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This article is concerned with the consensus‐based distributed filtering problem for a class of general nonlinear systems over sensor networks with communication protocols. In order to avoid data collisions, the stochastic protocol and the Round‐Robin protocol are respectively introduced to schedule the data transmission between each node and its neighboring ones. A consensus‐based unscented Kalman filtering (UKF) algorithm is developed for the purpose of estimating the system states over sensor networks subject to communication protocols. Moreover, the exponential boundedness of estimation error in mean square is proved for the proposed algorithm. Finally, compared with the extended Kalman filtering, an experimental simulation example is provided to validate the effectiveness of the consensus‐based UKF algorithm. [ABSTRACT FROM AUTHOR]

In the networks, a transport layer is responsible for reliable data with guaranteed Quality of Service. A Modified‐QUIC protocol provides improved throughput and reduced latency of the network. This work proposes a modification in handshaking mechanism for QUIC protocol to minimize overhead due to control signals and time required to update the congestion‐window size. This modification fine‐tunes the window update mechanism with the acknowledgment frame, which results in a smooth variation in the congestion‐window size. This leads to the control congestion by regulating traffic in the network. In the proposed mechanism, unnecessary time out events are avoided by updating the congestion‐window on receipt of the acknowledgment frame. This work has been carried out using two different testbed setups to verify the transport layer and a browser network performance. It has been observed that a Modified‐QUIC protocol is easy to deploy, improves the throughput and data rate by 35% and 3.43%, respectively, over the QUIC protocol. The average fairness index is increased with a file size and for the long live traffic. [ABSTRACT FROM AUTHOR]

Summary: This paper investigates the strictly dissipative stabilization problem for multiple‐memory Markov jump systems with network communication protocol. Firstly, for reducing data transmission, we put forward a novel mode‐dependent event‐triggered communication scheme based on aperiodically sampled data. Secondly, a Markov jump system with general transition rates is considered to make the result more applicable, where the transition rates of some jumping modes allow to be completely known, or partially known, or even completely unknown. Thirdly, a less restrictive Lyapunov‐Krasovskii functional, which is only required to be positive definite at end points of each subinterval of the holding intervals, is first introduced for event‐triggered control issue. Based on the above methods, a sufficient condition with less conservatism is obtained to ensure the stochastic stability and dissipativity of the resulting closed‐loop system. Meanwhile, an explicit design method of the desired controller is achieved. Finally, two numerical examples are presented to demonstrate the effectiveness and advantage of the proposed method. [ABSTRACT FROM AUTHOR]

Compared with traditional radio frequency identification (RFID), computational RFID (CRFID) tag has more powerful computing capability, but it shows poor performance when it follows EPC Class-1 Generation-2 protocol especially transmitting large amounts of data. In fact, the operation of the CRFID tag entirely depends on the state of energy. For this point, this study proposes an optimised protocol called LILAC. LILAC allows tags to select the communication time slot according to their current voltage value measured by using an analogue-to-digital converter rather than randomly selecting, the authors proposed a more reasonable time slot mapping algorithm for LILAC that increases the success rate of tag responding. In addition, they design a data transmission format that can be retransmitted to improve the uplink throughput. Finally, they implemented LILAC on a CRFID platform and practically measured several parameters to compare with the existing well-known protocol. The results of experiments show that LILAC increases the maximum communication distance by more than half in the access phase and doubles the goodput of the backscatter link in both the inventory and access phase. [ABSTRACT FROM AUTHOR]

The growing popularity of multimedia applications and services needs to support several quality of service metrics such as high throughput, low energy, and jitter, which is a challenging task in mobile ad hoc networks. Because of limited bandwidth, energy constraints, dynamic topology, transmission errors, and fluctuating link stability, the links between adjacent nodes are often not reliable and may break because of node mobility. Link breakage initiates the process of rerouting either at the sender node (the node at which the link breaks) or at the source node. In either case, it leads to packet loss, delivery delays, and increased control overheads. Hence, to attain a minimum quality of service, routing protocols must address the dynamic network topology. Uncertain and varying movement of nodes necessitates stability of the links between such nodes. The objective of this paper is to propose 2 protocols, the first based on link stability and the other based on route stability. Link stability identifies a stable link from the available links to the next hop and determines a stable end-to-end route. The probability of successful transmission of periodic packets is used as a link stability metric to assess the stable path. Acknowledgment -free packets are used to check connectivity in the network. Increased probability of successful transmission implies that the selected link is sustained for longer duration and can deliver packets more reliably or, as a consequence, results in a stable link to deliver a better data rate. With a stable link, there is a reduced possibility of retransmissions, reduced end-to-end delay, reduced control overheads, and enhanced data delivery ratio. Selection of the most stable route for data transmission improves the performance. Experimental results from simulations performed on EXata/Cyber v2.0 simulator reveal that our proposed protocols are an improvement over the existing protocols in terms of packet delivery ratio, average end-to-end delay, and average route lifetime, even without route optimization with the minor increase in control packets. A case study of the application of proposed protocols is also presented. [ABSTRACT FROM AUTHOR]

SUMMARY The efficiency of medium access control strongly influences the performance of distributed wireless networks. Collisions in the channel and unused timeslots result in poor performance. On the other hand, quantum superdense coding proved that entanglement-assisted quantum communications can improve the data transmission rates compared with classical systems. It allows sending two classical bits from one party to another in exchange of one quantum bit and a pre-shared entangled Bell pair. This paper introduces a new protocol, which involves a modified version of superdense coding to medium access control of distributed communication systems. Using a pre-shared entangled Bell pair and one classical bit, 2.5 classical bits can be transmitted in average between the parties in each timeslot. This means not only valuable increase in capacity, but the two-way distributed operation opens new fields of investigation. Copyright © 2014 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

This paper addresses the issue of real-time data transmission in vehicles to roadside environment by the hybrid coordination function controlled channel access (HCCA) scheme with Institute of Electrical and Electronics Engineers 802.11p protocol. The HCCA is one of the medium access protocols specified in 802.11p standard, which uses a polling scheme similar to the point coordination function to provide a reliable QoS. In this paper, we improve the HCCA polling scheme and design a new data transmission scheduling method. We also prepared a mathematical model to systematically evaluate the performance of the HCCA in terms of average delay. Moreover, we analyze the factors that could impact the average delay, throughput, and packet loss rate as well. The simulation results are compared with the original scheme specified in the standard. The simulation results demonstrate that the proposed delay model could estimate the average delay of the HCCA with a high precision, whereas the proposal of the enhancement of the HCCA has lower delay, lower loss rate, and higher throughput than the one specified in the standard. Copyright © 2013 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The ground plane of a handset device plays an important role in the radiation process. As a consequence, through a ground plane modification, it is possible to obtain a multiband performance while keeping small antenna dimensions. Numerical simulation using Method of Moments has been used to understand the effect of the slots and a prototype has been built to validate the solution. The proposal consists in a multiband planar monopole handset antenna featuring a small area of 13 mm × 11 mm with a slotted ground plane operating in multiple communication standards (GSM850, GSM900, DCS, PCS, UMTS, LTE2300, WiMAX, Bluetooth, and Wi-Fi). Thanks to the presence of the slots, the bandwidth and efficiency results have been significantly improved. © 2012 Wiley Periodicals, Inc. Microwave Opt Technol Lett 55:173-180, 2013; View this article online at wileyonlinelibrary.com. DOI 10.1002/mop.27232 [ABSTRACT FROM AUTHOR]

SUMMARY In this study, an optimal method of clustering homogeneous wireless sensor networks using a multi-objective two-nested genetic algorithm is presented. The top level algorithm is a multi-objective genetic algorithm (GA) whose goal is to obtain clustering schemes in which the network lifetime is optimized for different delay values. The low level GA is used in each cluster in order to get the most efficient topology for data transmission from sensor nodes to the cluster head. The presented clustering method is not restrictive, whereas existing intelligent clustering methods impose certain conditions such as performing two-tiered clustering. A random deployed model is used to demonstrate the efficiency of the proposed algorithm. In addition, a comparison is made between the presented algorithm other GA-based clustering methods and the Low Energy Adaptive Clustering Hierarchy protocol. The results obtained indicate that using the proposed method, the network's lifetime would be extended much more than it would be when using the other methods. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

SUMMARY The IEEE 802.15.4 standard is designed to provide a low-power, low data rate protocol offering a high reliability. As an amendment to this standard, IEEE 802.15.4a introduces new options for physical layer to enable precision ranging. In this work, we analyzed the theoretical throughput and delay bounds of the unslotted version of chirp spread spectrum PHY-based 802.15.4a. The formulae for transmission between one sender and one receiver for an ideal channel with no transmission errors are given. The throughput and delay bounds are derived for different frequency bands and data rates. Additionally, to measure spectral utilization, we measured the bandwidth efficiency for both the standards. We also compared our results with IEEE 802.15.4. The comparative analysis concludes that the performance of 802.15.4a exceeds 802.15.4 in terms of throughput and delay. The analytical results of throughput are verified by computer simulations. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In this paper, we explore the advantages of network coding and space-time coding in improving the performance of two-way-relayed communications where two terminals absent of direct links exchange information through a single relay in between. Network coding allows embracing the interference from other terminals thereby turning it into a capacity boost. The application of space-time codes yields higher capacity by exploiting the spatial diversity. The joint performance of both techniques is studied in this paper. Specifically, we consider the class of decode-and-forward (DF) relaying strategy, evaluated in terms of symbol error rate using BPSK and QPSK modulations by both theoretical analysis and simulation. Based on our results, DF outperforms the amplify-and-decode and partial-decode-and-forward protocols. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

Cities worldwide have planned and implemented large-scale wireless mesh network (WMN) deployments. These mesh deployments are expected to provide broadband mobile access to the Internet at a low cost to the user. This paper considers the fairness problem affecting nodes in multihop WMNs. Many existing approaches to coping with the fairness problem are unsuitable because they necessitate modifications to the IEEE 802.11 MAC protocol or node queueing operations. Such modifications require a change in hardware/firmware at every node in the WMN. Thus, these approaches are less favorable from the points of view of interoperability and cost. Without modifying the lower layer protocols, this study identifies TCP parameters that impact throughput fairness and proposes an adjustment to these parameters to reduce frame collisions and to improve throughput fairness. Using simple mathematical formulations and ns2 simulations, this study shows that the frame transmissions from each node can be effectively controlled by properly controlling the delayed ACK timer and by using a suitable advertised window. The proposed method, in addition to fairness, requires fewer buffer resources than other methods. Moreover, it is not sensitive to the carrier sense range. It is also simple and easy to deploy. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The outage probability in a network of cooperative unmanned airborne vehicles (UAVs) over generalized fading channels is studied analytically using finite mixture with expectation maximization technique. A relay-based topology with one ground control unit (GCU) is considered, where the cooperative UAVs can communicate with the GCU directly or through relay. The application the UAV is assigned for specifies the minimum required transmission rate the UAV should achieve. The outage probability of the system is defined as the probability that either the transmission rate over any of the links drops below a predefined minimum threshold for that link or the Relay-GCU link is not able to transmit the aggregate data from all relayed UAVs and the minimum rate required by the relay UAV itself. Throughout the paper, expressions for the outage probability and the average achievable bit rate of a cooperative multi-carrier system are derived over generalized fading channels. Finite mixture with expectation maximization algorithm is utilized to derive a simple approximate expression for the probability density function (pdf) of the achievable bit rate assuming adaptive M-ary quadrature amplitude modulation (M-QAM). This pdf is used to derive closed-form expressions for the outage probability and the average bit rate. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In this paper, we depart from our previously proposed Non-Congestive Queuing (NCQ) mechanism and propose a new scheduling discipline that allows for efficient interoperation of dedicated systems (such as sensor and voice networks) with the Internet. More precisely, our approach provides delay guarantees to applications that do not contribute noticeably to congestion because of their tiny packet sizes and low transmission rates. In addition, our approach uses a second level of prioritization that conditionally favors, in terms of delay, applications with slightly longer packets as well. Based on the NCQ concept, Non-Congestive Queuing Plus (NCQ+) promotes applications that require comparatively small service times, as long as their total service times cause insignificant delays to other packets in the queue. Therefore, we prioritize packets, and in turn corresponding flows, according to their impact on total delay. We evaluate NCQ+ using ns-2-based experiments. Results show that whenever prioritization buffer resources are sufficient, NCQ+ can provide non-detectable delays to sensor applications and conditionally improve the performance of VoIP applications, while maintaining the impact on the performance of the other flows insignificant. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The nodes in a wireless ad hoc network act as routers in a self-configuring network without infrastructure. An application running on the nodes in the ad hoc network may require that intermediate nodes act as routers, receiving and forwarding data packets to other nodes to overcome the limitations of noise, router congestion and limited transmission power. In existing routing protocols, the 'self-configuring' aspects of network construction have generally been limited to the construction of routes that minimize the number of intermediate nodes on a route while ignoring the effects that the resulting traffic has on the overall communication capacity of the network. This paper presents a context-aware routing metric that factors the effects of environmental noise and router congestion into a single time-based metric, and further presents a new cross-layer routing protocol, called Warp-5 (Wireless Adaptive Routing Protocol, Version 5), that uses the new metric to make better routing decisions in heterogeneous network systems. Simulation results for Warp-5 are presented and compared to the existing, well-known AODV ( Ad hoc On-Demand Distance Vector) routing protocol and the reinforcement-learning based routing protocol, Q-routing. The results show Warp-5 to be superior to shortest path routing protocols and Q-routing for preventing router congestion and packet loss due to noise. Copyright © 2011 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

High-speed communication networks, such as the Internet, have recently seen rapid development. This has facilitated the construction of control systems that employ these networks. However, these systems have two major problems: (i) bit rate limitation, and (ii) network unreliability, such as delays, data dropouts, etc. In this study, dynamic quantization is used to overcome the first problem. For the second problem, data-driven zero-order hold is used to suppress the effect of stochastic data dropout through an unreliable network. The stochastic stability of the system is analyzed and the sufficient condition for stability is derived in this study. Finally, the proposed method is validated through some experiments. © 2011 Wiley Periodicals, Inc. Electron Comm Jpn, 94(2): 1-8, 2011; Published online in Wiley Online Library (). DOI 10.1002/ecj.10329 [ABSTRACT FROM AUTHOR]

Precoding techniques can be introduced into multi-relay systems due to the similarity between cooperative communication systems and traditional multi-input-multi-output (MIMO) systems. In this paper, a channel state information (CSI) feedback scheme based on the zero-forcing (ZF) relaying protocol is proposed at first, where the information of relaying channel and noise related to each relay node can be compressed into two positive real parameters. Then, based on the proposed feedback scheme, the singular-vector-based local temporal precoder is presented at the source node through two continuous transmitted vectors, which is termed as distributed spatial-temporal precoding (DSTP). Moreover, various spatial data rates can be conveniently supported by DSTP. Based on the analysis on DSTP, it is better that the number of data streams is not larger than the number of antennas equipped at the source node. The unitary DSTP with the proposed feedback scheme outperforms not only the close-loop direct transmission but also the simple ZF relaying method. Copyright © 2010 John Wiley & Sons, Ltd. A CSI feedback scheme based on the ZF relaying protocol is proposed in the MIMO multi-relay system to support the singular-vector-based temporal precoder at the source node which is termed as DSTP. The theoretical analysis and simulation results demonstrate the number of data streams supported by DSTP and its advantage compared with the close-loop direct transmission and the simple ZF relaying method. Copyright © 2010 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In a next generation network, the IPv6-enabled IP multimedia subsystem (IMS) network may connect to an IPv4 network. When an IPv4/IPv6 dual-stack user equipment (UE) initiates a call by sending an IPv6 SIP INVITE message to an IPv4-only user agent (UA), the call cannot be established correctly. To resolve this problem, the IMS-application layer gateway solution, the redirect solution, and the interactive connectivity establishment solution have been proposed. In this paper, we propose an effective solution where only the IPv6 INVITE message is translated into an IPv4 INVITE message. Upon receipt of the IPv4 200 OK message replied from the IPv4-only UA, the dual-stack UE learns that the correspondent UA supports IPv4-only and utilizes IPv4 instead of IPv6 to send the subsequent SIP messages and real-time transport protocol (RTP) packets. The proposed solution is compared with the existing solutions in terms of network node modification, call setup complexity, and RTP transmission latency. Our study indicates that the proposed solution outperforms the other three solutions in the call setup and the RTP transmission. Copyright © 2009 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

RTP (Real-time Transport Protocol) has been designed for efficient transmission of real-time multimedia data on the Internet by Internet Engineering Task Force (IETF). Different applications tend to implement separate RTPs as a library. In order to resolve this problem, embeddedRTP was proposed as an RTP-based kernel-level protocol. This paper proposes the ExtendedERTP protocol based on the embeddedRTP protocol. This new protocol deals with a couple of issues, such as packet processing overhead and buffer requirement, and combines its Application Programming Interface (API) with Berkeley Software Distribution (BSD) socket's which have been widely utilized in network applications. This paper demonstrates that this integration makes it possible to transmit real-time multimedia data through the existing interface of BSD socket API with nominal extra overhead. This paper also shows that the time and memory requirement for packet processing can be reduced by about 15–20% and by 3.5%, respectively, compared with those of embeddedRTP. Copyright © 2010 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In the mobile Stream Control Transmission Protocol (mSCTP) for vertical handover, a mobile user may suffer from performance degradation due to the problems of packet reordering and retransmission timeout and due to the packet loss during handover. To solve these problems, we propose a new scheme of handover retransmission for mSCTP handover, in which the correspondent node retransmits the outstanding data packets to the mobile node over the new primary path. From simulation results, it is shown that the proposed scheme can avoid the packet reordering and retransmission timeout problems during handover. Moreover, we can see that the proposed scheme can significantly improve throughput of mSCTP handover, compared with the existing schemes. Copyright © 2009 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The lifetime of a network is the key design factor of mobile ad hoc networks (MANETs). To prolong the lifetime of MANETs, one is forced to attain a tradeoff of minimizing the energy consumption and load balancing. In MANETs, energy waste resulting from retransmission due to high bit error rate (BER) and high frame error rate (FER) of wireless channel is significant. In this paper, we propose two novel protocols termed multi-threshold routing protocol (MTRP) and enhanced multi-threshold routing protocol (EMTRP). MTRP divides the total energy of a wireless node into multiple ranges. The lower bound of each range corresponds to a threshold. The protocol iterates from the highest threshold to the lowest one and chooses those routes with bottleneck energy being larger than the current threshold during each iteration. This approach thus avoids overusing certain routes and achieves load balancing. If multiple routes satisfy the threshold constraint, MTRP selects a route with the smallest hop count to further attain energy efficiency. Based on MTRP, EMTRP further takes channel condition into consideration and selects routes with better channel condition and consequently reduces the number of retransmissions and saves energy. We analyze the average loss probability (ALP) of the uniform error model and Gilbert error model and give a distributed algorithm to obtain the maximal ALP along a route. Descriptions of MTRP and EMTRP are given in pseudocode form. Simulation results demonstrate that our proposed EMTRP outperforms the representative protocol CMMBCR in terms of total energy consumption and load balancing. Copyright © 2008 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In this paper we propose a new medium access mechanism for HomePNAv2 and HomePNAv3 standards. This mechanism uses a priority aggregation mechanism to avoid collisions, increasing these protocols throughput. Furthermore, this mechanism does not require modifying standards, since it can be implemented over their actual MAC sublayers. Simulation results show that the proposed mechanism is able to increase the throughput up to 44% for HomePNAv2 and 36% for HomePNAv3. Moreover, we also show how this mechanism affects network delay, analyzing average delay and jitter for all protocols presented. Copyright © 2008 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

To provide a multicasting service, several multicast protocols for mobile hosts (MHs) have been proposed. However, all of these protocols have faults, such as non-optimal delivery routes and data loss when hosts move to another network, resulting in insecure multicast data transmissions. Thus, this paper presents a new reliable and efficient multicast routing protocol for mobile IP networks. The proposed protocol provides a reliable multicast transmission by compensating the data loss from the previous mobile agent when a MH moves to another network. In addition, an additional function allows for direct connection to the multicast tree according to the status of agents, thereby providing a more efficient and optimal multicast path. The performance of the proposed protocol is confirmed based on simulations under various conditions. Copyright © 2008 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

We investigate the behaviour of TCP(α, β) protocols in the presence of wireless networks. We seek an answer to strategic issues of maximizing energy and bandwidth exploitation, without damaging the dynamics of multiple-flow equilibrium. We take a fresh perspective on protocol design: What is the return of the effort that a protocol expends? Can we achieve more gains with less effort? We study first the design assumptions of TCP(α, β) protocols and discuss the impact of equation-based modulation of α and β on protocol efficiency. We introduce two new measures to capture protocol behaviour: the ‘Extra Energy Expenditure’ and the ‘Unexploited Available Resource Index’. We confirm experimentally that, in general, smoothness and responsiveness constitute a tradeoff; however, we show that this tradeoff does not graft its dynamics into a conservative/aggressive behaviour, as it is traditionally believed. We uncover patterns of unjustified tactics; our results suggest that an adaptive congestion control algorithm is needed to integrate the dynamics of heterogeneous networks into protocol behaviour. Copyright © 2007 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

Packet reordering is an inevitable phenomenon on the Internet. An ideal metric for packet reordering should capture reordering accurately, provide insight into the nature of reordering, and help in the evaluation and analysis of reordering leading to mitigation of its adverse effects. Proposed metrics for packet reordering, namely, reorder density, reorder buffer-occupancy density, reordering extent, and n-reordering, overcome to various degrees the deficiencies of percentage reordering in capturing the nature and extent of reordering. These metrics vary widely in areas such as evaluation complexity, measurement technique, usage, and in the definition used for packet reordering itself. Metrics for reordering are evaluated using a framework consisting of a set of both essential and desirable attributes of reorder metrics. The attributes include the ability to capture reordering, sensitivity to lost and duplicate packets in reorder measurements, usefulness, simplicity, and evaluation complexity. Finally, the characterization of packet reordering using these metrics is discussed, using sets of measurements carried over the Internet. Copyright © 2007 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In this paper, we improve On-Demand Multicast Routing Protocol (ODMRP), which is a multicast routing protocol for Mobile Ad hoc Network (MANET), and propose a new protocol decreasing the number of data forwarding without deteriorating communication reliability. The proposed protocol avoids unnecessary data forwarding by distinguishing each multicast source at every Forwarding Group (FG) Node used in ODMRP. Moreover, the proposed protocol utilizes Support Group (SG), which also distinguishes each multicast source, in order to retain substitute routes without any redundant data forwarding. Therefore, the proposed protocol can avoid deteriorating communication reliability. We evaluate the performance of ODMRP, ODMRP with SG, and the proposed protocol, and we show that the proposed protocol can decrease the number of data forwarding compared with the conventional protocols. We also show that the proposed protocol can restrain the deterioration of packet delivery ratio within allowable range except the case where the number of multicast sources and receivers is extremely large. Furthermore, we show that the proposed protocol can mitigate the deterioration of packet delivery ratio caused by traffic congestion in the case where the number of multicast sources is large. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 90(11): 36–47, 2007; Published online in Wiley InterScience (www. interscience.wiley.com). DOI 10.1002/ecja.20382 [ABSTRACT FROM AUTHOR]

Circuit emulation service (CES) allows time-division multiplexing (TDM) services (T1/E1 and T3/E3 circuits) to be transparently extended across a packet network. With circuit emulation over IP, for instance, TDM data received from an external device at the edge of an IP network is converted to IP packets, sent through the IP network, passed out of the IP network to its destination, and reassembled into TDM bit stream. Clock synchronization is very important for CES. This paper presents a clock synchronization scheme based on a double exponential filtering technique and a linear process model. The linear process model is used to describe the behaviour of clock synchronization errors between a transmitter and a receiver. In the clock synchronization scheme, the transmitter periodically sends explicit time indications or timestamps to a receiver to enable the receiver to synchronize its local clock to the transmitter's clock. A phase-locked loop (PLL) at the receiver processes the transmitted timestamps to generate timing signal for the receiver. The PLL has a simple implementation and provides both fast responsiveness (i.e. fast acquisition of transmitter frequency at a receiver) and significant jitter reduction in the locked state. Copyright © 2006 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

We propose an extension of the classical CSMA/CD protocol that eliminates its three main drawbacks. The new protocol, called dual-mode CSMA/CD (CSMA/DM), operates in two modes, light mode and heavy mode. The light mode of CSMA/DM is almost the same as the original CSMA/CD protocol and is primarily used when the LAN load is light. The heavy mode is a collision free mode and is applied when the LAN load is heavy. The proposed modification to CSMA/CD is minimal while performance gain is significant. CSMA/DM automatically switches between its two modes based on the observed LAN load. Under heavy mode, the monitor station in a CSMA/DM LAN allocates bandwidth to those busy stations without collisions. The newly added priority scheme in the heavy mode allows stations to reserve bandwidth for their high priority frames. Compared with the current collision-free fast Ethernet and Gigabit Ethernet, CSMA/DM has the advantage of not needing any extra switches or hubs while maintaining comparable performance. Its priority scheme provides more flexibility on bandwidth distribution than in fast Ethernet and Gigabit Ethernet. It can also be easily adapted for high-speed wireless LANs. More importantly, CSMA/DM should be a good alternative of the widely used CSMA/CA (collision avoidance) in mobile ad hoc networks (MANET) and sensor networks. Copyright © 2005 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In Mobile IP networks, the home agent manages the destination information of all mobile terminals. It receives as proxy the packets destined for the mobile terminal and transfers them to the new location of the terminal. When the number of terminals in the network is increased, the home agent must manage a large amount of information, which increases the retrieval processing load in packet transfer. It is possible that a packet may be lost by an overloaded home agent. This paper proposes a method of eliminating the retrieval processing of the home agent in packet transfer by utilizing ARP, and of reducing the load of the whole packet transfer process. It is shown experimentally that even if sufficient mobile terminal entries to completely fill the location information list for 200,000 entries are registered, the home agent can transfer all packets without any packet loss in Fast Ethernet. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(6): 32–40, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20249 [ABSTRACT FROM AUTHOR]

We propose a novel explicit rate-based congestion avoidance scheme. The scheme is similar to ATM available bit rate (ABR) services but its complexity is largely reduced to enable implementation in IP networks. In our proposed scheme, sources can adapt their sending rate according to network status. Adaptation of sending rates converges to max–min fairness with minimal rate guarantee. Furthermore, routers do not maintain per flow state; they use FIFO packet scheduling enhanced by an explicit rate feedback mechanism with estimations, including number of active flows, number of bounded flows and total bounded rate, based on a simple algorithm motivated by Bloom filter. We present and discuss simulations on the performance under various network conditions. Copyright © 2005 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

FRN (flexible radio network) is an ad hoc network product that can realize a wide-area data collection network. This paper discusses a performance improvement method when TCP is applied to the transport layer in FRN. It is shown that even when the terminal is not moving as in FRN, the throughput can be improved in radio link failure due to noise or obstacles by combining ELFN (explicit link failure notification), which is proposed as a method for TCP performance improvement in ad hoc networks, with routing information. The proposed method is evaluated by simulation in this paper. A further objective is to reduce the performance degradation due to packet collision. The delayed ACK and the ACK retransmission with priority in the case of packet collisions are analyzed and the effectiveness of the method is demonstrated. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(11): 99–107, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10134 [ABSTRACT FROM AUTHOR]

Recent research has shown that long-range dependence is found not only in network traffic volume, but also in the propagation delay time of packets. The round-trip propagation time of packets, which is known as the round-trip time (RTT), is one of the most important parameters for determining TCP performance. The ability to understand its characteristics can contribute to an improvement in TCP performance. Therefore, in this paper, to determine the impact of the long-range dependence of the RTT on TCP performance, the authors looked at the two congestion control phases known as congestion avoidance and slow start in TCP window control and used computer simulations to evaluate performance. First, the evaluation related to the impact on the congestion avoidance phase showed that higher long-range dependence and larger transfer files caused greater variability in the file transfer completion time. Next, the evaluation related to the impact on the slow start phase, which evaluated the impact of the long-range dependence of the RTT on the TCP retransmission timeout (RTO), showed that higher long-range dependence causes unnecessary timeouts, which end up causing the resending of packets that have not actually been lost, to occur more frequently. The authors also performed investigations for mitigating the impact of the long-range dependence of the RTT and made a number of modifications to the existing RTO estimation algorithm to show that TCP performance can be improved. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(6): 52–60, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10073 [ABSTRACT FROM AUTHOR]

Ad hoc networks have excellent capabilities for data transmission in an environment with an incomplete communications infrastructure, and in mobility and portability of terminals. Routing protocols between terminals in an ad hoc network can be broadly divided into location-aware routing, based on location information on the terminals, and non-location-aware routing, which is not based on location information. In the former, routing can be made accurate and easy by effectively utilizing location information. Hitherto, the use of the global positioning system (GPS) has been most often treated as the means of acquiring terminal location information. However, in location estimation by GPS, electromagnetic waves from multiple location estimation satellites must be received, and the method cannot be applied effectively in an environment in which that condition is not satisfied. This paper proposes the following method of location estimation. Fixed access points whose absolute locations are already known are placed in the network. Then, by using self-organizing maps (SOM), the mutual location information among the terminals is organized, and the location of the home terminal is successively inferred. The feasibility and accuracy of estimation in the proposed method are examined by a computer simulation. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(4): 73–81, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10098 [ABSTRACT FROM AUTHOR]

When bulk data is transferred between end hosts over an IP network, a known problem is the limited throughput obtained at the application level due to the memory copy processing load between the kernel buffer and the user buffer depending on the operating system. To solve this problem, in this paper we propose a high-speed bulk data transfer architecture called Network Direct Memory Access (Network DMA) and describe its design, implementation, and test results. The purpose of Network DMA is to be a transport function installed on a network interface card (NIC) and to achieve high-speed and reliable data transfers over a network between the user buffers of the end hosts without requiring operating system-specific processing. An application will use the data transfer function provided by a new interface capable of directly specifying the send destination host and the target transfer buffer. Based on the proposed architecture, we developed a 622-Mbit/s ATM NIC and Gigabit Ethernet NIC incorporating the Network DMA. The ATM NIC we developed achieved a continuous transfer throughput of 540 Mbit/s (99% bandwidth usage efficiency at the application level) in data transfer tests between general-purpose PCs. The Gigabit Ethernet NIC achieved an 800 Mbit/s continuous transfer throughput. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(11): 57–65, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10107 [ABSTRACT FROM AUTHOR]

In this paper, we present a detailed description of the Reliable Multicast using Adaptive Caching (RMAC) system and protocol, and describe its performance evaluation. From the idea that support based on functional processing in the network is essential to efficiently perform reliable multicast, routers that process the RMAC protocol are introduced by RMAC. RMAC can improve retransmission efficiency by packet caching and retransmission at the routers. The required cache size can be decreased by distributing the caching among the routers to adapt to the congestion. The performance evaluation by a simulation showed that RMAC has lower recovery latency compared to conventional techniques and improved scalability with respect to the number of receivers. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(10): 89–99, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1179 [ABSTRACT FROM AUTHOR]

In this paper, we describe a communication control mechanism that can dynamically adapt to the quality changes of a communication channel. Specifically, we propose a communication control mechanism that adapts to the quality changes of the communication channel. To implement this mechanism, we model the communication flow and derive equations for the communication time and the optimum window size that minimizes the communication time. The problems of existing communication control protocols are explained, and a new communication control protocol that solves these problems is presented. Furthermore, to implement a communication control mechanism employing the optimum window size, we propose a simple equation for the optimum window size and demonstrate its effectiveness by evaluating the implementation of the communication control mechanism. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 85(11): 1–12, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10015 [ABSTRACT FROM AUTHOR]

There have been many proposals for media access control protocols in packet radio networks (PRN) in which transmission efficiency is improved by using a directional antenna. This paper assumes that the topology of the PRN and the data to be transmitted among the stations are specified, and discusses the essential effectiveness of using the directional antenna, as independent of the protocol, using the shortest time needed to complete all data transmissions as a measure. It is shown that the number of directional antennas in each station is sufficient if it is equal to the number of neighboring stations, and that the effectiveness of the directional antenna is more pronounced when the PRN has a high density. The effectiveness when the communication buffer used to store the data to be sent or received is combined with the directional antenna is also discussed. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 85(9): 51–58, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1125 [ABSTRACT FROM AUTHOR]

TCP error control mechanism lacks the ability to detect with precision the nature of potential errors during communication. It is only capable of detecting the results of the errors, namely that segments are dropped. As a result, the protocol lacks the ability to implement an appropriate error recovery strategy cognizant of current network conditions and responsive to the distinctive error characteristics of the communication channel. TCP sender always calls for the sending window to shrink. We show that probing mechanisms could enhance the error detection capabilities of the protocol. TCP could then flexibly adjust its window in a manner that permits the available bandwidth to be exploited without violating the requirements of stability, efficiency and fairness that need to be guaranteed during congestion. Our experiments have three distinct goals: First, to demonstrate the potential contribution of probing mechanisms. A simple probing mechanism and an immediate recovery strategy are grafted into TCP-Tahoe and TCP-Reno. We show that, this way, standard TCP can improve its performance without requiring any further change. Second, to study the performance of adaptive strategies. An adaptive TCP with probing is used, that is responsive to the detected error conditions by alternating slow start, fast recovery and immediate recovery. An adaptive error recovery strategy can yield better performance. Third, to study the design limitations of the probing device itself. The aggressive or conservative nature of the probing mechanisms themselves can determine the aggressive or conservative behaviour of the protocol and exploit accordingly the energy/throughput trade-off. Copyright © 2002 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In this paper, we propose a new method for fast and efficient detection of communication protocol errors with less memory requirements and processing time when there exists an excess/deficiency of definition or deadlock in signal transmission/reception. Practically, a new method of protocol validation is presented where the protocol errors are detected by dividing the protocol inspection into two stages: one is inspection centered on the operations of each process and the second is inspection of all process operations at the same time. The characteristic of this method of validation is that in the former stage the excess/deficiency of transmission/reception is detected without enumerating the states of the whole system and in the second stage, deadlock is detected by enumerating the minimum number of system states. As a result, when the number of processes is large, the number of system states to be enumerated, can be greatly decreased as compared to the conventional methods. To evaluate the proposed method of protocol validation we present application results of various protocol examples. We show that memory requirements and processing time are greatly reduced and large-scale protocols with large number of processes which could not be fully inspected by the conventional methods, can also be validated completely. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 83(7): 26–39, 2000 [ABSTRACT FROM AUTHOR]

Local am network (LAN) is used widely at pres- eat, and the radio LAN also is utilized worldwide in practice. This paper focuses on the performance evaluation of the radio LAN in actual operation. It is shown that the method based (Al simple network management protocol (SNMP) is useful. [ABSTRACT FROM AUTHOR]

In ATM networks, CLAD functions convert the information source signal to cells and disassemble the ATM cells to yield the original signal. This paper implements a CLAD utilizing synchronous residual time stamp (SRTS) from the one in the method based upon AAL type 1. That transmits signals whose timing is independent of that of the network. It is clarified that the SRTS method generates low-frequency jitter of the same order as the stuff multiplexing method. The SRTS jitter of the prototype CLAD is measured. A high-quality NTSC motion picture is transmitted as DS3 (44.736 Mbits/s) coded signals. Furthermore, communication between two FDDI LANs is tested using DS3 data routers. These tests confirm that the CLAD works well.

This paper evaluates the performance of a fast select connection oriented (FS-CO) logical link control (LLC) protocol in slotted ring networks. The FS-CO-LLC protocol is one of light-weight protocols. Is employs a method of implicit connection establishment and release to reduce overhead of connection control. This paper analyzes the performance (throughput and average response time) of a slotted ring network in which a single slot circulates by means of Equilibrium Point Analysis. The performance in the case of plural slots is evaluated by simulation. This paper also compares the FS-CO-LLC protocol with connection-oriented (CO) and connection-less (CL) LLC protocols. Numerical results show that the performance of FS-CO-LLC and CL-LLC protocols is superior to that of CO-LLC protocol when the average message length is smaller than the buffer size. On the other hand, when the average message length is larger than the buffer size, CO-LLC protocol has the best performance. It is also shown that the performance of FS-CO-LLC protocol does not degrade but CL-LLC protocol does, even if the average message length becomes larger. [ABSTRACT FROM AUTHOR]

A signal lug functional model (separated model) that assumes the functional separation of call control (CC) and bearer control (BC) must be considered for B-ISDN because B-ISDN deals with various levels of service quality, various speeds, and various connection configurations. This paper provides the basic structure of the forementioned separated model and the functions of the CC and BC based on actual B-ISDN service specifications. Also, system design guidelines are proposed for connection-type bandwidth selection and interworking between broadband ISDN and narrowband ISDN, which are key issues in the study of B-ISDN signaling protocol. [ABSTRACT FROM AUTHOR]

Pipelined protocol processors are used in high-speed networks such as FDDI for achieving high-throughput data communications between end users. In this paper, we present a model for computing the maximum throughput that can be delivered by such processors. The parameters of the model are: input packet length, length of PDU (Protocol Data Unit) and SDU (Service Data Unit) headers, packet processing time of pipeline stages, capacity of the FIFO channels that inter-connect stages, and average number of packets assembled/segmented at a stage. As an example, the maximum throughput of an idealized pipelined processor that executes the FDDI, LLC Class I, IP, and TCP protocols is computed. Also, we show that the formulas given in [1] and [2] are particular cases of the formulas presented in this paper. In a typical application of the proposed model, some desired, or measured, throughput characteristics of a protocol processor will be given, and the parameters of the processor will be computed. [ABSTRACT FROM AUTHOR]

This paper presents a BCMP-type closed queuing network which models the error recovery defined as HDLC based protocols such as LAPB, LAPD, and LLC type II. Throughput and average end-to-end delay against the I-frame loss probability owing to bit errors are obtained by using the queuing network. With the I-frame length independence assumption, agreement between analytical and simulation results is shown to be fairly good. [ABSTRACT FROM AUTHOR]

With the increased use of personal computers (PCs) in communication environments have come the expectation of reliable communication protocols. This paper describes the implementation details of two very similar packet communication protocol interfaces on PCs. One is the X.25 interface for packet-switched public data networks, and the other is the X.32 interface for public-switched telephone networks. The implementation of these protocols is performed with consideration of the future introduction of OSI higher-layer protocols. This paper describes also that the addition of the assigning mechanism of level 2 address to the Rec. X.32 enables powerful personal computer communication through public-switched telephone networks. Furthermore, the results of the performance evaluation are described which show the high throughput of more than 90 percent on various speeds of the physical circuit. [ABSTRACT FROM AUTHOR]

This paper discusses a simple go-back-N- ARQ scheme in which the satellite is provided not with the repeater function but also with the processing function. The satellite detects the error in the data block sent from the transmitter. It returns ACK/NACk to the transmitter and relays the correctly received date block to the receiver. The receiver detects the error in the received data block and returns ACK/NACk to the transmitter. The transmitter examines two acknowledgement signals returned from the satellite and the receiver, and transmits the data block based on the basic go-back-N ARQ protocol. This paper analyzes the scheme in which the hardware complexity of the satellite is minimized. It is shown that the throughput can be increased and the average transmission delay can be decreased. [ABSTRACT FROM AUTHOR]

This paper considers the logical link control (LLC) sublayer in the CSNAICD local area network, and compares analytically the performance of connection-oriented (CO) and connectionless (CL) protocols. As the medium access protocol, slotted nonpersistent CSMA/CD with a priority assigned to some types of frames is employed. In comparison, the overhead for connection establishment/ release in the CO-LLC protocol, and the receiver-buffer overflow in the CL-LLC protocol are especially considered, and the two kinds of protocols are compared. In the CO-LLC protocol, a connection is established corresponding to a message. Employing the technique of equilibrium point analysis, the throughput and the average response time of the two kinds of MAC protocols are derived. To compare the performance of the two kinds of protocols, the receiver-buffer size is kept constant, and the average number of data frames in a message is varied. As a result, it is shown that the CL-LLC protocol is better than the CO-LLC protocol when the average number of data frames is sufficiently less than the receiver-buffer size, but the order is reversed when the average number of data frames is increased. [ABSTRACT FROM AUTHOR]

For the OSI message exchange protocol, the MRS (message handling system) was recommended in 1984 by CCITT. The implementation of this protocol is being considered by net- work service enterprises and computer manufacturers in various countries. MRS is constructed as an application of OSI common application protocol. ROS (remote operations service) and RTS (reliable transfer service), specified by MHS, and performing the acknowledgement of send/receive of application data and the interruption/recovery, can also be used in common with other applications of OSI. This paper presents an analysis and evaluation of ROS and RTS from the viewpoint of the number of transmitted/ received packets and the transfer time. The relation between the window size and the transfer efficiency in RTS is derived. For the system combining ROS and RTS, the method for controlling the window size corresponding to the response time, as well as the method for setting the session data length corresponding to the interruption rate, are presented. As to the method of combining ROS and RTS, it is shown that a large amount of application data can be transferred in a stable way, even if the interruption rate is large, by applying RTS to ROS. [ABSTRACT FROM AUTHOR]

The shared media local area networks (LAN) such as Ethernet/FDDI has the bottleneck of bandwidth scalability. ATMLAN based on asynchronous transfer mode (ATM) switch is considered interesting at present as the next-generation multimedia LAN to dissolve this bottleneck. The research and development of ATM has mostly been promoted for application to the public network. With respect to LAN, a new investigation is needed which includes the architecture since there are various requirements such as compatibility to the existing protocol, the simplification of the network control, and the drastic reduction of the system cost. This paper proposes the ATM-oriented multimedia LAN (ATOMLAN) architecture which meets the forementioned requirements. The proposed architecture has the following three features: (I) the ATMLAN addressing, considering the affinity to the existing protocol and the efficient internetworking; (2) the high-speed connection control to realize a high throughput for the connectionless-type data transfer; and (3) the simple and efficient traffic control for the high-speed burst-multimedia service. Using the proposed architecture, a seamless high-speed multimedia network can be realized, from the desk-top to the broadband network. [ABSTRACT FROM AUTHOR]