This paper addresses the interference and load imbalance problems in multi-radio infrastructure mesh networks where each mesh node is equipped with multiple radio interfaces and a subset of nodes serve as Internet gateways. To provide backbone support, it is necessary to reduce interference and balance load in Wireless Mesh Networks (WMNs). In this paper, we propose a new Load-Aware Routing Metric, called LARM, which captures the differences in transmission rates, packet loss ratio, intra/inter-flow interference and traffic load in multi-radio mesh networks. This metric is incorporated into the proposed load-balancing routing, called LBM, to provide load balancing for multi-radio mesh network. Simulation results show that LARM provides better performance compared to WCETT and hop-count routing metrics in LBM routing protocol.

In this paper, we propose a novel Hidden-terminal and Exposed-terminal Interference aware routing metric (HEI-ETT) for Multi-Radio and Multi-Rate wireless mesh networks in which each stationary mesh node is equipped with multi-radio interfaces that relays traffic to the extend networks by using multi-hop transmissions. We have two main design goals for HEI-ETT. First, we will characterize interferences as Hidden-terminal Interference and Exposed-terminal Interference regardless of inter- or intra-flow interference and should take into account both interference effects while computing the path metric. Second, an efficient transmission rate adaptation should be employed in HEI-ETT to enhance the network throughput. We incorporated our metric in well known Optimized Link State Routing protocol version 2 (OLSRv2) which is one of the two standard routing protocols for MANETs and evaluated the performance of our metric by simulation. The results show that our metric outperforms existing metrics such as ETX, ETT and WCETT.

With quick topology changes due to mobile node movement or signal fading in MANET environments, conventional routing restoration processes are costly to implement and may incur high flooding of network traffic overhead and long routing path latency. Adopting the traditional shortest path tree (SPT) recomputation and restoration schemes used in Internet routing protocols will not work effectively for MANET. An object of the next generation SPT restoration system is to provide a cost-effective solution using an adaptive learning control system, wherein the SPT restoration engine is able to skip over the heavy SPT computation. We proposed a novel SPT restoration scheme, called Cognitive Shortest Path Tree Restoration (CSPTR). CSPTR is designed based on the Network Simplex Method (NSM) and Sensitivity Analysis (SA) techniques to provide a comprehensive and low-cost link failure healing process. NSM is used to derive the shortest paths for each node, while the use of SA can greatly reduce the effort of unnecessary recomputation of the SPT when network topology changes. In practice, a SA range table is used to enable the learning capability of CSPTR. The performance of computing and communication overheads are compared with other two well-known schemes, such as Dijstra's algorithm and incremental OSPF. Results show that CSPTR can greatly eliminate unnecessary SPT recomputation and reduce large amounts of the flooding overheads.

The research on Delay Tolerant Networks (DTN) has been activated aiming at a variety of potential applications toward ubiquitous society. The vehicular network is one of the promising areas for applying DTN. In this paper, the end-to-end delay characteristics for vehicular ad-hoc broadcast DTN is analyzed, where the dynamic vehicle mobility model is introduced. The analysis is applied to the two realistic road models, that is, one-way and two-way traffic models. The simulation study demonstrates validity of our analysis for use in generic vehicle networks.

The usable throughput of an IEEE 802.11 system for an application is much less than the raw bandwidth. Although 802.11b has a theoretical maximum of 11 Mbps, more than half of the bandwidth is consumed by overhead leaving at most 5 Mbps of usable bandwidth. Considering this characteristic, this paper proposes and analyzes a real-time distributed scheduling scheme based on the existing IEEE 802.11 wireless ad-hoc networks, using USC/ISI's Power Aware Sensing Tracking and Analysis (PASTA) hardware platform. We compared the distributed real-time scheduling scheme with the real-time polling scheme to meet deadline, and compared a measured real bandwidth with a theoretical result. The theoretical and experimental results show that the distributed scheduling scheme can guarantee real-time traffic and enhances the performance up to 74% compared with polling scheme.

In multi-hop wireless networks, communication quality depends on the selection of a path between source and destination nodes from several candidate paths. Exploring how path selection affects communication quality is important to characterize the best path. To do this, in [1], we used expected transmission count (ETX) as a metric of communication quality and theoretically characterized minimum route ETX, which is the ETX of the best path, in a static one-dimensional random multi-hop network. In this paper, we characterize minimum route ETX in static two-dimensional multi-hop networks. We give the exact formula of minimum route ETX in a two-dimensional network, assuming that nodes are located with lattice structure and that the ETX function satisfies three conditions for simplifying analysis. This formula can be used as an upper bound of minimum route ETX without two of the three conditions. We show that this upper bound is close to minimum route ETX by comparing it with simulation results. Before deriving the formula, we also give the formula for a one-dimensional network where nodes are located at constant intervals. We also show that minimum route ETX in the lattice network is close to that in a two-dimensional random network if the node density is large, based on a comparison between the numerical and simulation results.

The integration of multihop relays with orthogonal frequency-division multiple access (OFDMA) cellular infrastructures can meet the growing demands for better coverage and higher throughput. Resource allocation in the OFDMA two-hop relay system is more complex than that in the conventional single-hop OFDMA system. With time division between transmissions from the base station (BS) and those from relay stations (RSs), fixed partitioning of the BS subframe and RS subframes can not adapt to various traffic demands. Moreover, single-hop scheduling algorithms can not be used directly in the two-hop system. Therefore, we propose a semi-distributed algorithm called ASP to adjust the length of every subframe adaptively, and suggest two ways to extend single-hop scheduling algorithms into multihop scenarios: link-based and end-to-end approaches. Simulation results indicate that the ASP algorithm increases system utilization and fairness. The max carrier-to-interference ratio (Max C/I) and proportional fairness (PF) scheduling algorithms extended using the end-to-end approach obtain higher throughput than those using the link-based approach, but at the expense of more overhead for information exchange between the BS and RSs. The resource allocation scheme using ASP and end-to-end PF scheduling achieves a tradeoff between system throughput maximization and fairness.

Recently, a promising packet forwarding architecture COPE was proposed to essentially improve the throughput of multihop wireless networks, where each network node can intelligently encode multiple packets together and forward them in a single transmission. However, COPE is still in its infancy and has the following limitations: (1) COPE adopts the FIFO packet scheduling and thus does not provide different priorities for different types of packets. (2) COPE simply classifies all packets destined to the same nexthop into small-size or large-size virtual queues and examines only the head packet of each virtual queue to find coding solutions. Such a queueing structure will lose some potential coding opportunities, because among packets destined to the same nexthop at most two packets (the head packets of small-size and large-size queues) will be examined in the coding process, regardless of the number of flows. (3) The coding algorithm adopted in COPE is fast but cannot always find good solutions. In order to address the above limitations, in this paper we first present a new queueing structure for COPE, which can provide more potential coding opportunities, and then propose a new packet scheduling algorithm for this queueing structure to assign different priorities to different types of packets. Finally, we propose an efficient coding algorithm to find appropriate packets for coding. Simulation results demonstrate that this new COPE architecture can further greatly improve the node transmission efficiency.

In this paper, we propose a cross-layer design of packet level cooperation for wireless LANs that support rate adaptation. While keeping compatibility with legacy wireless LANs, distributed control of multi-hop packet transmission is enabled without pre-negotiation of routing or pairing. These features are provided by prioritization scheme based on IEEE 802.11e EDCF in which we set the parameters according to the measured link condition at each terminal. Relaying packets with high transmission rate makes much efficient use of radio resource, and it leads not only to improve performance of the total system, but also to overcome the fairness issue known in rate adaptation, where the terminals with good link conditions cannot gain enough resource because of the time consumed by low data rate transmission. These advantages are confirmed through computer simulations considering packet error rate at each transmission which is assumed by receiving power calculated from the distance between source, relay, and destination terminals. Furthermore, we also discuss about the fairness between cooperation-enabled and legacy terminals when they coexist in the same system, in order to make gradual deployment feasible.

As a flexible, cost effective solution for a large-scale access network to the Internet, we have studied the design optimization of the Wireless Internet-access Mesh NETwork (WIMNET). WIMNET consists of multiple access points (APs) that have wireless links between them mainly on the wireless distribution system (WDS). In WIMNET, the links around the Internet gateway can be bottlenecks because every packet passes through it after multihop link activations. Besides, the link quality may be degraded by obstacles in indoor environments. Thus, the proper allocation of APs is essential in WIMNET, so that the communication quality should be ensured while the installation and management cost be minimized. In this paper, we formulate this AP allocation problem for indoor environments in WIMNET with the proof of the NP-completeness of its decision version. Then, we present its two-stage heuristic algorithm composed of the initial greedy allocation and the iterative improvement. The effectiveness of our proposal is verified through extensive simulations in three indoor environments.

A typical feature of MANETs is that network topology is dynamically changed by node movement. When we execute state transition testing for such protocols, first we draw the Finite State Machine (FSM) with respect to each number of neighbor nodes. Next, we create the state transition matrix from the FSMs. Then, we generate test cases from the state transition matrix. However, the state transition matrix is getting much large because the number of states and the number of transitions increase explosively with increase of the number of neighbor nodes. As a result, the number of test cases increases, too. In this paper, we propose a new method to reduce the number of test cases by using equivalent division method. In this method, we decide a representative input to each state, which is selected from equivalent inputs to the states. By using our proposed method, we can generate state transition matrix which is hard to affect increasing the number of neighbor nodes. As a consequence, the number of test cases can be reduced.

Broadband wireless networks are rapidly expanding over the world. To provide wireless broadband services for Shanghai Expo 2010, pre-research is launched and a test-bed has been developed, in which WiMAX and Wi-Fi mesh are involved. The test-bed shows that Wi-Fi mesh integrated with WiMAX is highly suitable for large-scale activities like the Olympic Games and the World Expo.

In reliable multicast, feedback and recovery traffic limit the performance and scalability of the multicast session. In this paper, we present an improvement to the many-to-many reliable multicast protocol, Group-Aided Multicast protocol (GAM), with a local-group based recovery by making use of forward error correction (FEC) locally in addition to NACK/retransmission. In contrast to the original GAM, which only makes use of NACK-based recovery, our scheme produces FEC packets and multicasts the packets within the scope of a local group in order to correct uncorrelated errors of the local members in each group of the multicast session, which reduces the need for NACK/retransmission. By using our scheme, redundancy traffic can be localized in each group within a multicast session, and the overall recovery traffic can be reduced.

Gigabit Ethernet Passive Optical Networks (GE-PON) systems have been deployed widely as a broadband access solution for the optical access network, the so-called FTTx networks. 10 Gigabit EPON (10G-EPON) is about to be standardizing by a task force (802.3av). To provide the next FTTx solution with 10G-EPON systems, one of the key technologies is how to migrate from 1G-based to 10G-based. In this paper, we present Dynamic Bandwidth Allocation (DBA) algorithm which considered a fair policy for 1G/10G coexistence EPON system to achieve a smooth migration.

Optical burst switching (OBS) is a promising approach for the realization of future flexible high-speed optical networks. In particular, a multicast forwarding method for optical bursts is important if an efficient high-speed grid computing network is to be realized. In OBS networks, the number of burst replicas generated at each node is strongly restricted due to optical power impairment of multicast bursts. Moreover, unrestricted replication of multicast bursts at each OBS node may not be advantageous because an increase in the number of multicast bursts within the network causes more frequent deflection forwarding of both multicast and unicast bursts. This paper proposes an efficient hop-by-hop multicast forwarding method for optical bursts, where idle output ports are selected based on scores simply calculated using a routing table that each OBS node holds. This method can mitigate increases in loss rate and transfer delay of multicast bursts, even if the number of burst replicas generated at each OBS node is strongly restricted. Moreover, this method can efficiently mitigate an increase in the number of multicast bursts within the network by avoiding unnecessary replication of multicast bursts at each OBS node. Simulation results show that the proposed method can actually mitigate degradation of the loss rate and transfer delay for multicast bursts under the restricted number of burst replicas at each OBS node. Moreover, when the arrival rate of multicast bursts is large relative to that of unicast bursts, the proposed method is able to improve the loss rates of both multicast and unicast bursts by switching the forwarding method for the multicast bursts to the simple unicast forwarding method without burst replication.

A receiver access control scheme is proposed to protect the multicast distribution tree from DoS attack induced by unauthorized use of IGMP, by extending the security-related functionality of IGMP. Based on a specific network and business model adopted for commercial deployment of IP multicast applications, a key management scheme is also presented for bootstrapping the proposed access control as well as accounting and billing for CP (Content Provider), NSP (Network Service Provider), and group members.

In this paper, we propose a subcarrier resource allocation algorithm for managing the video quality degradation for multiuser orthogonal frequency division multiplex (OFDM) systems. The proposed algorithm exploits the unequal importance existing in different picture types for video coding and the diversity of subcarriers for multiuser systems. A model-based performance metric is first derived considering the error concealment and error propagation properties of the H.264 video coding structure. Based on the information on video quality enhancement existing in a packet to be transmitted, we propose the distortion management algorithm for balancing the subcarriers and power usages for each user and minimizing the overall video quality degradation. In the simulation results, the proposed algorithm demonstrates a more gradual video quality degradation for different numbers of users compared with other resource allocation schemes.

Slotted wireless ad hoc networks are drawing more and more attention because of their advantage of QoS (Quality of Service) support for multimedia applications owing to their collision-free packet transmission. Time slot assignment is an unavoidable and important problem in such networks. The existing time slot assignment methods have in general a drawback of limited available bandwidth due to their local assignment optimization without the consideration of directions of the radio wave transmission of wireless links along the routes in such networks. A new time slot assignment is proposed in this paper in order to overcome this drawback. The proposed assignment is different from the existing methods in the following aspects: a) consideration of link directions during time slot assignment; b) largest bandwidth to be achieved; c) feasibility in resource limited ad hoc networks because of its fast assignment. Moreover, the effectiveness of the proposal is confirmed by some simulation results.

The authors propose a user-centric seamless handover (HO) scheme, which is a kind of a vertical HO from a new perspective, toward a next generation network where heterogeneous access networks converge. The users' experience-oriented scheme allows users to enjoy the optimal service quality for real-time applications in respective access networks. In addition, the scheme sustains on-going sessions during the vertical HO. The proposed scheme consists of two methods -- the bicasting of Different Quality-level Streams (DiffQS) and the Delay Difference Absorption (DDA) method. Initially, the authors propose two plausible methods for the SIP-based bicasting of DiffQS. This document introduces a SIP-capable network element named the HO Assistive Server (HOAS). HOAS controls bicasting of DiffQS and provides users with the optimal service quality for real-time applications via respective access networks as well as avoiding packet loss during HO. The DDA method is also proposed to prevent a service interruption and smoothly continue a real-time service during HO. Evaluation results show that the scheme achieves the seamless service continuity from the users' perspective for HO between cellular and high-speed wireless access via the implementation of a prototype system.

This paper proposes an adaptive transmission control method for massive and intensive telecommunication traffic generated by communication-broadcasting integrated services. The proposed method adaptively controls data transmissions from viewers depending on the congestion states, so that severe congestion can be effectively avoided. Furthermore, it utilizes the broadcasting channel which is not only scalable, but also reliable for controlling the responses from vast numbers of viewers. The performance of the proposed method is evaluated through experiments on a test bed where approximately one million viewers are emulated. The obtained results quantitatively demonstrate the performance of the proposed method and its effectiveness under massive and intensive traffic conditions.

In this paper, we propose a new method using dynamic linkage with IP-based infrastructure to compose a wide-area MANET. The proposed method extends AODV and enables searches for destinations in the IP-based infrastructure. The proposed method settles the low scalability problem by creating shortcuts in the IP network. In addition, to overcome the geographical dependence problem, the proposed method connects terminals that are out of radio range with each other by using the IP-based infrastructure. In this way, the proposed method can flexibly establish a wide-area MANET that is scalable and free from geographical constraint.

MPLS-based path technology shows promise as a means of realizing reliable IP networks. Real-time services such as VoIP and video-conference supplied through a multi-domain MPLS network must be able to guarantee end-to-end QoS of the inter-domain paths. Thus, it is important to allocate an appropriate QoS class to the inter-domain paths in each domain traversed by the inter-domain paths. Because each domain has its own policy for QoS class allocation, it is necessary to adaptively allocate the optimum QoS class based on estimation of the QoS class allocation policies in other domains. This paper proposes two kinds of adaptive QoS class allocation schemes, assuming that the arriving inter-domain path requests include the number of downstream domains traversed by the inter-domain paths and the remaining QoS value toward the destination nodes. First, a measurement-based scheme, based on measurement of the loss rates of inter-domain paths in the downstream domains, is proposed. This scheme estimates the QoS class allocation policies in the downstream domains, using the measured loss rates of path requests. Second, a state-dependent type scheme, based on measurement of the arrival rates of path requests in addition to the loss rates of paths in the downstream domains, is also proposed. This scheme allows an appropriate QoS class to be allocated according to the domain state. This paper proposes an application of the Markov decision theory to the modeling of state-dependent type scheme. The performances of the proposed schemes are evaluated and compared with those of the other less complicated non-adaptive schemes using a computer simulation. The results of the comparison reveal that the proposed schemes can adaptively increase the number of inter-domain paths accommodated in the considered domain, even when the QoS class allocation policies change in the other domains and the arrival pattern of path requests varies in the considered domain.

We present a recovery scheme based on Self-protected Spanning Tree (SST), which recovers from failure all by itself. In the recovery scheme, the links are assigned birthdays to denote the order in which they are to be considered for adding to the SST. The recovery mechanism, named Birthday-based Link Replacing Mechanism (BLRM), is able to transform a SST into a new spanning tree by replacing some tree links with some non-tree links of the same birthday, which ensures the network connectivity after any single link or node failure. First, we theoretically prove that the SST-based recovery scheme can be applied to arbitrary two-edge connected or two connected networks. Then, the recovery time of BLRM is analyzed and evaluated using Ethernet, and the simulation results demonstrate the effectiveness of BLRM in achieving fast recovery. Also, we point out that BLRM provides a novel load balancing mechanism by fast changing the topology of the SST.

We address the issue of MIMO channel estimation with the aid of a priori temporal correlation statistics of the channel as well as the spatial correlation. The temporal correlations are incorporated to the estimation scheme by assuming the Gauss-Markov channel model. Under the MMSE criteria, the Kalman filter performs an iterative optimal estimation. To take advantage of the enhanced estimation capability, we focus on the problem of channel estimation from a partial channel measurement in the MIMO antenna selection system. We discuss the optimal training sequence design, and also the optimal antenna subset selection for channel measurement based on the statistics. In a highly correlated channel, the estimation works even when the measurements from some antenna elements are omitted at each fading block.

This paper proposes a pilot-based channel state information (CSI) feedback from a terminal to a base station (BS), considering the terminal's co-channel interference in a time-division duplex/multi-input multi-output system. In the proposed method, the terminal determines a precoding matrix according to the terminal's co-channel interference characteristics and transmits the precoded pilot signals on uplink. Using the responses of the precoded pilot signals, the BS determines appropriate weight vectors for downlink transmit beams considering the terminal's interference characteristics. Furthermore, the BS can predict the terminal's output signal-to-interference-plus-noise power ratio (SINR) for the downlink data stream. Numerical results show that the BS can achieve efficient transmission and accurate SINR prediction using the proposed CSI feedback.

The IEEE802.11b WLAN standard employs direct-sequence/spread-spectrum (DS/SS) modulation. With a fractional sampling RAKE receiver, it is possible to achieve diversity and reduce the BER in DS/SS communication. In order to realize the diversity through fractional sampling, the impulse response of the channel must be estimated. In this paper, a channel estimation scheme for a RAKE receiver with fractional sampling in IEEE802.11b WLAN system is investigated through a computer simulation and an experiment. In order to estimate the impulse response of the channel, a pseudo-inverse matrix with a threshold is employed. Numerical results indicate that the channel can be estimated with an optimum threshold in both the simulation and the experiment.

Similar to orthogonal frequency-division multiplexing (OFDM) systems, orthogonal frequency-division multiple access (OFDMA) is vulnerable to carrier frequency offset (CFO). Since the CFO of each user is different, CFO compensation in OFDMA uplink is much more involved than that in OFDM systems. It has been shown that the zero-forcing (ZF) compensation method is a simple yet effective remedy; however, it requires the inversion of a large matrix and the computational complexity can be very high. Recently, we have developed a low-complexity iterative method to alleviate this problem. In this paper, we consider the theoretical aspect of the algorithm. We specifically analyze the output signal-to-interference-plus-noise-ratio (SINR) of the algorithm. Two approaches are used for the analysis; one is simple but approximated, and the other is complicated but exact. The convergence problem is also discussed. In addition to the analysis, we propose a pre-compensation (PC) method enhancing the performance of the algorithm. Simulations show that our analysis is accurate and the PC method is effective.

Recently, there has been growing interest in the design of wireless cooperative protocol to achieve higher diversity-multiplexing tradeoff among single antenna devices. We propose an automatic request for cooperation (ARC) scheme for wireless networks which can achieve higher order diversity by selecting the best relay. In this scheme, a source transmits a data packet towards a destination and a group of relays. The destination tries to decode the information from the source and if the detection is correct the process will stop. Otherwise, the destination transmits an ARC towards the relays. We utilize this ARC signal for selecting the best relay from the set of relays that have successfully decoded the source packet. The selected relay generates and transmits redundant information for the source packet. The destination combines the two packets received from the source and the best relay to improve the reliability of the packet. We analyze the packet error rate, spectral efficiency and diversity-multiplexing tradeoff of our proposal and compare them with some existing protocols. Analysis shows that our proposal can achieve higher diversity multiplexing tradeoff than conventional cooperative protocols.

In this paper, we derive the exact average symbol error probability (SEP) of M-ary phase-shift keying and quadrature amplitude modulation signals over Stacy fading channels. The Stacy fading is modelled by a three-parameter generalized gamma or physically α-µ fading distribution, spanning a wide range of small-scale fading such as Rayleigh, Nakagami-m, and Weibull fading. The average SEP is generally expressed in terms of (generalized) Fox's H-functions, which particularizes to the previously known results for some special cases. We further analyze the diversity order achieved by orthogonal space-time block codes in multiple-input multiple-output (MIMO) Stacy fading channels.

As a candidate for the transmission technology of next generation mobile communication systems, time-domain spreading MC-CDMA systems have begun to attract much attention. In these systems, data and pilot symbols are spread in the time domain and code-multiplexed. To combat fading issues, we need to conduct channel estimation by using the code-multiplexed pilot symbols. Especially in next generation systems, frequency bands higher than those of current systems, which raise the maximum Doppler frequency, are expected to be used, so that a more powerful channel estimation method is expected. Considering this, we propose a channel estimation method for highly accurate channel estimation; it is a combination of a two-dimensional channel estimation method and an impulse response-based channel estimation method. We evaluate the proposed method by computer simulations.

In this paper, we investigate a coded solution to compensate for the nonlinear distortion of TDMA satellite waveforms. Based on a Volterra-type channel model and the turbo principle, we present a turbo-like system that includes a simple rate-1 encoder at the transmit side in addition to a conventional channel encoder; the receive side iteratively equalizes the nonlinear channel effect and decodes the received symbols. Some other design alternatives are also explored and computer simulated performance is presented. Numerical results show that significant improvement over conventional approaches can be achieved by the proposed turbo system.

This paper presents the regulation and dynamic characteristics of the dc-dc converter with digital PID control, the minimum phase FIR filter or the IIR filter, and then the design criterion to improve the dynamic characteristics is discussed. As a result, it is clarified that the DC-DC converter using the IIR filter method has superior performance characteristics. The regulation range is within 1.3%, the undershoot against the step change of the load is less than 2% and the transient time is less than 0.4 ms with the IIR filter method. In this case, the switching frequency is 100 kHz and the step change of the load R is from 50 Ω to 10 Ω . Further, the superior characteristics are obtained when the first gain, the second gain and the second cut-off frequency are relatively large, and the first cut-off frequency and the passing frequency are relatively low. Moreover, it is important that the gain strongly decreases at the second cut-off frequency because the upper band pass frequency range must be always less than half of the sampling frequency based on the sampling theory.

In this paper, we consider a dual-hop wireless cooperative network with amplify-and-forward (AF) relaying. The output signal-to-noise ratio (SNR) at the destination of the AF cooperative networks is in the form of the sum of harmonic mean of the source-relay channel SNR and the relay-destination channel SNR. Instead of deriving the exact probability density function (PDF) of the output SNR, we study the series expansion of this PDF around zero. This result is then applied to evaluate the performance of the AF cooperative systems over Nakagami-m fading channels, and closed-form high-SNR approximations of the average symbol error rate (SER) and the outage probability are derived. Next, we investigate the optimal power allocation (OPA) among the source node and the relays to minimize the approximate SER as well as the outage probability. It is shown that the optimal power allocation depends on the channel m parameters and the ratio of the source-relay channel gain to the relay-destination gain. In addition to the optimal power allocation, we also propose a low complexity sub-optimal power allocation (SubOPA) scheme. The performance improvement with optimal and sub-optimal power allocation is analyzed and validated by numeric results. It is shown that equal power allocation is near optimal when the relays are close to the source, while significant performance improvement is observed by both the optimal and sub-optimal power allocation schemes when the relays are close to the destination.

The transmission of correlated messages over strong interference channels is examined. The result is the proposal of a single-letter characterization of the sum-rate capacity of strong interference channels with correlated messages. It is shown that if the messages are independent, the sum-rate capacity is equal to that of [1] obtained by Costa and El Gamal. However, it can be larger than that of [1] if the messages are correlated. It is also shown that, in terms of the sum-rate, the achievable rate region in [2] is indeed the sum-rate capacity.

Frame loss of the optical layer protection switching using Plumbum Lanthanum Zirconium Titanium (PLZT) optical switch is evaluated. Experimental results show that typically 62 µs guard time is required for commercially available non-burst mode 10 Gigabit Ethernet modules.

Transmission delay in audio communications is a well-known obstacle to achieving smooth communication. However, it is not known what kinds of effects are caused by small delays. We hypothesized that the small delay in the listener's responses disturbs the speaker's "verbal conditioning," where the verbal behavior of the speaker varies in accordance with the listener's responses. We examined whether the small delays in the listener's responses disturb the speaker's verbal conditioning using an artificial-grammar learning task. The results suggested that a 300-ms delay disturbed the participants' verbal conditioning although they were not adequately aware of the delay.

In RFID systems, collision resolution is a significant issue in fast tag identification. This letter presents a dynamic frame-slotted ALOHA algorithm that uses a collision factor (DFSA-CF). This method enables fast tag identification by estimating the next frame size with the collision factor in the current frame. Simulation results show that the proposed method reduces slot times Required for RFID identification. When the number of tags is larger than the frame size, the efficiency of the proposed method is greater than those of conventional algorithms.

A Policy-based Network Management (PBNM) in Mobile Ad-hoc Networks (MANETs) should be efficient and reliable. In this letter, we propose a mechanism for the policy-based management in ad hoc networks and we discuss methods to discover the Policy Decision Point (PDP), set the management area, and manage the movements of nodes in the PBNM system. Finally, we assess the results through simulations.

In this letter, we propose a computationally effient equalization technique that employs block minimum mean squared error (MMSE) depending on LDL^H factorization. Parallel interference cancellation (PIC) is executed with pre- obtained output to provide more reliable symbol detection. In particular, the band structure of the frequency domain channel matrix is exploited to reduce the implementation complexity. It is shown through computer simulation that the proposed technique requires lower complexity than the conventional algorithm to obtain the same performance, and that it exhibits better performance than the conventional counterpart when the same complexity is assumed.

We describe a channel-vector quantization scheme that is suitable for multiple stream transmission per user in zero-forcing beamforming (ZFBF) multiuser multiple-input and multiple output (MU-MIMO) systems with finite rate feedback. Multiple subsets of a channel matrix are quantized to vectors from random vector codebooks for finite rate feedback. The quantization vectors with an angle difference that is closer to orthogonal are then selected and their indexes are fed back to the transmitter. Simulation results demonstrate that the proposed scheme achieves a better average throughput than that serving a single stream per user when the number of active users is smaller than the number of transmit antennas and that it provides an average throughput close to that serving a single stream per user when the number of users is equal to the number of transmit antennas.

In this letter, we propose a multi-cell cooperation method for broadcast packet transmission in the OFDM-based cellular system with multiple transmit antennas. In the proposed method, to transmit two streams of spatially demultiplexed or transmit diversity coded symbols, we divide a coded packet into subparts to each of which different cell group and antenna pairs are assigned. It is shown that the proposed method reduces the outage probability with only negligible increase in channel estimation.

This letter corrects some errors on a previous letter concerning the derivation of the covariance matrix of phase noise. This derivation doesn't affect the results of the previous letter.

In this letter, we propose a computationally efficient search space for QRM-MLD that is used for spatially multiplexed multiple antenna systems. We perform a set of computer simulations to show that the proposed method achieves a performance that is near to that of the original QRM-MLD, while its computational complexity is near to that of rank-QRM-MLD.

In this letter, a closed form approximation for the average sum rate of random beamforming is derived. It provides a near-exact approximation for arbitrary numbers of beams, users, and received SNR. The approximation is also applied to an average-sense multimode random beamforming scheme which optimizes the number of random beams without any type of instantaneous channel information. The proposed scheme shows better sum rate performance than random beamforming as well as an existing dual mode random beamforming scheme based on instantaneous channel information, while the number of feedback bits for beam index is reduced compared to random beamforming.

A novel closed-form algorithm is presented for estimating the 3-D location (azimuth angle, elevation angle, and range) of a single source in a uniform circular array (UCA) with a center sensor. Based on the centrosymmetry of the UCA and noncircularity of the source, the proposed algorithm decouples and estimates the 2-D direction of arrival (DOA), i.e. azimuth and elevation angles, and then estimates the range of the source. Notwithstanding a low computational complexity, the proposed algorithm provides an estimation performance close to that of the benchmark estimator 3-D MUSIC.

This letter proposes a high precision ranging scheme based on the time of arrival estimation technique for the IEEE 802.15.4a chirp spread spectrum system. The proposed scheme consists of a linear channel impulse response estimation process with the zero forcing or minimum mean square error technique and the multipath delay estimation process with matrix pencil algorithm. The performance of the proposed scheme is compared with that of a well known MUSIC algorithm in terms of computational complexity and ranging precision. Simulation results demonstrate that the proposed scheme outperforms the MUSIC algorithm even though it has comparatively lower computational complexity.

A method is proposed for estimating code and carrier phase parameters of GNSS reflected signals in low SNR (signal-to-noise ratio) environments. Simulation results show that the multipath impact on code and carrier with 0.022 C/A chips delay can be estimated in 0 dB SNR in the condition of 46 MHz sampling rate.