Fiber access network architectures such as active optical networks (AONs) and passive optical networks (PONs) have been developed to support the growing bandwidth demand. Whereas particularly Swedish operators prefer AON, this may not be the case for operators in other countries. The choice depends on a combination of technical requirements, practical constraints, business models, and cost. Due to the increasing importance of reliable access to the network services, connection availability is becoming one of the most crucial issues for access networks, which should be reflected in the network owner's architecture decision. In many cases protection against failures is realized by adding backup resources. However, there is a trade off between the cost of protection and the level of service reliability since improving reliability performance by duplication of network resources (and capital expenditures CAPEX) may be too expensive. In this paper we present the evolution of fiber access networks and compare reliability performance in relation to investment and management cost for some representative cases. We consider both standard and novel architectures for deployment in both sparsely and densely populated areas. While some recent works focused on PON protection schemes with reduced CAPEX the current and future effort should be put on minimizing the operational expenditures (OPEX) during the access network lifetime.

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We propose a scalable parallel interface that provides an ideal aggregated bandwidth link for an application. The scalable parallel interface uses time information to align packets and allows dynamic lane and/or path change, a large difference in transmission delays among lanes, and so on. The basic performance of the scalable parallel interface in 10Gb/s × 2 lanes is verified using an estimation board that is newly developed to evaluate the basic functions used in a Terabit LAN. The evaluation shows that the scalable parallel interface achieves a very low delay variation that is almost the same as that under back-to-back conditions. The difference in the delay variation between the scalable parallel interface and the back-to-back condition is approximately 10ns when the transmission delay time varies from 10µs to 1s.

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This paper proposes an enhanced method for estimating the a priori Signal-to-Disturbance Ratio (SDR) to be employed in the Acoustic Echo and Noise Suppression (AENS) system for full-duplex hands-free communications. The proposed a priori SDR estimation technique is modified based upon the Two-Step Noise Reduction (TSNR) algorithm to suppress the background noise while preserving speech spectral components. In addition, a practical approach to determine accurately the Echo Spectrum Variance (ESV) is presented based upon the linear relationship assumption between the power spectrum of far-end speech and acoustic echo signals. The ESV estimation technique is then employed to alleviate the acoustic echo problem. The performance of the AENS system that employs these two proposed estimation techniques is evaluated through the Echo Attenuation (EA), Noise Attenuation (NA), and two speech distortion measures. Simulation results based upon real speech signals guarantee that our improved AENS system is able to mitigate efficiently the problem of acoustic echo and background noise, while preserving the speech quality and speech intelligibility.

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Orthogonal frequency division multiplexing (OFDM) is a critical technology in 3G evolution systems, which can effectively avoid intra-cell interference, but may bring with serious inter-cell interference. Inter-cell interference cancellation is one of effective schemes taken in mitigating inter-cell interference, but for many existing schemes in inter-cell interference cancellation, various generalized spatial diversities are taken, which always bring with extra interference and blind spots, or even need to acquire extra information on source and channel. In this paper, a novel inter-cell interference mitigation method is proposed for 3G evolution systems. This method is based on independent component analysis in blind source separation, and the input signal to interference plus noise ratio (SINR) is set as objective function. By generalized eigenvalue decomposition and algorithm iterations, maximum signal noise ratio (SNR) can be obtained in output. On the other hand, this method can be worked with no precise knowledge of source signal and channel information. Performance evaluation shows that such method can mitigate inter-cell interference in a semi-blind state, and effectively improve output SNR with the condition that lower input SINR, higher input SNR and longer lengths of the processing frame.

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In a Direct-Sequence/Spread-Spectrum (DS/SS) system, a RAKE receiver is used to improve a bit error rate (BER) performance. The RAKE receiver can collect more signal energy through independent paths and achieve path diversity. The RAKE receiver obtains further diversity gain through fractional sampling. However, the power consumption of the RAKE receiver increases in proportion to a sampling rate and does not always maximize the signal-to-noise ratio (SNR). Therefore, sampling rate selection schemes have been proposed to reduce the average sampling rate without degrading the BER. These schemes select the tap positions and the sampling rate depending on channel conditions and the power consumption can be reduced. In this paper, sampling rate selection schemes for the DS/SS system are investigated through an experiment since there have been no numerical results through an experiment. Numerical results show that the power consumption can be reduced even through the experiment without the degradation of the BER.

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In the modern day, MPEG-4 image compression technique have been commonly applied in various indoor wireless communication systems. The efficient system design mostly relies on the joint source channel coding algorithms, which aim to reduce the complexity of channel coding process, while maintaining the quality of the receiving images. In this paper, we design the MAP source-controlled channel decoders with both random and semirandom interleavers for Rician slow flat block-fading channels. The MAP-Viterbi decoder employs the residual redundancy from zerotree symbol sequences of MPEG-4 HFS packets. The interleaving processes are done after the overall channel coding process to combat the block-fading effects. The computer simulations summarize the system performance in terms of average WER and PSNR (dB). With the interleavers, the significant improvement in PSNR of about 15-17dB is obtained for both ML and MAP decoding. Also in many cases, we obtain more improvement of about 0.2-0.4dB for using MAP decoding with the interleavers.

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This paper proposes bidirectional packet aggregation and coding (BiPAC), a packet mixing technique which jointly applies packet aggregation and network coding in order to increase the number of supportable VoIP sessions in wireless multi-hop mesh networks. BiPAC applies network coding for aggregated VoIP packets by exploiting bidirectional nature of VoIP sessions, and largely reduces the required protocol overhead for transmitting short VoIP packets. We design BiPAC and related protocols so that the operations of aggregation and coding are well-integrated while satisfying the required quality of service by VoIP transmission, such as delay and packet loss rate. Our computer simulation results show that BiPAC can increase the number of supportable VoIP sessions maximum by around 87% as compared with the case when the packet aggregation alone is used, and 600% in comparison to the transmission without aggregation/coding. We also implement BiPAC in a wireless testbed, and run experiments in an actual indoor environment. Our experimental results show that BiPAC is a practical and efficient forwarding method, which can be implemented into the current mesh hardware and network stack.

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In this letter, a low complexity transmitter is proposed for the downlinks of orthogonal frequency code division multiplexing (OFCDM) systems. The principle is based on a joint time-frequency spreading and inverse fast Fourier transform (TFS-IFFT), which combines the frequency spreading with partial stages of IFFT, so as to simplify the real-time processing. Compared with the conventional one, the proposed OFCDM transmitter is of lower real-time computational complexity, especially for those with large spreading factor or low modulation level. Furthermore, the proposed TFS-IFFT can also be applied to other frequency spreading systems, such as MC-CDMA, for complexity reduction.

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Localization is an important problem for Wireless Sensor Networks (WSN). The localization method can be categorized as range-free or range-based schemes. Since sensor nodes are usually cheap and small, the range-based schemes that require range measurement unit are unsuitable in WSN. The DV-hop algorithm is one of the range-free localization algorithms in which average hop-distance and hop counts are used for range estimation. But it requires heavy communication cost if the number of nodes increases in the network. Therefore, we propose a simple algorithm to reduce the communication cost and its performance is verified via computer simulations.

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Cognitive radio systems offer the opportunity to improve the spectrum utilization by detecting unused frequency bands while avoiding interference to primary users. This paper proposes a new algorithm for spectrum sensing, which is an energy detector using a hybrid (adaptive and fixed) threshold, in order to compensate the weak points of the existing energy detector in the distorted communication channel environment. Simulation results are presented which show that the performance of the new proposed scheme is better than the existing scheme using a fixed threshold or an adaptive threshold. Additionally, the performance is investigated in terms of several parameters such as the mobile speed and the probability of false alarms. The simulation results also show that the proposed algorithm makes the detector highly robust against fading, shadowing, and interference.

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When we design a robust vector quantizer (VQ) for noisy channels, an appropriate index assignment function should be contrived to minimize the channel-error effect. For relatively high rates, the complexity for finding an optimal index assignment function is too high to be implemented. To overcome such a problem, we use a structurally constrained VQ, which is called the sample-adaptive product quantizer (SAPQ) [12], for low complexities of quantization and index assignment. The product quantizer (PQ) and its variation SAPQ [13], which are based on the scalar quantizer (SQ) and thus belong to a class of the binary lattice VQ [16], have inherent error resilience even though the conventional affine index assignment functions, such as the natural binary code, are employed. The error resilience of SAPQ is observed in a weak sense through worst-case bounds. Using SAPQ for noisy channels is useful especially for high rates, e.g., > 1 bit/sample, and it is numerically shown that the channel-limit performance of SAPQ is comparable to that of the best codebook permutation of binary switching algorithm (BSA) [23]. Further, the PQ or SAPQ codebook with an affine index assignment function is used for the initial guess of the conventional clustering algorithm, and it is shown that the performance of the best BSA can be easily achieved.

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In this paper, for systematically evaluating estimation methods of node characteristics, we first propose a social network generation model called LRE (Linkage with Relative Evaluation). LRE is a network generation model, which aims to reproduce the characteristics of a social network. LRE utilizes the fact that people generally build relationships with others based on relative evaluation, rather than absolute evaluation. We then extensively evaluate the accuracy of the estimation method called SSI (Structural Superiority Index). We reveal that SSI is effective for finding good nodes (e.g., top 10% nodes), but cannot be used for finding excellent nodes (e.g., top 1% nodes). For alleviating the problems of SSI, we propose a novel scheme for enhancing existing estimation methods called RENC (Recursive Estimation of Node Characteristic). RENC reduces the effect of noise by recursively estimating node characteristics. By investigating the estimation accuracy with RENC, we show that RENC is quite effective for improving the estimation accuracy in practical situations.

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How to minimize the number of mirroring resources under a QoS constraint (resource minimization problem) is an important issue in content delivery networks. This paper proposes a novel approach that takes advantage of the parallelism of dynamically reconfigurable processors (DRPs) to solve the resource minimization problem, which is NP-hard. Our proposal obtains the optimal solution by running an exhaustive search algorithm suitable for DRP. Greedy algorithms, which have been widely studied for tackling the resource minimization problem, cannot always obtain the optimal solution. The proposed method is implemented on an actual DRP and in experiments reduces the execution time by a factor of 40 compared to the conventional exhaustive search algorithm on a Pentium 4 (2.8GHz).

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This paper analyzes transient behaviors of the polariza-tion-mode-dispersion (PMD) vector for the Foschini and Poole's birefringence vector model. We find an asymptotic solution of the corresponding Fokker-Planck equation representing the solution as a superposition of angular components characterized by the Legendre polynomials. The distribution tail for the PMD vector magnitude evolves slowly to the Maxwellian owing to the residual couplings between adjacent angular components. Of particular interest, the distribution tail for the PMD vector magnitude lies well below the Maxwellian fit during the transient.

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With the wide expansion of voice services over the IP networks (VoIP), the volume of this delay sensitive traffic is steadily growing. The current packet schedulers for IP networks meet the delay constraint of VoIP traffic by simply assigning its packets the highest priority. This technique is acceptable as long as the amount of VoIP traffic is relatively very small compared to other non-voice traffic. With the notable expansion of VoIP applications, however, the current packet schedulers will significantly sacrifice the fairness deserved by the non-voice traffic. In this paper, we extend the conventional Deficit Round-Robin (DRR) scheduler by including a packet classifier, a Token Bucket and a resource reservation scheme and propose an integrated packet scheduler architecture for the growing VoIP traffic. We demonstrate through both theoretical analysis and extensive simulation that the new architecture makes it possible for us to significantly improve the fairness to non-voice traffic while still meeting the tight delay requirement of VoIP applications.

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We propose a novel adaptive segment repair mechanism to improve traditional MPLS (Multi-Protocol Label Switching) failure recovery. The proposed mechanism protects one or more contiguous high failure probability links by dynamic setup of segment protection. Simulations demonstrate that the proposed mechanism reduces failure recovery time while also increasing network resource utilization.

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Mobile Ad Hoc Networks (MANETs) have inherently dynamic topologies. Due to the distributed, multi-hop nature of these networks, random mobility of nodes not only affects the availability of radio links between particular node pairs, but also threatens the reliability of communication paths, service discovery, even quality of service of MANETs. In this paper, a novel Markov chain model is presented to predict link availability for MANETs. Based on a rough estimation of the initial distance between two nodes, the proposed approach is able to accurately estimate link availability in a random mobility environment. Furthermore, the proposed link availability estimation approach is integrated into Max-Min d-clustering heuristic. The enhanced clustering heuristic, called M⁴C, takes node mobility into account when it groups mobile nodes into clusters. Simulation results are given to verify the approach and the performance improvement of clustering algorithm. It also demonstrates the adaptability of M⁴C, and shows that M⁴C is able to achieve a tradeoff between the effectiveness of topology aggregation and cluster stabilities. The proposed algorithm can also be used to improve the availability and quality of services for MANETs.

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Efficient mobility management is one of the most important challenges in mobile networks. Since roaming between wireless cells can cause long handoff latency and high packet loss, it can result in poor quality of services for delay-sensitive applications and for applications requiring reliable transmission. In this paper, we propose a new seamless micro-mobility management framework for Micro Mobile MPLS (called MiM-MPLS) to overcome the packet loss problem while maintaining low handoff latency. In addition, we propose a combination of MiM-MPLS and the existing MFC-Micro Mobile MPLS [3],[4] (called MiM-MFC-MPLS) to further enhance the handoff performance. MiM-MPLS and MiM-MFC-MPLS provide efficient handoff mechanisms through the use of pre-established Label Switched Paths (LSPs), the L2 trigger, and local registration. The handoff mechanisms for (L2 and L3) intra handoffs and for an inter handoff in these frameworks are described in detail. The performance is analyzed in terms of the signaling cost of registration updates, handoff latency, packet loss, buffer size requirement, and delay jitter. Compared to existing handoff frameworks, the results show that the proposed MiM-MPLS and MiM-MFC-MPLS can provide no packet loss while maintaining low signaling cost of registration updates, handoff latency, and delay jitter at the cost of some buffer space. In particular, MiM-MFC-MPLS has the best handoff performance (i.e., the lowest signaling cost of registration updates, handoff latency, and delay jitter) among all the frameworks that we considered at the cost of some buffer space.

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Mobility-aware quality of service (QoS) signaling is crucial to provide seamless multimedia services in the ambient environment where mobile nodes may move frequently between different wireless access networks. The mobility of an IP-based node in ambient networks affects routing paths, and as a result, can have a significant impact on the operation and state management of QoS signaling protocols. In this paper, we first analyze the impact of mobility on QoS signaling protocols and how the protocols operate in mobility scenarios. We then propose an efficient mobility-aware QoS signaling protocol which can operate adaptively in ambient networks. The key features of the protocol include the fast discovery of a crossover node where the old and new paths converge or diverge due to handover and the localized state management for seamless services. Our analytical and simulation/experimental results show that the proposed/implemented protocol works better than existing protocols in the IP-based mobile environment.

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In this paper, the performance of the Wireless Sensor Network (WSN) using fixed relay nodes and Multiple-Input Multiple-Output (MIMO) technology was evaluated based on the correlated channel capacity of MIMO system and the number of sensor node served by the system. Moreover, the performance evaluation of the proposed algorithm, which is used to find the optimum distance to place the relay nodes from sink node, is done not only with AF relaying and spatial correlation effect, but also with Decode-and-Forward (DF) relaying scheme. The results show that the relay gain (a ratio between the maximum number of sensors satisfying the required channel capacity in 7-cell topology to the number of sensor nodes in sink cell) is affected strongly by the spatial correlation at high required channel capacity but little at low required channel capacity. The results also show that the relay gain can be improved remarkably by using the DF relaying scheme, and that the validity of the proposed algorithm holds for any relaying scheme, spatial correlation effect and different antenna size.

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In this paper, we analyze the stability of XCP (eXplicit Control Protocol) in a network with heterogeneous XCP flows (i.e., XCP flows with different propagation delays). Specifically, we model a network with heterogeneous XCP flows using fluid-flow approximation. We then derive the conditions that XCP control parameters should satisfy for stable XCP operation. Furthermore, through several numerical examples and simulation results, we quantitatively investigate effect of system parameters and XCP control parameters on stability of the XCP protocol. Our findings include: (1) when XCP flows are heterogeneous, XCP operates more stably than the case when XCP flows are homogeneous, (2) conversely, when variation in propagation delays of XCP flows is large, operation of XCP becomes unstable, and (3) the output link bandwidth of an XCP router is independent of stability of the XCP protocol.

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Compared with traditional static anonymous communication networks, the P2P architecture can provide higher anonymity in communication. However, the P2P architecture also leads to more challenges, such as route, stability, trust and so on. In this paper, we present WindTalker, a P2P-based low-latency anonymous communication network. It is a pure decentralized mix network and can provide low-latency services which help users hide their real identity in communication. In order to ensure stability and reliability, WindTalker imports “seed nodes” to help a peer join in the P2P network and the peer nodes can use gossip-based protocol to exchange active information. Moreover, WindTalker uses layer encryption to ensure the information of relayed messages cannot be leaked. In addition, malicious nodes in the network are the major threat to anonymity of P2P anonymous communication, so WindTalker imports a trust mechanism which can help the P2P network exclude malicious nodes and optimize the strategy of peer discovery, tunnel construction, and relaying etc. in anonymous communications. We deploy peer nodes of WindTalker in our campus network to test reliability and analyze anonymity in theory. The network measurement and simulation analysis shows that WindTalker can provide low-latency and reliable anonymous communication services.

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This paper proposes a new antenna array calibration technique which uses frequency selection in orthogonal frequency division multiple access (OFDMA)/time division duplexing (TDD) systems. In the proposed method, subbands or frequencies of good channel conditions are initially selected for channel measurements. The relative calibration is performed at the selected subbands, which compensates for mismatch of analogue gains in multiple antennas using the measured uplink and downlink channel parameters. Furthermore, the calibration parameters are interpolated in the frequency domain for the whole bandwidth. The proposed calibration maintains accurate channel reciprocity for the whole bandwidth compared to the conventional calibration which does not use the frequency selection. The proposed calibration technique is effective in exploiting channel reciprocity at both base station and terminals with feasible amount of feedback and low-cost operation.

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Multiuser — Multiple Input Multiple Output (MU-MIMO) techniques were proposed to increase spectrum efficiency; a key assumption was that the Mobile Terminals (MTs) were simple with only a few antennas. This paper focuses on the Block Diagonalization algorithm (BD) based on the equal power allocation strategy as a practical MU-MIMO technique. When there are many MTs inside the service area of the access point (AP), the AP must determine, at each time slot, the subset of the MTs to be spatially multiplexed. Since the transmission performance depends on the subsets of MTs, the user selection method needs to use the Channel State Information (CSI) obtained in the physical layer to maximize the Achievable Transmission Rate (ATR). In this paper, we clarify the relationship between ATR with SU-MIMO and that with MU-MIMO in a high eigenvalue channel. Based on the derived relationship, we propose a new measure for user selection. The new measure, the eigenvalue decay factor, represents the degradation of the eigenvalues in null space compared to those in SU-MIMO; it is obtained from the signal space vectors of the MTs. A user selection method based on the proposed measure identifies the combination of MTs that yields the highest ATR; our approach also reduces the computational load of user selection. We evaluate the effectiveness of user selection with the new measure using numerical formulations and computer simulations.

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This paper proposes a preamble structure that raises the throughput of the physical layer of a MIMO-OFDM system that complies with IEEE 802.11n. The proposed preamble is based on two important criteria that have not been considered in conventional preamble structures but have an important bearing on realization of IEEE 802.11n. One is backward compatibility with legacy devices, such as IEEE 802.11a or IEEE 802.11g. The other is accurate AGC for MIMO data payload of the packet. Computer simulations and a laboratory test demonstrate that the proposed preamble is superior to the other preambles with respect to those two important criteria. The results were submitted to the task group N under the IEEE 802.11 working group and contributed to development of the current draft of IEEE 802.11n.

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For phased and adaptive arrays of antennas, an optimal arrangement of antenna elements is essential to avoid grating lobes in the visible angular region of the array. Large sidelobes cause degradation in signal-to-noise ratio; grating lobes, in the worst case, cause malfunctions. One method of evaluating sidelobe level is square integration. For a given set element positions, evaluation by square integration of the sidelobes involves Fourier transform and numerical integration. For faster evaluation, we developed an equivalent transform algorithm that requires no numerical Fourier transform or integration. Using this new algorithm, we introduced a fast trial-and-error algorithm that iteratively applies random perturbation to the array, evaluates the function, and minimizes it. A number of separate runs of this algorithm have been conducted under the constraint of 3-fold rotational symmetry for stability. The optimal output, for which the function is minimized, is a uniformly spaced equilateral-triangular-type arrays that, unfortunately, has unwanted grating lobes. However the algorithm also yields variations trapped at local minima, some of which do not have grating lobes and whose sidelobe peaks are sufficiently low within a wide angular region. For the case N=12, a characteristic triagular-rectangular-type array often arises, which has not only better sidelobe properties as evaluated by square-integration and peak sidelobe, but also sufficient element-to-element clearance. For the case N=36, one of the results achieves a peak-sidelobe level of -8dB, with a minimum element-to-element separation of 0.76 wavelength.

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This paper presents the analysis and design of a reflection-cancelling transverse slot-pair array antenna with baffles by using the Spectrum of Two-Dimensional Solutions (S2DS) method. For the transverse slot array, the slot spacings with more than one free-space wavelength cause the grating-lobes. The baffles suppress the grating-lobes effectively. A one-dimensional slot array is extracted from the 2D array with in-phase excitation by assuming periodicity in the transversal direction. The uniform excitation over the finite array is synthesized iteratively to demonstrate the fast and accurate results by S2DS. A unit design model with the baffles is introduced to determine the initial parameters of the slot-pairs, which greatly accelerate the iterations process. Experiments at 25.3GHz demonstrate the suppression of the grating lobes to the level less than -20.0dB and also the good uniformity of the aperture field distribution.

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This letter proposes a new retransmission persistence management scheme for selective repeat automatic repeat request (SR-ARQ). By considering the overall traffic load that has to be managed by SR-ARQ, the proposed scheme arbitrates the retransmission persistence to prevent an abrupt delay increment due to excessive link-level local retransmissions. OPNET simulations show that SR-ARQ performs better with the proposed scheme than with a fixed value of retransmission persistence in terms of the throughput of transmission control protocol (TCP).

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This letter presents the optimal routing by the intermediate model; a construction that lies between the pipe and hose models. We show that it is a practical way of realizing optimal routing. A formulation extended from the pipe model to the intermediate model can not be solved as a regular linear programming (LP) problem. Our solution, the introduction of a duality theorem, successfully turns our problem into an LP formulation that can be easily solved. Numerical results show that the intermediate model has better routing performance than the hose model.

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In this letter we show that the dynamic optimal PCID allocation problem in LTE systems is NP-complete. Subsequently we provide a near-optimal solution using SON which models the problem using new merge operations and explores the search space using a suitable randomized algorithmic approach. Two feasible options for dynamic auto-configuration of the system are also discussed. Simulation results point out that the approach provides near-optimal auto-configuration of PCIDs in computationally feasible time.

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An ICI (Inter-Cell Interference) mitigation algorithm for exploiting macroscopic diversity for an up-link OFDMA (Orthogonal Frequency Division Multiple Access) system is proposed. To reduce the influence of carrier collision, the order of resource allocation is coordinated based on the location of each MS (Mobile Station) and the associated carrier group. This consideration significantly reduces ICI and enhances throughput at the boundary region.

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A new random access channel (RACH) preamble detection scheme using variable coherent correlation intervals (CCIs) is proposed. It is shown first that it is enough to employ two CCIs for supporting a user equipment (UE) velocity of 300km/h, and then a CCI selection criterion is proposed. Computer simulation results indicate that the proposed scheme can provide a robust detection performance in time-varying fading channel environments.

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Combining the sphere decoding (SD) algorithm and the sequential detection method, we propose an adaptive group detection (AGD) scheme based on the sort-descending QRD (S-D-QRD) for V-BLAST architectures over an i.i.d. Rayleigh flat fading channel. Simulation results show that the proposed scheme, which encompasses the SD algorithm and the sequential detection method as two extreme cases in a probability sense, can achieve a very flexible tradeoff between the detection performance and computational complexity by adjusting the group parameter.

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In conjunction with a first-order Taylor series approximation of the spatial scanning vector, this letter presents an iterative multiple signal classification (MUSIC) direction-of-arrival (DOA) estimation for code-division multiple access signals. This approach leads to a simple one-dimensional optimization problem to find each iterative optimal search grid. It can not only accurately estimate DOA, but also speed up the estimating process. Computer results demonstrate the effectiveness of the proposed algorithm.

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In this paper, we propose a novel frequency-hopping scheme in order to improve the BER (Bit Error Rate) performance of the Partial Block MC-CDMA (PB/MC-CDMA) system. The joint carrier distribution and frequency hopping (JDFH) scheme achieves the optimal frequency diversity gain while avoiding interference. By contrast, the conventional FH scheme only avoids interference, and the frequency interleaving scheme achieves only frequency diversity. The JDFH scheme thus performs better than conventional schemes, such as carrier FH, block FH, or frequency interleaving. Through computer simulations, we confirmed the superior performance of the PB/MC-CDMA system when using the JDFH scheme.

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If ranging processes are not frequent in an uplink OFDMA system, timing synchronization between the base and mobile stations may not be maintained and the performance may be degraded. This paper proposes a random access scheme in which a short OFDMA symbol is transmitted to maintain the orthogonality with timing offsets. A short symbol is constructed by inserting zero-padding to an OFDMA symbol.

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In this letter, Doppler spread estimation in different Doppler spectra is investigated and some efficient methods are proposed to calculate the maximum Doppler frequency from autocorrelation function easily.

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In this letter, we present a closed-form bound of the average bit error rate (BER) performance for the multi-hop amplify-and-forward (AF) relaying systems with fixed gain in Rayleigh fading channel. The proposed bound is derived from the probability density function (PDF) of the overall multi-hop relay channel under the assumption of asymptotic high signal-to-noise ratio (SNR) at every intermediate relays. When intermediate relays actually operate at finite SNR, the proposed BER bound becomes looser as the SNR of the last hop increases. In order to reflect the effect of all the noise variances of relay links to the BER evaluation, a hop-indexed recursive BER approximation is proposed, in which the proposed bound of BER under asymptotic high SNR is used. The simulation results manifest that the proposed hop-indexed recursive BER approximation can not only guarantee accuracy regardless of the SNR of the last hop but also provide higher accuracy than the previous works.

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