With the big success of 802.11 wireless networks, there have been many proposals addressing end-to-end QoS guarantees in 802.11 WLAN. However, we have found that current end-to-end QoS architectures lack of one or more important properties such as cross-layer interaction, end-to-end integration, reconfigurability and modularity. In this work, we present an end-to-end reference QoS architecture for 802.11 WLAN that encapsulates in an unifying fashion software-based QoS components (mechanisms, algorithms, services), proposed in the literature. To show the usefulness and correctness of the reference architecture, we present three case studies of end-to-end QoS architectures addressing different QoS requirements such as bandwidth and delay with different approaches such as differentiated services and integrated services. We will give an architectural comparison and performance evaluation of these architectures. We believe the reference QoS architectures can help QoS designers to understand the importance and the complexity of various QoS components during the design phase and thus choose these QoS components appropriately.

We propose a parametric packet-layer model for monitoring audio quality in multimedia streaming services such as Internet protocol television (IPTV). This model estimates audio quality of experience (QoE) on the basis of quality degradation due to coding and packet loss of an audio sequence. The input parameters of this model are audio bit rate, sampling rate, frame length, packet-loss frequency, and average burst length. Audio bit rate, packet-loss frequency, and average burst length are calculated from header information in received IP packets. For sampling rate, frame length, and audio codec type, the values or the names used in monitored services are input into this model directly. We performed a subjective listening test to examine the relationships between these input parameters and perceived audio quality. The codec used in this test was the Advanced Audio Codec-Low Complexity (AAC-LC), which is one of the international standards for audio coding. On the basis of the test results, we developed an audio quality evaluation model. The verification results indicate that audio quality estimated by the proposed model has a high correlation with perceived audio quality.

This paper describes non-reference objective quality evaluation for noise-reduced speech. First, a subjective test is conducted in accordance with ITU-T Rec. P.835 to obtain the speech quality, the noise quality, and the overall quality of noise-reduced speech. Based on the results, we then propose an overall quality estimation model. The unique point of the proposed model is that the estimation of the overall quality is done only using the previously estimated speech quality and noise quality, in contrast to conventional models, which utilize the acoustical features extracted. Finally, we propose a non-reference objective quality evaluation method using the proposed model. The results of an experiment with different noise reduction algorithms and noise types confirmed that the proposed method gives more accurate estimates of the overall quality compared with the method described in ITU-T Rec. P.563.

In this paper, we discuss QoE (Quality of Experience) requirements for MVV (Multi-View Video) and audio transmission over IP networks and study the effect of the playout buffering time, contents and viewpoint change interfaces on the QoE and user's behavior. Unlike previous works, which mainly discuss MVV transmission from aspects of video codecs, we study MVV and audio transmission under various IP traffic and delay conditions by experiment. We compare two schemes: a scheme that the user watches from a single viewpoint and the one that he/she can choose one viewpoint from many ones. As a result, we show that the users prefer the scheme where they can choose one viewpoint from many ones. We have found that when using proper buffering time, the users feel faster viewpoint changes; it improves their satisfaction compared to that when they watch on a single viewpoint. We have also noticed that the user pays more attention to the degradation of the video when watching on a single viewpoint. We have observed that the users tend to change the viewpoint more frequently in light traffic and low delay.

This paper proposes a cross-layer packet scheduling scheme for QoS support in audio-video transmission with IEEE 802.11e HCCA and assesses application-level QoS and QoE of the scheduling scheme under lossy channel conditions. In the proposed scheme, the access point (AP) basically allocates transmission opportunity (TXOP) for each station in a service interval (SI) like the reference scheduler of the IEEE 802.11e standard, which is referred to as the TGe scheme in this paper. In the proposed scheme, however, the AP calculates the number of MAC service data units (MSDUs) arrived in an SI, considering the inter-arrival time of audio samples and that of video frames, which are referred to as media units (MUs), at the application layer. The AP then gives additional TXOP duration in the SI to stations which had audio or video MAC protocol data units (MPDUs) in their source buffers at the end of the previous TXOP. In addition, utilizing video frame information from the application layer, we propose video frame skipping at the MAC-level of a source station. If a station fails to transmit a video MPDU, it drops all the following video MPDUs in the source buffer until the next intra-coded frame comes to the head of the buffer. We compare the reference scheduler (TGe scheme), the proposed packet scheduling scheme with and without the video frame skipping at the source in terms of application-level QoS and QoE. We discuss the effectiveness of the proposed packet scheduling scheme from a viewpoint of QoE as well as QoS. Numerical results reveal that the proposed packet scheduling scheme can achieve higher quality than the TGe scheme under lossy channel conditions. We also show that the proposed scheduling scheme can improve the QoS and QoE by using the video frame skipping at the source. Furthermore, we also examine the effect of SI on the QoS and QoE of the proposed packet scheduling scheme and obtain that the appropriate value of SI is equal to the inter-arrival time of video frame.

VoIP service is expected as one of the key applications of Mobile WiMAX, but the speech quality of VoIP service often suffers deterioration due to the fluctuating transmission delay called jitter. This is commonly ameliorated by a de-jitter buffer, and we aim to find the optimal size of de-jitter buffer to achieve speech quality comparable to PSTN. We developed a new model of the packet drops at the de-jitter buffer and the end-to-end packet delay which takes account of the additional delay introduced by the WiMAX power-saving mode. Using our model, we analyzed the optimal size of the de-jitter buffer for various network parameters, and showed that the results obtained by analysis accord with simulation results.

In differentiated services, packet classification is used to categorize incoming packets into multiple forwarding classes based on pre-defined filters and make information accessible for quality of service. Although numerous algorithms have presented novel data structures to improve the search performance of packet classification, the performance of these algorithms are usually limited by the characteristics of filter databases. In this paper, we use a different approach of filter preprocessing to enhance the search performance of packet classification. Before generating the searchable data structures, we cluster filters in a bottom-up manner. The procedure of the filter clustering merges filters with high degrees of similarity. The experimental results show that the technique of filter clustering could significantly improve the search performance of Pruned Tuple Space Search, a notable hash-based algorithm. As compared to the prominent existing algorithms, our enhanced Pruned Tuple Space Search also has superior performance in terms of speed and space.

With the increasing demand for services provided by communication networks, quality and reliability of such services as well as confidentiality of data transfer are becoming ones of the highest concerns. At the same time, because of growing hacker's activities, quality of provided content and reliability of its continuous delivery strongly depend on integrity of data transmission and availability of communication infrastructure, thus on information security of a given IT landscape. But, the amount of resources allocated to provide information security (like security staff, technical countermeasures and etc.) must be reasonable from the economic point of view. This fact, in turn, leads to the need to employ a forecasting technique in order to make planning of IT budget and short-term planning of potential bottlenecks. In this paper we present an approach to make such a forecasting for a wide class of information security related incidents (ISRI) -- unambiguously detectable ISRI. This approach is based on different auto regression models which are widely used in financial time series analysis but can not be directly applied to ISRI time series due to specifics related to information security. We investigate and address this specifics by proposing rules (special conditions) of collection and storage of ISRI time series, adherence to which improves forecasting in this subject field. We present an application of our approach to one type of unambiguously detectable ISRI -- amount of spam messages which, if not mitigated properly, could create additional load on communication infrastructure and consume significant amounts of network capacity. Finally we evaluate our approach by simulation and actual measurement.

In this letter, we propose distorted scenes enhancement algorithm in order to provide end user perceptual QoE-guaranteed IPTV service. The block edge detection with weight factor and partition-based local color values method can be applied for the degraded video frames which are affected by network transmission errors such as out of order, jitter, and packet loss to improve QoE efficiently. Based on the result of quality metric after using the distorted scenes enhancement algorithm, the distorted scenes have been restored better than others.

In this letter, we propose a Throughput-aimed MAC Protocol with Quality of Service (QoS) provision (T-MAC) for cognitive Ad Hoc networks. This protocol operates based on the Time Division Multiple Access (TDMA) slot assignments and the power control mechanism, which can improve the QoS provision and network throughput. Our simulation results show that the T-MAC protocol can efficiently increase the network throughput and reduce the access delay.

In Vehicular Ad hoc Networks (VANETs), general purpose ad hoc routing protocols such as AODV cannot work efficiently due to the frequent changes in network topology caused by vehicle movement. This paper proposes a VANET routing protocol QLAODV (Q-Learning AODV) which suits unicast applications in high mobility scenarios. QLAODV is a distributed reinforcement learning routing protocol, which uses a Q-Learning algorithm to infer network state information and uses unicast control packets to check the path availability in a real time manner in order to allow Q-Learning to work efficiently in a highly dynamic network environment. QLAODV is favored by its dynamic route change mechanism, which makes it capable of reacting quickly to network topology changes. We present an analysis of the performance of QLAODV by simulation using different mobility models. The simulation results show that QLAODV can efficiently handle unicast applications in VANETs.

The mobility control of mobile nodes can be an alternative to the transmitting power adjustment in case that fixed transmitting power is just used in the topology control. Assuming the controllable mobility of nodes, we propose four distributed mobility control algorithms assuring the network connectivity and the capacity improvement. We compare the throughput of each algorithm with the widely accepted capacity scale law considering the energy consumption. The proposed mobility-based topology control algorithms are named according to its operational characteristics; RP (Rendezvous Point), NNT (Nearest Neighbor Tracking), DM (Diffusion Model), and GP (Grid Packing). Through extensive simulations, we show that all the proposed algorithms successfully change a partitioned random network topology into a connected network topology without the power control. Furthermore, the topology reconfigured by the mobility control has the improved network capacity beyond that of the initial network. In the newly defined performance metric, effective capacity, the simulation results show that GP provides more improved and stable performance over various node densities with the short completion time.

One of the important technologies for the Future Internet is the delay-tolerant network, which enables data transfers even when mobile nodes are connected intermittently. Routing algorithms for a delay-tolerant network generally aim to increase the message delivery rate and decrease the number of forwarded messages in the situation of an intermittent connection. A fame-based strategy for delay-tolerant network routing is suggested in this work. The number of contacts of a node with other nodes, known as the fame degree in this work, is counted to rank the fame degree of the node. By utilizing the fame degree, the proposed routing algorithm determines the probability of forwarding the messages of a node to the contact node. Due to the characteristics of the proposed algorithm, it can be combined harmonically with the PROPHET routing algorithm. Through experiments on well-known benchmark datasets, the proposed algorithms shows better delivery rates with much lower number of forwarded messages and lower average hop counts of delivered messages compared to Epidemic, PROPHET and SimBet.

Broadband wireless access networks are promising technology for providing better end user services. For such networks, designing a scheduling algorithm that fairly allocates the available bandwidth to the end users and maximizes the overall network throughput is a challenging task. In this paper, we develop a centralized fair scheduling algorithm for IEEE 802.16 mesh networks that exploits the spatio-temporal bandwidth reuse to further enhance the network throughput. The proposed mechanism reduces the length of a transmission round by increasing the number of non-contending links that can be scheduled simultaneously. We also propose a greedy algorithm that runs in polynomial time. Performance of the proposed algorithms is evaluated by extensive simulations. Results show that our algorithms achieve higher throughput than that of the existing ones and reduce the computational complexity.

The dynamic channel allocation (DCA) scheme in multi-cell systems causes serious inter-cell interference (ICI) problem to some existing calls when channels for new calls are allocated. Such a problem can be addressed by advanced centralized DCA design that is able to minimize ICI. Thus, in this paper, a centralized DCA is developed for the downlink of multi-cell orthogonal frequency division multiple access (OFDMA) systems with full spectral reuse. However, in practice, as the search space of channel assignment for centralized DCA scheme in multi-cell systems grows exponentially with the increase of the number of required calls, channels, and cells, it becomes an NP-hard problem and is currently too complicated to find an optimum channel allocation. In this paper, we propose an ant colony optimization (ACO) based DCA scheme using a low-complexity ACO algorithm which is a kind of heuristic algorithm in order to solve the aforementioned problem. Simulation results demonstrate significant performance improvements compared to the existing schemes in terms of the grade of service (GoS) performance and the forced termination probability of existing calls without degrading the system performance of the average throughput.

For multiple-input multiple-output (MIMO) precoded transmission that has individual constraints on the maximum power of each transmit antenna or a subset of transmit antennas, the transmit power optimization problem is a non-linear convex optimization problem with a high level of computational complexity. In this paper, assuming the use of the interior point method (IPM) to solve this problem, we propose two efficient techniques that reduce the computational complexity of the IPM by appropriately setting its parameters. Based on computer simulation, the achieved reductions in the level of the computational complexity are evaluated using the proposed techniques for both the fairness and the sum-rate maximization criteria assuming i.i.d Rayleigh fading MIMO channels and block diagonalization zero-forcing as a multi-user MIMO (MU-MIMO) precoder.

This paper presents a pilot-aided channel estimation method which is particularly suitable for mobile WiMAX 802.16e Downlink Partial Usage of Subchannel mode. Based on this mode, several commonly used channel estimation methods are studied and the method of least squares line fitting is proposed. As data of users are distributed onto permuted clusters of subcarriers in the transmitted OFDMA symbol, the proposed channel estimation method utilizes these advantages to provide better performance than conventional approaches while offering remarkably low complexity in practical implementation. Simulation results with different ITU-channels for mobile environments show that depending on situations, enhancement of 5 dB or more in term of SNR can be achieved.

This study focuses on system throughput by taking into account the channel interference in IEEE 802.15.3c WPAN, which is based on the hybrid multiple access of CSMA/CA and TDMA, namely CSMA/CA-TDMA. To study the system throughput, we construct a novel analytical model by taking into consideration the channel interference caused by the hidden networks in CSMA/CA-TDMA. The obtained results show that the system throughput achieved by TDMA is highly affected by frame transmission in CSMA/CA. Furthermore, we show that channel interference, which causes a degradation in the system throughput, is a very significant problem in the IEEE 802.15.3c WPAN.

Although the topic of multiple-input multiple-output (MIMO) based orthogonal frequency division multiplexing (OFDM) over different fading channels is well investigated, its closed form symbol error rate (SER) expressions and performance results employing orthogonal space time block codes (OSTBCs) over uncorrelated frequency-selective Nakagami-m fading channels are still not available. The closed form expressions are extremely useful for evaluating system's performance without carrying out time consuming simulations. Similarly, the performance results are also quite beneficial for determining the system's performance in the sense that many practical wireless standards extensively employ MIMO-OFDM systems in conjunction with M-ary quadrature amplitude modulation (M-QAM) constellation. This paper thus, derives exact closed form expressions for the SER of M-ary Gray-coded one and two dimensional constellations when an OSTBC is employed and N_t transmit antennas are selected for transmission over frequency-selective Nakagami-m fading channels. For this purpose, first an exact closed-form of average SER expression of OSTBC based MIMO-OFDM system for M-ary phase shift keying (M-PSK) using traditional probability density function (PDF) approach is derived. We then compute exact closed form average SER expressions for M-ary pulse amplitude modulation (M-PAM) and M-QAM schemes by utilizing this generalized result. These expressions are valid over both frequency-flat and frequency-selective Nakagami-m fading MIMO channels and can easily be evaluated without using any numerical integration methods. We also show that average SER of MIMO-OFDM system using OSTBC in case of frequency-selective Rayleigh fading channels remains independent to the number of taps, L of that fading channel and the performance of the same system for two-tap un-correlated Rayleigh and Nakagami-m fading channels is better than that of the correlated one. Moreover, Monte Carlo simulation of MIMO-OFDM system using multiple transmit and receive antennas for different modulations is presented to validate our theoretical results. Finally, due to availability of closed form expressions, we further provide the performance results of MIMO-OFDM system over frequency-selective Nakagami-m fading channels employing (M-QAM) using OSTBCs under the transmission rate equal to 1, 2 and 3 bit(s)/s/Hz, respectively.

A diversity scheme with Fractional Sampling (FS) in an OFDM receiver has been investigated recently. Through FS, it is possible to separate multipath components and obtain diversity gain in OFDM systems. Enlargement of the bandwidth of the total frequency response between transmit and receive baseband filters allows the FS scheme to achieve path diversity. However, the transmit filter has to be designed according to the spectrum mask of the wireless standards such as IEEE802.11a/g to avoid interference to the other communication systems and the frequency response of the composite channel including the transmit and receive filters has often been set to minimal bandwidth to eliminate adjacent channel signals. In order to achieve the maximum signal-to-noise ratio (SNR), the same filter is commonly used in the transmitter and the receiver. In this paper, the trade-off among the SNR deterioration, adjacent channel interference, and the diversity gain due to the enlargement of the bandwidth of the receive filter is investigated. Numerical results from computer simulations indicate that the BER performance with wider bandwidth in the receiver shows better performance than that with the minimal bandwidth for maximizing the SNR in certain conditions.

The CSMA/CA driven MAC protocols withhold packet transmissions from exposed stations when they detect carrier signal above a certain threshold. This is to avoid collisions at other receiving stations. However, this conservative scheme often exposes many stations unnecessarily, and thus minimizes the utilization of the spatial spectral resource. In this paper, we demonstrate that remote estimation of the status at the active receivers is more effective at avoiding collisions in wireless networks than the carrier sensing. We apply a new concept of the interference range, named as n-tolerant interference range, to guarantee reliable communications in the presence of n (n≥ 0) concurrent transmissions from outside the range. We design a distributed interference preventive MAC ( IP-MAC ) using the n-tolerant interference range that enables parallel accesses from the noninterfering stations for an active communication. In IP-MAC, an exposed station goes through an Interference Potentiality Check (IPC) to resolve whether it is potentially interfering or noninterfering to the active communication. During the resolve operation, IPC takes the capture effect at an active receiver into account with interfering signals from a number of possible concurrent transmissions near that receiver. The performance enhancement offered by IP-MAC is studied via simulations in different environments. Results reveal that IP-MAC significantly improves network performance in terms of throughput and delay.

Free probability theory, which has become a main branch of random matrix theory, is a valuable tool for describing the asymptotic behavior of multiple systems, especially for large matrices. In this paper, using asymptotic free probability theory, a new cooperative scheme for spectrum sensing is proposed, which shows how the asymptotic free behavior of random matrices and the property of Wishart distribution can be used to assist spectrum sensing for cognitive radio. Simulations over Rayleigh fading and AWGN channels demonstrate the proposed scheme has better detection performance than the energy detection techniques and the Maximum-minimum eigenvalue (MME) scheme even for the case of a small sample of observations.

Previously, we have proposed an access point (AP) allocation algorithm in indoor environments for the Wireless Internet-access Mesh NETwork (WIMNET) using one gateway (GW) to the Internet. WIMNET consists of multiple APs that are connected wirelessly mainly by the Wireless Distribution System (WDS), to expand the coverage area inexpensively and flexibly. In this paper, we present two extensions of this algorithm to enhance the applicability to the large-scale WIMNET. One is the multiple GW extension of the algorithm to increase the communication bandwidth with multiple GWs, where all the rooms in the network field are first partitioned into a set of disjoint GW clusters and then, our previous allocation algorithm is applied to each GW cluster sequentially. The APs in a GW cluster share the same GW. The other is the dependability extension to assure the network function by maintaining the connectivity and the host coverage, even if one link/AP fault occurs, where redundant APs are added to the AP allocation by our previous algorithm. The effectiveness of our proposal in terms of the number of APs and the throughput is verified through simulations using the WIMNET simulator.

A novel analysis model for post-wall waveguide T-junctions is proposed. Equivalent solid-walls for the post-walls to have equal guided wavelength are corrected in the analysis model so that the wall thickness for the coupling windows is set to the difference in the width between the post-wall and the solid-wall waveguides. The accuracy of the proposed model is confirmed by comparing it to an HFSS analysis for the real structure of the post-wall waveguide T-junction including the post surfaces. 61.25 GHz model antennas are fabricated for experimental verification. The reflection of the antenna designed by the modified analysis model is suppressed to below -15 dB over a 5.6 GHz bandwidth, while that in the antenna designed by the conventional model is larger than -15 dB around the design frequency.

Future high-resolution short-range automotive radar will have a higher false alarm probability than the conventional low-resolution radar has. In a high-resolution radar, the reception signal becomes sensitive to the difference between intended and unintended objects. However, automotive radars must distinguish targets from background objects that are the same order of size; it leads to an increase in the false alarm probability. In this paper, a CFAR circuit for obtaining the target mean power, as well as the background mean power, is proposed to reduce the false alarm probability for high-resolution radars working in automotive environments. The proposed method is analytically evaluated with use of the characteristic function method. Spatial correlation is also considered in the evaluation, because the sizes of the both target and background objects approach the dimension of several range cells. Result showed the proposed CFAR with use of two alongside range cells could reduce the ratio of 6.4 dB for an example of an automotive situation.

We propose a simplified model of real-time joint source-channel coding, which can be used to adaptively determine the quality-optimal code rate of forward error correction (FEC) coding. The objective is to obtain the maximum video quality in the receiver, while taking time-varying packet loss into consideration. To this end, we propose a simplified model of the threshold set of the residual video packet loss rate (RVPLR). The RVPLR is the rate of residual loss of video packets after channel decoding. The threshold set is defined as a set of discrete RVPLRs in which the FEC code rate must be changed in order to maintain minimum distortion during increases or decreases of channel packet loss. Because the closed form of the proposed model is very simple and has one scene-dependent model parameter, a video sender can be easily implemented with the model. To train the scene-dependent model parameters in real-time, we propose a test-run method. This method accelerates the test-run while remaining sufficiently accurate for training the scene-dependent model parameters. By using the proposed model and test-run, the video sender can always find the optimal code rate on the fly whenever there is a change in the packet loss status in the channel. An experiment shows that the proposed model and test-run can efficiently determine the near-optimal code rate in joint source-channel coding.

In this letter, a new exact expression of the ergodic channel capacity for a Rician fading channel is provided that is written in terms of exponential integral and incomplete gamma function. Also, a good approximation of the Rician fading channel capacity is derived from the exact expression and its accuracy is numerically verified.

Transmission power control (TPC) is an important aspects of underlay transmission in the cognitive radio (CR) networks since it is able to avoid the extra interference from secondary transmission which can let the CR user coexist with the primary systems around. However, due to the different coverage of the primary signal and CR signal, combined with the complexity of the wireless communication, the scanning CR transmitter may not detect the existence of the primary systems. It will cause inaccurate TPC which will severely disrupt the primary service. In this letter, we propose a dynamic neighbor coordinated power control scheme that can avoid the misdetection of the primary signal and provide relatively accurate TPC related distance estimation. Simulation results show that the proposed scheme greatly reduces interference to the primary systems while enhancing overall CR network throughput.

This letter investigates the space complexity of the sender buffer in a TCP variant, TCP-PR, to deal with packet reordering. Our finding is that with the SACK option used, TCP-PR requires the sender buffer of (β+1) pipesize where β indicates the number of RTTs that must pass before packet loss is detected.

In this letter, we develop an analytical model for the drive-thru applications based on the IEEE 802.11p WAVE. The model shows that prioritizing the bitrates via the 802.11e EDCA mechanism leads to significant throughput improvement.

In Proxy Mobile IPv6 (PMIPv6), when a Mobile Node (MN) enters a PMIPv6 domain and attaches to an access link, the router on the access link detects attachment of the MN by the link-layer access. All elements of PMIPv6 including the router then provide network-based mobility management service for the MN. If the MN moves to another router in this PMIPv6 domain, the new router emulates attachment to the previous router by providing same network prefix to the MN. In other words, PMIPv6 provides rapid mobility management based on layer-2 attachment and transparent mobility support to the MN by emulating layer-3 attachment with respect to intra-domain roaming. However, when the MN moves to other PMIPv6 domains, although the domains also provide the network-based mobility management service, the MN should exploit the host-based mobility management protocol, i.e. Mobile IPv6 (MIPv6), for the inter-domain roaming. Hence, this letter proposes the rapid and transparent inter-domain roaming mechanism controlled by the networks adopting PMIPv6.

This paper proposes a pattern partitioning algorithm that maps multiple target patterns onto homogeneous memory-based string matchers. The proposed algorithm adopts the greedy search based on lexicographical sorting. By mapping as many target patterns as possible onto each string matcher, the memory requirements are greatly reduced.

This paper proposes a multicast delivery system using base station diversity for cellular systems. Conventional works utilize single wireless link communication to achieve reliable multicast. In cellular systems, received signal intensity declines in cell edge areas. Therefore, wireless terminals in cell edge areas suffer from many transmission errors due to low received signal intensity. Additionally, multi-path fading also causes dynamic fluctuation of received signal intensity. Wireless terminals also suffer from transmission errors due to the multi-path fading. The proposed system utilizes multiple wireless link communication to improve transmission performance. Each wireless terminal communicates with some neighbor base stations, and combines frame information which arrives from different base stations. Numerical results demonstrate that the proposed system can achieve multicast data delivery with a short transmission period and can reduce consumed wireless resource due to retransmission.

The bit-error-rate (BER) performance predicted by the semi-analytical evolution technique proposed by Li Ping et al. becomes inaccurate for parallel concatenated coded interleave-division multiple-access (PCC-IDMA) systems. To solve this problem, we develop a novel evolution technique of such systems. Numerical results show that the predicted performance agrees well with the simulation results, and that this technique is useful for system optimization.

This letter proposes an Ultra WideBand (UWB) channel gain estimation algorithm that is based on frequency response correlation. From data measured in indoor and outdoor environments, the frequency correlation statistics of the UWB channel are characterized by correlation coefficients, which are represented as functions of the frequency separation according to the propagation environments. We use empirically established correlation coefficient models to develop the proposed channel gain estimation algorithm. In this algorithm, unknown channel gains of intermediate frequencies are estimated from known channel gains of neighboring reference frequencies. The performance of the proposed algorithm is evaluated against various configurations of reference frequencies and compared with that of a conventional estimation algorithm using a linear interpolation scheme.

In this letter, a new approximation of log-likelihood ratio (LLR) for soft input channel decoding is proposed. Conventional simplified LLR using log-sum approximation can degrade the performance of bit interleaved coded modulation (BICM) systems employing hybrid automatic repeat request (HARQ) at low SNR. The proposed LLR performs as well as the exact LLR, and at the same time, requires only a small number of elementary operations.

A new estimation algorithm of clock drift in symbol duration for high precision ranging, based on multiple symbols of chirp spread spectrum (CSS) is proposed. Since the permissible error of a crystal oscillator in CSS is relatively high given the need to lower device costs, ranging results are perturbed by clock drift. We establish the phenomenon of clock drift in multiple symbols of CSS, and estimate the clock drift in symbol duration based on phase difference between adjacent symbols. The proposed algorithm is analyzed, and verified by Monte Carlo simulations.

We propose 3-hop cooperative diversity using QOSTBC (Quasi-Orthogonal Space-Time Block Code), which offers 3-hop cooperative diversity without signal separation in relay nodes. The key of our proposed scheme is encoding signal sequence in different signal unit according to relay stage. This letter explains details of the proposed scheme and shows that it offers interference reduction among streams and space diversity gain by result of simulations.

For multiple-input multiple-output (MIMO) spatial multiplexing, signal separation/detection is one of the most important signal processing parts, so that signal separation/detection schemes are being vigorously researched. As a promising signal separation/detection scheme, frequency-domain iterative soft interference cancellation (FD-SIC) has been proposed. Although iterative FD-SIC can provide the transmission performance close to lower bound for QPSK, the accuracy of signal separation/detection significantly degrades in case of high level data modulation. Therefore, in this paper, we propose layered soft interference cancellation (LSIC). We consider single-carrier (SC)-MIMO spatial multiplexing with frequency domain equalization (FDE). The achievable frame error rate (FER) performances with LSIC are evaluated by computer simulation to show that LSIC can provide better FER performance than iterative FD-SIC.

Dual-hop wireless transmission is a technique by which the channel from the source to the destination is divided into two shorter links using relays. By using this transmission, the communications reliability is improved whereas the throughput might go down since the dual-hop transmission is performed by two times. In this letter, we propose a scheme that uses hierarchical modulation at the source and adaptive modulation based on cyclic redundancy check (CRC) code at the relays. The proposed scheme is evaluated in terms of the bit error rate (BER) and throughput performances.

Based on game theory, a distributed power control algorithm with sequential subchannel nulling is proposed for ad-hoc networks. It is shown that the proposed method, by sharing subchannels appropriately according to the interference profiles, can reduce the power consumption of the network while satisfying the target rate of each link.

In this letter, we study cell cooperation in the downlink OFDMA cellular networks. The proposed cooperation scheme is based on fractional frequency reuse (FFR), where a cooperation group consists of three sector antennas from three adjacent cells and the subchannels of each cooperation group are allocated coordinately to users. Simulation results demonstrate the effectiveness of the proposed schemes in terms of throughput and fairness.

We present an adaptive power saving (APS) scheme to reduce downlink energy consumption of the transmit power in the cellular relay network (CRN). In the APS scheme, some cell topologies operating in 2-hop mode using relay stations (RSs) are converted to that of 3-hop mode and others are simultaneously converted to that of single-hop mode when the offered traffic load becomes very low, especially during night periods. By this means, we show the APS scheme outperforms the conventional CRN (CCRN) scheme in terms of energy consumption.