OFDM系统中基于改进阈值的DFT信道估计算法

在正交频分复用(Orthogonal Frequency Division Multiplexing,OFDM)系统中,传统的离散傅里叶变换(Discrete Fourier Transform,DFT)信道估计算法对循环前缀之内的噪声没有得到抑制,只是抑制了循环前缀之外的噪声.以循环前缀之外的噪声点的能量值来设置阈值对于循环前缀之内噪声的滤除不是很理想,为此本文通过将循环前缀内外的样本点分别排序,分别选取其中一部分样本能量值的平均值,并乘以适当系数,以两部分和作为一种新的时域阈值来抑制噪声.仿真结果表明,本文改进的算法能很好地抑制循环前缀之内的噪声,保留了有效的信号,误码率和均方误差均比得到进一步的降低.     

在无线双向通信中基于CRC-NC抵抗污染攻击方案

本文提出一种CRC-NC (Cyclic Redundancy Check-Network Coding）方案．在无线双向通信网络中,该方案通过结合网络编码和循环冗余校验码技术,对接收到的消息的进行可信度检测,能有效地降低目的节点解码误码率,并抵抗污染攻击．该方案中节点与节点在中继节点的辅助下相互发送消息,并利用对方节点和中继节点发送的消息解码．若节点直接从对方节点获得的消息中有S个消息正确,该节点将对方节点发送的消息和中继节点发送的消息进行组合,并计算组合消息的汉明重量,从中选择K-S个最小的汉明重量所对应的由中继节点发送的消息解码．通过仿真结果表明,与基于随机选取方案和加密方案相比,该方案能有效的降低节点解码差错概率．      

Saturation throughput analysis of IEEE 802.11 in the presence of non ideal transmission channel and capture effects

In this paper, we provide a saturation throughput analysis of the IEEE 802.11 protocol at the data link layer by including the impact of both transmission channel and capture effects in Rayleigh fading environment. Impacts of both non-ideal channel and capture effects, specially in an environment of high interference, become important in terms of the actual observed throughput. As far as the 4-way handshaking mechanism is concerned, we extend the multi-dimensional Markovian state transition model characterizing the behavior at the MAC layer by including transmission states that account for packet transmission failures due to errors caused by propagation through the channel. This way, any channel model characterizing the physical transmission medium can be accommodated, including AWGN and fading channels. We also extend the Markov model in order to consider the behavior of the contention window when employing the basic 2-way handshaking mechanism. Under the usual assumptions regarding the traffic generated per node and independence of packet collisions, we solve for the stationary probabilities of the Markov chain and develop expressions for the saturation throughput as a function of the number of terminals, packet sizes, raw channel error rates, capture probability, and other key system parameters. The theoretical derivations are then compared to simulation results confirming the effectiveness of the proposed models.     

Enhanced spectrum sensing using a combination of energy detector,matched filter and cyclic prefix

In this paper, we propose a sensing scheme based on energy detection, matched filter and cyclic prefix. Both Equal Gain Combining (EGC) and optimal combination of the aforementioned detectors are investigated in cooperative and non-cooperative spectrum sensing scenarios. In packet transmission systems such as OFDM (Orthogonal Frequency Division Multiple access) systems, the proposed scheme takes advantage of utilizing more samples than individual detectors, i.e., cyclic prefix, training or pilot samples, and data payload samples. The proposed combine-sensing scheme offers higher detection probability and lower false alarm probability, as compared with the performance of individual detectors over the same frame duration. Simulation results are congruent with the theoretical curves and confirm the validity of our derivations.     

TCP Window-Based Flow-Oriented Dynamic Assembly Algorithm for OBS Networks

In transport control protocol (TCP) over optical burst switching (OBS) networks, TCP window size and OBS parameters, including assembly period and burst dropping probability, will impact the network performance. In this paper, a parameter window data dropping probability(WDDP), is defined to analyze the impact of the assembly and the burst loss on the network performance in terms of the round trip time and the throughput. To reduce the WDDP without introducing the extra assembly delay penalty, we propose a novel TCP window based flow-oriented assembly algorithm dynamic assembly period (DAP). In the traditional OBS assembly algorithms, the packets with the same destination and class of service (CoS) are assembled into the same burst, i.e., the packets from different sources will be assembled into one burst. In that case, one burst loss will influence multiple TCP sources. In DAP, the packets from one TCP connection are assembled into bursts, which can avoid the above situation. Through comparing the two consecutive burst lengths, DAP can track the variation of TCP window dynamically and update the assembly period for the next assembly. In addition, the ingress node architecture for the flow-oriented assembly is designed. The performance of DAP is evaluated and compared with that of fixed assembly period (FAP) over a single TCP connection and multiple TCP connections. The results show that DAP performs better than FAP at almost the whole range of burst dropping probability.     

Performance evaluation of a new BPSK-OFDM timing estimation algorithm

Previously, the timing estimation in OFDM system employs cyclic prefix has been presented with the assumption of independent identified distributed OFDM symbol. The information used to find the symbol synchronization depends on the length of cyclic prefix. Actually, the data in the BPSK-OFDM modulated symbol, transferred by inverse fast Fourier transform, is with a complex symmetry character. With this character, more information from the whole OFDM symbol could be provided for the symbol timing. In this paper, the proposed algorithm uses the autocorrelation of the received signal to obtain the symbol timing. The simulation of Beek’s algorithm and the proposed scheme are presented. The performances of these algorithms are evaluated based on the missing symbol probability and the estimator mean square error. The simulation results show that the proposed algorithm is suitable to achieve a better symbol synchronization.     

Packet dropping statistics of a data-link protocol for wireless local communications

We consider a data transmission system over a wireless channel, where packets are queued at the transmitter. Automatic repeat request is employed as the error control scheme, by which packets corrupted due to channel impairments are immediately retransmitted. We study the statistics of the packet dropping process due to buffer overflow. In particular, we propose a Markov approximation to model such losses. The delay performance of the packets that are admitted to the queue is studied, and extensions of the analysis to general nonindependent arrival processes are also presented.     

Two-Way Handshaking Circular Sequential $k$-Out-of- $n$ Congestion System

Many communication systems require a two-way, or three-way handshaking process to improve their dependability & authenticity in order to achieve a more successful operation. In this paper, we present a new two-way handshaking reliability model based upon threshold-based cryptography systems. Such systems require a two-way handshaking process to i) establish a group of participated servers in the first handshaking process, and ii) calculate a cipher with successfully connected servers collaboratively in the second handshaking process. When the servers are attempted, each server has three known connection probabilities in the following three states: i) successful, ii) breakdown, and iii) congested. These connection probabilities are unchanged in both handshaking processes. During the first handshaking process, we establish connections that more than servers are willing to participate. For the second handshaking process, the system becomes successful as soon as we can connect these servers successfully again. Because we need to connect servers successfully in the second handshaking process, we would rather connect additional servers besides the servers required to be connected successfully in the first handshaking process. This preference will minimize the chance that the system breaks down when fewer than servers can be reconnected successfully in the second handshaking process. We refer to this system as a Two-Way Handshaking Circular Sequential-out-of-Congestion (TWHCSknC) system. In this paper, we derived analytical formulas for the system's successful probability & average stop length, and we showed that the TWHCSknC system is a communication system with an efficient two-way handshaking process.     

Channel Allocation for GPRS with Buffering Mechanisms

General Packet Radio Service (GPRS) provides mobile users end-to-end packet-switched services by sharing the radio channels with voice and circuit-switched services. In such a system, radio resource allocation for circuit-switched and packet-switched services is an important issue, which may affect the QoS for both services significantly. In this paper, we propose two algorithms: Dynamic Resource Allocation with Voice and Packet queues (DRAVP) and Dynamic Resource Allocation with Packet and Voice queues (DRAPV) for channel allocation of the voice calls and packets. We propose analytic and simulation models to investigate the performance of DRAVP and DRAPV in terms of voice call incompletion probability, packet dropping probability, average voice call waiting time, and average packet waiting time. Our study indicates that the buffering mechanism for GPRS packets significantly increase the acceptance rate of GPRS packets at the cost of slightly degrading the performance of voice calls.     

An Efficient Cyclic Prefix Reconstruction Technique for MIMO Single-Carrier Frequency-Domain Equalization

In this letter, an efficient cyclic prefix reconstruction (CPR) technique with turbo equalization is developed for a multi-input multi-output (MIMO) single-carrier frequency-domain equalization (SC-FDE) system. The proposed method consists of pre-processing estimation (PPE) and residual inter-carrier interference suppression (RICIS). The PPE is employed to compute initial values of MIMO turbo equalization, and the RICIS is used to mitigate residual intercarrier interference (ICI) after each iteration of the CPR. By applying the proposed method to MIMO SC-FDE system with insufficient cyclic prefix (CP), we can significantly improve its error performance, obtaining both the benefits of multiplexing gain and spectral efficiency gain     

光突发交换网络中一种基于LAUC的批调度算法

为了降低突发丢失率和调度复杂度,提出了一种基于LAUC的批调度算法.该算法首先缓存多个突发控制包,当最大缓存时间到达时,根据突发到达顺序批调度处理缓存中的突发控制包,并采用LAUC算法来为突发预留信道资源.其优势在于:计算复杂度与LAUC一样低,因为它只增加了突发控制包缓存和改变了LAUC算法的调度顺序;突发丢失率比较低,仿真结果表明它的突发丢失率比LAUC算法要低,当最大缓存时间大于偏置时间的最大差值时,它的突发丢失率比LAUC-VF算法还要低.     

一种基于OFDM电力线通信系统信道估计方法

通过实际测量给出低压电力线的信道模型,在此基础之上提出了利用循环前缀对信道进行估计的方法,即在梳状导频模式时,使用导频获得导频子信道频率响应后,对其在频域方向进行插值得到数据子信道响应,在其后没有导频的序列的数据周期内,使用循环前缀对信道进行估计,从而起到改善系统误比特性能的目的。     

Modeling OFDM irreducible BER with impact of CP length and CFO in multipath channel with small delay dispersion

In this paper, we propose an analytical model for estimating the irreducible bit error ratio (BER) in multipath channel with small‐extent delay dispersion, such as indoor, where the signal‐to‐noise ratio is high, implying dominance of inter‐symbol interference as error‐generating mechanism. Both channel and overall orthogonal frequency‐division multiplexing (OFDM) symbols are modeled stochastically, resulting with novel expression for the residual BER prediction that is shown to analytically distinguish power delay profiles with equal delay spreads but having different profile shapes. In addition, the model could simply accommodate insertion of cyclic prefix onto the OFDM symbol, providing a means for either testing adequacy of any applied (standard) cyclic prefix length or finding its optimal value as a compromise between the performance enhancement achieved by inserting cyclic prefix and the consequently added redundancy. Finally, the model was modified as to include the analysis of effects of subcarrier frequency inaccuracy or Doppler shift, by adding additional equivalent delay dispersion with equal effect on BER degradation, while considering the system virtually free of carrier frequency offset. All analytically achieved results and conclusions are tested and successfully verified by conducted extensive Monte Carlo simulations. Copyright © 2015 John Wiley & Sons, Ltd.     

OFDM系统中符号和频偏联合同步算法研究

正交频分复用系统（OFDM）对同步误差十分敏感,当存在同步误差时．会引起严重的载波间干扰（ICI）和符号间干扰（ISI）。在AWGN信道以及频率选择衰落信道下,对基于循环前缀最大似然算法进行了仿真。并提出了基于循环前缀叠加子导频的符号和频偏联合同步新算法,实现了整数倍频偏（IFO）估计,仿真分析表明该算法具有较好的性能,并且计算量小。     

Packet throughput in slotted ALOHA DS/SSMA radio systems withrandom signature sequences

Packet throughput figures are obtained for direct sequence spread spectrum multiple access (DS/SSMA) slotted ALOHA radio systems where all users employ random signature sequences from bit-to-bit within all transmitted packets. These calculations use an improved Gaussian approximation technique which gives accurate bit error probabilities and also incorporates the effect of bit-to-bit error dependence within each packet in the multiaccess interference environment. Numerical results are given for packets which employ varying amounts of block error control, and a comparison is made with results obtained by other methods which ignore the effects of bit-to-bit error dependence within each packet in the multiaccess interference environment. Numerical results are given for packets which employ varying amount of block error control, and a comparison is made with results obtained by other methods which ignore the effects of bit-to-bit error and/or employ less-accurate Gaussian approximations to the probability of data bit error. Maximum throughput per unit bandwidth figures are calculated which compare favorably to similar figures for narrowband signaling techniques     

Implication of dropping packets from the front of a queue

When congestion occurs in a packet queuing system, packets can be dropped from the rear or the front of the queue. It is demonstrated that the probability of a packet being dropped is the same in systems with rear and front packet dropping. It is shown that the probability of a packet being delayed longer than a given value in a system with front dropping is less than or equal to that in a system with rear dropping. It is further illustrated that front dropping not only improves the delay performance on an internodal link, but also provides the overall loss performance for time constrained traffic such as packet voice     

DRED‐MP: Queue management with multiple levels of drop precedence

There is current interest in differentiated service architectures where packets with different priorities can share the same queue. In the case of congestion, packets marked with higher drop probability are preferentially dropped in order to make buffer room for packets marked with lower drop probability. Active queue management (AQM) based on randomized packet dropping has become a key component of this packet forwarding model. This paper extends our previously developed AQM algorithm called DRED with multiple packet drop precedence to allow for priority treatment of traffic in a network. The main advantage of using the DRED algorithm is the lower parameter configuration complexity it offers and the ease of configuration for a wide range of network conditions. Copyright © 2004 John Wiley & Sons, Ltd.     

AAL2分组话音复接器缓冲器队列门限值的确定方法

该文研究了带比特丢弃的AAL2分组话音复接器缓冲器队列门限值的确定方法,提出用话音分组作为缓冲器队列门限值的单位,给出了确定门限值的计算公式,并对输出链路容量为384kb/s的情况进行了计算机仿真。仿真结果表明,作者提出的门限值的确定方法可获得较小的平均分组时延和较低的平均分组丢失率,计算简便,易于实现,是一种很好的确定缓冲器队列门限值的方法。     

Modified CAZAC Sequence Based Timing Synchronization Scheme for OFDM System

Wireless Personal Communications - Active queue management schemes are used to reduce the number of dropped packets at the routers. Random early detection uses dropping probability which is...     

Design and performance evaluation of an RRA scheme for voice‐data channel access in outdoor microcellular environments

In PCS networks, the multiple access problem is characterized by spatially dispersed mobile source terminals sharing a radio channel connected to a fixed base station. In this paper, we design and evaluate a reservation random access (RRA) scheme that multiplexes voice traffic at the talkspurt level to efficiently integrate voice and data traffic in outdoor microcellular environments. The scheme involves partitioning the time frame into two request intervals (voice and data) and an information interval. Thus, any potential performance degradation caused by voice and data terminals competing for channel access is eliminated. We consider three random access algorithms for the transmission of voice request packets and one for the transmission of data request packets. We formulate an approximate Markov model and present analytical results for the steady state voice packet dropping probability, mean voice access delay and voice throughput. Simulations are used to investigate the steady state voice packet dropping distribution per talkspurt, and to illustrate preliminary voice-data integration considerations. This revised version was published online in June 2006 with corrections to the Cover Date.