This paper outlines and analyses major P2P technologies currently being actively studied at various research organisations or already in use in both business and personal environments. Various technical features of P2P as well as its history and market trends are shown. P2P principles focussing on discovery and network-organisational mechanisms of major P2P systems such as JXTA, SIONet, Freenet, and Chord are first summarised. Also, other technical issues related to delivery, network control, security, digital rights management, and distributed computing are described. Then, P2P technologies to promote next-generation community networks with autonomous and intelligent characteristics are addressed. Based on these, P2P applications are analysed and some examples are shown. Studies of P2P traffic measurement, which is difficult by using traditional evaluation methods, are taken up to consider total network design in the P2P era.

Density evolution has recently been used to analyze the iterative decoding of Low Density Parity Check (LDPC) codes, Turbo codes, and Serially Concatenated Convolutional Codes (SCCC). The density evolution technique makes it possible to explain many characteristics of iterative decoding including convergence of performance and preferred structures for the constituent codes. While the analytic density evolution methods were applied to LDPC codes, the simulation based density evolution methods were used for Turbo codes and SCCC due to analytic difficulties. In this paper, several density evolution ideas in the literature are used to analyze common code structures and it is shown that those ideas yield consistent results. In order to do that, we derive expressions for density evolution of SCCC with a simple 2-state constituent code. The analytic expressions are based on the sum-product and min-sum algorithms, and the thresholds are evaluated for both message passing algorithms. Particularly, for the min-sum algorithm, the density evolution with Gaussian approximation is derived and used to analyze the effect of scaling soft information. The scaling of extrinsic information slows down the convergence of soft information or avoids an overestimation effect of it and results in better performance, and its gain is maximized in particular constituent codes. Similar approaches are made for LDPC code. We show that the scaling gain is noticeable in the LDPC code as well. This scaling gain is analyzed with both density evolution and simulation performance. The expected scaling gain by density evolution matches well with the achievable scaling gain from simulation results. These results can be extended to the irregular LDPC codes based on the degree distribution for the min-sum algorithm. All density evolution algorithms used in this paper are based on the Gaussian approximation for the exchanged messages.

The hybrid fiber coax (HFC) technology enables the conventional cable-television (CATV) network to provide subscribers with Internet access services. In this paper, we propose a new preemptive priority scheme (PPS) for IEEE 802.14 hybrid fiber coax (HFC) networks with the intelligent nodes (INs). The INs are placed between the headend controller and stations. By using INs, that stand for downstream subscribers to contend for the demand resources, the collision probability, and the collision resolving period can be reduced. In this paper, we further extend such network architecture to support multi-priority access. In each IN or individual station, the proposed PPS will prevent a higher priority request from colliding with requests of lower priority. Moreover, in PPS, the granted bandwidth for lower priority requests can be preempted by the waiting request with higher priority. This will speedup the channel capture by priority data. The efficiency of PPS is investigated by simulations. Simulation results show that by adopting INs with PPS to be an agent for subscribers can not only shorten the collision resolving period but also minimize the average request delay of priority data.

Recently, there have been a number of studies on wavelength division multiplexing (WDM) systems designed to increase transmission capacity and flexibility. This has in turn made it necessary to design the optical loss for optical fiber networks over a wide wavelength range. However, there is no general optical loss design method that allows us to employ for an optical fiber network over a wide wavelength range. In this paper, we propose a novel optical loss design and estimation method based on multiple regression analysis and statistical loss design technology for designing new optical fiber networks and for upgrading those that have already been installed. We show that the method can be used for highly accurate optical loss design and estimation from the O- to the L-band (1260 nm to 1625 nm), except for the E-band, by using easily obtainable optical loss values at wavelengths of 1310 and 1550 nm, which are usually used for communication. Moreover, we also show that the accuracy of our proposed method can be improved by selecting proper additional predictor wavelengths.

All-optical switching is of considerable interest, since it enables the construction of large-capacity networks with protocol- and bit-rate-independent transmission. In this paper, we determine the most desirable of three all-optical architectures for a backbone network, by comparing the following architectures: the wavelength-routed network, the slotted wavelength-routed network, and the optical burst switching network. After proposing an optical path accommodation algorithm that minimizes the total fiber length, we evaluate the total network cost in order to compare the availability of the first two architectures. We then compare the architectures in terms of the burst blocking probability in order to clarify the effectiveness of the third architecture.

We have proposed and theoretically verified a 2R (reshaping and regeneration) limiter circuit using continuous wave (CW) holding beam for cross-gain modulation (XGM) wavelength converter, through simulation. The gain clamping effect of semiconductor optical amplifier (SOA), which is caused by CW holding beam injected into SOA, was used to obtain the accurate optical gain and phase conditions for SOA's in 2R limiter circuit. XGM wavelength converter with the proposed 2R limiter circuit provides higher extinction ratio (ER) as well as more enhanced operation speed than any other wavelength converter. Our numerical results show that after the wavelength-converted signal from XGM wavelength converter passed the 2R limiter circuit, it was re-inverted with the improved ER of 30 dB at 5 Gb/s. In case of high-speed operation, great enhancement to decrease power penalty of about 12 dB was shown at 10 Gb/s.

The operation of a polarization mode dispersion (PMD) compensator using a polarizer and a Faraday rotator-based polarization controller (FRPC) is analyzed in detail, and the compensation performance is experimentally evaluated in 40 Gbit/s operation. The evaluation results show that a wide range of differential group delay over a bit period can almost be completely compensated using the PMD compensator. The characteristics of electrical spectrum-based signal monitoring methods are investigated in detail, and the results shows advantages of a low frequency band monitoring method that produces about double the wider dynamic range than a fundamental repetition frequency monitoring method. The automated PMD compensator using a polarizer and a FRPC driven by the low frequency band monitoring method is experimentally investigated using a terrestrial 40 Gbit/s wavelength division multiplexing system involving 350-km installed single-mode fibers. The PMD compensator produces highly stable signal performance in the field environment for a long term and reduces the standard deviation of the Q-factor distribution.

The block-based fast transmission scheme, which is one of typical stored video delivery schemes, is reasonable in terms of its bandwidth efficiency and tolerance to the delay jitter, etc. However, it causes packet loss because of its burst data transmission method. Thus, we propose a slotted multicast scheme for MPEG video based on the block transmission scheme to maintain a higher quality and to include time constraints. We define two delivery units, the "GoPs Group" and the "Frame Type," on the basis of the MPEG characteristics with periodical NACK feedback from the clients. The former is tolerant to burst packet loss, and the latter gives priority to important frames. Our block multicast has two phases: a "Transmission Phase" and a "Retransmission Phase." In the former, a server multicasts a block, and in the latter, a server retransmits lost packets using multicast according to the proper delivery unit. We evaluate our proposal from some viewpoints with a computer simulation. We also measure the quality of the video reflected the result of a computer simulation. From these results, we confirm performance effectiveness of our proposal.

In this paper, we propose a novel traffic engineering architecture for IP networks with MPLS backbones. In this architecture, two link-disjoint label switched paths, namely the primary and secondary paths, are established among every pair of IP routers located at the edges of an MPLS backbone network. As the main building block of this architecture, we propose that primary paths are given higher priority against the secondary paths in the MPLS data plane to cope with the so-called knock-on effect. Inspired by the ABR flow control mechanism in ATM networks, we propose to split traffic between a source-destination pair between the primary and secondary paths using explicit rate feedback from the network. Taking into consideration the performance deteriorating impact of packet reordering in packet-based load balancing schemes, we propose a traffic splitting mechanism that operates on a per-flow basis (i.e., flow-based multipath routing). We show via an extensive simulation study that using flow-based multipath traffic engineering with explicit rate feedback not only provides consistently better throughput than that of a single path but is also void of out-of-order packet delivery.

This paper proposes a bandwidth allocation algorithm and a demand accommodation algorithm guaranteeing utility max-min fairness under bandwidth constraints. We prove that the proposed algorithms can fairly split network resources among connections and achieve call admission control considering the fairness among different types of applications. We then formulate three different network design problems to maximize the total utility of all customers, the number of users accommodated in the network, and the average utility of the customers accommodated in the network. To solve the problems, we extend the conventional network design algorithms considering utility max-min fair share, and numerically evaluate and compare their performance. Finally, we summarize the best algorithms to design the utility max-min fair share networks considering the operation policy of network providers.

The window flow control based end-to-end TCP congestion control may cause unfair resource allocation among multiple TCP connections with different RTTs (round trip times) at a bottleneck link. In this paper, in order to improve this unfairness, we propose the active ECN which is an ECN based active queue mechanism (AQM). A bottleneck router with the proposed mechanism marks TCP segments with a probability which depends on the RTT of each connection. By enabling the TCP senders to reduce their transmission rate when their packets are marked, the proposed mechanism can realize the same transmission rate among TCP connections with different RTTs. Furthermore, the active ECN can directly mark ACKs from TCP receivers, while the conventional ECN marks TCP segments coming from the TCP senders. As a result, the queue length distribution at the bottleneck link gets stabilized, because the sender can quickly react to the marking according to variation of the queue length.

A multi-hop wireless local area network (LAN) is an ad-hoc wireless network that connects to the Internet backbone via an access point. Routing paths between mobile hosts and a fixed host can be divided into two sub-paths, wireless and wired. In this paper, we apply the Hierarchical Routing Tree (HRT) concept to finding wireless sub-paths. That is, by constructing an HRT, each mobile host can find a routing path to an access point (i.e., the HRT root) quickly and thus gain the access to Internet. In addition, we choose the Ad-hoc On-demand Distance Vector (AODV) routing protocol as a point-to-point routing method for sources and destinations located in the same ad-hoc network in order to improve upon a weakness in the HRT method. Numerical experiments are given to show the effectiveness of the hybrid routing method.

We propose in this paper the integrated power and admission control (IPAC) algorithm for multimedia CDMA wireless networks. The distributed power control with partial power assignment (DPC-PPA) algorithm, which can speed up the process in finding a feasible power set, is chosen as the power control algorithm for the IPAC algorithm. We prove that given a feasible configuration, the power set of the DPC-PPA algorithm will converge to a feasible power set which achieves equality in a set of linear equations. After a pre-specified number of iterations of power control, if the QoS requirements are not all satisfied, the IPAC algorithm uses the admission control procedure to reduce the outage probability. We derive an admission criterion, which gives a necessary and sufficient condition for a group of users to be admissible for the network, for the admission control procedure. Furthermore, we also found from simulation results that the IPAC algorithm results in smaller outage probability than the distributed power control algorithms without admission control.

For an efficient software download in cellular CDMA systems, location dependant session admission control (LDSAC) is presented. In the LDSAC scheme, a mobile that is located near cell center can request software download session, but the mobile that is located far from cell center can request session only after approaching near the cell center. Performance is analyzed in terms of handoff rate, mean channel holding time, session blocking probability and handoff forced termination probability. Numerical results show handoff rate between cells in the proposed scheme is reduced to 30-250% compared to conventional scheme, according to traffic characteristics such as terminal speed, session duration time and the size of the allowable zone area in a cell for the initiation of the session. And new session blocking probability decreases slightly, but handoff session forced termination probability decreases drastically.

In mobile cellular networks it is crucial to be able to use the available radio spectrum as efficiently as possible while providing a certain level of Quality of Service (QoS) for the users. Emergence of miscellaneous services has dramatically increased the complexity of this problem by creating a heterogeneous traffic environment. In this paper an efficient resource allocation scheme between two real-time services with different bandwidth requirements has been proposed for cellular networks with multimedia offered traffic and highly mobile users, which combines classical intera-cell resource borrowing concept with a novel inter-cell resource-sharing scheme between diverse classes of traffic. By assuming the heterogeneous offered traffic to be a combination of audio and video traffic types, through extensive simulations it will be shown that HCBA-UCB is capable of significantly improving audio teletraffic performance of the system while preserving fairness in service provision, i.e. without imposing additional expenses upon video QoS performance.

This paper proposes a SNR estimation scheme based on Gibbs sampler. This scheme can estimate SNR using a very short received sequence, and does not require any prior information of the transmitted symbol. Compared with the existing estimators, the performance of this method is better when real SNR is larger than 5 dB in both single path channel and multi-path channel.

This paper discusses the implementation of multi-band digital intermediate frequency (IF) for wideband CDMA (W-CDMA) transceiver. The majority of the implemented module in hardware is composed of wideband analog-to-digital converters (ADCs), digital-to-analog converters (DACs), and field programmable-gated-arrays (FPGA). And in software, it is coded by VHSIC hardware description language (VHDL) for realizing digital filters and numerically controlled oscillator, etc. To cope with the hardware limitation such as the number of gates in FPGA, the overall digital filter embedded in transceiver is constructed via a cascading a series of decimation and interpolation filters. At transmitter, in order to upconvert the multi-band baseband channels simultaneously, two-stage digital complex quadrature modulation (DCQM) is utilized. The relevant up-and-down conversion of the numerically controlled oscillator (NCO) is designed in the form of a look-up-table (LUT), having samples associated with a sampled sinusoidal with period of 1/4. At receiver, to avoid the usage of surface acoustic wave (SAW) filter, the high-performance digital filter is implemented subject to satisfying band rejection ratio prescribed in blocker and adjacent channel specification. This paper provides the performance of the implemented digital IF module by revealing the results taken from the measurement instruments. Moreover, to confirm its validity computer simulations are simultaneously conducted.

To improve the DS-CDMA signal transmission performance in a frequency-selective fading channel, the frequency-domain equalization (FDE) can be applied, in which simple one-tap equalization is carried out on each subcarrier component obtained by fast Fourier transform (FFT). Equalization weights for joint FDE and antenna diversity combining based on maximal ratio combining (MRC), zero-forcing (ZF), and minimum mean square error (MMSE) are derived. The conditional bit error rate (BER) is derived for the given set of channel gains in a frequency-selective multipath fading channel. The theoretical average BER performance is evaluated by Monte-Carlo numerical computation method using the derived conditional BER and is confirmed by computer simulation. Performance comparison between DS- and multi-carrier (MC)-CDMA both using FDE is also presented.

Supporting quality of service (QoS) capabilities for multi-media applications is one of the major issues in medium access control (MAC) research. In distributed contention-based MAC algorithms, it is a challenging task to support the desired QoS because of the inherent random access characteristics. In this paper, we propose an efficient prioritized fast collision resolution algorithm. The MAC protocol with this new algorithm attempts to provide significantly high throughput performance for data services and support QoS for real-time services. We incorporate the priority algorithm based on service differentiations with the fast collision resolution algorithm, and show that this algorithm can simultaneously achieve high throughput and good QoS support for real-time and data services.

In this paper, we evaluate the performance of single- and multi-antenna multi-carrier code division multiple access (MC-CDMA) downlink (base station to mobile terminal) systems in single- and multi-cell environments. We first propose a minimum mean square error (MMSE) filter with a Gaussian approximation for a single input single output (SISO) MC-CDMA downlink system. Then, we apply it to a SIMO (single input multiple output) system with a conventional turbo coding. Furthermore, we compare the performance of SISO (11) and SIMO (12) MC-CDMA systems with that of a multiple input multiple output (MIMO) (22) system employing space-time turbo coded modulation (STTuCM) in a multi-cell environment with 7 cells by computer simulation. Based on the computer simulation results, it is found that the considered MIMO system can achieve twofold capacity with the same transmission power in the multi-cell environment.

We analyze the performance of code division multiple access (CDMA) systems with orthogonal frequency division multiplexing (OFDM). We obtain the probability density function (pdf) of the multiple access interference (MAI) of CDMA systems and extend the results to OFDM-CDMA systems to determine the pdf of the MAI and inter-carrier interference (ICI) in terms of the number of users, the spreading length, the crosscorrelation of spreading sequences, the number of sub-carriers and the frequency offset. We consider the synchronous downlink of cellular multi-carrier CDMA systems and derive a Gaussian approximation of the MAI and ICI. The results show that the overall effect of frequency offset varies with system loading for a given crosscorrelation. The performance of OFDM-CDMA in frequency selective fading channels is analyzed in terms of the joint probability of the fading parameter in each sub-carrier.

While a multicarrier approach of achieving frequency diversity performs well in the presence of partial-band interference, it suffers from the effects of intermodulation distortion (IMD) due to power amplifier (PA) nonlinearity. On the other hand, transmit diversity using multiple transmit antennas has the benefit of no IMD effects, but can suffer from a larger performance degradation due to partial-band interference (e.g., jamming or narrowband signals in a overlay system) compared to the multicarrier approach. Hence, hybrid diversity schemes which use both multicarrier as well as multiple transmit antennas are of interest. Techniques to suppress IMD effects in such hybrid diversity schemes are important. In this paper, we propose and evaluate the performance of a minimum mean square error (MMSE) receiver to suppress the intermodulation distortion in a coded multicarrier multiple transmit antenna (P transmit antennas) DS-CDMA system with M subcarriers on each transmit antenna, for both BPSK and QPSK modulation. The system uses rate-1/M convolutional coding, interleaving and space-time coding. We compare the performance of a (M = 4,P = 2) scheme and a (M = 2,P = 4) scheme, both having the same diversity order. We show that the proposed MMSE receiver effectively suppresses the IMD effects, thus enabling to retain better antijamming capability without much loss in performance due to IMD effects.

This paper is concerned with the application of ray tracing method (RTM) to field analysis in bended and branched tunnels. On the line of sight, direct wave from a source and reflected waves at the tunnel walls are dominant compared with diffracted waves, but off the line of sight, diffraction can not be ignored especially beyond an abrupt bending. As a result, a detailed attention should be focused on the RTM analysis when dealing with propagation in the region off the line of sight. In this paper, we take into account of the diffraction rays which are originally reflection rays, next diffracted at a bending edge, and again converted to reflection rays. It is shown that numerical results are in good agreement with the experimental data.

This paper proposes a method to detect buildings and houses whose walls are not parallel to Synthetic Aperture Radar (SAR) flight path. Experimental observations show that it is difficult to detect these targets because of small backscattering characteristics. The detection method is based on the correlation coefficient in the circular polarization basis, taking full advantage of Polarimetric SAR (POLSAR) data. Since the correlation coefficient is real-valued for natural distributed targets with reflection symmetry and for non-natural targets orthogonal to illumination direction, and it becomes a complex number for non-natural targets aligned not orthogonal to radar Line-Of-Sight (LOS), the value seems to be an effective index for detection of obliquely aligned non-natural targets. The detection results are shown using the X-band Polarimetric and Interferometric SAR (Pi-SAR) single-path data set in conjunction with other polarimetric indices.

This paper presents an application-level QoS comparison of three inter-destination synchronization schemes: the master-slave destination scheme, the synchronization maestro scheme, and the distributed control scheme. The inter-destination synchronization adjusts the output timing among destinations in a multicast group for live audio and video streaming over the Internet/intranets. We compare the application-level QoS of these schemes by simulation with the Tiers model, which is a sophisticated network topology model and reflects hierarchical structure of the Internet. The comparison clarifies their features and finds the best scheme in the environment. The simulation result shows that the distributed control scheme provides the highest quality of inter-destination synchronization among the three schemes in heavily loaded networks, while in lightly loaded networks the other schemes can have almost the same quality as that of the distributed control scheme.

In providing video programs to a number of clients through networks, a broadcasting approach is more appropriate than a true-video-on-demand approach in efficient use of bandwidth. However, the broadcasting approach also needs excessively wide bandwidth if many video programs are to be transmitted. This study presents a very simple but novel architecture called dynamic channel broadcasting, for video-on-demand systems. The proposed architecture uses both static and dynamic broadcasting channels to improve the efficiency of channels. The proposed architecture eliminates the necessity of dividing each video into segments and switching channels frequently unlike the pyramid broadcasting, the skyscraper broadcasting and the harmonic broadcasting. Also this new architecture needs a smaller buffer size. The numerical results demonstrate that the newly proposed approach in some cases requires only 14% of the bandwidth required for the conventional broadcasting while maintaining the start-up latency.

In packet networks including the Internet and commercial 3G wireless bearers, the network states that a streaming media application experiences are not known a priori and exhibit time-varying characteristics. For such dynamic environments, network-adaptive techniques are essential to efficiently deliver video data. In this paper, we propose a frame-based optimal scheduling algorithm which incorporates a MAP (Maximum A Posteriori) framework for adapting to varying network loss rate. The optimal transmission schedule is determined such that effective frame-rate is maximized at playback. Also, for multiple packets per frame, frame-based selection of delivery order greatly reduces computational complexity for a server scheduler when compared with packet-based scheduling techniques. In addition, by dynamically estimating instantaneous packet loss probability, the proposed scheduler performs network-adaptive transmission for streaming video over time-varying packet networks. Simulation results for test video sequence show that the proposed scheduling algorithm outperforms conventional ARQ-based schemes from a view point of reconstructed video quality as well as playable frame-rate. In particular, the proposed scheduling algorithm exhibits significant improvements of frame-rate over highly lossy channels.

Unitary space-time modulation (USTM) is well-tailored for flat fading channels where neither the transmitter nor the receiver knows the fading coefficients. A systematic approach to designing USTM is proposed by Hochwald et al., which is not guaranteed full diversity and difficult to put into practice due to the large number of signal points in the two-dimensional constellation. Consequently, we modify the systematic method to search new unitary space-time coded 8PSK with full diversity. Simulation results indicate that the error performances of the searched codes are better than or at least the same as those of the USTM in [2].

In 2002, Hwang and Yeh showed that Peyravian-Zunic's password authentication schemes are not secure and proposed an improvement by using the server public key. Since applying the server public key results in the additional burden, we propose secure password authentication schemes without using the server public key in this paper.

One of the categories of decoding techniques for DFT codes in erasure channels is the class of iterative algorithms. Iterative algorithms can be considered as kind of alternating mapping methods using the given information in a repetitive way. In this paper, we propose a new iterative method for decoding DFT codes. It will be shown that the proposed method outperforms the well-known methods such as Wiley/Marvasti, and ADPW methods in the decoding of DFT codes in erasure channels.

This study proposes a feasible method to successfully improve probabilistic packet marking (PPM) used to trace back the original DoS attacker. PPM is modified by compensating for the remarked marked packets to achieve the optimal marked packets required for reconstructing the complete attack path.

A deniable authentication protocol is used to identify the source of a received message for a receiver, but the receiver is unable to prove to a third party the source of the received message. Recently, Fan et al. proposed a deniable authentication protocol based on Diffie-Hellman algorithm. In this paper, we show that Fan et al.'s protocol does not possess the deniable property as they claimed. A cheating receiver can prove the source of the received message to a third party. In addition, we also present a modification of Fan et al.'s protocol to overcome the security flaw.

This paper proposes a new FEC scheme using backup channel to send redundant information instead of piggybacking the main packet. This is particularly applicable to the modern IP networks which are distributed all over the world. In this method only one source coder for both the main and the redundant payload is used to reduce the overall computational complexity. The Gilbert loss model (GLM) is employed to verify the improvement of the packet loss probability in this new method compared with that in a single path FEC scheme. It is shown, through simulation results that using our proposed backup channel can considerably improve the packet loss and delay performance of the VoIP networks.

This letter proposes a hierarchical label-switched path (LSP) setup scheme, called ConSet, for multi-layer generalized multi-protocol label switching (GMPLS) networks. ConSet allows a Path message to be transmitted to the downstream neighbor node without waiting for the establishment of the higher-order LSP. Confirmation of the establishment of the higher-order LSP is performed at the ingress node of the higher-order LSP before a Resv message of the lower-order LSP is transmitted to the upstream neighbor node. ConSet is able to set up hierarchical LSPs faster than the sequential scheme.

In this letter, we address the problem of how to measure the link quality in terms of signal-to-interference ratio (SIR) in rapidly time varying channels which are due to fast variation of the interference or the bursty nature of the traffic. We propose a new SIR estimation approach for the TD-CDMA system, that is based on a pre-selection tentative decision device which discards and selects the estimated symbol on the basis of MAP ratio. The simulation results show that the proposed SIR estimator works and is superior to other known estimators in rapidly time varying channels.

In this study, a fourth-order cumulants based iterative algorithm for blind channel equalization is introduced, which is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum phase characteristic of the channel. The transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel inputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple reordering and scaling. In simulation studies, both a closed-form and a stochastic version of the proposed algorithm are tested with three-ray multi-path channels, and their performances are compared with the methods based on conventional second-order statistics and higher-order statistics (HOS) as well. Relatively good results with fast convergence speed are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

In this paper, we presented a new interference suppression method based on groupwise decorrelation for the multirate wideband-code division multiple access (W-CDMA) downlink. Code grouping in the proposed method is performed according to the correlation property between the mother code and the child code in the orthogonal variable spreading factor (OVSF) code tree. The decorrelation process based on the grouped codes, so called groupwise decorrelation, is then performed to suppress the interference induced in the downlink propagation. We demonstrate that the proposed method can enhance the performance significantly, with lower computational complexity and higher operational efficiency in which any information about interference users (code, data, amplitude) is not required to know in prior at the detection stage.

Efficient schemes to enhance the performance of the optimum beamforming for DS/CDMA systems are proposed. The main focus of the proposed schemes is to enhance the practical estimation of an array response vector used at the weight vector for the optimum beamforming. The proposed schemes for the performance enhancement of the optimum beamforming are the Complex Toeplitz Approximation (CTA) and the real Toeplitz-plus-Hankel Approximation (RTHA) which have the theoretical property of an overall noise-free signal. It is shown through several simulation results that the performance of the optimum beamforming using the proposed schemes is much superior to that of a system using the conventional method under several simulation environments, i.e., the number of users, the SNR value, the number of antenna elements, the angular spread, and Nakagami fading parameter.

We design a DAB system using space-time coding for reliable transmission in the multipath fading channel. This letter assess the performance evaluation of space-time coded DAB system. For channel estimation in the space-time coded DAB system, we introduce two training sequences and derive the mean square error of each training sequence. We then represent the average symbol error rate of the space-time coded DAB system for the specific cases of two, three, four transmit antennas.

In this letter, we enhance Complementary Code Keying (CCK) modulation with Pseudo-Periodic Sequence. It has been proved that the new Pseudo-Periodic CCK modulation is more efficient and robust fighting against multi-path interference. In order to support our new scheme, we design and implement the corresponding simulation. The in-depth analysis of the reason why Pseudo-Periodic Sequence can do a favor to CCK is also presented and emphasized.

A random transmit power control (TPC) is applied to DS-CDMA/TDD packet mobile radio, which controls the transmit power so as to intentionally vary the received signal power in order to obtain the large capture effect. The uplink capacity with the random TPC in a frequency-selective fading channel is evaluated by computer simulation. The simulation results show that the random TPC provides larger link capacity than slow TPC.

A full-duplex radio-on-fiber (ROF) transport system based on optical single sideband (SSB) modulation, wavelength-division-multiplexing (WDM) and optical add-drop multiplexing techniques is proposed and demonstrated. A 1.5-dB RF power degradation due to the chromatic dispersion was achieved by employing optical SSB modulation scheme in the system, in which resulting in low bit error rate (BER) and third order intermodulation distortion to carrier ratio (IMD₃/C) values. Such a proposed full-duplex ROF transport system is suitable for the long-haul microwave optical link.

In this paper, an experimental result of complex refractive index of soda-lime glass at 30-GHz obtained by transmission method is presented at first. Secondly, a simple empirical formula of complex refractive index of soda-lime glass over frequency range from 0.1-GHz to 1000-GHz is derived using the present experimental result together with data previously reported in literatures by various researchers.

Estimating the parameters of a statistical distribution from measured sample values forms an essential part of many signal processing tasks. K-distribution has been proven to be an appropriate model for characterising the amplitude of sea clutter. In this paper, a new method for estimating the parameters of K-Distribution is proposed. The method greatly lowers the computational requirement and variance of parameter estimates when compared with the existing non-maximum likelihood methods.