The AODV (Ad Hoc On-Demand Distance Vector) protocol is one of the typical reactive routing protocols in Ad Hoc networks, in that mobile nodes initiate routing activities only in the presence of data packets in need of a route. In this paper, we focus upon the local repair mechanism of AODV. When a link is broken, the upstream node of the broken link repairs the route to the destination by initiating a local route discovery process. The process involves the flooding of AODV control messages in every node within a radius of the length from the initiating node to the destination. In this paper, we propose an efficient local repair scheme for AODV, called AELR (AODV-based Efficient Local Repair). AELR utilizes the existing routing information in the downstream intermediate nodes which have been on the active route to the destination before a link break occurs. AELR can reduce the flooding range of AODV control messages and the route recovery time for route recovery because it can repair a route through the nearby downstream intermediate nodes. The performance results show that AELR can achieve faster route recovery than the local repair mechanism of AODV.

We propose an efficient IP address assignment protocol in mobile ad hoc networks, which use a proactive routing protocol. In this method, which is termed the Bisected-Range based Assignment (BRA), a node repeatedly broadcasts an Agent Request to ask for address assignment. If there are one or more neighbor MANET nodes, one of them becomes an agent to select and assign an IP address to the requesting node. We use address location in the IP address space so that each agent maintains its own exclusive address range to be used for address selection, resulting to decrease the possibility of address conflict. If the requesting node cannot discover any neighbor MANET node over pre-determined random agent-search time, it selects by itself an IP address at random from the given address block. We evaluate performance of the basic and enhanced BRAs by computer simulation. It is shown that the basic and enhanced BRAs can reduce address conflict compared with random assignment. It is also shown that the enhanced BRA is superior in terms of control traffic overhead as well as address assignment delay over the random assignment with the strong Duplicate Address Detection.

The single path routing protocol, known as the Ad Hoc On-demand Distance Vector, has been widely studied for use in mobile ad hoc networks. AODV requires a new route discovery whenever a path breaks. Such frequent route discoveries cause a delay due to route discovery latency. To avoid such inefficiency, a multipath routing protocol has been proposed that attempts to find link-disjoint paths in a route discovery. However, when there are two or more common intermediate nodes on the path, the protocol can not find a pair of link-disjoint paths even if the paths actually exist. To reduce this route discovery latency, it is necessary to increase the opportunities for finding a pair of link-disjoint paths. In this paper, we focus on AODV and propose an AODV-based new multipath routing protocol for mobile ad hoc networks. The proposed routing protocol uses a new method to find a pair of link-disjoint paths by selecting a route having a small number of common intermediate nodes on its path. Using simulation models, we evaluate the proposed routing protocol and compare it with AODV and the existing multipath routing protocol. Results show that the proposed routing protocol achieves better performance in terms of delay than other protocols because it increases the number of cases where a pair of link-disjoint paths can be established.

This paper presents a path compression protocol for on-demand ad hoc network routing protocols, which is called dynamic path shortening (DPS). In DPS, active route paths adapt dynamically to node mobility based on the "local" link quality estimation at each own node, without exchanging periodic control packets such as Hello messages. Each node monitors its own local link quality only when receiving packets and estimates whether to enter the "proximity" of the neighbor node to shorten active paths in a distributed manner. Simulation results of DPS in several scenarios of various node mobility and traffic flows reveal that adding DPS to DSR which is the conventional prominent on-demand ad hoc routing protocol significantly reduces the end-to-end packet latency up to 50-percent and also the number of routing packets up to 70-percent over the pure DSR, in heavy traffic cases. We also demonstrate the other simulation results obtained by using our two novel mobility models which generate more realistic node mobility than the standard random waypoint mobility model: Random Orientation Mobility and Random Escape Mobility models. Finally, simple performance experiments using DPS implementation on FreeBSD OS demonstrate that DPS shortens active routes in the order of milliseconds (about 5 ms).

Since a radio channel is shared among terminals in an ad hoc network, packet collisions are frequent. In case of transmitting packets especially using TCP, data and ACK packets are transmitted in opposite directions on the same radio channel. Therefore, frequent collisions are unavoidable, and this seriously degrades TCP throughput. It is possible to transmit to two or more nodes which adjoin from a certain node simultaneously on the radio channel. To reduce the likelihood of packet collisions when an intermediate node transmits both data and ACK packets, these two types of packet can be combined and transmitted at the same time to increase the efficiency of radio channel utilization. In this paper, we propose a new technique to improve TCP performance by combining data and ACK packets. Our proposed technique is applicable to generic ad hoc networks easily. By means of a simulation using networks with various topologies, we have found that throughput can be improved by up to 60% by applying our proposed technique.

Along with expansion of utilization of mobile ad hoc networks, it is expected that the network size becomes large. However, design of current typical routing protocols supposes at most several hop routes between source and destination nodes. When messages are delivered along long hop routes in the networks, such routing protocols tend to degrade performance. Previously, we have proposed an autonomous clustering scheme for constructing and maintaining hierarchical structure in mobile ad hoc networks, which are adaptive to node movement. This paper proposes a class of hierarchical routing protocols Hi-TORA, Hi-DSR and Hi-AODV, all of which are based on the autonomous clustering scheme, compares them with their corresponding flat routing protocols TORA, DSR and AODV, respectively, and shows effectiveness of these hierarchical routing protocols by simulation experiments.

Sensor nodes are prone to failure and have limited power capacity, so the evaluation of fault tolerance and the creation of technology for improved tolerance are among the most important issues for wireless sensor networks. The placement of sensor nodes is also important, since this affects the availability of nodes within sensing range of a target in a given location and of routes to the base station. However, there has been little research on the placement of sensor nodes. Furthermore, all research to date has been based on deterministic node placement, which is not suitable when a great many sensor nodes are to be placed over a large area. In such a situation, we require stochastic node placement, where the sensor-positions are in accord with a probability density function. In this paper, we examine how fault tolerance can be improved by stochastic node placement that produces scale-free characteristics, that is, where the degree of the nodes follows a power law.

This paper presents a mobility management scheme that combines host-based routing (HBR) with prefix routing to achieve balanced loading of network nodes in a distributed hierarchically arranged mobile IPv6 access network. This allows the higher-level nodes to be less loaded than in pure host based routing schemes, where the root node presents a capacity bottleneck to the system. As a result, this scheme achieves good savings in memory by reducing host-specific caches, and thus enhances network scalability. A direct consequence of reduced database entry is reduced processing latencies at the nodes, which reduces delay and improves on network performance. Our hybrid HBR scheme performs better than the pure HBR schemes in memory conservation and increased network capacity.

Proposed CCRSVP (CandidateCasting RSVP) algorithm is a new fast handoff method for IEEE 802.11 Wireless LAN (WLAN) environments. It shows good performance in the handoff latency and the bandwidth efficiency aspect and guarantees QoS because it uses an advanced multicasting method and RSVP. CCRSVP uses L2 information (BSSID) of WLAN and starts reserving resources and multicasting packets before L2 handoff completes. Therefore, the proposed algorithm can reduce L3 handoff latency more than other methods. To show performance of CCRSVP algorithm, we calculate handoff latency and packet loss ratio of each algorithm. Also we model handoff process which uses RSVP mechanism to confirm resource efficiency. Proposed handoff model uses parameters which can distinguish each handoff algorithm. We introduce Markov chain which can analyze handoff model and analysis method which uses iteration method. In this article, the results show that the proposed algorithm shows superior bandwidth efficiency than existing L3 handoff algorithms using RSVP. To analyze bandwidth efficiency of each algorithm, we compare the blocking probability which occurs in case of absence of resource, resource usage which shows reservation quantity, the average number of ongoing session which really uses resource reserved and resource utilization. We can confirm that CCRSVP algorithm has better performance than other algorithms at each comparative item.

In this article we propose a new framework to support QoS and 4G enabled mobility management. The key feature of this framework is twofold; the Policy based Handover control, and the adaptive handover mechanisms. On one hand, the adaptive handover mechanism will allow the selection of different handover schemes based on the user service level specification in order to differentiate the handover quality provided for the required service level, thus optimizing the network resources usage and providing the agreed service level to users. On the other hand, the proposed handover control mechanism considers resource availability and other constraints during the handover decision in order to provide support for QoS aware mobility on the contrary of the classical naïve mobility management that considers only signal strength fading. This decision mechanism will provide mainly the interaction between the mobility and QoS management frameworks but also a 4G handover control. In addition, the QoS aware handover will provide the knowledge necessary to achieve a proactive handover's procedures control.

Recently, mobile SCTP (mSCTP) has been proposed as a transport layer approach for supporting mobility. mSCTP is based on the 'multi-homing' feature of Stream Control Transmission Protocol (SCTP), and utilizes the functions to dynamically add or delete IP addresses of end points to or from the existing connection in order to support mobility. In this paper, we propose a transport layer mobility supporting scheme, which is based on mSCTP and utilizes the link layer signal strength information in order to determine when to add or delete end-point IP addresses of mobile node and how to change data delivery paths when handover happens. Exploiting the fact that the transport layer is aware of the mobility in the proposed scheme, we also propose error and congestion control enhancement to cope with handover efficiently. The simulation results show that the performance of proposed scheme is competitive compared to the traditional network layer mobility supporting approach. Especially, when the moving speed of mobile node is fast or new path acquisition takes long, it shows better performance than the traditional network layer approach.

Hierarchical Mobile IPv6 (HMIPv6) has been proposed to manage the mobility of Mobile Terminals (MTs) hierarchically to reduce packet losses during local handover. HMIPv6 uses a mobility manageable router in a domain visited by the MTs to manage the micromobility of the MTs. The router is called Mobility Anchor Point (MAP). As a hierarchical mobility management scheme based on HMIPv6, we have already proposed a multilevel hierarchical distributed IP mobility management scheme to manage the mobility of MTs in a decentralized manner using multiple MAPs. Our scheme manages the mobility of an MT using a MAP having a suitable management domain. This usage of MAPs aims to efficiently decentralize the load of mobility management. Our scheme estimates the movement speed of the MT and then estimates the mobility of the MT based on the estimated movement speed of the MT to achieve the objective. However, recent simulation results obtained with more realistic mobility model indicate that our scheme has two problems in estimating the mobility of MTs: One is that our current scheme misestimates the movement speed of an MT in some cases, and the other is that our current scheme does not notice the changes in the mobility of an MT when the MT decelerates and stays in the same access area for a long time. Thus, an enhanced mobility estimation method is proposed in this paper. The enhanced method has an ability to estimate the movement speed of MTs more correctly and an ability to urge decelerated MTs to degrade their MAP quickly. Finally, the performance of the proposed mobility estimation method is evaluated using simulation experiments. The simulation results show that the enhanced method allows our scheme to estimate the mobility of MTs more correctly and so achieve more efficient load sharing.

Hierarchical Mobile IPv6 (HMIPv6) has been proposed to improve the performance capability of Mobile IPv6 at handover. In HMIPv6, local entities named Mobility Anchor Points (MAPs) are distributed throughout a network to localize the management of intra-domain mobility. In particular, multi-layered MAP has been proposed to improve performance. MAPs reduce the number of Binding Updates to the Home Agent and improve the communication quality at handover. These conventional methods that manage a multi-layered MAP cannot, however, select an appropriate MAP because they use the virtual mobility speed. As a result, they increase the signaling traffic in a multi-layered MAP. Moreover, they may cause the load to concentrate at a specific MAP. In this paper, we propose a location management method for Hierarchical Mobile IPv6 using the MN's mobile history. In this method, when a MN performs a handover, the Access Router calculates the area-covered rate of each upper MAP from the MN's mobile history and selects the MAP that best manages the MN in accordance with its rate. Thus, the proposed method reduces both the number of Binding Updates to the Home Agent and the signaling traffic because it reduces the frequency of changing the MAP. We evaluate the performance of the proposed method by simulation.

The next-generation IP-based mobile network must accommodate various kinds of wireless access technologies, including W-CDMA. Although the soft handover technique should be supported if W-CDMA is used, redundant paths will be created by the soft handover scheme employed by the 3rd generation mobile communication system. This paper proposes the Network Distributed Soft Handover (NDSHO) method, which achieves soft handover control in an IP network but without any redundant paths. NDSHO continuously optimizes all routing paths by relocating the data copy point dynamically during communication according to the movement of the mobile terminal. To achieve the proposed method, this paper introduces a copy point seamless relocation method and an optimal point selection method which takes advantage of OSPF. Furthermore, we show quantitatively that NDSHO makes more efficient use of system resources than the 3rd generation system.

In this paper, we propose the Synchronized Mobile Multicast (SMM) scheme for the real-time multimedia service to achieve three most important characteristics that the traditional Home Subscription (HS) and Remote Subscription (RS) mobile schemes cannot support. First, the SMM scheme supports the scalable one-to-many and many-to-many synchronized multimedia multicast on mobile IP networks to achieves seamless playback of continuous media streams even when both the mobile sender and receivers handoff simultaneously. Second, it analyzes the minimal buffer requirements of the mobile sender, the core router, the foreign agents and the mobile receivers in the multicast tree and formulates the initial playback delay within a handoff Guarantee Region (GR). Further, combined with the fine granularity scalability (FGS) encoding approach in the MPEG-4 standard, the SMM scheme achieves superior multimedia QoS guarantees and unlimited numbers of handoffs of the mobile sender and receivers only at the cost of degraded video quality for a short period after handoff with minimal extra bandwidth.

In this paper, we propose a set of broadcast streaming protocols designed for unidirectional radio channels. Considering the limited size and implementation overhead on a mobile terminal, the proposed protocol set is almost compliant with the current mobile streaming protocols, i.e. 3GPP PSS (Packet-switched Streaming Service), except for that the proposed protocols are designed to work on a unidirectional downlink channel. This protocol set enables flexible layout rendering by SMIL (Synchronized Multimedia Integration Language) in combination with SDP (Session Description Protocol), and reliable and synchronized static media (including still image and text) delivery by RTP (Real-time Transport Protocol) carousel. We present the prototype of this protocol set and measure its performance of video quality and waiting time for video presentation through a W-CDMA radio channel emulator and header compression nodes. From the experimental results, we show 1) trade-off between video quality and waiting time, 2) advantage and disadvantage of header compression, 3) effectiveness of synchronized transmission of SDP, SMIL, and I-frames of video objects, and 4) reliability of RTP-carousel. This protocol set is applicable to 3G MBMS (Multimedia Broadcast/Multicast Service) streaming service.

Integrating Multi Protocol Label Switching (MPLS) and Mobile IP (MIP) protocols is a challenge to support Quality of Services (QoS) as well as mobility. Existing approaches may suffer from (1) high blocking rate and handover delay due to the path re-establishment, and (2) twice MPLS label PUSH and POP procedures due to the triangle routing. In this paper, we propose a novel approach to integrating MPLS and MIP. The proposed approach maintains an efficient QoS path and solves the above drawbacks. We also design a path extension procedure to reduce blocking probability and a path optimization procedure to maintain the shortest transmission path under delay constraints with the lowest cost. The simulation results show that the proposed approach reduces handover delay, connection dropping rate, and data loss rate comparing to the original MIPs over MPLS. Furthermore, we can also tune the system performance using an extension counter for the trade-off between management cost and data transmission delay.

Three representative protocols are proposed to support mobility for IPv6 in IETF: Mobile IPv6, Hierarchical Mobile IPv6, and Fast Handovers for Mobile IPv6. Recently, IEEE 802.11 network has been widely deployed in public areas for mobile Internet services. In the near future, IPv6 mobility support over IEEE 802.11 network is expected to be a key function to actualize the pure IP-based mobile multimedia service. The IPv6 mobility support protocols have their characteristics in terms of signaling, handover latency, lost packets, and required buffer size. In this paper, we analyze the performance of the protocols over IEEE 802.11 network. We define a packet-level traffic model and a system and mobility model. Then, we construct a framework for the performance analysis. We also make cost functions to formalize each protocol's performance. Lastly, we investigate the effect of varying parameters used to show diverse numerical results.

In wireless environment, TCP suffers from significant performance degradation due to bit errors on wireless link and handovers because it responds to all packet losses by invoking congestion control even though packet losses are not related to congestion. Several schemes have been proposed to improve the performance degradation due to each cause. They have been evaluated in a specific network environment where either bit errors or handover occurs, i.e. they do not occur at the same time. In this paper, we reveal the packet recovery mismatch problem in an environment where both of bit errors and handover can cause the performance degradation. We pick up one scenario that TCP traffic is transmitted in the situation that ARQ (Automatic Repeat reQuest) and packet forwarding are implemented together. They are proposed to reduce the influence of bit errors and handover respectively and are natural approaches from the viewpoint of protocol layering. Computer simulation shows that in that scenario both techniques cannot perform efficiently due to interaction of each other. We also propose two buffer control approaches to resolve the packet recovery mismatch problem in our scenario according to applicability of cross-layer operation between layer 2 and layer 3.

In this letter, a resource allocation scheme is proposed to efficiently utilize the resource of CDMA systems with respect to the throughput. When we let the throughput be a function of various data rates for multimedia traffics, the scheme is to find the optimum set of data rates for concurrent user groups with which we can achieve the maximum throughput. It is also shown that the optimum data rate set always includes one non-trivial rate while all the others keep the minimum required data rate.

Since IEEE 802.11 has many problems such as hidden node, exposed node problem, larger sensing function and BEB (Binary Exponential Back-off), it is not suitable for use in multi-hop wireless ad hoc network. When an on-demand routing protocol is used with 802.11 DCF (Distributed Coordination Function), the route to transmit the packet will be formed by nodes which have less competition (fewer neighbors) than others for the medium access. This effect will make that the routing path will be longer and decrease network performance. Therefore, we propose a new MAC (Medium Access Control) protocol that makes a shorter routing path, enabling better performance in multi-hop wireless ad hoc networks. The protocol modifying IEEE 802.11 DCF gives priority to the node with more neighbors and with less transmission. Through simulations, we have demonstrated that the proposed algorithm improves performance in terms of transmission rate, transmission delay and total consumption energy.

This paper presents a new approach based on current mode circuits for fast and accurate optical level monitoring with wide dynamic range of a gigabit burst-mode laser driver chip. Our proposed solution overcomes the drawbacks that voltage mode implementations show at higher bit rates or in other technologies. The main speed-limiting factor of the level monitoring circuitry is the parasitic capacitance of the back facet monitor photodiode. We propose the use of an active-input current mirror to reduce the impact of this parasitic capacitance. The mirror produces two copies of the photo current, one to be used for the "0" level measurement and another for the "1" level measurement. The mirrored currents are compared to two reference currents by two current comparators. Every reference current needs only one calibration at room temperature. A pattern detection block scans the incoming data for patterns of sufficiently long consecutive 0's or 1's. At the end of such a pattern a valid measurement is present at the output of one of the current comparators. Based on these measurements the digital Automatic Power Control (APC) will adjust the bias (I_BIAS) and modulation current (I_MOD) setting of the laser driver. Tests show that the chip can stabilize and track the launched optical power with a tolerance of less than 1 dB. In these tests the pattern detection was programmed to sample the current comparators after 5 bytes (32 ns at 1.25 Gbps) of consecutive 1's and 0's. Automatic power control on such short strings of data has not been demonstrated before. Although this laser transmitter was developed for FSAN GPON applications at a speed of 1.25 Gbps upstream, the design concept is generic and can be applied for developing a wide range of burst mode laser transmitters. This chip was developed in a 0.35 µm SiGe BiCMOS process.

The WDM optical networks currently being deployed are opaque optical networks, in which each link is optically isolated by transponders. To reduce the number of expensive transponders and switching ports, a hierarchical optical architecture consisting of all-optical waveband switching and opaque OEO switching has been proposed. Although this architecture requires fewer transponders and ports, it also requires a large number of wavelength (waveband) multiplexers and demultiplexers. Switching the optical path solely at the fiber level (i.e., by using fiber cross-connects, or FXCs) is desirable as a way to reduce the total node cost. If all the core nodes in an optical network are FXCs, however, the grooming of wavelengths for the optical fibers is only possible at the edge nodes. This leads to poor utilization of wavelength resources when there is only demand for small numbers of wavelengths, and as a result, the link cost increases. This problem can be solved by adding an OEO grooming function to some of the FXCs. In this paper, we propose an algorithm for designing optical cross-connect (OXC) functions on the basis of the FXC, thus minimizing the total network cost.

In this paper, we study a work-conserving multicast scheduling with fanout splitting in a switch, which routes incoming packets asynchronously without fragmentation into cells. A new switch architecture is proposed, which distributes the input links to P variable length packet switching fabrics (VPS) with every G input links sharing GR inlets of the VPS. The system performance is analyzed by queueing analysis to express the maximum throughput and packet delay in terms of the system parameters and traffic characteristics. A practical switch design is also proposed to realize almost the same scheduling as the work-conserving one. We have surveyed how the fanout distribution affects the performance of the switch through Fanout Function, which is defined and studied to help the design of a multicast switch. We show how Fanout Function determines the maximum throughput and packet delay. Various fanout distributions are compared. The mixed fanout distribution exhibits better performance while the deterministic fanout can be used as a bound in the design of a multicast switch. We optimize R and P to attain 100% maximum throughput under limited switch complexity. When the mean fanout size is large, we can use less hardware to achieve the optimal performance by using our architecture. The proposed realization of this switch can be implemented easily due to its modular design. It is scalable because distributed output contention resolution and routing are used instead of a central arbitrator. Its performance is verified by simulation. The result matches the theoretical work-conserving scheduling very well.

This paper proposes two types of network architectures using UMTS Release 5 architecture and wireless LAN suitable for context-aware information delivery and personal communication services, and it reports on preliminary evaluations of their performance. The first type of network architecture is the NCA (network-centric architecture) and the second is the ECA (end-user-centric architecture). The two architectures are modeled with a queuing network and their response times are compared through theoretical analysis and simulation. The results indicate that with low-performance servers, the response times of the ECA are generally shorter or almost the same as those of the NCA. However with high-performance servers, the response times of the NCA are generally shorter except during high server utilization.

This paper proposes dynamic distributed unicast and multicast routing algorithms for multiple classes of QoS guaranteed networks. Each link in such a network is assumed to be able to provide multiple classes of QoS guarantee by reserving various amounts of resource. A distributed unicast routing algorithm, DCSP (Distributed Constrained Shortest Path), for finding a QoS constrained least cost path between each O-D (Originating-Destination) pair, is proposed first. Two class reduction schemes, the linear and logarithmic policies, are develpoed to prevent exponential growth of the number of end-to-end QoS classes. Based on DCSP, two distributed multicast routing algorithms, DCSPT (Distributed Constrained Shortest Path Tree) and DTM (Distributed Takahashi and Mutsuyama), are proposed to find QoS constrained minimum cost trees. Numerical results indicate that DCSP strongly outperforms previously proposed centralized algorithms and it works better with the linear class reduction method. For the multicast routing algorithms, the DCSPT with linear class reduction method yields the best performance of all multicast routing algorithms.

With the explosive growth of the Internet system, demands for broadband communication networks have rapidly increased to provide high quality network services. For this purpose, the IEEE 802.6 MAC standard protocol defines the distributed-queue dual bus (DQDB) for metropolitan area networks (MANs). The isochronous channel reuse problem (ICRP) has been studied for efficient use of DQDB by finding proper channel assignments to incoming connection requests. In this paper, we first define the generalized isochronous channel reuse problem (GICRP) as a generalization of ICRP, to afford demands of simultaneously satisfying plural connection requests such as for multicast applications, where certain sets of connection requests must be assigned channels simultaneously. We prove the NP-completeness of its decision problem. Then, we propose a minimum dead space (MDS) algorithm as a heuristic approach to GICRP. The extensive simulation results show that with shorter computation time, our MDS algorithm can always find better channel assignments reducing the waiting time for packet transmissions than the best existing algorithm for conventional ICRP.

We propose a new end-to-end transport protocol called Multi-path Transmission Control Protocol (M/TCP) and its two robust acknowledgement (ACK) schemes. Our protocol is designed as an alternative TCP option to improve reliability and performance of the Internet. The M/TCP sender simultaneously transmits data via multiple controlled paths to the receiver. Our protocol requires no modification in IP layer. Two M/TCP endpoints establish multiple paths between them by subscribing to multiple ISPs. The two robust ACK schemes proposed in this paper aim at improving M/TCP performance over the Internet with high packet loss in ACK channels. Performances between our protocol and TCP Reno are compared in terms of throughput and fairness by using ns2 simulator. Simulation results indicate that M/TCP achieves higher throughput than TCP Reno in situation of random drop and burst traffic with small buffer size. When there is network congestion on reverse path, M/TCP with the proposed robust ACK schemes performs better than M/TCP with the conventional immediate ACK scheme.

In future high-speed networks, provision of diverse multimedia services with strict quality-of-service (QoS) requirements, such as bandwidth, delay and so on, is desired. QoS routing is a possible solution to handle these services. Generally, a path selection for QoS routing is formulated as a shortest path problem subject to multiple constraints. However, it is known to be NP-complete when more than one QoS constraint is imposed. As a result, many heuristic algorithms have been proposed so far. The authors proposed a path selection algorithm Fallback+ for QoS routing, which focuses not only on the path selection with multiple constraints but also on the efficient use of network resources. This paper proposes an enhanced version of Fallback+, named Enhanced Fallback+, where in a shrewd way, it keeps tentative paths produced in the conventional Fallback algorithm with Dijkstra's algorithm. Simulation experiments prove the excellent performance of Enhanced Fallback+, compared with the original Fallback+ and other existing path selection algorithms.

In a severe frequency-selective fading channel, the bit error rate (BER) performance of orthogonal multicode DS-CDMA is severely degraded since the orthogonality property of spreading codes is partially lost. The frequency-selectivity of a fading channel can be exploited by using frequency-domain equalization to improve the BER performance. Further performance improvement can be obtained by using transmit diversity. In this paper, joint transmit diversity and frequency-domain equalization is presented for the reception of orthogonal multicode DS-CDMA signals in a frequency-selective fading channel. As for transmit diversity, delay transmit diversity (DTD) and frequency-domain space-time transmit diversity (STTD) are considered. The achievable BER performance of multicode DS-CDMA in a frequency-selective Rayleigh fading channel is evaluated by computer simulation. It is shown that the frequency-domain STTD significantly improves the BER performance irrespective of the degree of the channel frequency-selectivity while DTD is useful only for a weak frequency-selective channel.

Orthogonal Frequency Division Multiplexing (OFDM) is a promising technique for achieving high bit-rate transmission in radio environments. Various techniques to estimate channel attenuation have been proposed for OFDM transmission. In these techniques, the added pilot semi-blind (APSB) channel estimation has been proposed, which does not require any additional bandwidth. For an OFDM packet transmission that includes preambles, it is difficult to track the channel variation of the latter of packet due to time-varying channels in fast fading environments. We expect that the channel estimation is improved by applying the APSB channel estimation technique to the last symbols of packets without any additional bandwidth or degradation of bit rate. In this paper, we propose an OFDM packet transmission system with APSB channel estimation technique where this technique is applied to the last symbols of packets. We expect that the proposed system can track the fast time-variant channels without decrease of the data-rate, and the packet error rate (PER) is improved. We show that our proposed system is effective for improving the accuracy of the channel estimate in fast fading channels.

The unfairness problem among TCP connections has been proved to be very severe in the IEEE 802.11-based wireless ad hoc networks because the hidden station problem still exists and the binary exponential backoff algorithm always favors the latest successful station. In this paper, a novel protocol, neighbor-medium-aware MAC (NEMA-MAC), is proposed to improve the TCP fairness. By adding a medium (channel) state field in the head of the traditional IEEE 802.11 MAC frame, the NEMA-MAC protocol provides a communication mechanism to resolve the hidden station problem. In addition, when a collision occurs, the new backoff algorithm makes the senders cooperatively adjust the contention window according to their local and neighbors' channel usage indexes. The simulation results show that TCP sessions can acquire satisfying fairness and increase the throughput in the NEMA-MAC-based multihop ad hoc networks.

This paper proposes the joint beamforming space-time block coding (JBSTBC) scheme for multi-input multi-output (MIMO) communication systems. To enhance the order of spatial diversity in presence of deteriorate fading correlations as well as inter-substream interferences, the proposed JBSTBC method employs joint eigen-beamforming technique together with the block-ordered layered detector (BOLD) for MIMO-STBC. In order to confirm the superiority of the proposed JBSTBC method, computer simulations are conducted in highly correlated fading situations while providing detailed mathematical derivations for clarifying functionality of the proposed scheme.

The Multiple Input Multiple Output (MIMO) technique is motivating the world-wide researchers to realize the next generation wireless LANs with the higher channel capacity and higher signal quality. This paper proposes and analyzes the Space Division Multiplexing Single Carrier OFDM (SDM-SCOFDM) system with adaptive modulation (AM) method over MIMO channels. The salient features of the proposed method are to enable the significant reduction of the number of required feedback adaptive modulation information (AMI) bits and the improvement of the PAPR (peak to average power ratio) performance at a cost of little degradation of channel capacity as comparing with the AM aided SDM-OFDM system. We also propose the Carrier-to-Noise Power Ratio (CNR) estimation method for SDM-SCOFDM signals over multi-path fading channels, which can be used for the assignment of the optimal modulation scheme in each transmit antenna. This paper presents the various computer simulation results to verify the proposed method under a typical wireless LAN environment.

In December 2001, the authors started two kinds of experiments on the meteor burst communication (MBC) in Antarctica to study the ability of MBC as a communication medium for data collection systems in that region. In the first experiment, a continuous tone signal is transmitted from Zhongshan Station. The signal received at Syowa Station (about 1,400 km apart) is recorded and analyzed. This experiment is aimed to study basic properties of the meteor burst channel in that high latitude region. On the other hand, the second experiment is designed to estimate data throughput of a commercial MBC system in that region. A remote station at Zhongshan Station tries to transfer data packets each consisting of 10 data words to the master station at Syowa Station. Data packets are generated with five minutes interval. In this paper, we explain the experiments, briefly examine the results of the first year (from April 2002 to March 2003), and put forward the plan for the experiments in the second and third year. From the data available thus far, we can see that 1) the sinusoidal daily variation in the meteor activity typical in middle and low latitude regions can not be clearly seen, 2) non-meteoric propagations frequently dominate the channel especially during night hours, 3) about 60% of the generated data packets are successfully transferred to the master station within two hours delay even though we are now operating the data transfer system only for five minutes in each ten minutes interval, etc.

A multimode horn with both a low cross-polarization component less than -30 dB and good VSWR characteristics has been realized at frequency bands 8.6-9.8 GHz and 10.75-11.15 GHz. The improved performance of the proposed horn is verified by comparing with the previous dual-band double-flared horn. The design method for such a horn is based on the mode-matching approach combined with the optimization procedure. This paper proposes an objective function taking account of a spill-over loss and a rotational symmetry in the radiated field instead of an ideal radiation pattern. The effectiveness of our horn is verified by comparison between experimental results in the X band and predicted ones.

In this paper, we examine the effect of random steering errors on the signal-to-interference-plus-noise-ratio (SINR) at the output of the recently addressed wavelet-based generalized sidelobe canceller (GSC). This new beamformer employs a set of P-regular M-band wavelet bases for the design of the blocking matrix of the GSC. We first carry out a general expression of the output SINR of the GSC with multiple interferers present. With this expression, we then examine the analysis of wavelet-based GSC by expressing the SINR in terms of parameters such as the regularity of wavelet filters, the number of bands of wavelet filters, the length of adaptive weights, and the input signal-to-noise ratio (SNR). Some simulation results verify the analytically predicted performance.

In this letter, we analyze blocking probabilities for prioritized multi-classes in optical burst switching (OBS) networks. The blocking probability of each traffic class can be analytically evaluated by means of class aggregation and iteration method. The analytic results are validated with results garnered from simulation tests.

In this letter, we propose the reducing method of feedback information for transmitting adaptable data rate in multi-user OFDMA/FDD system. In order to transmit the downlink channel information to Base-Station (BS) by using the limited uplink control channel, the proposed algorithm uses the channel variation level which describes the similarity among the adjacent clusters and uses just one modulation and coding scheme (MCS) level which represents the channel information of all clusters. The performance was investigated in one-cellular environment. It has a similar overhead for feedback information with conventional algorithm and has better performance than the conventional algorithm.

In this letter, we investigate the performance of using subband adaptive loading for the combination of orthogonal frequency division multiplexing (OFDM) and adaptive antenna array. The frequency-domain adaptive loading is very effective to deal with the frequency-selective fading which is inevitable in broadband wireless communications. However, almost all of the existing adaptive loading algorithms are based on "subcarrier-to-subcarrier" mode which may results in awfully large signaling overhead, since every subcarrier needs its own signaling loop between the transmitter and receiver. We investigate the performance of the combination of OFDM and adaptive antenna array when a subband adaptive loading algorithm is used to decrease the signaling overhead. It is shown by simulation results that at the cost of some tolerable performance loss, the signaling overhead of adaptive loading can be greatly reduced.

This paper presents a simplified maximum likelihood detection (MLD) scheme for multiple-input and multiple-output spatial division multiplexing (MIMO-SDM) systems. In the scheme, ordered successive detection (OSD) is applied to multiple symbol candidates retained in the preceding stage to limit the number of symbol vector candidates. Accordingly, the subsequent MLD searches for the most likely signal vector among the limited symbol-vector candidates. Simulation results demonstrated that the proposed scheme provides the bit error rate performance close to that achieved by MLD while reducing the computational complexity.

An empirical formula of resonant frequency of bow-tie microstrip antennas is presented, which is based on the cavity model of microstrip patch antennas. A procedure to design a bow-tie antenna using genetic algorithm (GA) in which we take the formula as a fitness function is also given. An optimized bow-tie antenna by genetic algorithm was constructed and measured. Numerical and experimental results are used to validate the formula and GA. The results are in good agreement.

In this paper, we propose a new fragile watermarking scheme that can be used to verify three-dimensional (3-D) polygonal models by detecting unauthorized alterations. In order to generate a binary watermark sequence, we combine a binary random sequence generated by the user key with a binary logo sequence using the bit-wise exclusive OR operation. The binary watermark sequence is embedded into 3-D triangle strips by perturbing their vertex coordinates. The proposed watermarking scheme can identify specific locations that have been changed. Changes of the vertex coordinates in the watermarked 3-D model are reflected in the extracted watermark image, which indicates the attacked area. Experimental results demonstrate that we can detect various modifications of the watermarked 3-D model: translation, rotation, scaling, randomization of points, polygon simplification, geometry compression, and local deformation.

We propose a feedback controller to efficiently control web requests especially on overloaded networks. The controller is designed based on a feedback closed loop that prevents overload of web server and enforces target CPU utilization via controlling the amount of input web requests. The main contribution of this letter is the use of feedback control theory to design the controller that delicately regulates web requests even under the dynamic changes in processing power of web server. In contrast with many previous heuristic methods, the proposed controller uses a systematic approach to adaptively control web requests taking account of the dynamic behavior of web server. Simulation results performed in overloaded cases show that the proposed controller lets web server effectively control input web requests and reach its CPU utilization to desired levels in relatively small settling times.