While mobile networks have been enhanced to support a variety of mobile multimedia services such as video telephony and rich data content delivery, a new challenge is being created by the remarkable development of micro-device technologies such as micro processor-chips, sensors, and RF tags. These developments suggest the rapid emergence of the ubiquitous computing environment; computers supporting human life without imposing any stress on the users. The combination of broadband global networks and ubiquitous computing environment will lead to an entirely new class of services, which we call ubiquitous networking services. This paper discusses how to create ubiquitous service environments comparing global networking approaches which are based on fixed and mobile networks. It is shown that the mobile approach is better from service applicability and reliability viewpoints. Networking architecture is proposed which expand 4G mobile cellular networks to real space via gateways on the edges of the mobile network (i.e. mobile terminals). A new set of technical requirements will emerge via this approach, which may accelerate the paradigm shift from the current mobile network architecture and even from the Internet of today.

Future wireless/mobile system is expected to have heterogeneous wireless overlay networks for ubiquitous multimedia communication. In a such network environment, mobile users are likely to try to get attached to higher bandwidth network as bandwidth-hungry multimedia applications are increasing. However, the users have to perform vertical handoff to lower bandwidth network, as high bandwidth network become unavailable due to various reasons (such as its limited coverage, network congestion, etc.). In this paper, we discuss the problem of vertical handoff from a user's perspective. For this purpose, we formulate user satisfaction as a function of bandwidth utility and handoff latency. Then, we investigate the effect of call holding time, user movement probability, etc. on the satisfaction that a user derives from the use of network service for multimedia applications. In addition, based on the evaluation, we present an algorithm for selecting a wireless network, which maximizes the effective user satisfaction.

One of the main issues of Mobile IPv6 is handover latency that causes service disruption time. Although plenty of proposals significantly reduce the service disruption time, they suffer from redundant routing that causes packet misordering and bandwidth consumption during the process of inter-domain handover. In this paper, we propose a new scheme that minimizes the redundant routing during the process of inter-domain handover by utilizing forwarding routers for each correspondent node. Our proposed scheme consists of forwarding router discovery and proactive handover. We evaluate our proposed scheme in the view of packet misordering and bandwidth consumption, and clarify the efficiency of our proposed scheme. We also evaluate the impact of the forwarding routers' capacity since routers have limited resources. By strategically locating forwarding routers, e.g. next to the router that has peering to another domain, the redundant routing caused by inter-domain handover will be efficiently suppressed.

Mobile multimedia services such as TV-call or video streaming are gradually becoming popular in the 3rd generation mobile network (IMT-2000). Multimedia traffic is expected to continue increasing into the coming years, and IP technology is considered to be capable of efficiently transporting such huge volumes of multimedia traffic. Aimed at ALL-IP mobile networks, we are proposing the IP-based IMT Network Platform (IP²), which represents an evolution from IMT-2000. In this paper, we propose the Edge Intelligent Mobility Management Protocol (EIM) to enable IP mobility in future ALL-IP mobile networks to meet mobile operators' requirements. EIM employs the so-called network-controlled mobility management approach and conceals IP mobility from mobile nodes by processing user packet and mobility control at the edge node, access router. To verify feasibility and scalability, we have implemented EIM on our experimental system and conducted a series of tests. The results showed that processing delays for basic mobility operations are considerably small. It was also confirmed that the EIM is scalable regarding the increase of the number of mobile nodes and correspondent nodes.

In this paper, we propose an architecture for the MPLS (Multi-Protocol Label Switching)-based micro-mobility management including label switched path setup, packet forwarding, handoff processing, and paging. In order to prevent packet loss during handoff, we propose two packet recovery mechanisms, namely: buffer time-based packet recovery and medium access control (MAC) layer assisted packet recovery schemes. Simulation results show that the MAC layer assisted packet recovery scheme has a better performance than buffer time-based scheme. Our proposed scheme provides a higher throughput when compared with other IP micro-mobility protocols including Cellular IP, HAWAII, and Hierarchical Mobile IP.

In recent years, the Internet has come to be able to be used at higher speed and more cheaply everywhere as the broadband service and the wireless LAN service have been provided by various ISPs. In such a ubiquitous environment, a demand for mobile computing environments in the intranet and easy access from remote sites has been increasing greatly as well as the Internet. To meet this requirement, we have proposed a Logical Office service, which is a kind of the mobile VLAN, and enables terminal plug and play, layer two tunneling, and ubiquitous communication. This paper newly proposes that the offer of the IP mobility to not only the movement of the terminal but also existing network services be possible the Logical Office service which we developed. As a result, it was verified to operate existing network services such as wirelesses LAN on the Logical Office service, and for newly existing network services to able to have the IP mobility.

In future, mobile terminals may be linked in various types of local network where the whole network is moving. Mobile networks will need to provide global connectivity to such moving networks and manage their mobility. A moving network consists of mobile terminals and a mobile router, which acts as the gateway to the mobile network. To manage the mobility of the moving network, it is important to minimize the packet overhead, to optimize routing, and to reduce the volume of handoff signals over the mobile network and air interface. This paper proposes a new routing mechanism using hierarchical mobile network prefix assignment, home agent concatenation, hierarchical address management, and hierarchical re-routing. In hierarchical mobile network prefix assignment, a mobile router is assigned a mobile network prefix, which is used as a prefix when allocating the location addresses of mobile terminals in the moving network, so allowing them to be managed in a hierarchical manner. Home agent concatenation limits the number of home agents which need to be updated during handoff by enabling one home agent hold information relating to others, while hierarchical address management minimizes the volume of handoff signals by managing the location addresses of all mobile terminals in a hierarchical manner. Hierarchical re-routing introduces a local anchor router in order to localize handoff and to optimize routing. Simulation results show that our proposed routing method is better than the conventional solutions in terms of efficiency of data transmission including data transmission delay, and handoff performance.

This paper compared the approaches concerning the pinball routing problem that occurs in the nested network in network mobility environment and developed the analytic framework to model. Each model was evaluated of transmission latency, memory usage, and BU's occurrence number at routing optimization process. The estimation result showed that the optimization mechanism achievement overhead existed in each model, and the full optimization of the specific model was not attained because of it. Therefore, the most appropriate approach for routing optimization in nested NEMO can be determined only after a careful evaluation, and the proposals must consider using it in combination with other approaches. The modeling framework presented in this paper is intended to quantity the relative merits and demerits of the various approaches.

This paper describes two way satellite system environments on geostationary orbit (GEO) and performance enhancement mechanisms which reduces round trip time (RTT) and supports real-time services. We use performance enhancing proxy (PEP) for reducing round trip time and user-level real-time scheduler for reducing deadline violation tasks. The user-level real-time scheduling method classifies priority of user process into four types and those are reflected in kernel. With these dual performance enhancement mechanisms, we can improve quality of service (QoS) of end-user who connects to the DVB-RCS system.

This paper proposes "Multipath Control and Proactive Control" to realize a robust QoS control system for mobile multimedia communication in an IP-based cellular network. In this network, all kinds of traffic will share the same backbone network. This requires a QoS system that differentiates services according to the required quality. Though DiffServ is thought to be a promising technique for achieving QoS, an effective path control scheme and a technique that is suitable enough for rapid traffic changes are not yet available. Our solution is multipath control using linear optimization combined with proactive control using traffic anomaly detection. Simulation results show that multipath control and proactive control improve system performance in terms of throughput and packet loss when rapid traffic change takes place.

The technological development of wireless environment has made real-time multimedia communications possible in wireless networks. Many studies have been done on real-time communications in wireless networks in order to overcome a higher bit error rate in wireless channels. However, none of work deals with firm real-time communications which can be applied to multimedia communications. In this paper, we propose an adaptive error correcting scheme for firm real-time multimedia communications in wireless networks in order to maximize the expected net profit. The proposed scheme adaptively selects an error correcting code under the current air state and the message state of a message stream. Throughout simulation results, we show that the suggested scheme provides more profit than single error-correcting code schemes.

Radio resource is the bottleneck for current multimedia wireless networks. Intelligent traffic control strategies can be enforced to optimize resource allocation so as to enhance network performance. In this study, dynamic control scheme for non-real-time traffic and autonomic control schemes for multimedia traffic are proposed to guarantee the required quality of service (QoS) in the inference-dominated high-speed wireless environment. Both handoff priority and terminal mobility are also taken into consideration. The performance of the state-dependent multidimensional birth-death process is derived by the efficient matrix-analytic methods (MAMs). Compared with the previous results, this paper shows that the proposed control methods can be used for both real-time and non-real-time multimedia traffic in order to meet the required performance without degrading the quality of multimedia services. These results are also important for the design of evolving multimedia wireless systems as well as network optimization.

In the adaptive modulation and coding (AMC)-based orthogonal frequency division multiple access (OFDMA) system for broadband wireless service, a large number of users with short packets cause a serious capacity mismatch problem, which incurs resource under-utilization when the data rate of subchannel increases with a better channel condition. To handle the capacity mismatch problem, we propose an AMC-based subchannel multiplexing (ASM) scheme, which allows for sharing the same subchannel among the different simultaneous flows of the same user. Along with the ASM scheme, we also consider multi-class scheduling scheme, which employs the different packet scheduling algorithm for the different service class, e.g., packet loss rate-based (PLR) scheduling algorithm for real-time (RT) service and modified minimum bit rate-based (MMBR) scheduling algorithm for non-real-time (NRT) service. In the typical integrated service scenario with voice, video, and data traffic, we have shown that the proposed schemes significantly improve the overall system capacity.

In multicast congestion control, the receiver of the worst congestion level is selected as the representative and transmission rate of the sender is adjusted to TCP throughput of the representative. This approach has high scalability and TCP friendliness. However, when this approach is applied in wireless communications, wireless-caused packet loss will cause to frequent change of the representative. This is because degradation of wireless channel quality causes bursty packet loss at a corresponding receiver. Fading, one of main reasons of wireless channel degradation, is expected to be recovered after short time period, which leads to frequent change of the representative. This frequent change of representative makes the sender adjust its transmission rate to the tentative worst receiver, which brings severe performance degradation to wireless multicast. We call this technical problem, the wireless-caused representative selection fluctuation problem. Wireless-caused representative selection fluctuation problem is one of scalability problems in the wireless multicast, because this problem remarkably happens for large scale multicast. We propose two possible solutions for this problem, an end-to-end approach and a network support approach. Performance evaluation in various situation show that an end-to-end approach is sensitive for its inferring error but a network support approach shows good performance improvement.

IP multicasting has naturally been considered the ideal technique to be used with multimedia communications but is difficult to mobile hosts due to the addition of mobility in the host group. In this paper, we propose a novel multicast algorithm, namely time and distance-based multicast algorithm (TDBMA), for IPv6-based mobile networks. TDBMA intends to trade off between the shortest delivery path and the frequency of the multicast tree reconfiguration. When a mobile host (MH) moves into a new foreign network which is managed by a foreign network router and re-subscribes to an existing multicast tree, the foreign network router uses the TDBMA to determine whether the MH joins the multicast tree directly or receives the tunneled multicast packets from its home agent (HA). The TDBMA is subject to two key factors, called time and distance. The visited foreign network router can join the multicast tree directly only when the value of both time and distance is true. The simulation results have shown that TDBMA has better performance than other algorithms, especially in the case where the number of MHs is large.

To offer location based services, service providers need to have access to Location Information (LI) regarding the users which they wish to serve; this is a potential privacy threat. We propose the use of constraints, i.e. statements limiting the use and distribution of LI, that are securely bound to the LI, as a means to reduce this threat. Constraints may themselves reveal information to any potential LI user--that is, the constraints themselves may also be a privacy threat. To address this problem we introduce the notion of a LI Preference Authority (LIPA). A LIPA is a trusted party which can examine LI constraints and make decisions about LI distribution without revealing the constraints to the entity requesting the LI. This is achieved by encrypting both the LI and the constraints with a LIPA encryption key, ensuring that the LI is only revealed at the discretion of the LIPA.

Today, pedestrian navigation systems for mobile phones use 2-dimensional maps as a navigation media in general. But 3-dimensional maps or scenery images are easier to understand for users than 2-dimensional maps. To use 3-dimensional maps or scenery guidance, the measuring accuracy of user position is essential to understand guidance images. In this paper, we will present a pedestrian navigation system using real scenery photographs as a navigation media, and based on experiments we will report the evaluation result of influence of positioning accuracy on the understandability of navigation. It is shown that 3 meters or less error of positioning is tolerable for pedestrian navigation systems using scenery images.

The latest video coding standard, H.264/AVC, adopts 44 approximate transform instead of 88 discrete cosine transform (DCT) to avoid the inverse transform mismatch problem. However, that is only one of the factors that make it difficult to transcode pre-coded video contents with the previous standards to H.264/AVC in the common domain without causing cascaded pixel-domain transcoding. In this paper, to support the existent DCT-domain transcoding schemes and to reduce computational complexity, we propose an efficient algorithm that converts the quantized 88 DCT block into four newly quantized 44 transformed blocks. The experimental results show that the proposed scheme reduces computational complexity by 5-11% and improves video quality by 0.1-0.5 dB compared with the cascaded pixel-domain transcoding scheme that exploits inverse quantization (IQ), inverse DCT (IDCT), DCT, and re-quantization (re-Q).

This paper describes how to integrate interaction in wallpaper using vector graphics solution VIS (Vector Image Solution). By using dedicated key action for toggling input mode and VIS functionality such as action script support, pause-resume interface, etc., we can achieve direct interactivity for rich media services without deteriorating stability of the integrated program of mobile handset.

In this letter, we propose a complexity-scalable DCT-based video encoder which works for the highly resource-limited terminals, such as cellular phone, PDA, handheld terminals, etc. The basic concept in the proposed method is for DCT operations to be adaptively changing the complexity like buffer control algorithm in the CBR (Constant Bit-Rate) video encoder.

Resilient Packet Ring (RPR) is a new technology currently being standardized in the IEEE 802.17 working group. The existed bandwidth allocation algorithms for RPR networks are not able to provide satisfactory solutions to meet the performance requirements. In this paper we propose one fair bandwidth allocation algorithm, termed PID-RPR, which satisfies the performance goals of RPR networks, such as fairness, high utilization and maximal spatial reuse. The algorithm is operated at each RPR node in a distributive way; the proportional, integral and differential (PID) controller is used to allocate bandwidth on the outgoing link of the node for the flows over the link in a weighted manner. To achieve the global coordination, one control packet containing every node's message runs around the ring in order to update the relevant message for all nodes on the ring. When the packet reaches one node, this node adjusts its own rate according to its own message in the control packet; in the meantime it updates other nodes' control message in the control packet. As the control packet propagates around the ring, each node can eventually adjust its sending rate to reach its fair share according to the fairness criterion, and the buffer occupancy at each node is kept within the target value. Our algorithm is of distributed nature in the sense that upstream ring nodes inject traffic at a rate according to congestion and fairness criteria downstream. The simulation results demonstrate that satisfactory performance of RPR networks can be achieved under the proposed bandwidth allocation scheme.

In an empowered Internet with end-to-end Quality of Service (QoS) facilities, it is essential for content servers to minimize reserved network resources to achieve a reduction in transmission cost for the use of QoS. Resource reservation usage charging forces customers to efficiently use network resources. In this paper, we analyze a model that optimizes the total cost when semi-elastic traffic flows are delivered in a client-server scenario. The client uses the proposed analytical model to easily control its buffer occupancy and to determine at any time if it is needed to utilize resource reservation or best-effort transmission mode.

With the occurrence of new applications such as Voice over IP (VoIP) and multimedia conference, there is an ongoing discussion about realizing QoS in the Internet today. Because of its potential scalability in support of QoS guarantees, the Differentiated Service (DiffServ) architecture with aggregate packet scheduling has recently attracted much attention in the networking community as a feasible solution for providing Internet QoS. Thus, it is important to understand delay bound of an individual flow in the DiffServ architecture in order to provide delay-sensitive applications. In this paper, we study, via both analysis and simulation, the deterministic bound on edge-to-edge delay of a flow in a DiffServ network domain with FIFO aggregation and a class-based Latency Rate (LR) server that provides guaranteed performance with rate reservation for a traffic class. We derive edge-to-edge delay bound for a single flow as a function of allocated service rate for a traffic class, token bucket parameters adopted for flows at the network ingress, and information about joining and leaving flows. We compare the obtained delay bound with previous works using analytic results, and then conduct simulation to confirm the results. The derived bound is less than that of previous studies in all cases.

This paper presents an adaptive control design for the ABR traffic congestion control in ATM networks. Firstly, we consider a control-based mathematical model to the ABR traffic congestion control problem. Then the feedback pole placement control design is applied to the ATM ABR traffic congestion control problem for the case of known delays. Finally, by using the online plant parameter estimation algorithm and modifying the controller parameters adaptively in real time, a method to treat the case of unknown time-varying delays is proposed. Several design modifications are introduced to solve practical control issues such as bounded command rate constraint, output buffer saturation and bounded values to the plant parameter estimation algorithm. Simulations are implemented to verify the proposed control design. It is shown that while considering these practical control issues, the control method satisfies the requirements of fairness to users, network efficiency, unknown time-varying delays, queue length control and good convergence performance at an acceptable computation effort.

The multichannel switch is an architecture widely used for ATM (Asynchronous Transfer Mode). It is known that the fault tolerant characteristic can be incorporated in into the multichannel crossbar switching fabric. For example, if a link belonging to a multichannel group fails, the remaining links can assume responsibility for some of the traffic on the failed link. On the other hand, if a fault occurs in a switching element, it can lead to erroneous routing and sequencing in the multichannel switch. We investigate several fault localization algorithms in multichannel crossbar ATM switches with a view to early fault recovery. The optimal algorithm gives the best performance in terms of time to localization but is computationally complex, which makes it difficult to operate in real time. We develop an online algorithm which is computationally more efficient than the optimal one. We evaluate its performance through simulation. The simulation results show that the performance of the online algorithm is only slightly suboptimal for both random and bursty traffic. There are cases where the proposed online algorithm cannot pinpoint down to a single fault. We explain the causes and enumerate those cases. Finally, a fault recovery algorithm is described which utilizes the information provided by the fault localization algorithm. The fault recovery algorithm adds extra rows and columns to allow cells to detour the faulty element.

Our goal in this paper is to provide a complete detection analysis for the OS processor along with OSGO and OSSO modified versions, for M postdetection integrated pulses when the operating environment is nonideal. Analytical results of performance are presented in both multiple-target situations and in regions of clutter power transitions. The primary and the secondary interfering targets are assumed to be fluctuating in accordance with the Swerling II target fluctuation model. As the number of noncoherently integrated pulses increases, lower threshold values and consequently better detection performances are obtained in both homogeneous and multiple target background models. However, the false alarm rate performance of OSSO-CFAR scheme at clutter edges is worsen with increasing the postdetection integrated pulses. As predicted, the OSGO-CFAR detector accommodates the presence of spurious targets in the reference window, given that their number is within its allowable range in each local window, and controls the rate of false alarm when the contents of the reference cells have clutter boundaries. The OSSO-CFAR scheme is useful in the situation where there is a cluster of radar targets amongst the estimation cells.

Orthogonal modulation provides low probability of bit error, however its bandwidth efficiency is very low. Biorthogonal code may double the bandwidth efficiency, but its required bandwidth grows exponentially with the number of input bits as in orthogonal modulation. In this paper, we propose a multi-code biorthogonal code keying (MBCK) scheme that significantly reduces the signal bandwidth with the benefit from orthogonal waveform coding maintained. The system consists of multiple waveform coding blocks, and the sum of output codewords is transmitted. A problem with MBCK is that output signal is multi-level, which requires amplifier with high linearity. So it may not be an appropriate scheme for portable unit where power efficiency is highly important. We also propose a modified MBCK scheme that guarantees constant amplitude output. The transmitter of the proposed scheme contains a redundant waveform coder whose input is generated by encoding the information bits. Adding the codewords from all constituent waveform coding blocks, the composite signal has constant amplitude. It is also shown that the redundant bits are not only used to make constant amplitude signal but also used to improve the BER performance at the receiver.

The basic carrier sense multi-access control scheme for multicast communications in wireless ad hoc networks suffers from the well know hidden terminal problem. The data packet collision probability is relatively high, and the packet delivery ratio is sensitive to the network topology, nodes distribution and traffic load. In this paper, we generalize the virtual carrier sense collision avoidance approach to reduce packet collisions in multicast communications. The sender and receivers exchange RTS and CTS packets to reserve the channel. When more than one receivers reply with CTS packets, the sender will detect an "expected" collision which may be interpreted as a valid "clear-to-send" signal provided the collision satisfies the given timing requirements. Together with a receiver-initiated local recovery mechanism, the reliability and packet delivery ratio can be improved to close to 100%.

Smart antennas, with spatial processing, used in code division multiple access (CDMA) multiuser communications can enhance range, reliability and capacity. Moreover, the adaptive beamforming technologies can remove unwanted noise and interference from the received signal. In this paper, we propose four adaptive antenna receiver structures for direct-sequence (DS) CDMA multiuser environment with multipath fading channels. Narrowband and wideband adaptive array receiver structures are considered in this paper. LMS and RLS algorithms are employed in both narrowband and wideband receiver structures to adjust the spreading code coefficients. We call these new schemes as Wiener code filters. The weights of the adaptive beamformer and the spreading code are updated every symbol interval, so the computational complexity is very low. Simulation examples are given to compare the performances of the proposed receiver structures. Analyses are also given for these structures.

In this paper, downlink performance of multicarrier CDMA (MC-CDMA) systems under correlated fading channels is analytically investigated. Under code hopping, a signal to interference plus noise ratio (SINR) and bit error rate (BER) performance are derived in multi-cell environments. In addition, specific conditions to validate the commonly used Gaussian approximation of an interference plus noise distribution in MC-CDMA systems are discussed. It is proved that the approximation is adequate in case of low correlation between subcarriers and a large spreading factor (SF). The proposition is confirmed through comparison between analytical and simulation results.

In this paper, a pilot-assisted channel estimation using adaptive interpolation (in which, different interpolation filter tap weights is used for different symbol position) is proposed. Each set of tap weights is updated using the normalized least mean square (NLMS) algorithm, the reference signal for which is obtained by decision feedback and reverse modulation of the received data symbol. In order to reduce the number of tap weight sets and to achieve fast convergence, the conjugate centrosymmetry property of the tap weight set is used. The average bit error rate (BER) performance in a frequency-selective Rayleigh fading channel is evaluated by computer simulation. Also evaluated is the robustness against the frequency offset between a transmitter and a receiver.

In mobile communication systems, mobility tracking operations are executed to maintain known the whereabouts of each mobile terminal (MT). In this article, we propose and analyse two new versions of the original time-based registration method, namely the Stop and Wait (S W) and the Reset and Wait (R W) time-based strategies. We also propose and study the modified S&W and R&W schemes as a result of combining the S&W and R&W policies with the enhanced time-based method in [3]. When the MT uses one of the four schemes proposed, it sends less location update (LU) messages. As a result of fewer LUs, which results in less contacts with the network, the uncertainty of the MT position increases in the proposed time-based schemes. Although the MT paging (PG) cost is lightly increased, a significant reduction in the LU cost is achieved. As it is shown, the net effect is a saving in the total location management cost per call arrival. In order to analyse the four strategies and compare them with the original and enhanced time-based schemes, a performance evaluation method based on Markovian standard tools is proposed here. The location management cost for all these time-based policies are evaluated analytically here by using this evaluation method.

Ad hoc networks have recently become a hot topic. In ad hoc networks, battery power is an important resource, since most terminals are battery powered. Terminals consume extra energy when their network interfaces are in the idle state or when they overhear packets not destined for them. They should, therefore, switch off their radio when they do not have to send or receive packets. IEEE802.11 features a power saving mechanism (PSM) in Distributed Coordination Function(DCF). In PSM for DCF, nodes must stay awake for a fixed time, called ATIM window (Ad-Hoc Traffic Indication Map window). If nodes do not have data to send or receive, they enter the doze state except for during ATIM window. However, ad hoc networks with PSM have longer end-to-end delays to deliver packets and suffer lower throughput than the standard IEEE802.11. To solve this problem, this paper proposes a protocol that reduces delay and achieves high throughput and energy efficiency. Simulation results show that our proposal outperforms other PSMs in terms of throughput, end-to-end delay and energy efficiency.

Many reports have investigated TCP performance over wireless links, where a high and time-invariant frame error rate is assumed for cellular systems. However, the frame error rate is temporally and geographically changeable by fading and interference in cellular systems. On the other hand, SINR-based transmission power control, which is employed for the randomization of frame errors in DS-CDMA cellular systems, can not always work properly depending on the control parameters or the channel characteristics. In this paper, we investigate the TCP performance over the wireless links in a DS-CDMA cellular system by computer simulation. From the simulation results, it has been found that the assumption of random frame error is valid only for a part of the TCP performance even in the system with an SINR-based transmission power control scheme.

Taking the uplink and downlink cochannel interference and noise into account, we determine the error probability in detecting a binary phase-shift keying (BPSK) signal transmitted over a satellite system containing two high power amplifiers (HPA). The first one is the constituent part of the transmitting ground station and the second one is the constituent part of the satellite station. The emphasis is placed on determining the system performance degradation imposed by the influence of the nonlinear characteristic of the HPA at the transmitting ground station in combination with the negative influences of the uplink and downlink cochannel interference, as well as the nonlinear characteristic of HPA at the satellite station.

The effect of antenna correlation on the Multiple-Input Multiple-Output (MIMO) channel capacity in the real propagation environment is a topic of interest. In this paper, we present the results of a measurement campaign conducted in an indoor Line-Of-Sight (LOS) office environment. Channel responses were taken with varying distance in a static indoor environment. Results showed measurements with high received Signal-to-Noise Ratio (SNR) and a high level of correlation among the antenna elements. Further analysis of the results showed that MIMO systems can achieve sufficient channel capacity compared to the Single-Input Single-Output (SISO) system, despite high antenna correlation. Theoretical analysis reveals that when the SNR is sufficiently high, the loss in channel capacity due to high antenna correlation is relatively low. Therefore it is shown that in the indoor LOS environment, MIMO systems can be sufficiently efficient because the MIMO channel is more robust to antenna correlation when the SNR is high.

In order to assess the effect of multipath fading on Orthogonal Frequency Division Multiplex (OFDM) signal transmission when the delay profile exceeds the guard interval, a simple prediction model is developed based on the Equivalent Transmission-Path (ETP) model. This model, which is described in this paper, is referred to as the ETP-OFDM-statistical model. The validity of the model is demonstrated by comparing the calculated digital transmission characteristics to results obtained by computer simulation. Using the newly developed ETP-OFDM-statistical model, digital transmission characteristics of the OFDM signal in a multipath environment when the delay profile exceeds the guard interval are shown as a function of delay spread, guard interval and OFDM symbol period.

Crosstalk or coupling phenomenon between two parallel conductors with a ground or reference conductor is well known in electromagnetic compatibility and microwave circuit fields. In this paper we consider the characteristics of a long two-conductor line embedded in an inhomogeneous medium while taking account of the difference between two phase constants in independent propagation modes of two parallel lines. The characteristics of near-end and far-end crosstalk and of through-end transmission are discussed for frequencies up to more than the frequency where the line length is 10 times the wavelength. Interesting properties, such as very strong and null crosstalk phenomena that occur at the far end and non-transmission at the through end, appear in the envelope characteristics. The reason for these properties is clarified by making theoretical and experimental investigations.

In designing a video-on-demand system, one of the major challenges is how to reduce the client's waiting time maintaining the concurrently used channels. For this reason, the hybrid architectures which integrate the multicast streams with the unicast streams were suggested in order to improve channel efficiency in recent years. In combining multicast with unicast, the ways to group the channels together are important so that more clients can share the multicast transmission channels. This paper proposes a hybrid video-on-demand system which gathers the unicast and multicast transmission channels efficiently by using dynamic channel allocation architecture. The newly proposed architecture can reduce the average client's waiting time significantly. The numerical results demonstrate that the dynamic channel allocation architecture in some case (e.g., 100-channel and 10-video system at 0.5 requests/second) achieves performance gain of 551% compared to existing architecture. This paper presents procedure of channel release and reuse, performance analysis, and simulation results of the dynamic channel allocation architecture.

A robust time and frequency synchronization scheme is proposed for the high rate OFDM-based wireless local area networks (WLANs). The IEEE 802.11a standardized preamble is efficiently utilized and makes the proposed scheme practical. Simulation results under different channel environments are presented to illustrate the effectiveness of the proposed scheme.

In this letter, we propose a closed-form joint space-time channel and direction of arrival (DOA) blind estimation algorithm for space-time coded MC-CDMA systems equipped with a uniform linear array (ULA) at the base station in frequency-selective fading environments. We prove that the signal subspaces defined by the receive data covariance matrix can be determinately separated into an equivalent set of signal subspaces spanned by the space-time channel vector of an individual user. From these signal subspaces, the space-time channels of multiple users are estimated using the subspace method.

One of the drawbacks of multicarrier transmission schemes is that the transmitted signals have high peak-to-average power ratio (PAPR). When PAPR of the signal is larger than the input-back off (IBO) of the high power amplifier (HPA), signal powers larger than the saturation threshold of the HPA cause the saturation events so that the error rate performance is degraded. To improve the error-rate performance degraded by the nonlinear distortion, not only the signal power above the saturation threshold but also the interval of the signals causing the saturation events at HPA should be reduced. In this paper, we propose the total exceeding power (TEP) as a new criterion for improving the error rate performance degraded by the nonlinear amplifier for multicarrier transmission.

We propose an iterative tuning process of blind/group-blind detector based on constant modulus property of digitally modulated signal. By adopting the iterative tuning process after applying blind/group-blind detectors, significant gains are attained compared to the original blind/group-blind detectors. The signal to interference and noise ratio (SINR) approaches to ideal MMSE detector even with reasonably small number of data samples.

This letter discusses the prediction of the time-varying bit error rate (BER) for a transmitting channel using recent transmissions and retransmissions. Depending on the predicted BER, we propose a maximum frame size control to improve the goodput in wireless networks. It is shown, using simulation, that when the maximum frame size is controlled relative to the time-varying BER the goodput of the network is improved.

This letter compares orthogonal space time codes and quasi-orthogonal codes when the wireless channels are fast fading. It is well known that a orthogonal space-time code is better than a quasi-orthogonal code in high signal-to-noise ratio (SNR) range and that a quasi-orthogonal code is better in low SNR range. In this letter, we show that a quasi-orthogonal space-time code is a better choice even in high SNR range when the channels are fast fading.

In this paper, we propose an OFDM scheme with pre-IDFT/DFT and the frequency domain equalization on frequency-selective Rayleigh fading channels. In this scheme, a two-dimensional block interleaving is used to randomize the correlated noise caused by the frequency domain linear equalizer. Then, the pre-DFT averages the interleaved noise enhancement and improves the error performance of the proposed scheme. Computer simulations confirm the bit error probability of the proposed scheme for multilevel modulations.

Severe frequency-selective fading, encountered in a broadband wireless mobile communication, significantly degrades the bit error rate (BER) performance of direct sequence spread spectrum (DSSS) signal transmission with rake combining. In this paper, frequency-domain pre-equalization transmission, called pre-FDE transmission, is presented for orthogonal multicode DSSS signal transmission. It is confirmed by the computer simulation that pre-FDE transmission can achieve a BER performance almost identical to that attainable by FDE reception.

In an Orthogonal Frequency Division Multiplexing (OFDM) systems, the Peak to Average power Ratio (PAR) is high. The clipping signal scheme is a useful and simple method to reduce the PAR. However, it introduces additional noise that degrades the systems performance. We propose an oversampling scheme to deal with the received signal in order to reduce the clipping noise by using finite impulse response (FIR) filter. Coefficients of the filter are obtained by correlation function of the received signal and the oversampling information at receiver. The performance of the proposed technique is evaluated for frequency selective channel. Results show that the proposed scheme can mitigate the clipping noise significantly for OFDM systems and in order to maintain the system's capacity, the clipping ratio should be larger than 2.5.

An approach based on signal subspace analysis is proposed to blind estimation of the PN (Pseudo Noise) sequence from lower SNR (Signal to Noise Ratios) DS/SS (Direct Sequence Spread Spectrum) signals. The received signal is divided into vectors according to a temporal window, from which an autocorrelation matrix is computed and accumulated. The PN sequence can be reconstructed from principal eigenvectors of the matrix.

In this letter, we propose a 2-D receiver structure for multicarrier code division multiple access (MC-CDMA) systems with the reduced-rank multistage Wiener filter. Due to the fast convergence property of the reduced-rank processing, it outperforms MMSE-based receivers with the classical Wiener solution, which is estimated by using a limited number of samples.

Digital Terrestrial Television (DTV) services began in Japan in December 2003. This paper proposes a novel on-glass antenna for mobile reception of terrestrial television. The gain of the proposed antenna is 4.7 dB higher than commercial monopole antennas when installed on a vehicle. Other merits of this antenna are a broad input impedance bandwidth across the UHF band (470-710 MHz), and the fact that it does not spoil vehicle appearance. Field experiments have confirmed that a diversity system using four of the proposed antennas is capable of mobile DTV reception.

In this letter, we study a hybrid Multimedia-on-Demand (MoD) system which provides broadcast, batch and interactive services. An analytical model for such an MoD system is provided. Numerical results show that with proper design, the system can provide better performance than those systems which only provide any subset of two services.