Quantifying user satisfaction is essential, because the results can help service providers deliver better services. In this work, we propose a generalizable methodology, based on survival analysis, to quantify user satisfaction in terms of session times, i. e., the length of time users stay with an application. Unlike subjective human surveys, our methodology is based solely on passive measurement, which is more cost-efficient and better able to capture subconscious reactions. Furthermore, by using session times, rather than a specific performance indicator, such as the level of distortion of voice signals, the effects of other factors like loudness and sidetone, can also be captured by the developed models. Like survival analysis, our methodology is characterized by low complexity and a simple model-developing process. The feasibility of our methodology is demonstrated through case studies of ShenZhou Online, a commercial MMORPG in Taiwan, and the most prevalent VoIP application in the world, namely Skype. Through the model development process, we can also identify the most significant performance factors and their impacts on user satisfaction and discuss how they can be exploited to improve user experience and optimize resource allocation.

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In heterogeneous network environments, it is mandatory to measure the grade of the video quality in order to guarantee the optimal quality of the video streaming service. Quality of Service (QoS) has become a key issue for service acceptability and user satisfaction. Although there have been many recent works regarding video quality, most of them have been limited to measuring quality within temporal and Signal-to-Noise Ratio (SNR) scalability. H.264/AVC Scalable Video Coding (SVC) has emerged and has been developed to support full scalability. This includes spatial, temporal, and SNR scalability, each of which shows different visual effects. The aim of this paper is to define and develop a novel video quality metric allowing full scalability. It focuses on the effect of frame rate, SNR, the change of spatial resolution, and motion characteristics using subjective quality assessment. Experimental results show the proposed quality metric has a high correlation to subjective quality and that it is useful in determining the video quality of SVC.

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We present a method for the objective quality evaluation of noise-reduced speech in wideband speech communication services, which utilize speech with a wider bandwidth (e. g., 7kHz) than the usual telephone bandwidth. Experiments indicate that the amount of residual noise and the distortion of speech and noise, which are quality factors, influence the perceived quality degradation of noise-reduced speech. From the results, we observe the principal relationships between these quality factors and perceived speech quality. On the basis of these relationships, we propose a method that quantifies each quality factor in noise-reduced speech by analyzing signals that can be measured and assesses the over-all perceived quality of noise-reduced speech using values of these quality factors. To verify the validity of the method, we perform a subjective listening test and compare subjective quality of noise-reduced speech with its estimation. In the test, we use various types of background noise and noise-reduction algorithms. The verification results indicate that the correlation between subjective quality and its objective estimation is sufficiently high regardless of the type of background noise and noise-reduction algorithm.

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Image quality assessment method is a methodology that measures the difference of quality between the reference image and its distorted one. In this paper, we propose a novel reduced-reference (RR) quality assessment method for JPEG-2000 compressed images, which exploits the statistical characteristics of context information extracted through partial entropy decoding or decoding. These statistical features obtained in the process of JPEG-2000 encoding are transmitted to the receiver as side information and used to estimate the quality of images transmitted over various noisy channels at the decompression side. In the framework of JPEG-2000, the context of a current coefficient is determined depending on the pattern of the significance and/or the sign of its neighbors in three bit-plane coding passes and four coding modes. As the context information represents the local property of images, it can efficiently describe textured pattern and edge orientation. The quality of transmitted images is measured by the difference of entropy of context information between received and original images. Moreover, the proposed quality assessment method can directly process the images in the JPEG-2000 compressed domain without full decompression. Therefore, our proposed can accelerate the work of assessing image quality. Through simulations, we demonstrate that our method achieves fairly good performance in terms of the quality measurement accuracy as well as the computational complexity.

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This paper considers the problem of contention resolution algorithm for multi-class with quality of service (QoS) constrained for wireless communication. Five different channel reservation schemes are proposed, namely, UNI+MLA, UNI+DS, UNI+DS+MLA, Partial UNI and Partial UNI+MLA schemes for multimedia traffic, all are extensions of our recently proposed UNI scheme for single-class traffic. The goal is to achieve the highest system performance and enable each traffic type to meet its QoS requirements. We evaluate the performance of each scheme by mathematical analysis. The numerical results show that our proposed schemes are effective in enabling each traffic type to achieve the best successful rate possible in this kind of environment. Finally when comparing between our proposed schemes and conventional technique in terms of both throughput performance and QoS requirements it is found that the UNI+MLA, UNI+DS+MLA and Partial UNI+MLA schemes are relatively efficient and suitable for practical applications.

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Managing the performance at the flow level through traffic measurement is crucial for effective network management. With the rapid rise in link speeds, collecting all packets has become difficult, so packet sampling has been attracting attention as a scalable means of measuring flow statistics. In this paper, we firstly propose a method of estimating TCP flow rates of sampled flows through packet sampling, and then develop a method of detecting performance degradation at the TCP flow level from the estimated flow rates. In the method of estimating flow rates, we use sequence numbers of sampled packets, which make it possible to improve markedly the accuracy of estimating the flow rates of sampled flows. Using both an analytical model and measurement data, we show that this method gives accurate estimations. We also show that, by observing the estimated rates of sampled flows, we can detect TCP performance degradation. The method of detecting performance degradation is based on the following two findings: (i) sampled flows tend to have high flow-rates and (ii) when a link becomes congested, the performance of high-rate flows becomes degraded first. These characteristics indicate that sampled flows are sensitive to congestion, so we can detect performance degradation of flows that are sensitive to congestion by observing the rate of sampled flows. We also show the effectiveness of our method using measurement data.

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It is important to monitor routing protocols to ensure IP networks and their operations can maintain sufficient level of stability and reliability because IP routing is an essential part of such networks. In this paper, we focus on Open Shortest Path First (OSPF), a widely deployed intra-domain routing protocol. Routers running OSPF advertise their link states on Link State Advertisements (LSAs) as soon as they detect changes in their link states. In IP network operations, it is important for operators to ascertain the location and type of a failure in order to deal with failures adequately. We therefore studied IP network failure identification based on the monitoring of OSPF LSAs. There are three issues to consider in regard to identifying network failures by monitoring LSAs. The first is that multiple LSAs are flooded by a single failure. The second is the LSA delay, and the third is that multiple failures may occur simultaneously. In this paper, we propose a method of network failure identification based on a detailed analysis of OSPF LSA flooding that takes into account the above three issues.

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We propose an algorithm for finding heavy hitters in terms of cardinality (the number of distinct items in a set) in massive traffic data using a small amount of memory. Examples of such cardinality heavy-hitters are hosts that send large numbers of flows, or hosts that communicate with large numbers of other hosts. Finding these hosts is crucial to the provision of good communication quality because they significantly affect the communications of other hosts via either malicious activities such as worm scans, spam distribution, or botnet control or normal activities such as being a member of a flash crowd or performing peer-to-peer (P2P) communication. To precisely determine the cardinality of a host we need tables of previously seen items for each host (e. g., flow tables for every host) and this may infeasible for a high-speed environment with a massive amount of traffic. In this paper, we use a cardinality estimation algorithm that does not require these tables but needs only a little information called the cardinality summary. This is made possible by relaxing the goal from exact counting to estimation of cardinality. In addition, we propose an algorithm that does not need to maintain the cardinality summary for each host, but only for partitioned addresses of a host. As a result, the required number of tables can be significantly decreased. We evaluated our algorithm using actual backbone traffic data to find the heavy-hitters in the number of flows and estimate the number of these flows. We found that while the accuracy degraded when estimating for hosts with few flows, the algorithm could accurately find the top-100 hosts in terms of the number of flows using a limited-sized memory. In addition, we found that the number of tables required to achieve a pre-defined accuracy increased logarithmically with respect to the total number of hosts, which indicates that our method is applicable for large traffic data for a very large number of hosts. We also introduce an application of our algorithm to anomaly detection. With actual traffic data, our method could successfully detect a sudden network scan.

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This paper proposes two traffic control schemes to support the communication quality of multimedia streaming services such as VoIP and audio/video over IEEE 802.11 wireless LAN systems. The main features of the proposed scheme are bandwidth control for each flow of the multimedia streaming service and load balancing between access points (APs) of the wireless LAN by using information of data link, network and transport layers. The proposed schemes are implemented on a Linux machine which is called the wireless traffic controller (WTC). The WTC connects a high capacity backbone network and an access network to which the APs are attached. We evaluated the performance of the proposed WTC and confirmed that the communication quality of the multimedia streaming would be greatly improved by using this technique.

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This paper presents a new QoS model for end-to-end service provisioning in multi-hop wireless networks. In legacy IEEE 802.11e based multi-hop wireless networks, the fixed assignment of service classes according to flow's priority at every node causes priority inversion problem when performing end-to-end service differentiation. Thus, this paper proposes a new QoS provisioning model called Dynamic Hop Service Differentiation (DHSD) to alleviate the problem and support effective service differentiation between end-to-end nodes. Many previous works for QoS model through the 802.11e based service differentiation focus on packet scheduling on several service queues with different service rate and service priority. Our model, however, concentrates on a dynamic class selection scheme, called Per Hop Class Assignment (PHCA), in the node's MAC layer, which selects a proper service class for each packet, in accordance with queue states and service requirement, in every node along the end-to-end route of the packet. The proposed QoS solution is evaluated using the OPNET simulator. The simulation results show that the proposed model outperforms both best-effort and 802.11e based strict priority service models in mobile ad hoc environments.

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As one of the promising techniques in Broadband Wireless Access (BWA), IEEE 802.16 also namely WiMax provides wide-area, high-speed, and non-line-of-sight wireless transmission to support multimedia services. Four service types are defined in the specification of IEEE 802.16 for QoS support. In order to achieve end-to-end multimedia services, 802.16 QoS must be well integrated with IP QoS. In this paper, we propose a framework of cross-layer QoS support in the IEEE 802.16 network. Two novel mechanisms are proposed in the framework for performance improvement: Fragment Control and Remapping. Fragment Control handles the data frames that belong to the same IP datagram in an atomic manner to reduce useless transmission. Remapping is concerned with the mapping rules from IP QoS to 802.16 QoS and is designed to reduce the impact of traffic burstiness on buffer management. Simulation study has shown that the proposed scheme has higher goodput and throughput, and lower delay than the contrast.

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In this letter, we propose a shared adaptive packet loss concealment scheme for the high quality guaranteed Internet telephony service which connects multiple users. In order to recover packet loss efficiently in the all-IP based convergence environment, we provide a robust signal recovery scheme which is based on the shared adaptive both-side information utilization. This scheme is provided according to the average magnitude variation across the frames and the pitch period replication on the 1-port gateway (G/W) system. The simulated performance demonstrates that the proposed scheme has the advantages of low processing times and high recovery rates in the all-IP based ubiquitous environment.

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Wireless communication systems usually employ a concatenated error control coding scheme consisting of an outer error detection code and an inner error correction code. Traditionally, these two codes are decoded separately. When the sub-block structure is used, each data block (input sequence) at the inner encoder consists of several sub-blocks and each of these sub-blocks is protected with the error detection code. The sub-block structure is used in the Wideband CDMA (WCDMA) system specified by the 3rd Generation Partnership Project (3GPP). In this paper, a sub-block recovery scheme is proposed for this concatenated error control coding scheme to utilize the error detection capability introduced by the outer code in the decoding of the inner code. We demonstrate that, if the inner code is a turbo code with a highly structured interleaver and iterative sub-optimal decoding is used, the sub-block recovery scheme is helpful in correcting a typical error pattern, which helps to improve the block error rate performance. We analyze the decoding performance when sub-block recovery is used together with the maximum likelihood (ML) algorithm as well as the log maximum-a-posteriori probability (Log-MAP) and the soft output Viterbi algorithm (SOVA) and demonstrate gains introduced by the sub-block recovery in the latter two cases using computer simulations.

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The purpose of this study is to design, analyze and calibrate a pre-modulation filter with a variable cutoff frequency property for the PCM/FM transmission system. The effect of the pre-modulation filter reduces the energy contained in the sidebands of the modulated carrier. Most existing filters are based on a constant cutoff frequency specification. The onboard units of the launch vehicle which have been performed by the functional and environmental tests should not be revised by any hardware modifications from a reliability point of view. In case of the VCU in this paper, it is necessary to change the PCM bit rate and the cutoff frequency of the pre-modulation filter with software programming in order to improve the picture quality in the allowable bandwidth even if the unit has already been manufactured. In addition, the link margin around the satellite separation event is very low because the flight range is more than 2000km. Under the same RF power condition, the reduction of the transmitting data rate will result in an increase of link margin. For this, the pre-modulation filter should have the variable cutoff frequency property with software programmable function according to 0.7 times the data rate. The variable cutoff frequency pre-modulation filter consists of a digital FIR filter, a DAC system and a tunable 2nd order LPF to meet the magnitude frequency response of an analog 7th order Bessel LPF. The attenuation requirements of three sub-blocks are generated by the frequency response analysis and the linear phase features are also taken into account for each stage. And the analog conversion part including a DAC system and a tunable 2nd order LPF is verified and calibrated using a tone generator.

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We propose a new Clock and Data Recovery (CDR) circuit for burst-mode applications. It can recover clock signals after two data transitions and endure long sequence of consecutive identical digits. Two Digital Phase Aligners (DPAs), triggered by rising or falling edges of input data, recover clock signals, which are then combined by a phase interpolator. This configuration reduces the RMS jitters of the recovered clock by 30% and doubles the maximum run length compared to a previously reported DPA CDR. A prototype chip is demonstrated with 0.18-μm CMOS technology. Measurement results show that the chip operates without any bit error for 1.25-Gb/s 2³¹-1 PRBS with 200-ppm frequency offset and recovers clock and data after two clock cycles.

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Failure avoidance capability is a desired feature for telecommunication networks, such as the Internet. However, not all failures can be promptly bypassed on the Internet because routing systems that are responsible for detecting and avoiding failures cannot detect all failures. Consequently, failures can interrupt internet communications for a long time, such as a few hours. This paper proposes a novel routing architecture called Drouting that enables flexible failure avoidance. In Drouting, routers calculate multipaths from a source to a destination by constructing Directed Acyclic Graphs (DAGs) that include all links in the intra-domain network graph. IP packets carry packet tags that are set by the end host. The packet tags are used to select a network path from the multipath routes. In this paper, the failure avoidance property of Drouting architecture is evaluated through comparison with another proposal, Deflection, using simulations. Simulations were performed on inferred and synthetic topologies. Drouting exhibits similar performance with Deflection in terms of the number of nexthops, the number of paths and the length of paths, while Drouting outperforms Deflection in the probability of success of failure avoidance.

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Tag collision is a major problem in the field of multi-tag identification in RFID systems. To solve this problem, many RFID systems adopt their own collision arbitration algorithms based on framed-structure slotted Aloha (FSSA) due to the simplicity of implementation. The frame size, meaning the number of slots in a frame, is a very important factor to inventory tags' responses in the FSSA. How to assign the frame size is therefore crucial to the collision arbitration performance. Since the existing collision arbitration methods do not consider the slot times of each slot when assigning frame size, they may increase overall identification time. By involving the slot times, we improve the collision arbitration performance of the conventional methods. Simulation results show that collision arbitration based on the proposed method is superior to that based on the conventional methods, irrespective of the number of tags.

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Vertical handoff is a new type of handoff that is triggered when a mobile node moves over heterogeneous wireless networks with each proving different access bandwidth, transmission latency, and coverage. A mobile node can achieve higher throughput by accessing a higher bandwidth providing wireless network. However, TCP has to experience drastic changes of the bandwidth and the latency due to the vertical handoff which must be recognized as a network congestion, and this degrades end-to-end performance. In this paper, we propose a TCP context switching scheme, named Context-Switching TCP, that maintains TCP variables separately for different types of wireless networks. Through simulations, Context-Switching TCP shows higher performance than TCP SACK for vertical handoff. Especially, it shows much higher performance gain when vertical handoff occurs frequently.

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Wireless Mesh Network (WMN) is a promising model with benefits in coverage extension and throughput improvement. In WMN, multiple channels are available for improving system performance through concurrent transmission. For maximum utilization, per-node channel quality and inter-channel interference should be considered in multi-channel assignment. We propose a new multi-channel assignment method. First, we model the mesh network connectivity after a multi-graph which has multiple edges between two nodes. From this connectivity graph, we generate a multi-channel conflict graph, then we allocate multiple channels so that they do not overlap, using list coloring algorithm. We also propose a new sub-graph list coloring algorithm to enhance channel allocation performance. From computer simulations, we verify the performance of the algorithm.

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In this paper, a routing protocol referred to as Directed Diffusion with Stepwise Interest Retransmission (DD/SIR) for wireless sensor networks is proposed to mitigate power consumption considering node mobility. In DD/SIR, a sink retransmits interest. The propagation areas of the interest are narrowed stepwisely. In addition, according to the number of hops between the sink and sensor nodes, the data transmission timing is controlled sequentially. By both theoretical analysis and computer simulation, we evaluate the performance of DD/SIR. We show that DD/SIR can mitigate control overhead and realize low power operation without degrading data reachability to the sink. Especially, at a small number of data sending nodes, DD/SIR is more effective than the conventional routing.

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In this paper, we propose a simple peak power reduction (PPR) method based on adaptive inversion of parity-check block of codeword in BCH-coded OFDM system. In the proposed method, the entire parity-check block of the codeword is adaptively inversed by multiplying weighting factors (WFs) so as to minimize PAPR of the OFDM signal, symbol-by-symbol. At the receiver, these WFs are estimated based on the property of BCH decoding. When the primitive BCH code with single error correction such as (31,26) code is used, to estimate the WFs, the proposed method employs a significant bit protection method which assigns a significant bit to the best subcarrier selected among all possible subcarriers. With computer simulation, when (31,26), (31,21) and (32,21) BCH codes are employed, PAPR of the OFDM signal at the CCDF (Complementary Cumulative Distribution Function) of 10^-4 is reduced by about 1.9, 2.5 and 2.5dB by applying the PPR method, while achieving the BER performance comparable to the case with the perfect WF estimation in exponentially decaying 12-path Rayleigh fading condition.

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IEEE 802.11 wirelesses LANs (WLANs) have been rapidly deployed in enterprises, public areas, and households. Voice-over-IP (VoIP) and similar applications are now commonly used in mobile devices over wireless networks. Recent works have improved the quality of service (QoS) offering higher data rates to support various kinds of real-time applications. However, besides the need for higher data rates, seamless handoff and load balancing among APs are key issues that must be addressed in order to continue supporting real-time services across wireless LANs and providing fair services to all users. In this paper, we introduce a novel access point (AP) with two transceivers that improves network efficiency by supporting seamless handoff and traffic load balancing in a wireless network. In our proposed scheme, the novel AP uses the second transceiver to scan and find neighboring STAs in the transmission range and then sends the results to neighboring APs, which compare and analyze whether or not the STA should perform a handoff. The initial results from our simulations show that the novel AP module is more effective than the conventional scheme and a related work in terms of providing a handoff process with low latency and sharing traffic load with neighbor APs.

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Code acquisition performance in the Direct-Sequence Code-Division Multiple-Access (DS/CDMA) communication system is strongly related to the quality of the communication systems. The performance is assessed by (i) code acquisition time; (ii) precision; and (iii) complexity for implementation. This paper applies the method of maximum likelihood (ML) to estimation of propagation delay in DS/CDMA communications, and proposes a low-complexity method for code acquisition. First, a DS/CDMA system model and properties of outputs with a passive matched-filter receiver are reviewed, and a statistical problem in code acquisition is mentioned. Second, an error-controllable code acquisition method based on the maximum likelihood is discussed. Third, a low-complexity ML code acquisition method is proposed. It is shown that the code acquisition time with the low-complexity method is about 1.5 times longer than that with the original ML method, e. g. 13 data periods under 4.96dB.

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UHF radio frequency identification (RFID) has gathered significant interest in the field of long-distance automatic identification applications. Since UHF RFID shares the frequency band with other RFID and/or other wireless systems, it is important to determine how much interference can be applied without causing a significant degradation of anti-collision speed. In this paper, the permissible link quality for RFID anti-collision in a practical environment is discussed by considering an erroneous communication link, taking into account of bit encoding and the type of interference. We approach the quantification of permissible link quality experimentally along with protocol simulations and the mathematical analyses. An international standard protocol, employing frame slotted ALOHA, was used as the air protocol. For these investigations, the present authors developed a protocol simulator. The simulation results were compared with analytical values based on Poisson distribution. The investigation in the return (tag to reader) link, and the forward (reader to tag) link, were analyzed separately. As result of the protocol simulation, it is generally important to secure the Pulse Error Rate 10^-4 or better in both return and forward links for the anti-collision of 64 or less tags. The quality of the return link may be relaxed when the application does not require fast anti-collision. The degradation of the forward link, on the other hand, may entail loss of important commands, resulting in extremely slow anti-collision. It is measured experimentally that the required link quality can be relaxed by up to 10dB in the return links and by 5dB in the forward link when the primary source of interference originates in the interfering readers.

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This paper proposes a new minimum BER (MBER) criterion precoding method that is robust to imperfect channel state information (CSI) for MIMO-OFDM mobile communications. The proposed MBER precoding aims to minimize BER of the maximum likelihood detection (MLD), on the condition that the transmitter can obtain only imperfect CSI owing to channel estimation and quantization errors of the feedback CSI. The proposed scheme controls its precoding parameters under a transmit power constraint by minimizing an upper bound of BER which is derived from the pairwise error probability and averaged with respect to the CSI error. In contrast with a conventional power allocation MBER precoding method that is also robust to the CSI error, the proposed scheme does not make any assumption on the precoding parameters so as to reduce complexity. Thus, it is expected to outperform the conventional scheme at the cost of higher complexity. Computer simulations demonstrate that the proposed precoding method outperforms a conventional nonrobust MBER precoder and the conventional robust power allocation MBER precoding method when the amount of the CSI error is not considerable.

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In this paper, we propose a reduced-complexity radial basis function (RBF)-assisted decision-feedback equalizer (DFE)-based turbo equalization (TEQ) scheme using a novel extended fuzzy c-means (FCM) algorithm, which not only is comparable in performance to the Jacobian RBF DFE-based TEQ but also is low-complexity. Previous TEQ research has shown that the Jacobian RBF DFE TEQ considerably reduces the computational complexity with similar performance, when compared to the logarithmic maximum a posteriori (Log-MAP) TEQ. In this study, the proposed reduced-complexity RBF DFE TEQ further greatly reduces the computational complexity and is capable of attaining a similar performance in contrast to the Jacobian RBF DFE TEQ in the context of both binary phaseshift keying (BPSK) modulation and 4 quadrature amplitude modulation (QAM). With this proposal, the materialization of the RBF-assisted TEQ scheme becomes more feasible.

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In this paper, we evaluate BER (bit error rate) performance and diversity gain when employing a transmission technique utilizing LC (Linear Combination) diversity using 2 time slots with QPSK channels in 2×2 MIMO (Multiple-Input Multiple-Output) spatial multiplexing systems by comparing it with the upper and lower bound on BER. This evaluation shows that this transmission technique realizes high diversity gain and high transmission rate in LOS (line-of-sight) and NLOS (non line-of-sight) environments.

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In mobile communications, the channel consists of many resolvable paths with different time delays, resulting in a severely frequency-selective fading channel. The frequency-domain equalization (FDE) can take advantage of the channel selectivity and improve the bit error rate (BER) performance of the single-carrier (SC) transmission. Recently, multi-input multi-output (MIMO) multiplexing is gaining much attention for achieving very high speed data transmissions with the limited bandwidth. Eigenbeam space division multiplexing (E-SDM) is known as one of MIMO multiplexing techniques. In this paper, we propose frequency-domain SC E-SDM for SC transmission. In frequency-domain SC E-SDM, the orthogonal transmission channels to transmit different data in parallel are constructed at each orthogonal frequency. At a receiver, FDE is used to suppress the inter-symbol interference (ISI). In this paper, the transmit power allocation and adaptive modulation based on the equivalent channel gains after performing FDE are applied. The BER performance of the frequency-domain SC E-SDM in a severe frequency-selective Rayleigh fading channel is evaluated by computer simulation.

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Very high-speed data services are demanded in the next generation wireless systems. However, the available bandwidth is limited. The use of multi-input multi-output (MIMO) multiplexing can increase the transmission rate without bandwidth expansion. For high-speed data transmission, however, the channel becomes severely frequency-selective and the achievable bit error rate (BER) performance degrades. In our previous work, we proposed the joint use of iterative frequency-domain parallel interference cancellation (PIC) and two-dimensional (2D) MMSE-FDE for the non-spread single-carrier (SC) transmission in a frequency-selective fading channel. The joint use of PIC and 2D MMSE-FDE can effectively suppress the inter-path interference (IPI) and the inter-code interference (ICI), resulting from the channel frequency-selectivity, and the interference from other antennas simultaneously. An iterative PIC with 2D MMSE-FDE has a high computational complexity. In this paper, to well suppress the interference from other antennas while reducing the computational complexity, we propose to replace 2D MMSE-FDE by 1D MMSE-FDE except for the initial iteration stage and to use multicode spread-spectrum (SS) transmission instead of the non-spread SC transmission. The BER performance of the proposed scheme in a frequency-selective Rayleigh fading channel is evaluated by computer simulation to show that the proposed scheme can basically match the BER performance of 2D MMSE-FDE with lower complexity.

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In this paper, we design a new coded cooperation protocol utilizing superposition modulation together with iterative decoding/detection algorithms. The aim of the proposed system is to apply “dirty paper coding” theory in the context of half-duplex relay systems. In the proposed system, the node transmits a superposed signal which consists of its own coded information and other node's re-coded information. The destination node detects and decodes the signal using the received signals at two continuous time-slots with iterative decoding algorithm. Moreover, the destination node detects the received signal using the results of decoding, iteratively. This paper provides the outage probability of the proposed system under the assumption that the proposed system can ideally perform dirty paper coding, and it is shown from the comparison between outage probabilities and simulated results that the proposed system can get close to the dirty paper coding theory.

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In multiple-input multiple-output (MIMO) cellular networks, certain schedulers have two independent phases: the first selects a group of users based on the scheduler criterion, and the second assigns the selected users to the transmit antennas by using an assignment scheme taking into consideration capacity maximization. Other schedulers directly select among the available channels between users and the base station in a centralized way. The schedulers of the first category can be implemented with lower complexity compared with the schedulers of the second category. For the first category, we propose three near-optimal assignment schemes with low complexities. We conducted a simulation in which the mobility of users was considered that demonstrated the superior performance of our assignment schemes. Furthermore, we analytically demonstrate their efficiency.

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Phased array antennas are attractive in terms of low cost and power consumption. This paper proposes a controlling scheme based on a bisection method for phased array antennas employing phase shifters with slow switching speed, which is typical for Micro Electro Mechanical Systems (MEMS) switches. Computer simulation results, assuming the IEEE 802.11a Wireless Local Area Network (WLAN) standard, show that the proposed scheme has good gain enhancement capability in multipath fading channels.

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In virtual cellular network (VCN), proposed for high-speed mobile communications, the signal transmitted from a mobile terminal is received by some wireless ports distributed in each virtual cell and relayed to the central port that acts as a gateway to the core network. In this paper, we apply the multi-route MHMRC diversity in order to decrease the transmit power and increase the multi-hop link capacity. The transmit power, the interference power and the link capacity are evaluated for DS-CDMA multi-hop VCN by computer simulation. The multi-route MHMRC diversity can be applied to not only DS-CDMA but also other access schemes (i. e. MC-CDMA, OFDM, etc.).

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MUSIC-based estimation of direction of arrival (DOA) using universal steering vector (USV) is experimentally studied. A four-element array antenna and a four-channel receiver are employed for the experiment. In order to improve the accuracy of DOA estimation, USV which has already included the effect of mutual coupling between array elements and effect of array elements themselves is compensated to further include the electric delay and loss of four channels in the receiver. The compensated USV (C-USV) approach proposed in this paper does not need the time-consuming measurement of array element pattern because the compensating matrix for USV is obtained by measuring the S parameters between RF input ports of the feeding cables and IF output ports of the receiver. The experimental results of MUSIC-based DOA estimation show that C-USV approach is an accurate, effective and practical method for the MUSIC-based DOA estimation.

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The multistage Wiener filter (MWF) outperforms the full rank Wiener filter in low sample support environments. However, the MWF adaptive process should be stopped at an optimum stage to get the best performance. There are two methods to stop the MWF adaptive process. One method is to calculate until the final full-stage, and the second method is to terminate at r-stage less than full-stage. The computational load is smaller in the latter method, however, a performance degradation is caused by an additional or subtractive stage calculation. Therefore, it is very important for the r-stage calculation to stop an adaptive process at the optimum stage. In this paper, we propose a simple method based on a cross-correlation coefficient to stop the MWF adaptive process. Because its coefficient is calculated by the MWF forward recursion, the optimum stage is determined automatically and additional calculations are avoided. The performance was evaluated by simulation examples, demonstrating the superiority of the proposed method.

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In air navigation, the rotation of aircraft results in the discontinuous tracking of GNSS signals. As the platform rotates, the GNSS signals are subject to blanking effects. To solve this problem, a ring-type antenna array is used to prevent signal discontinuity and a hypothesis-test based detection scheme is developed so that the correct antenna combination can be formed to provide uninterrupted reception of GNSS signals in the presence of blanking, noise, and interferences. A fixed threshold detection scheme is first developed by assuming that the statistics of the noise are known. It is shown that the scheme is capable of differentiating signal from noise at each antenna element. To account for the interference effect, a multiple hypothesis test scheme, together with an adaptive selection rule, is further developed. Through this detection and selection process, it is shown, through simulations, that the desired GNSS signals can be extracted and successfully tracked in the presence of blanking and co-channel interference.

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In this paper, we propose a method of resizing images in the integer DCT domain employed by H.264/AVC. To accomplish this, we first derive the matrix scaling the image resolution, and then factorize the scaled DCT blocks and the post scaling factors (PF) from the matrix obtained from the multiplication of the scaling matrix and the original integer DCT blocks. Then, we separate the scaled DCT blocks into the integer scaling matrix and the scaled integer DCT blocks. The experiments show that the proposed method produces nearly the same performance as those operating in the real DCT domain.

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This letter presents a new TR-UWB receiver exploiting frequency components of UWB pulses. This is accomplished by separating frequency-components of UWB pulses into real and imaginary parts, independently correlating and effectively combining them. We analytically show this scheme improves the output SNR compared with a conventional one using complex correlation. This will also be justified by simulation results.

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Spectrum and Bit Error Rate (BER) performance of a coded π/2 phase shift Non-Square (NS) 8 Quadrature Amplitude Modulation (QAM) system are studied in this letter. The modulation process of this scheme removes all 180° phase shift between adjacent constellation points and contains inherent memory which can be treated as a type of inner coding in coded system. Simulation results show that this modulation scheme has much lower spectrum regrowth and better BER performance when passing through nonlinear channel compared with conventional mode.

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We propose an Adjustable Parallel TCP (AP-TCP) which is a new scheme to control the aggregate throughput of parallel TCP flows. The AP-TCP can adjust the aggregate throughput to be any desired level irrespective of the parallel size (the number of parallel TCP flows). To adjust the aggregate throughput, we modify the increment factor of each parallel TCP flow to K²/N² where N is the number of parallel TCP flows and K is a value equivalent to any desired level for the aggregate throughput. Once K is given, the AP-TCP attempts to have K times more bandwidth than a single TCP flow when they are competing on the same network path. Another feature of the AP-TCP is its self-adjustment scheme. There is no central coordination or control overhead for parallel TCP flows. We analyze the model of the AP-TCP theoretically and evaluate it by using NS-2 simulation.

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A control frame can be piggybacked onto a data frame to increase channel efficiency in wireless communication. However, if the control frame including global control information is piggybacked, the delay of the data frame from a access point will be increased even though there is only one station with low physical transmission rate. It is similar to the anomaly phenomenon in a network which supports multi-rate transmission. In this letter, we define this phenomenon as “the piggyback problem at low physical transmission rate” and evaluate the effect of this problem with respect to physical transmission rate and normalized traffic load. Then, we propose a delay-based piggyback scheme. Simulations show that the proposed scheme reduces average frame transmission delay and improves channel utilization about 24% and 25%, respectively.

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We propose a token-bucket based rate control algorithm that satisfies both maximum and minimum rate constraints with computational complexity of O(1). The proposed algorithm allocates the remaining bandwidth in a strict priority queuing manner to the flows with different priorities and in a weighted fair queuing manner to the flows within the same priority.

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This paper presents a coexistence model of IEEE 802.15.4 with IEEE 802.11b interference in fading channels and proposes two adaptive channel allocation schemes. The first avoids the IEEE 802.15.4 interference only and the second avoids both of the IEEE 802.15.4 and IEEE 802.11b interferences. Numerical results show that the proposed algorithms are effective for avoiding interferences and for maximizing network capacity since they select a channel which gives the maximum signal to noise ratio to the system.

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In this letter, the effect of distorted constellation shapes of 16-ary modulation due to the power saturation channel is analyzed. In particular, error bounds for 16-QAM and 16-APSK with distorted constellations are derived, and optimal operating points in terms of E_s/N₀ are presented. The result can be used to accurately predict the performance of these modulation schemes with a given level of the constellation distortion, as well as to determine the amount of input power to the saturation channel which minimizes the probability of modulation symbol error.

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Due to the computational complexity of the optimum maximum likelihood detector (OMD) growing exponentially with the number of users, suboptimum techniques have received significant attention. We have proposed the particle swarm optimization (PSO) for the multiuser detection (MUD) in asynchronous multicarrier code division multiple access (MC-CDMA) system. The performance of PSO based MUD is near optimum, while its computational complexity is far less than OMD. Performance of PSO-MUD has also been shown to be better than that of genetic algorithm based MUD (GA-MUD) at practical SNR.

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The computation involved in multiuser detection (MUD) for multicarrier CDMA (MC-CDMA) based on maximum likelihood (ML) principle grows exponentially with the number of users. Particle swarm optimization (PSO) with soft decisions has been proposed to mitigate this problem. The computational complexity of PSO, is comparable with genetic algorithm (GA), but is much less than the optimal ML detector and yet its performance is much better than GA.

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Cognitive Radio is an advanced enabling technology for efficient utilization of vacant spectrum due to its ability to sense the spectrum environment. Various detection methods have been proposed for spectrum sensing, which is the key function in implementing cognitive radio. However most of the existing methods put their interests in detecting TV signal and wireless microphone signals. In this paper, we explore the periodicity of the equally spaced pilot subcarriers in OFDM signal. Simulations in various fading environments show that the proposed cyclostationarity based detection method works well for OFDM signal.

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The MIMO system can meet the growing demand for higher capacity in wireless communication fields. So far, the authors have reported that, based on channel measurements, uncoded performance of narrowband MIMO spatial multiplexing in indoor line-of-sight (LOS) environments generally outperforms that in non-LOS (NLOS) ones under the same transmit power condition. In space-frequency coded MIMO-OFDM spatial multiplexing, however, we cannot expect high space-frequency diversity gain in LOS environments because of high fading correlations and low frequency selectivity of channels so that the performance may degrade unlike uncoiled cases. In this letter, we present the practical performance of coded MIMO-OFDM spatial multiplexing based on indoor channel measurements. The results show that an LOS environment tends to provide lower space-frequency diversity effect whereas the MIMO-OFDM spatial multiplexing performance is still better in the environment compared with an NLOS environment.

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In this letter, a simple but effective antenna selection algorithm for orthogonal space-time block codes with a linear complex precoder (OSTBC-LCP) is proposed and compared with two conventional algorithms in temporally and spatially correlated fading channels. The proposed algorithm, which minimizes pairwise error probability (MinPEP) with an error codebook (EC) constructed from the error vector quantization, is shown to provide nearly the same performance of MinPEP based on all possible error vectors, while keeping the complexity close to that of antenna selection algorithm based on maximum power criterion (Max-power).

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