction error. To increase the prediction accuracy, the decimated DCT coefficients in each subband are interpolated from the overlapped blocks.

This paper examines the effect of segmentation mismatch on audio-video transmission by Bluetooth. We focus on the segmentation mismatch caused by the difference between the RFCOMM Maximum Frame Size and the baseband packet payload size. By experiment, we assessed the maximum throughput and media synchronization quality for various types of ACL packets. In the experiment, a media server transferred stored video and audio streams to a single terminal with point-to-point communication; we supposed no fading environment and added white noise by which interference from DSSS systems is modeled. The experiment showed that the effect of segmentation mismatch is large especially when the total bit rate of the two streams is near the channel transmission rate. We also observed that the media synchronization control is effective in compensating for the disturbance by the segmentation mismatch in noisy environments.

To provide multimedia services, a server system can use a cluster of servers to serve many users concurrently. In the literature, several assignment algorithms have been proposed to assign the media objects to these servers. The popularities of the media objects may change with time, and the service provider may add new objects or delete obsolete objects. In these cases, it is necessary to execute an assignment algorithm to find a new assignment. To realize the new assignment from the current one, it is necessary to relocate some objects. In this paper, we formulate and solve this relocation problem, and our objective is to minimize the relocation time for minimal service interruption. We propose a two-stage algorithm to solve this relocation problem. In the first stage, we introduce the concept of server relabelling, and apply the Hungarian method to find the largest common assignment between the new and current assignments. In the second stage, we propose a chain relocation schedule by which every object is directly relocated from its original server to the new server without deadlock. We use computer simulation to demonstrate the effectiveness of the optimal relocation method.

Recently, Yeh, Shen, and Hwang proposed a smartcard-based one-time password authentication scheme as an improved version of S/KEY, and claimed that their scheme is superior to other similar schemes in security and efficiency. In this letter, we show that Yeh-Shen-Hwang's scheme is still vulnerable to a stolen-verifier attack that may cause serious security problems.

This letter describes consecutive zero and one bits detection circuits designed for a 1.25 Gbit/s burst mode laser driver realized in a SiGe 0.35 µm BiCMOS technology with 3.3 V power supply. The architecture is based on a frequency divider and a delay line counting per four consecutive zero or one bits. The detector was designed with high-speed split-output stage flip-flops modified to have a reset input. Experimental results validate the design of the detector.

We propose a novel wavelength assignment algorithm that can establish lightpaths requiring the least wavelength conversions by chaining a minimum number of wavelength-continuous segments. Simulations show that our algorithm outperforms both first-fit and most-used schemes with large margins. Besides, moderate computational requirement and insignificant signaling overhead are also advantages of our algorithm.

In this paper, we propose a Time Interleave (TI)-Ultra Wideband (UWB)-Impulse Radio (IR) system where N_f monocycle pulses per information symbol are transmitted not continuously but discontinuously through the time interleaver to get more time diversity. Moreover, we compare the diversity gains of the UWB-IR systems on block fading channels with the fading block size q: the UWB-IR system with a single transmit antenna, the Space Time (ST)-UWB-IR system with N_t transmit antennas, and the TI-UWB-IR system with a single transmit antenna. Simulation results show that when the fading block size q = 1, the diversity gains of all the systems are same. When q is larger than or equal to 2 (q 2) and when both the number of transmit antennas N_t of the ST-UWB-IR system and the time interleaver size N_TI of the TI-UWB-IR system are larger than or equal to q (N_t, N_TI q), we show the following; both the ST-UWB-IR system and the TI-UWB-IR system achieve the same diversity gain that is constant regardless of N_t and N_TI. On the other hand, when q 2 and when N_t and N_TI < q, we show the following; when N_t = N_TI, the ST-UWB-IR system and the TI-UWB-IR system get the same diversity gain; when N_t > N_TI, the diversity gain of the ST-UWB-IR system is larger than that of the TI-UWB-IR system; and vice when N_t > N_TI.

In this proposed method, CRC coding is only applied to data unit, not to padded zeros that are to fill up the fixed length of data unit. So improved frame error rate and protocol efficiency are possible instead of increment of receiver complexity.

In this letter, Multicarrier Code Division Multiple Access (MC-CDMA) technique combined with Space Time Block Code (STBC) is analyzed in the case of two transmit antennas. A multicarrier transmit diversity scheme with antenna selection is proposed. The transmission power is allocated onto the antenna which has larger channel gain instead of the two antennas uniformly. Simulation results show that the new scheme has considerable performance gain compared to the Alamouti's scheme.

A scheme to evaluate the number of users and cell coverage of a WCDMA supporting multi-rate traffic is newly presented through calculation of the realizable Erlang capacity from a derived blocking probability and the path loss from the COST231 Walfisch-Ikegami (WI) model. Based on this analytical scheme, we evaluate the voice-data Erlang capacities at various data rates of 15 kbps to 480 kbps and the relationship between the cell coverage and the number of active users from them. When the value of E_b/I_o is low from 4 dB to 3 dB under voice user capacity of 50 Erlang at 8 kbps, the result shows that the data user capacity is increased to 10 Erlang at low rate of 15 kbps and the cell coverage is enlarged to 100 m, and it is also shown that its capacity is increased to 0.2 Erlang at high rate of 480 kbps and its coverage to 50 m.

In this letter, adaptive multi-stage parallel interference cancellation (PIC) receiver is considered for multi-rate DS-CDMA system. In each stage of the adaptive multi-stage PIC receiver, multiple access interference (MAI) estimates are obtained by the sub-bit estimates from the previous stage and the adaptive weights for the sub-bit estimates. The adaptive weights are obtained by minimizing the mean squared error between the received signal and its estimate through normalized least mean square (LMS) algorithm. It is shown that the adaptive multi-stage PIC receiver achieves smaller BER than the matched filter receiver, multi-stage PIC receiver, and multi-stage partial PIC receiver for the multi-rate DS-CDMA system in a Rayleigh fading channel.

In this letter, DAB system employing multiple antenna scheme is considered for high capacity transmission. In the MIMO-DAB system, the transmission rate is augmented efficiently by increasing the number of antennas, and thus the multi-channel estimation process is required. So, this letter proposes a new phase reference symbol structure which can be adapted to the multi-channel estimation, and investigates the channel estimation performance based on the semi-blind processing using the time-domain windowing. Simulation results have been shown that the MIMO-DAB system with multi-antennas can achieve the high-rate transmission for multimedia broadcasting.

In this letter, we propose an iterative decoding with LDPC based unitary matrix modulated OFDM with splitting the diagonal components over the coherence bandwidth. The proposed system can obtain a frequency diversity gain by splitting the diagonal components of unitary matrix modulated symbols, and also obtain large coding gain by using LDPC code.

Arrival of the packet data is acknowledged by detecting the preambles of the received bursty packets. To increase system throughput and reduce transmission latency, a simple and robust signal detection algorithm is required. Furthermore, the threshold for signal detection must be set adaptively for different channel conditions. In this paper, we design and compare three signal detection algorithms for bursty packet systems.

We propose a novel handover initiation algorithm based on available bit rate and timing constraint criterion for multimedia capable cellular systems. Computer simulations are performed to evaluate the handover rate and handover initiation delay. Numerical results show that handover must be initiated at different positions for different services to maintain the required quality of service requirements.

Transmission performance of a multi-input multi-output (MIMO) antenna system (i.e., antenna diversity, adaptive antenna array, and space division multiplexing) highly depends on the arrival angle distribution of propagation paths. In this letter, a multipath fading simulator based on distributed scattering model is presented. The impacts of the arrival angle distribution of propagation paths on the bit error rates (BER) performance are measured using an implemented fading simulator and compared with the theoretically predicted performance.

We propose a new output arbitration method for an input buffered switch with a buffered crossbar. In the proposed method, each output selects the first nonempty buffer from the starting point. The starting points of output are determined to minimize the synchronization phenomenon that more than one input module sends cells destined for a same output. Using an approximate analysis of the synchronization phenomenon, we show the uniqueness of the starting points improves the switch performance. Finally, using computer simulations, we verify the proposed method outperforms the previous methods under the uniform and burst traffic.

This letter describes the design and performance of a new ATM Adaptation Layer (AAL-UDP) for time-critical traffic over wireless ATM networks. The key ideas in the design consist of no discard at the AAL level and header protection with sequence number mechanism. The UDP/IP header is repeated for reliability, because it contains the most important information such as address and port number. The simulation results show that the AAL-UDP provides significant improvement in throughput as well as in application-level performance compared to the conventional AAL 5 case.

We propose a new dynamic routing algorithm that uses traffic and network state information to minimize the blocking rate and link congestion level. Our scheme uses the currently available link capacity in calculating the link weight by modifying Wang's approach, and computes the shortest path when a new call comes into the network. We consider the blocking count based update mechanism and the timer based mechanism, and conclude that the former is better than the latter in terms of efficiency and complexity.

In finding the optimal solution of virtual-path bandwidth allocation for large-scale networks, existing searching algorithms frequently call the process which calculate the bandwidth for given call blocking probability (CBP) and traffic loads. This is an inverse process of calculating CBP for given traffic loads and bandwidth. Because there is no analytic expression of calculating CBP, the process of calculating bandwidth with given CBP and traffic adopts an iteration algorithm. It leads to a tedious computation process. In this letter, a fast bandwidth evaluation algorithm is proposed and applied to the field of virtual path bandwidth allocation that aims at minimizing the worst call blocking probabilities in the network. The algorithm is proved to be accurate and fast. Finally, we provide comparison curves for the exact optimal CBPs obtained in the case of using OPBM against that of DCLPBM aided by the fast bandwidth evaluation algorithm.

Many algorithms have been introduced to obtain giga-bit routing performance by reducing searching time. As most of them, however, have not considered the importance of update time and memory requirement seriously, they couldn't work well in real networks. We propose a flexible and fast IP lookup algorithm, named FFILA, considering these factors and compare the performance of our scheme with that of the conventional scheme of Patricia trie.

In this paper the correlation spectrum of antenna array is introduced. Based on the relationship between the correlation spectrum and space spectrum of MUSIC, we proposed a novel approach to improve the DOA estimation by arranging the linear antenna array elements using genetic algorithm (GA) in optimizing the correlation spectrum. The DOA estimation performance of the optimized array is validated by Monte Carlo simulation and Cramer-Rao bound (CRB), which are improved compared with that of the traditional uniform linear array and the Minimum-Redundancy array (MRA).

Matching Pursuits (MP), a technique for signal decomposition using a dictionary of functions, is applied to ground penetrating radar (GPR) signals in order to remove noise and clutter included in the signals and to extract target responses. A wave-based dictionary composed of wavefronts and resonances is employed. Noise reduction performance and the removal of ground-surface reflection are evaluated through numerical simulations. The results show that the MP approach performs well and offers an effective method for feature extraction from GPR signals.