The main blind channel identification methods based on second order statistics include linear prediction algorithm (LPA), outer product decomposition algorithm (OPDA), multi-step linear prediction algorithm (MSLP), least square smoothing algorithm (LSS), and constrained minimum output energy algorithm (CMOE). But the simulations show that the channel couldn't be identified. This paper will analyze these algorithms and point out the reason. The problem happens at the second step. The method based on second order statistics induces an unidentified sign and an indeterminable scale that result in the failure of channel identification. This problem can be resolved in the digital communication system.

A new frequency synchronization method is proposed in this paper. It uses PN (pseudo-noise) sequences and cyclic prefix for synchronization in OFDM (orthogonal frequency division multiplexing) system. Several subcarriers estimation range can be reached using the new method. When cyclic prefix is used in fine frequency synchronization, the MSE (mean square error) of the proposed method is less than the method only used PN sequences for the span of the correlated data is large. This has been verified by simulations in AWGN channel and Rayleigh channel.

It is a difficult problem to detect signal in low signal to noise ratio condition. A signal detection algorithm based on AR model is presented. The coefficients of AR model are found using RLS algorithm. The maximum magnitude of coefficients is summed to make the decision variable and compared to threshold. This algorithm has many advantages, such as small requirement of computation, high speed and independent of signal form, etc. Detecting complex sinusoid, QPSK and GMSK signals in white Gaussian noise under low signal to noise ratio condition are simulated. Results show that this algorithm can achieve better performance.

In the downlink of WCDMA systems, the different users are multiplexed synchronously to transmission channel by short orthogonal spreading codes that user specific and a long overlay scrambling code that is base-station specific. It is well known that in a multiple-user synchronous DS-CDMA system with multi-path fading, the widely used rake receiver is sub-optimal. Moreover, as the chip rate increases in high rate wireless applications, the underlying multi-path channels become time dispersive, and introduce frequency selective effects. For WCDMA downlink, receivers based on channel equalization at chip level have been proposed to ensure adequate performance even with a high number of active users. These receivers equalize the channel prior to the dispreading, thus restoring the orthogonality of users and resulting multiple access interference (MAI) suppression. The chip-level equalizer has proven to be one of the most promising terminal receivers for WCDMA/FDD downlink. In this paper, we summarize the technology of the chip-level equalization for WCDMA downlink.

Orthogonal Frequency Division Multiplexing (OFDM)is one of the best candidates for the future mobile communication systems. This paper analyzes channel estimation algorithms for OFDM systems not only for the downlink but also for the uplink. With reasonable constraint and well-designed preambles for each user, the DFT-based uplink channel estimation algorithm on the uplink can achieve good estimation accuracy without sacrificing much system capacity. Computer simulation demonstrates effectiveness of channel estimation algorithms and conclusion is followed.

Due to scarce radio bandwidth, efficient mobility management method is necessary for the PCS. This paper presents a new mobility management method for the PCS networks. It consists of three parts: cell coding scheme, location update strategy and paging strategy. The cell code scheme, which provides information about the topology of the network to mobile terminals, is the basis for the location update strategy in this paper. The location update strategy is a simplified derivative of distance-based scheme. An adaptive distance threshold is obtained by exploiting the mobile terminal’s individual call and mobility pattern. Since the location update strategy is not designed for certain mobility model such as random walk or fluid flow model, the computation of distance threshold is not only simplified but also better matches the real moving behavior of different mobile terminals. The paging strategy in this method is two-step. The moving direction vector is introduced here to optimize the paging procedure. Two-step paging will significantly reduce the signaling cost for paging. And it can be easily generalized to multi-step paging within the call setup delay constraint. A numerical analysis between mobile terminals with various moving and call patterns is provided. The mobility management method helps optimize the location update cost for all kinds of users.

Hierarchical mobile IPv6 makes the mobility management localized. Registration with HA is only needed while MN moving between MAP domains. This paper proposed an adaptive mobility management scheme based on the hierarchical mobile IPv6. The scheme focuses on the MN operation as well as MAP operation during the handoff. Adaptive MAP selection algorithm can be used to select a suitable MAP to register with once MN moves into a new subnet while MAP can thus adaptively changing his management domain. Furthermore, MAP can also adaptively changes its level in the hierarchical referring on the service load or other related information. Detailed handoff algorithm is also discussed in this paper.

Position information of individual node is useful in implementing functions such as routing and querying in ad-hoc networks. But adding GPS receivers or other sophisticated sensors to every node can be expensive. In this paper, we present the distributed, hop by hop positioning algorithm for solving the problem of positioning nodes within an ad hoc, wireless network of low powered sensor nodes, low mobility and large size without the aid of GPS aid. It works as an extension of both DV-hop propagation and GPS positioning in order to provide absolute position estimate for all nodes in a network where only a limited fraction of nodes have self location capability we call anchors. The procedure is partitioned into three phases: global location, local refinement and Kalman filter optimization, each phase is described in detail. The simulation environment used to evaluate this algorithm is also explained, including details about the specific implementation of each phase and experimental results, which demonstrate that the algorithm is more robust to measurement error than previous proposals, and it needs much fewer anchor nodes than previous algorithms to get comparable results.

To minimize the interruption during the handovers or relocations invoked by subscribers moving is a very critical factor to enhance the performance of the UMTS systems. We know that the 2G systems have been optimized to minimize the interruption of speech during handovers by two main technologies: one is the bi-casting for the DL traffic and the other is the fast radio resynchronization by the UE for the UL traffic. In the UMTS systems, we have also implemented lossless relocations for non real-time services with high reliability by data buffering in the source RNC and target RNC for the UE. However, the UMTS systems support four QoS classes traffic flow: conversational class, streaming class, interactive class and background class. The main distinguishing factor between these QoS classes is how delay sensitive the traffic is: Conversational and Streaming classes are mainly used to carry real-time traffic flows, like video telephony, interactive and background classes are mainly used by traditional Internet applications like WWW, E-mail and FTP. It’s essential to provide the solutions for supporting real-time services to meet the requirement for QoS in UMTS systems. Apparently, the Data buffering mechanism is not adapted to real-time services because of it’s delay may exceed the basic requirement for real-time services. Under this background, the paper discussed two data forwarding solutions for real-time services from the PS domain in the UMTS systems: packet duplication and Core Network bi-casting. The former mechanism does not require any new procedures, messages nor information elements. The later mechanism requires that the GGSN or SGSN is able to bi-cast the DL traffic to the target RNC according to the relocations involving two SGSNs or just involving one SGSN. It also implicitly shows that we need change procedures at the nodes SGSN, GGSN and RNC which are involved in the relocation procedure based on existing procedures that we have already designed if adopt the later solution. In a detail way, the paper analyzed the characteristic for these two solutions respectively, concentrated on the packet flows and the message flows in those nodes involved in relocations. Additionally, also gave out the impact on present transport technologies in the wireless communication systems. However we shall minimize the impact of evolution of transport mechanism and utilize the resource efficiently according to the general requirements for QoS in UMTS systems.

In this paper, the Time-Frequency Differential Phase Shift Keying (TFDPSK) modulation has been presented. Firstly, the Mathematical modeling is developed. Then the circuitry modeling is investigated. Finally, the simulation of the π/2-TFDPSK signal is described. The conclusion about TFDPSK is drawn through the analysis of the power spectral density.

In this paper, the performance of the LDPC coded MIMO system scheme is evaluated through numerical simulations, and according to the belief propagation decoding algorithm of LDPC codes, an iterative joint detection/decoding MAP algorithm based on the factor-graph representation is proposed. Simulation results show that LDPC codes can increase the effective diversity gain by utilizing both the space and the time diversity effects of the MIMO system.

The prime radio spectrum is a non-depleting but finite natural resource that is highly congested in a small set of bands -- such as the unlicensed and “PCS” bands -- but unbelievably underutilized over most of the other allocated frequencies. Enter Ultra-Wideband. Ultra-wideband may be the most over-hyped technology to surface within the past few years, but with 7.5 GHz of available and unlicensed spectrum and a little imagination, it poses a tremendous opportunity to drive a whole new generation of wireless devices. This paper, which is primarily informational in nature, highlights some of the fundamental issues surrounding the use of Ultra-wideband for wireless communications and also discusses its forecasted role in the evolution of future wireless systems.

The multiple access interference is the main reason that limits the UWB system capacity. The flexibility of pulse selection is a prominent property which is different from the conventional communications. In this paper, a new multiple-access scheme: Waveform Division Multiple Access (WDMA) based on the orthogonal wavelet function shapes is presented. In this scheme, the scaling function φ(t) and the wavelet function ψ(t) representing the UWB pulses are equally assigned to different users to realize multiple access. Because of the better orthogonality of φ(t) and ψ(t), the multiuser interference will be suppressed more effectively. The performance analysis and simulation results show that WDMA system can accommodate more users than TH-PPM UWB and DS-BPSK UWB system.

We give the definition of USB wireless laser communication system (WLCS) and the brief introduction to the protocol of USB, the standard of hardware is also given. The paper analyses the hardware and software of USB WLCS. Wireless laser communication part and USB interface circuit part are discussed in detail. We also give the periphery design of the chip AN2131Q, the control circuit to realize the transformation from parallel port to serial bus, and the circuit of laser sending and receiving of laser communication part, which are simply, cheap and workable. And then the four part of software are analyzed as followed. We have consummated the ISR in the firmware frame to develop the periphery device of USB. We have debugged and consummated the 'ezload,' and the GPD of the drivers. Windows application performs functions and schedules the corresponding API functions to let the interface practical and beautiful. The system can realize USB wireless laser communication between computers, which distance is farther than 50 meters, and top speed can be bigger than 8 Mbps. The system is of great practical sense to resolve the issues of high-speed communication among increasing districts without fiber trunk network.

In this paper, a hybrid DS/TH multiple access scheme is proposed for UWB impulse radio system on the basis of TH-PPM theoretical research. This scheme makes use of time hopping for multiple access and BPSK for data modulation. It utilizes the characteristics of PN sequence signal of DS to suppress further the multi-user interference of TH-PPM system. Results of performance analysis show that DS/TH scheme has higher system capacity than TH-PPM scheme.

The future proliferation of UWB-enabled devices raises legitimate concerns for the impact of its use on the performance of incumbent radio systems. In this paper, we present analytical models that quantify the blocking probability and the increase in forward traffic channel power allocation in the presence of noise-like UWB interference. The cdma2000 forward link has a fixed maximum transmit power but can control the fraction of total forward link power allocated to each traffic channel. If a terminal experiences a loss in sensitivity due to the presence of UWB energy, the base station can raise its power allocation to compensate for these conditions, up to a limit. If, however, the conditions at the handset require more power than that which is allowed by the system, then the traffic channel is blocked. In this study, we consider system performance as a function of the UWB power spectral density and its separation from the victim cdma2000 1X terminal. Although further study is pending, our analysis generally shows that protection of the cdma2000 forward link for indoor reception requires the UWB indoor emission limit to at least be below 20 dB below the currently specified emission limit when there is a single radiating UWB source.

Channel estimation is an important topic in wireless communication. In this paper, channel estimation schemes using pilot signals in the context of MIMO-OFDM systems are investigated. For the existing methods, in order to estimate the channel character between every transmitter and receiver antenna in MIMO-OFDM system, the different transmitter antennas usually send pilot signals alternately in different OFDM symbols. This means that only one antenna sends pilot signals every time, and other antennas must be vacant, just like the pilot module in IEEE802.16c. This means inefficient and inflexible, especially in mobile wireless channel. The proposed method can avoid the above defects. The proposed optimal training sequences can be inserted into every transmitter antenna pilot module synchronously without vacancy based on the MMSE. We can get more information about the mobile wireless channel. Through simulation, it is shown that the optimal pilot sequences derived in this paper outperform both the random and orthogonal pilot sequences that appear to be suboptimal. Furthermore, the proposed method can work even when some transmitters or receivers are disabled, but the existing methods can't.

A modified time-frequency transform algorithm is developed in this paper, the continuous pilot signal in time domain is transformed into time domain, through peak-detection processing, the possible peaks relative to multipath components are found and a proper window function is adopted to localize the peaks and minimize the noisy impact of non-peak area, the filtered pilot signal is then interpolated and transformed back to frequency domain. This simple but effective algorithm kicks out the additive noise and approaches the real channel response without additive noise more closely, improving the SNR of received signal after the deconvolution based on the channel estimation. The simulation results adopting this algorithm show the expected improvement of SNR of received signal.

In multipath channels, for orthogonal frequency division multiplexing (OFDM) system, the bit error rate (BER) of symbol-by-symbol differential detection (SSDD) is analyzed, the comparison of BER performance of SSDD and multi-symbol differential detection (MSDD) is obtained by simulation, the results indicate that when the whole bandwidth of the symbols used to MSDD is much less than the coherent bandwidth of the channel, the BER performance of MSDD is better than that of SSDD; when the whole bandwidth of the symbols used to the MSDD is closed to the coherent bandwidth of the channel, the BER performance of SSDD is better than that of MSDD.

The paper considers adaptive blind equalization problem of multichannel systems in digital communication. A feedforward neural network with lateral connections is introduced as the equalizer to estimate the source symbols from the received signals only. The lateral connections of the network are able to perform the self-orthogonalization, and thus the network can improved the performance. Simulation results have demonstrated promising performance of the proposed neural approach.

In this paper, we propose a fair scheduling algorithm based on proportional compensation mechanism for wireless networks. The proposed algorithm is capable of differentiate application-level services and use different queuing strategies for each class of service. We also introduce a compensation mechanism and a resource re-allocation method to achieve fairness.

Wireless networks are being rapidly installed in enterprise networks. There are several issues that are critical with the current specifications and the prominent ones are bandwidth utilization, cell capacity, propagation delay, power efficiency and quality of service aspects of the IEEE802.11 Medium Access Control (MAC) layer. IEEE 802.11 specifications for wireless LAN use Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) scheme. Although this scheme is widely successful due to its simplicity, it is inefficient in utilizing the physical bandwidth. Wireless media being highly limited in bandwidth and power, a Time Division Multiple Access (TDMA) may help to increase the utilization of the channel bandwidth. Many satellite networks have used Demand Assignment Multiple Access-TDMA (DAMA-TDMA), where the time slots are allocated dynamically. In this paper, a variation of this protocol has been proposed, called as Demand Based Bandwidth Assignment (DBBA) protocol. One of the main challenges that this protocol resolves is to avoid collision to the maximum extent during the demand request contention period. In order to analyze the DBBA, a test bench has been developed to simulate the traffic. Based on the simulation result, it has been observed that the proposed DBBA protocol has higher bandwidth utilization, supports larger number of stations in one cell, lesser propagation delay, more power efficient and has the ability to provide better quality of service. Since there is no collision, the efficiency of the system can be increased without affecting the overall throughput of the system.

With the popularity of wireless communication, much attention has been drawn to the reliable transmission for the fading and shadowing channels. In this paper, after a careful analysis of several existing schemes of hybrid ARQ with rate compatible punctured turbo (RCPT) codes, a novel hybrid ARQ scheme is proposed to fit for the unideal feedback channel in the worse wireless communication environment, which results in many advantages and improves performance of data transmission in practical wireless network.

Because of the various constraints imposed on ad hoc mobile wireless networks, a cost efficient routing protocol is crucial to the success of an ad hoc mobile wireless network, especially in a dynamic environment such as battlefields. In this paper, we present a cost efficient routing scheme for ad hoc mobile networks by taking advantages of the benefits of traffic aggregation. The central idea of our proposed routing scheme is motivated by the observation that traffic aggregation can reduce the effects of traffic variations. Based on similar studies in the fixed networks, we device a more realistic cost model in ad hoc mobile wireless networks, and propose a routing scheme based on this cost model that effectively promotes the traffic sharing in ad hoc mobile wireless networks. A much higher traffic variation factor than in fixed networks is used to account for the higher traffic fluctuations in an ad hoc mobile wireless network environment.

As the channel is accessed randomly in IEEE 802.11 WLAN systems, it’s difficult to support Quality of Services for real time applications. To solve the issue, there are already some approaches proposed in different network layers. However, when those QoS approaches are implemented in different network environments, the performance may vary significantly. In this paper, we will compare two major classes of QoS schemes for IEEE 802.11 WLAN in MAC layer and network layer respectively. The performances are compared in both one hop and multi-hop ad hoc network environments. It’s observed that MAC layer QoS scheme is very efficient in one hop network environments, while network layer QoS schemes is only efficient for multi-hop ad hoc networks. Based on the observations, it’s necessary to design a new hybrid scheme to efficiently support QoS in IEEE 802.11 WLAN system.

This paper investigates the QoS routing in TDMA/CDMA ad hoc networks. Since the network topology may constantly change and the available bandwidth is very limited in ad hoc networks, it's quite often to see a call is blocked when a path with required bandwidth cannot be found in the system. Therefore, we try to find multiple paths whose aggregated bandwidth can meet the bandwidth requirement and whose delays are within the required delay bound and then use the multiple paths in parallel for the QoS transmission of the call. This QoS routing we proposed can significantly reduce the system blocking probability and thus make a better use of network resources. We discuss the process of searching multiple parallel paths and proposed three heuristics (according to three parameters: property of maximum bandwidth, property of shortest path, property of maximum ratio of bandwidth to hops) to choose a group of paths whose total bandwidth satisfies the requirement. Some simulations have been conducted and the simulation results have demonstrated the deference of blocking rate gained by using the proposed three heuristics and also shown the proposed algorithms out-perform one existent on-demand algorithm.

Turbo codes are now universally known as one of the most effective techniques for achieving performance very close to the Shannon theoretical limits in many transmission systems. This paper presents a speed optimized ASIC turbo decoder core's design. The proposed architectures achieve a complexity reduction. Because of the recursion algorithm, the result of recursion is used immediately in following cycle. A reasonable pipeline is adopted by averaged the critical path to eliminate this effect. Core is fit to realize not only in FPGA, but also can embedded into other DSP and the decode rate can reach 6 Mbps in 0.18 um technology.

In this paper, we firstly present turbo product codes (TPCs) for forward error correction (FEC) coding, including TPCs encoding process and decoding principle, and then compare TPCs with turbo convolutional codes (TCCs) error coding solution. The performance of TPCs is shown to be closer to the Shannon limit than TCCs. Secondly, we introduce TPCs’ application in the 4th generation (4G) mobile communication system which is being developed in our country at present. The concept of TPC-OFDM system which promises higher code rate than conventional OFDM is first modified. Finally, simulation results show that the simplified 4G uplink systems offer Bit Error Rate of nearly 0 over IMT-2000 channel at E_b/N₀ > 15dB.

This paper presents some new methods to estimate the start of the frame and coarse frequency offset jointly based on standardized 10 short training symbols in 802.11a. Through simulations, it is shown that the metric functions of the frame synchronization can form sharp peak to avoid the ambiguity introduced by multipath effects at low SNR. So it performs more accurately in the frame synchronization. It also jointly realizes coarse frequency offet estimation with wide estimation range.

Symbol guard interval is almost indispensable to deal with ISI caused by channel delay spreading in wireless OFDM transmission systems. However, the length of guard interval is usually fixed in conventional OFDM system, which decreases the power and spectrum efficiency considerably, especially in a mobile environment. To solve this problem, a novel conceptual adaptive OFDM system with variable guard interval (AOFDM-VGI) is proposed in this paper. This system first utilizes the preamble and pilot sub-carriers of each OFDM packet burst to estimate the channel RMS delay spread, and then the result is fed into the guard interval controller. Then, the controller uses a criterion to calculate the guard interval and control the OFDM transmitter. For a typical simulation model as illustrated in this paper, this system can transmit a data rate 11.5 Mb/s higher and consume a power 0.65 dB less than the conventional non-adaptive counterpart without sacrificing the BER performance.

Most wireless system now relies on the use of Orthogonal Frequency Division Multiplexing (OFDM) because of its high data rate and its capability to combat the multi-path interference. In this paper, we investigate how the guard Interval (GI) length dealing with the delayed waves creating by the multi-path fading environment, affect the system performance. We will compare the performances of the system by measuring Bit Error Rate (BER) with QPSK as modulation scheme over multi-path fading and Add White Gaussian Noise (AWGN) in different delay environments. Furthers we will try to determine the optimum values of the GI parameter to achieve a good performance according to the existing delay. The simulations result shows that the GI depends strongly on the maximum delay spread.

This paper evaluates the performances of the contention-based channel access mechanism of IEEE 802.11e, called enhanced distributed coordination function (EDCF), compared with the 802.11 legacy MAC in supporting voice, video and data applications through network simulation of a scenario of 802.11e. Then we discuss the effects of Contention Window (CW) and Arbitration Inter-Frame Space (AIFS) on service differentiation and total throughput. We also consider an optional feature of the EDCF, called contention-free burst (CFB). Through our simulation study, we can draw a conclusion that the EDCF with TXOP can provide better-differentiated channel access for different traffic types than EDCF without TXOP especially at high traffic load conditions. But the movements caused by the parameters in CFB seem a lot bouncing and instability when in different application and configuration.

An optimum rate allocation scheme for code division multiple access (CDMA) uplink, which minimizes the average transmission delay of data users under the system resource constraint, is derived and evaluated in this paper. The average transmission delay minimization issue is formulated as a constrained optimization problem and solved using the Lagrange multiplier method. It is found that the proposed allocation rule achieves the minimum average transmission delay of data users compared to the other two conventional rate allocation schemes. Moreover, the proposed approach obtains much less transmission delay against the other two schemes especially when the traffic load is higher, which indicates that our scheme is more efficient in the high load systems. Numerical examples also show that the system parameters such as outage probability constraint and power control errors have great impact on the delay performance. Since our work is based on the imperfect power control model, it is more appropriate to the practical CDMA systems, especially for the delay-sensitive services.

With convergence of network technologies, future network infrastructure will be based on IP and traditional wireless/mobile networks will also evolve inevitably to a unified all-IP mobile internet which can support voice, data and video simultaneously. But, the pattern served by IP is best effort. So IP mobility and QoS management have been the main challenges of all-IP mobile networks. Based on Mobile IP, IP QoS protocol, etc, ... The paper brings forward an end-to-end QoS management solution for the next mobile communication network, introduces a terminology of QoS Agent Server (QAS), which acts as resource manage server and participate in dynamic Service Level Specification (SLS) negotiation with mobile hosts (MH) or other QAS. Taking advantage of the mechanisms, discussed the process of QoS manage to support mobile hosts' fast handoff between cells. Additionally, adopted and self-adaptive control algorithm for non-real time service on interior nodes. Finally, conclusions are given and possible future works are described.

A general summary of jamming methods in wireless communication is investigated in this paper. According to these jamming means, we give emphasis to analyze anti-jamming solution and their applications.

This paper proposes a new handover strategy named Minimal-Hops Handover (MHH) strategy for the LEO satellite-based constellation networks equipped with Inter-Satellites Links (ISL). MHH strategy, which is based on the hops of the end-to-end connection paths and makes good use of the regularity of the constellation network topology, can appropriately combine the handover procedure with routing and efficiently solve the inter-satellites handover issue. Moreover, MHH strategy can provide QoS guarantees to some extent. The system performances of the MHH strategy, such as time propagation delay and handover frequency, are evaluated and also are compared with that of other previous strategies. The simulation results show that MHH strategy performs better than them.

This research mainly focuses on the performance evaluation and the improvement of the IEEE 802.11 DCF. The basic access method in the IEEE 802.11 MAC protocol is the distributed coordination function (DCF), which is a carrier sense multiple access with collision avoidance (CSMA/CA) MAC protocol. In this paper, we first evaluate the performance of DCF using the classical Markov analytical model proposed by Bianchi, and then present an improvement named TDCF. Both the theoretical computation and the simulation in our research prove that TDCF has better performance than DCF.

Mobile IPv6 was designed to enable an IPv6 terminal to continue communications seamlessly while changing its access to network. Decreasing communication and management cost is a key issue of the research of the Internet mobility management. Hierarchical Mobile IPv6 was proposed to reduce the number of management messages in backbone network. However, the resources consumptions inside a hierarchical domain are increased as expense according to our cost models. Based on the idea of integrated optimization, adaptive mobility management scheme (AMMS) is proposed in this paper, which decreases the total cost of delivering management messages and data payload on the viewpoint of entire network resources by selecting a suitable mobility management scheme adaptively for a mobile node. The results of simulation show that AMMS has better performance than unmixed Mobile IPv6 and Hierarchical Mobile IPv6.

IEEE 802.11 WLAN is used widely because of the flexibility of Wireless LAN. Subscribers hope that the Wireless LAN can supply Real-time application as the wired network, for example, video and voice service, these require service differentiation in networks. Service differentiation at the IP layer is useless without support of that in lower layers. This support is even more critical in wireless networks because of its instability and limited bandwidth. This paper aims at the promotion of data stream transfer through research in TCP/UDP transfer over MAC layer supported by EDCF. Applications impose requirements on communication parameters, such as data rate and delay. In this paper we present ideas to provide service differentiation support for 802.11e and compare it with the legacy DCF of IEEE802.11 by simulation, and analyze the result in performance of the schemes with TCP and UDP flows.

TCP congestion control mechanism, being based on the presupposition that network congestion is the only reason for data loss, is suitable for wired environments rather than for wireless environments with characteristics of higher bit error rate, lower bandwidth, longer delay, frequent mobility and so on, where link corruption can also cause data loss. To avoid unnecessary decrease of packet sending rate resulting in lower throughput and longer delay, some improved TCP congestion mechanisms have been presented. However, one of the disadvantages of these mechanisms is that a great deal of unnecessary packets lost by corruption can be caused while heavy data loss rate by corruption lasts short time. As well known, energy consumption of mobile hosts is one of the most important factors for wireless communication. How to save energy of mobile host has become an important task for wireless communication. One of the methods is to improve reliability to decrease overheads resulting from re-transmitting lost data. Obviously, it is necessary to decrease data sending rate, but how does it do? This paper answers the question, and then presents an improved TCP congestion control mechanism, namely TCP_IR (TCP for Improving Reliability) whose implementations are based on two algorithms. Algorithm I-based TCP_IR is driven by data sending periodic, while algorithm II-based TCP_IR is driven by the event of data lost by corruption. They can be very easily implemented with fewer overheads and can effectively improve TCP reliability with light influence to throughput and delay. The paper’s contribution is to give valuable references to further study TCP in wireless networks.

In this paper, we propose a novel method to better evaluate the performance of TCP over broadband satellite networks. We decouple the most crucial parts of TCP that impact its performance in broadband satellite environments, namely congestion control and error control mechanisms. And then we re-design these two function blocks and make them become two individual parts. With these re-designed modules, we have investigated the interactions between various currently existing TCP congestion control and error control schemes, as well as their impact on TCP performance over a geostationary broadband satellite link with long propagation delay and high bit error rate. Simulation results have shown that some combinations of different congestion control and error control mechanisms can waste satellite link bandwidth with large numbers of retransmission packets and unnecessary retransmission packets. And the modified TCP NewReno implementation can avoid high amount of retransmissions and unnecessary retransmissions.

The IEEE 802.11 medium access control (MAC) protocol provides shared access to wireless channel. Its primary MAC technique is called distributed coordination function (DCF) that includes two packet transmission schemes, namely, basic access and RTS/CTS access mechanisms. In a "hybrid" network combining the two schemes, packets with payload longer than a given threshold (RTS Threshold) are transmitted according to the RTS/CTS mechanism. Based on delicate mathematical model, the average time in a successful and unsuccessful transmission is analyzed in the assumption of idea channel. Then the relation of network saturation throughput and RTS threshold was found and expressed in theoretical formula. We present the numerical techniques to find out the optimum RTS threshold that can maximize the network capacity. An adaptive RTS threshold adjust algorithm (ARTA), with which a station can automatically adjust its RTS threshold to the current optimum value, is also presented in detail. A special procedure is also developed to help ARTA in determination of station numbers. All theoretical analysis and algorithm are validated through computer simulation.

Ultraviolet light communication is a new means of optical information transmission technology developed since 1990’s. In this paper, some key technologies developed from the experiment will be described in detail. These technologies include the design of high speed UV source frequency modulator, the design of high sensitive optical receiver system and the design of weak signal detect and process modular, etc. The UV communication system has accomplished good voice and high speed digital data bidirectional communication in semi-duplex way and its maximal communication baud rate is up to 9600 bps which is twice as the baud rate of the UV communication system made by GTE Company for navy USA navy in 2000. Furthermore, in some distance the system can achieves a non-line-sight communication.

With the communication development people are expecting to get better communication services with higher data rates at anywhere and anytime. Wireless local-area networks (WLAN) provide such a good chance that they have gained widely accepted. The IEEE 802.11b provides data rate up to 11 Mbps and has gained widely use. But the throughput is not usually the best. After discussion we find in this paper that the throughput can be improved by properly fragments. We simulate the best fragment of the 802.11b at different SNRs in the AWGN channel and Rayleigh channel.

Multi-carrier technique is usually explored for high-speed communication. Traditionally, CP (cyclic prefix) is used to remove inter-symbol-interference (ISI). However, in this paper, a robust multi-carrier system with low data speed is proposed, with the purpose for anti-jamming. A pseudo-noise (PN) code is used to spread the message bit among sub-carriers. Instead of embedding the long time cyclic prefix (CP), a simple coding over time is used to reduce the error rate due to ISI. The result of simulation has shown that this system performs well.

This paper introduces a systolic array structured adaptive decision feedback equalizer (DFE) based on extended QR-RLS (QR-decomposition-based Recursive Least Squares) algorithm. In the wireless mobile communication systems, the fast time-varying channel could be experienced, so that high-speed convergence along with precise tracking ability is a crucial requirement for the equalizers. To cope with this, the QR-RLS algorithm has been considered as an adaptation apparatus satisfying these requisitions with showing numerical stability in fixed-point operation. In the aspect of realization, the QR-RLS algorithm could be realized in a form of systolic array, which gives an opportunity of utilizing the parallel pipelined processing. Even the computational complexity can be reduced by avoiding the square-root operations. However, the conventional QR-RLS approach demands the extra weight extraction process, so that the equalization can be hardly accomplished by a full pipelined process. Against this difficulty, this paper exploits the extended QR-RLS algorithm to build up the systolic array structured DFE in efficient. In addition to affirmative features such as the fast convergence and the superb tracking ability, the proposed DFE architecture based on the extended QR-RLS yields a capability of full pipeline process from the input acquisition till the weight update without involving undesirable square-root operations.

Free space optical communication system is competent for its large bandwidth, the agility and short fixing time. The bandwidth is affected by many factors, and the optical system is one. This paper firstly introduces the FSO system and the constitution of its optical system, and then analyzes the impact of the lens dispersion on system communicating bandwidth. On the basis of the bandwidth limitation discussion, this paper puts forward a means to use a couple of cemented-doublets group as the transmitting lens, and use the dense whorl lens as the receiving lens. In the end, the use of such design in our own system proved the feasibly of it.

Wireless mouse is not restricted by cable’s length and has advantage over its wired counterpart. However, all the mice available in the market have detection range less than 2 meters and angular coverage less than 180 degrees. Furthermore, commercial infrared mice are based on track ball and rollers to detect movements. This restricts them to be used in those occasions where users want to have dynamic movement, such as presentations and meetings etc. This paper presents our newly developed infrared wireless mouse, which has a detection range of 6 meters and angular coverage of 180 degrees. This new mouse uses buttons instead of traditional track ball and is developed to be a hand-held device like remote controller. It enables users to control cursor with a distance closed to computer and the mouse to be free from computer operation.

A mobile ad hoc network (MANET) facilitates mobile hosts to communicate with one another based on wireless infrastructure. There are no base stations in ad hoc networks and the mobile hosts act as routers for data packets. Since the mobile hosts in a MANET keep moving, topological changes take place frequently in the network. This cause routing information kept by the nodes to get outdated easily. Effective ways of getting updated routing information is needed. This problem becomes more complex in wireless networks, where bandwidth is limited. This paper proposes an improved dynamic source routing (DSR) algorithm as a solution to this problem. The proposed algorithm, the Improved Dynamic Source Routing (I-DSR), has a hybrid feature, combined proactive and reactive, that enables all the hosts to get updated changing topology information with a low cost in communication instead of enabling only a few hosts to get the information when there is a route request for a new moved destination. In anticipation of the need of a route request to the new moved destination by other hosts, the operations in the algorithm are designed to eliminate these needs for route request flooding and eventually generate a small number of control packets. Through simulations, we demonstrate that I-DSR gives better network performance than the well-known on-demanding routing protocol, Dynamic Source Routing (DSR) Protocol. As it is improved from DSR, it just needs a few modification on DSR and is easy to implement.

Ad hoc networks do not depend on any predefined infrastructure or centralized administration to operate. Their security characters require more complex security preventions. As the second line of defense, Intrusion detection is the necessary means of getting the high survivability. In this paper the security characters of ad hoc networks and the related contents of intrusion detection are discussed. Mobile Agent and Multi-layer Integrated Distributed Intrusion Detection Model (MAMIDIDM) and a heuristic global detection algorithm are proposed tentatively by combining the mobile agent technology with the multi-layer conception. This heuristic global detection algorithm combines the mobile agent detection engine with the multi-layer detection engines and analyzes the results obtained by the corresponding detection engines. MAMIDIDM has the better flexibility and extensibility, can execute the intrusion detection in clustering ad hoc networks effectively.

This paper proposes a QoS routing with delay constrained and bandwidth guaranteed in TDMA/CDMA ad hoc networks. Because the wireless bandwidth is shared among adjacent nodes and the network topology changes as the nodes move, quality of service is more difficult to guarantee in ad hoc networks than in most other type of networks. QoS routing requires finding not only a route from a source to a destination, but also a route that satisfies the end-to-end QoS requirement, often given in terms of bandwidth or delay. We improve a distributed algorithm to realize the control of delays and give the method to calculate the bandwidth of a path that can be used to judge whether the bandwidth of the path satisfies the requirement. In TDMA/CDMA ad hoc networks, this is a new scheme to satisfy not only delay constrained but also bandwidth guaranteed. Some simulations have been conducted and the simulation results have demonstrated the deference of call success rate between our algorithm and other existing global algorithm or other localized algorithms that only consider delay constrained or bandwidth guaranteed.

An ad hoc network is a collection of wireless mobile nodes dynamically forming a temporary network without using any existing network infrastructure or centralized administration. Because of its acentric, self-organized, fast deployable and mobile property, people are paying more attention to the using of ad hoc network in the emergency. In such an environment, multiple networks "hops" may be needed for one node to exchange data with another across the network, due to the limited range of each mobile node’s wireless transmissions. So for an ad hoc network, a rational routing protocol is especially important. The paper will analyze the character of ad hoc network and the special requirement for communication protocols. Toward the routing protocols, this paper presents four multi-hop wireless ad hoc network routing protocols that cover a range of design choices: DSDV (Destination Sequenced Distance Vector), TORA (Temporally Ordered Routing Algorithm), DSR (Dynamic Source Routing), and AODV (Ad hoc On-demand Distance Vector). After thoroughly analyzed the network structure and routing protocols, the paper gives the proposal of a new hybrid routing protocol and the view for future work.

Capacity analysis and call admission control in wireless communication systems are essential for system design and operation. The capacity for imperfectly power controlled multimedia code division multiple access (CDMA) networks based on system outage probability constraint is presented and analyzed. A handoff prioritized call admission scheme is then developed based on the derived system capacity and evaluated using the K-dimensional birth-death process model. A more general assumption that average channel holding times for new calls and handoff calls are not equal and an effective approximate model are adopted in the performance analysis. Numerical examples are given to demonstrate the system performance in terms of blocking probabilities, resource utilization and average system throughput. It is shown that the system parameters such as outage probability constraint and power control errors have great impact on system capacity and performance.

It is important to estimate the channel quality of forward link in cdma2000 1x EV-DV for much better systems performance. Two channel quality indicator algorithms are proposed in this paper. The CQI prediction based on the nonlinear numerical approximation is to get the parameters of the nonlinear numerical approximation model according to several past values of CQI information. And then we can employ this model to predict the coming CQI value. Another CQI prediction algorithm is based on the N-LMS algorithm according to the past CQI values. The simulations of the novel algorithms are performed to compare with the analysis. It is shown that the two algorithms have better performance than the general CQI processing method.

In recent years, multimedia services are developing rapidly in mobile communication systems. The multimedia services consist of many kinds of data which have different characteristics. To enable these services to be provided, because of large amount of data, there needs more effective utilization of radio resources in limited bandwidth. One of the solutions is making an effective DLC (Data Link Control) for QoS (Quality of Service) assurance. In this paper, we propose a structure of radio protocol and the DLC protocol layer which consists of hierarchical component based functional entities and an effective radio link control. The DLC is operated by the state of current traffic volume. For the efficiency of radio resource utilization, different adequate components will be taken by the current traffic state. Especially fraction based data transmission buffer control method with ARQ scheme, which is dynamic parameter change scheme, based on logical channel priority and DLC state for the QoS assurance in Wireless LAN system.

Based on both non-uniform coverage zone and non-uniform traffic density, a novel dynamic cell configuration system with a new channel assignment scheme has been proposed. With the multi-beam base station, radiating beam is split in the vertical plane to configure a multi-annular zone cellular system. Under the condition non-uniform coverage and non-uniform users' position distribution, we analyze the performance carrier-to-interference ratio of this cellular system on the downlink. With the adaptive multi-beam base station antennas, each beam adjusts its coverage area according to users' position distribution to maintain the maximum average area carrier-to-interference and area spectrum efficiency. Analysis and simulation show the proposed dynamic cellular configuration is efficient.

The MC-DS-CDMA system, OFDM technology is applied in, has some good properties, such as high-speed transmission and restraining multi-path interference. However it has some shortcomings. Lots of resource of frequency is used in this system. Because traditional spreading spectral sequence doesn’t have good property of auto-correlation and cross-correlation in the meantime, it leads to multi-access interference. In this paper, a new communication system of MC-DS-CDMA system using modified complementary sequences as spreading spectral sequence will be proposed to solve those problems. Based on the theory of complementary sequences and OND, the concept of dummy chip is led in. According to the algorithm of dummy chips generation, the whole frame data will be processed and becomes the extended frame which is also called modified complementary sequence. The dummy chips in the extended frame become an undisturbed window. When received data is carried on the particular correlating computation of complementary sequence in the undisturbed window, the primitive transmitted data will be restored correctly. The performance of MC-DS-CDMA system based on modified complementary sequence has been analyzed and simulated. The result of simulation proved that this measure is proper and feasible. With the quotation of this technology, the system can efficiently eliminate multi-path interference and decrease multi-access interference to a low level under the condition of using less subcarriers relatively.

A wideband high frequency (HF) channel simulator is implemented in a software simulation of a direct-sequence spread-spectrum (DSSS) HF communication system. A RAKE receiver is used to mitigate the effects of multipath delay. The validity of the HF channel simulator has been demonstrated by the analysis of in-out signals of the channel simulator. It shows that the simulator accurately models a HF channel. New performance results of the system with convolutional codes and bit interleaving are advanced to demonstrate the potential of the system with proposed scheme.

In current research on mobile ad-hoc networks, protocol evaluation is always needed. However, building real test-beds are costly, and sometimes infeasible, especially for large-scale mobile ad-hoc networks. Therefore, simulation systems are provided. Unfortunately, exiting simulation systems have obvious defects. They can hardly support scalability or real-time simultaneously, neither changeable network nodes. To remedy these problems, we propose a test system named ManTS in this paper. With distributed architecture and some novel mechanisms, such as Direct Transmission and Virtual Application Traffic, etc., ManTS is able to build testbeds for large-scale mobile ad-hoc networks and guarantee real-time at the same time, and it also supports dynamic enrolment of network nodes. Furthermore, IP or upper layer protocols and applications implemented on Linux can run on ManTS without any modification.

Mobile communication is now shaping our daily lives. However, as the end-user devices become increasingly smaller, the limited screen size and keyboard space start to bring about lots of inconvenience. Therefore, these conventional input/output methods are challenged by more natural interfaces, such as voice interface. Speech synthesis, which is one important aspect of voice interface, has so far been successfully applied to mobile phones to read email, instant messages for us. Nevertheless, these text-to-speech (TTS) systems, which are primarily based on concatenation of acoustic units, usually suffer from audible discontinuities between two successive acoustic units. This paper presents a novel method that can effectively remove the audible discontinuities in the Chinese speech synthesis and highly maintain the speech naturalness.

In this paper, the matching between some key performance parameters such as field of view, optical aperture, control bandwidth and engineering issues including weight, power, volume, lift time in APT sybsystem of optical communication link are analyzed and discussed in detail according to the requirement of BER, communication distance, platform jitter, background light intensity and other system parameters, the two communication terminal carrier are the GEO satellite and airplane. The goal of the system is to fulfill optical communication at data rates of 622Mb/s from aircraft at the altitudes of 10000 ~ 25000 m to another terminal mounted on satellite on the orbit of GEO, they are 40000 km away from each other. Communication error rates is better than 10^-6; the airborne terminal must be able to operate within temperature ranges of -65°C to +55°C and be able to survive in humidity.

The paper gives a scheme of the algorithm and the architecture of the information processing system of IR beacon image in the APT system. The CMOS image sensor gets the IR beacon's image in the optical antenna. The FPGA implements the IR image preprocessing and the DSP offers the possibilities of processing large high-resolution images. The wavelet transform is used in the image preprocessing, which can decompose the image in multi-scale to select the different frequency components. The IR beacon's position is detected based on gravity center, and the precision servo system finishes the tracking of the IR beacons. Thus the system can implement the acquisition, pointing, tracking and establish the FSO communication link in real time.

A method of improved motion vector based on quantization errors in transcoding for wireless mobile Internet is proposed in this paper. The computational complexity of this scheme can be reduced. The control strategy of transcoder buffer is also discussed, and the number of coding bits on the picture layer in the whole is allocated, then different types of picture respectively are processed on the basis of frondose picture characteristics when each frame is coded. Finally optimal quantization parameter for every macroblock in a frame is select. The simulation results shows that this method can effectively speed up the transcoding without significant lose in quality. The output bit rate of the transcoder and the buffer fullness tend to be stabilized, and the peak signal-to-noise-ratio (PSNR) of the reconstructed picture is raised.

Recently the requirement for wireless ad hoc networks to support multimedia services like videoconference and VOD has become more and more urgent, and load balancing is recognized as an effective mechanism to satisfy the real-time requirements of these services. But many current ad hoc routing protocols are single-path and best-effort routing, which aren't fit for load balancing. In this paper a node-disjoint multi-path routing protocol, namely Enhanced Multi-path Routing (EMR), is first presented to fulfill the short end-to-end delay and low package loss rate requirements of real-time traffic. Then the policy of how to balance load over multiple paths is discussed based on EMR in the paper. According to some key factors that can affect the performance of load balancing, a load-balancing weight (LBW) equation is formed to describe our load-balancing algorithm quantitatively. The simulation results indicate that the proposed load-balancing mechanism decreases the end-to-end delay and the packet loss rate of real-time traffic.

Transmission scheduling algorithm for WWW traffic in GPRS needs to be optimized according to characteristics of Temporary Block Flow, TCP/IP and mobile access behavior. We analyze the radio resource allocation problem in GPRS and design a new scheduling algorithm FIFO-with-flow. It is simple and can adapt to temporal flows, and allocate bandwidth among temporal flows fairly. Based on the characteristics of mobile user access behavior, we determine the wireless WWW traffic model and its parameters reflecting TCP congest control. We simulate and compare the performance of FIFO-with-flow and FIFO. The FIFO-with-flow can support more WWW users than FIFO.

General Packet Radio Service (GPRS) is an upgrade development of GSM and an indispensable component of 3G that provides both circuit and packet switched data communication ubiquitously. Wireless LAN (WLAN) is a network technology that lacks the flexible mobility management nevertheless is relatively cheap and gives much higher bandwidth than 3G. Combining these two techniques allows users to get a large coverage area with 3G and relatively high-speed data flow in WLAN spots. The essence is to identify and validate different solutions for interworking. These solutions will differ in the level of integration. A typical solution of tight integration is to physically connect the WLAN segment with the 3G networks, by making a new node that behaves either like a SGSN or a BSC in 3G systems. Another more loose solution is based on the usage of the mobile IP as a bridging technology. It is the recommended solution for it requires less of an effort to implement. This paper investigates those different solutions and proposal a new approach to improve the performance of the interworking.

Recently there has been an increasing demand for providing multicasting service in mobile IP environment. However, it is still faced with great challenges, such as the established multicast delivery paths may frequently restructure when a node moves from one sub-network to another, incurring expensive overheads. This paper proposed a delay-constrained least-cost multicast routing algorithm to support multicast routing infrastructure for heterogeneous wire/wireless networks. The algorithm has three goals: (a) reduces the overall cost of the multicast tree, (b) reduces the join/leave latency when miobile node hand-off between wireless cells, (c) by using adaptive strategy, overcomes the shortcomings of traditional algorithms which deal with the cost and delay separately and needs lots of retrospect.

Novel design of single-feed circular microstrip antenna for polarization diversity is proposed. It is demonstrated that by incorporating two orthogonal slots of equal lengths in the circular patch and utilizing two PIN diodes to switch the slots on or off, right hand circular polarization (RHCP) or left hand circular polarization (LHCP) can be achieved using the same feeding probe. The antenna design and results of the CP microstrip antenna obtained by applying this design are presented and discussed.

In this paper, two wireless broadband access and accounting schemes were introduced. There are some differences in the client and the access router module between them. In one scheme, Secure Shell (SSH) protocol is used in the access system. The SSH server makes the authentication based on private key cryptography. The advantage of this scheme is the security of the user's information, and we have sophisticated access control. In the other scheme, Secure Sockets Layer (SSL) protocol is used the access system. It uses the technology of public privacy key. Nowadays, web browser generally combines HTTP and SSL protocol and we use the SSL protocol to implement the encryption of the data between the clients and the access route. The schemes are same in the radius sever part. Remote Authentication Dial in User Service (RADIUS), as a security protocol in the form of Client/Sever, is becoming an authentication/accounting protocol for standard access to the Internet. It will be explained in a flow chart. In our scheme, the access router serves as the client to the radius server.

A 3-D near-field source localization algorithm based on higher-order statistics using a small antenna array is presented in this paper. To suppress Gaussian color noises of unknown autocorrelation matrix in received array data, a specific fourth-order cumulant matrix instead of autocorrelation matrix is formed and an ESPRIT-like shift-invariance technique is used to estimate the phase differences among the antennas and ranges from the sources to the antennas simultaneously. Then a closed-form source location estimate is given by the solution of a set of linear equations. The proposed algorithm imposes no geometric constraint on the antenna array. Simulation results are provided to demonstrate the effectiveness and feasibility of this method.

During the distributed design of UMTS-SGSN system, the Dynamic Central Tasks Scheduling algorithm (DCTS) with good real-time property is researched to meet the hard requirements of bandwidth and latency of UMTS-SGSN system. Based on the assumption without task migration, Markov queuing network model is used to model UMTS-SGSN system instead of M/M/c queuing model. For single user with one PDP context, DCTS is optimal through the theoretic calculation and simulation. And for single user with not less than one PDP context, the performance of DCTS is discussed through the simulation. In the case of longer service time in data service, DCTS is compared with stochastic scheduling algorithm and cycle scheduling algorithm. Simulation results show that DCTS is better than other two algorithms, and when service time is shorter, three algorithms' performance is approximately same. The performances of the three algorithms in different service time are also analyzed by adjusting the parameter μ of the exponential distribution. Results show that exponential distribution's heavy-tailed effect of the service time leads to deteriorating the efficiency of load balancing and DCTS shall initiate less task migration and decrease system overhead.