Underwater wireless optical communication (UWOC) enables high-rate data offloading for underwater sensing systems, but its strong directionality makes multi-node networking vulnerable to misalignment, occlusion, and dynamic link disruptions. Existing control-plane-driven recovery is often too slow for such transient failures. We present A-SCAN, a data-plane self-healing mechanism that maintains neighbor-angle mappings and performs lightweight angle-guided recovery without triggering global routing updates. Based on the locally recovered topology, Q-SHARP performs quality-aware multi-hop path selection and backup optimization in the control plane. Together, they separate fast local link recovery from slow global routing optimization, enabling more stable self-healing communication in directional UWOC networks.

For multi-transmission rate environments, access point (AP) connection methods have been proposed for maximizing system throughput, which is the throughput of an entire system, on the basis of the cooperative behavior of users. These methods derive optimal positions for the cooperative behavior of users, which means that new users move to improve the system throughput when connecting to an AP. Typically, the arrival of two or more new users causes signal interference for other users. Specifically, traditional methods have not considered interference and the number of allowable mobile positions is so large that it is difficult to find the optimal position to maximize system throughput. In this paper, we propose an AP connection method that maximizes system throughput by considering interference between users, with a potential game model. This game can find at least one optimal user position. Moreover, numerical analyses show the effectiveness of our method.

This paper proposes an optimal dynamic transmission power allocation mechanism for multiple users to maximize mobile ad hoc networks (MANETs) energy efficiency. With the widespread adoption of the internet of things (IoT), energy efficiency is a crucial metric in MANETs constrained by the battery capacity of portable devices. Specifically, a practical scheme needs to be explored that ensures the quality of service (QoS) while minimizing battery costs. Previous studies mainly focus on designing power-manageable components and achieving predictive mechanisms at the circuit or system layer. In contrast to the traditional approaches, we jointly consider power management and multiuser cooperative mobility for energy efficiency maximization. On the one hand, we formulate the dynamic power allocation schemes with multiuser cooperative mobility strategies into the infinite-horizon time average problems. On the other hand, an algorithm based on Lyapunov optimization is developed to obtain the optimum, i.e., dynamic power allocation based on the multiuser cooperative mobility (DPA-MCM) algorithm. Numerical simulations show that the proposed algorithm improves the energy efficiency by up to 27.84 % compared to the previous method, demonstrating the effectiveness.

This paper proposes a retransmission strategy using the non-orthogonal multiple access (NOMA) in the PC5-based cellular-V2X mode 4 (called mode 4) to mitigate the channel congestion. The proposed strategy has two key ideas. The first idea is to allow each node (e.g., a car or a pedestrian) to retransmit its packet at lower transmission power than its original packet in the first transmission. The second idea is that each node aims to synchronize its low-powered retransmission with the first transmission of its neighbor node. The two key ideas boost the PRR through diversity effects by mitigating interference and providing power differences with widely deployed receivers. Our simulations highlighted that the proposed strategy outperformed the existing NOMA with retransmissions in mode 4 by 29%.

Three algorithms for multiuser mobility method are evaluated in terms of performance to derive the most suitable algorithms for different practical application scenarios. Traditional multiuser mobility methods focus on improving throughput performance only without considering algorithm execution overhead and complexity. In contrast to previous studies, this paper develops an analytical tradeoff framework for the throughput and algorithmic complexity evaluation based on multiuser cooperative mobility in Mobile Ad Hoc Networks (MANETs). On the one hand, we quantitatively derive and model the optimal strategy for optimization throughput. On the other hand, we jointly evaluate the complexity performance of proposed algorithms, which are demonstrated by the discrepancies in the number of users, movable range, and location distribution. Numerical simulations show that the game-theoretic algorithm provides 16.85% throughput gain with high user volume, the bound-algorithm offers 17.22% throughput gain at large movement ranges and the graph-algorithm can reduce 29.26% complexity, indicating the effectiveness of our proposed framework.

As the Internet of Things (IoT) has become a widespread phenomenon, promising sensor applications at the nanoscale have begun to emerge. One example is imaging via distributed massive nanoscale nodes (NSNs), which can be used to implement “invisible surveillance cameras” by, for example, painting liquids containing nanoscale sensors onto walls. This imaging method requires periodic data transfer from thousands of NSNs to a data collection node (DCN). An essential technique for handling such transfers is the media access control (MAC) protocol. However, existing protocols cannot support periodic transfer from numerous NSNs because of inefficient communication caused by a large amount of headers in the packets. In this article, we introduce an original MAC protocol and discuss its capability in terms of implementing imaging applications. The proposed protocol applies a time-division access (TDA) feature to reduce the amount of headers and a two-layered protocol to enable simultaneous transmission among nodes. Slot assignment is an essential function in TDA and requires communication among nodes. Unlike existing methods, our simple approach enables communication by exploiting the unique features of the focused application. The results of the numerical simulation reveal that the proposed MAC protocol allows for periodic imaging with more than three thousand nodes and produces high-quality images very close to those obtained using ideal communication. These results are achieved by employing an original design framework to determine appropriate key parameters, such as the number of clusters and frame rate.

The recent development of mobile sensing nodes, such as unmanned vehicles, has enabled novel sensing applications. In these applications, the nodes periodically send data required for control, such as the position and velocity as well as data sensed data while moving. Prioritized communication is preferable for securing reliable data transfer for control. However, existing methods used for mobile sensor networks do not offer prioritized periodical communication. Herein, we introduce prioritized periodical communication with timing-reservation multiple access. In this method, response frames are extended to communicate the priority level of the transmitted frame. Numerical results show that high-priority data can be transmitted more accurately using the proposed method compared with existing methods, even in a heavy traffic scenario.

In this paper, we propose a user cooperative mobility method for the high throughput of a specific target flow in dense multiple ad-hoc networks with multiple flows. Our basic idea is that the moving of a node in an ad-hoc network could result in improved performance, such as increased throughput, by avoiding interference and/or useless transmission to the bottleneck node. Distributing Transmission opportunity of Extra flow (DTE) method extends the previously proposed Indirect Method, which accelerates the throughput of a target flow by moving a node close to a source node in a single flow. The DTE method can apply both single and multiple target flows. We analyze which flow has the most effective node to be moved based on network topology. The analysis reveals that selecting the optimal flow and moving the optimal node should greatly improve the throughput. We conducted an NS3 simulation with a typical multiple flow model, the bidirectional flow model. The results show that the throughput of a single target flow was improved by 27.4% with the Direct Method, 13.1 % with the Indirect Method only, and 64.1% with the proposed DTE method when compared with a case with no mobility.

With the large-scale mobile Internet promotion, people frequently use social applications, such as Facebook, Instagram, etc., making the social network a valuable research subject in Device-to-Device (D2D) communication. Maintaining high throughput and quality of service (QoS) are vital metrics worthy of exploration. This paper proposes a social-relay ad hoc network model based on multi-user cooperative mobility, consisting of two issues: relay selection scheme and multi– user cooperative mobility strategy. To ensure link stability in the relay selection scheme, we jointly consider social links’ requirements and the probability of successful data transmission in the physical layer. Since the optimization problem is NP– hard, with game theory, we define two strategic games that include relay selection game (RSG) and multi-user mobility game (MMG), to obtain the optimal relay and the best mobility strategy for maximizing throughput. The numerical results show that the proposed potential-game approach yields 13.93%, 81.78%, and 240.83% throughput gain over the previous mid-point selection scheme, random relaying and mobility, and direct transmission, respectively.

This paper describes a framework and evaluation results for user cooperative mobility control in ad hoc networks and the research results already achieved. In order for ad hoc networks as extended D2D communications to be a good complement to 5G and Beyond 5G cellular networks, further improvements for ad hoc networks in communication performance are needed. User cooperative mobility control is a method that does not require any changes on existing protocols or devices but can improve the performance simply by moving users having ad hoc network node devices. However, in practical implementation, influenced by factors such as network architecture (WLAN, ad hoc network), topology (dynamic, static), mobility cost, etc., this paper presents an overall control framework to achieve higher QoS. Within this framework, the characteristics of user cooperative mobility control in different factors and scenarios are analyzed in detail. The results that have been achieved are shown. This work will contribute to the design and development of higher QoS for future mobile smart devices, particularly in the context of autonomous driving and AR/XR.

In this paper, we propose and evaluate a user-selective mobility control method to improve communication performance in social ad hoc networks. In the network, the connected user depends on the relationship among users and the dynamic routing that optimizes the quality of service. The proposed method is a heuristic method that identifies the movable users and determines the user mobility direction regarding connection and location relationships. The three selected movable users are the user with a maximum number of connectable users, the user with a maximum number of connected users, and the farthest distance connected user. The direction of the user's mobility is the direction of the center of the region consisting of connectable users. The effectiveness of the proposed method is evaluated by comparing the throughput improvement and accuracy against throughput improvement at the optimal location. The calculated throughput improvement was 1.52 and the accuracy was 0.90 when the user with a maximum number of connectable users was moved by 20 meters. The throughput improvement was 1.15 and the accuracy was 0.83 when the user with a maximum number of connected users was moved by 15 meters. The throughput improvement was 1.22 and the accuracy was 0.75 when the farthest distance connected user was moved by 20 meters.

By enabling direct communications between nearby user equipment (UE), device‐to‐device (D2D) communication has become one of the key technologies in 5th generation (5G) mobile networks. D2D communication brings new communication opportunities for mobile devices, especially in a highly dense network. In this paper, D2D communication in the unlicensed spectrum, namely, D2D‐Unlicensed (D2D‐U), is discussed. The use of unlicensed frequency bands can ease the shortage of spectrum resources and improve network performance. However, the D2D‐U in 5G has significant effects on the network performance of existing unlicensed networks sharing the same frequency bands, such as Wi‐Fi and Bluetooth. Therefore, it is necessary to design a fair coexistence scheme for D2D‐U. To understand the coexistence problem, in this paper, we first formulate the network performance of D2D‐U and Wi‐Fi under two different coexistence schemes, namely, listen before talk (LBT) and duty cycle mechanism (DCM). Then, we use computer simulations to investigate a mode selection scheme that switches between these two schemes and point out the best possible solution for the coexistence between D2D‐U and Wi‐Fi.

The Internet of things has reached the nanoscale, and many applications using the nanoscale size of sensors are emerging. One example is imaging by distributed massive nanoscale nodes. This technique could realize an "invisible surveillance camera" by, for example, painting a liquid that contains the nanoscale sensors on a wall. This imaging method requires periodical data transfer from thousands of nanoscale nodes to a data collection node. The essential technique for handling transmission is Media Access Control (MAC). However, existing MAC protocols, like IEEE 802.11 CSMA/CA, cannot support data transfer among thousands of nodes. In this paper, we introduce a time division multiple access-based MAC protocol and evaluate its imaging performance. Our proposed method is clustering-based, two-layered protocol. A key parameter in the protocol design is the number of clusters as it determines how many sensors can successfully transmit data. Our numerical investigation shows that, depending on the transmission period, a certain number of the cluster achieves a high quality obtained image. Under this setting, a high reception rate is achieved, which significantly contributes to reliable data transfer.

We investigate the analytical expressions of typical performance metrics, namely message reception probability, message reception interval, and information dissemination rate, for vehicle-to-everything (V2X) safety communication in urban scenarios. Each of the performance metrics can be expressed in a simple closed form under some moderate assumptions such as vehicles being positioned according to a Poisson process in a two-dimensional plane. Based on the obtained expressions, we analyze the dependence of these performance metrics on the message transmission interval, which is a typical parameter of congestion control of V2X communication. The results show that the optimal message transmission interval that minimizes the average message reception interval can be derived in a simple closed form. The information dissemination rate is insensitive to the message transmission interval.

Large demands for mobile traffic subject base stations to frequent short-term and sharp peak loads. Recent analysis of data traffic on commercial mobile networks reported that the traffic peaks can be reduced by an average of 40% without compromising the quality of experience provided to the end user, if a peak load can be shifted for at most 20 s. To reduce peak traffic, we previously proposed a method for off-peak data transfer, with which user equipment (UE) autonomously delays receiving data, and a peak load on a base station can be shifted. In terms of off-peak transfer of data, a significant problem is determining how each UE estimates available throughput. In this paper we propose a method of passively estimating available throughput of each UE. We evaluated the effectiveness of the proposed method through experiments on experimental and commercial LTE networks. The results indicate that our method obtains more than a 0.7 correlation between actual available throughput and estimated throughput.

This paper proposes a new concept Decentralized Broadcasting-NOMA (Non-Orthogonal Multiple Access) for PC5-based Cellular-Vehicle-to-Everything mode 4 (called mode 4) to accommodate more vehicles and pedestrians (called nodes) in Crash Warning System (CWS). The proposed concept is to apply NOMA to broadcast system with autonomous distributed random access. This paper develops two methods to enhance mode 4 in the proposed concept. The first method is that several nodes broadcast their signals in parallel with DB-NOMA. This method can boost the spectrum utilization efficiency. The second method is that relay nodes superpose their own transmission signal and other node's relay signal with DB-NOMA. This method can efficiently expand broadcast ranges without consuming the additional transmission slots. Simulations results revealed that the proposed method accommodated 119% more nodes compared with the original mode 4. Therefore, this paper concludes that the proposed concept is useful and the proposed methods based on the concept outperform the current mode 4.

To mitigate performance anomalies due to different transmission rates and improve throughput performance in ad-hoc networks, we propose a method called user cooperative mobility in which a node, i.e., a user of an ad-hoc network, moves (changes its geometrical position) for communication. Unlike previous research regarding user cooperative mobility, it is not easy to decide to where the user should move to obtain the best result because the move may change multiple transmission rates in multiple links. Thus, we show that the best position to move to is quite different from the intuitively best position on the way to the next hop. We first evaluate throughput characteristics at every position within a movable distance from the original position of the cooperative mobility user. Then, we compare differences in throughput and direction of movement between the following three movement strategies: the best move, the intuitive move and no move. In terms of throughput improvement, which is normalized by the throughput of no move, the best move results were 40% better than those of no move and 24% better than those of the intuitive move. The direction of movement significantly differed from the intuitive move, by 92 degrees at the maximum. We conclude that cooperative user mobility with the best move is beneficial for both cooperative mobility users and ad-hoc networks.

Decentralized communication technologies (i.e., ad hoc networks) provide more opportunities for emerging wireless Internet of Things (IoT) due to the flexibility and expandability of distributed architecture. However, the performance degradation of wireless communications with the increase of the number of hops becomes the main obstacle in the development of decentralized wireless IoT systems. The main challenges come from the difficulty in designing a resource and energy efficient multihop communication protocol. Transmission control protocol (TCP), the most frequently used transport layer protocol for achieving reliable end-to-end communications, cannot achieve a satisfactory result in multihop wireless scenarios as it uses end-to-end acknowledgment which could not work well in a lossy scenario. In this paper, we propose a multihop data delivery virtualization approach which uses multiple one-hop reliable transmissions to perform multihop data transmissions. Since the proposed protocol utilizes hop-by-hop acknowledgment instead of end-to-end feedback, the congestion window size at each TCP sender node is not affected by the number of hops between the source node and the destination node. The proposed protocol can provide a significantly higher throughput and shorter transmission time as compared to the end-to-end approach. We conduct real-world experiments as well as computer simulations to show the performance gain from our proposed protocol.

With increase of mobile wireless LAN systems, frequency contamination by the overlapping basic service set (OBSS) becomes a critical issue. In this paper, we focus on HCF controlled channel access (HCCA) to alleviate the OBSS problem. HCCA considers single BSS (SBSS) environment only and suppresses traffic of co-existing DCF (distributed coordination function) based WiFi access points (APs). We propose two methods that utilize capture effects to improve the coexistence capability with DCF networks, that we call “DCF friendliness”. The first method adjusts transmission timing in HCCA WLAN by applying inter-AP coordination. The second method enables simultaneous communication with surrounding DCF WLANs by changing the frame interval of HCCA to “DIFS+1 SlotTime.” Simulations show that both the proposed method can achieve higher throughput and better DCF friendliness.

Vehicular ad hoc networks (VANETs) have been attracting interest for their potential roles in intelligent transport systems (ITS). In order to enable distributed ITS, there is a need to maintain some information in the vehicular networks without the support of any infrastructure such as road side units. In this paper, we propose a protocol which can store the data in VANETs by transferring data to a new carrier (vehicle) before the current data carrier is moving out of a specified region. For the next data carrier node selection, the protocol employs fuzzy logic to evaluate instant reward by taking into account multiple metrics specifically throughput, vehicle velocity, and bandwidth efficiency. In addition, a reinforcement learning-based algorithm is used to consider the future reward of a decision. We use theoretical analysis and computer simulations to evaluate the proposed protocol.

An incentive problem of ad hoc networks still remains for relaying frames from other nodes. By utilizing the Social Network System such as Twitter and Facebook, it is possible to estimate the degree of intimacy of friendship links not only to direct friends but also indirect friends such as friends-of-friends. If we provide links of ad-hoc network in relation to direct/indirect friends instead of other incentive mechanisms, ad-hoc network can obtain infinite potentials to connect all nodes efficiently and cover wide areas by short range communication such as IEEE802 wireless LANs. This paper is devoted to discuss social network based (real) ad-hoc network concept and evaluation of its connection probability performance.

In this paper, QoS characteristics such as TCP throughput is investigated for densely deployed mobile wireless LANs (WLANs). Factors affecting throughput characteristics are discussed and evaluated by using real machines such as smartphones and portable APs. In IEEE 802.11 WLANs, a rate adaptation mechanism controls the transmission rate and one of the dominant factors for QoS. In order to understand the behavior of the rate adaptation control, 1 to 18 sets mobile WLANs are examined under different parameters. Since a behavior of the rate adaptation control is a vender specific one and it strongly depends on interference, signal strength and etc., the real terminals such as smartphones were used in the experiments. Performance anomaly drastically reduces the throughput not only in the WLAN which has a terminal with low transmission rate but also in the neighboring WLANs that share the same channel. In order to avoid unnecessary transmission rate degradation by the rate adaptation control, Context Aware multi Rate Control (CARC) is proposed and evaluated. In CARC, Turning the rate adaptation control on/off is controlled according to a context, for example, signal strength. The evaluation results show that the CARC can be cost-effectively implemented and improves the throughput performance of whole WLANs by 3.5 times than that without the application of CARC.

We propose a light-weight performance analysis of Wi-Fi offload using a simple Markov model. Although the proposed model does not describe the behavior of the contention window of each station in detail, it allows us to analyze the non-saturation throughput while considering interaction among interfering stations and the queueing behavior of each station. In addition to this, the proposed model does not need to use the assumption that frames arrive at relay station according to a Poisson process. In this paper, we show how to apply the mean field approximation to the model to significantly reduce the amount of computation. We show that the proposed model accurately estimates the throughput performance of the Wi-Fi offload, and that the mean-field approximation is still valid in the scenario of the Wi-Fi offload.

In IaaS cloud services, QoS of network applications (NW-Apps) may degrade due to location factors such as significant distance between a server-side application (server) of a NW-App at a data center and a client-side application (client) of the NW-App at a client terminal. In order to shorten the distance and to improve the QoS, server migration services (SMSes) have been proposed. In SMSes, servers may migrate between different computers (called work places, WPs) on a network to prevent QoS degradation caused by the changes of client locations. Although server migrations can improve QoS of NW-Apps, they also generate a huge amount of traffic (server migration traffic) in the network. This paper focuses on a server location decision problem where the location of a server is decided in an on-line manner so that QoS of a NW-App is improved under the constraint that the server migration traffic has to be suppressed below an acceptable level. For the problem, we propose a practical on-line algorithm. The key idea behind the proposed algorithm is that the location of the server is decided with consideration of the QoS degradation in the future. The algorithm defines the averagely good location for the server where the QoS is expected to be relatively good for various client locations. Then, it keeps the range of the server's migration within the returnable range where the server can soon come back to the averagely good location. As a result, the QoS can be always kept as good as the one under the averagely good location. Simulation results show that the proposed algorithm improves QoS of the NW-App by up to 30% compared to a greedy algorithm.

The primary challenges with wireless sensor networks are how to construct the shortest spanning tree and how to determine the optimal sink node position in terms of minimizing the data transmission times and their variances for gathering data from all sensor nodes to a sink node. To solve these two problems, we previously proposed a basic algorithm that uses the polygonal affine shortening algorithm with flow aggregation in a fully distributed manner. In this paper, we proposed an algorithm that is more than 25% faster than the previously proposed algorithm and also proves that the proposed algorithm works in wireless sensor networks. The proposed algorithm was implemented on a well-known multi-agent simulator called Net logo.

Public areas, such as train stations and airports, providing wireless Local Area Network (WLAN) services are increasing and expanding because of the rapid development of WLANs based on IEEE 802.11 standard. Moreover, because of the advances in smartphone tethering technology, portable access points (APs) such as mobile Wi-Fi routers are being utilized more frequently. Consequently, there are increasing circumstances where a user needs to select and connect to one of the many APs. AP selection significantly determines the quality of service of the subsequent communication session. Existing AP selection algorithms consider user movement but not AP movement. We propose an AP selection method that handles both types of movement effectively. Moreover, we show that the proposed method improves the throughput significantly compared to the existing method.

This paper investigates Quality of Service (QoS) of the personal mobile wireless LANs (m-WLANs). The situations in the m-WLANs differs from the situations in the normal use of WLANs; the access point (AP) and the associated terminals (TEs) are in proximity. In the m-WLANs, the capture effect (CE) significantly affects on the throughput performance. To measure the impact on the QoS by the CE, the experimental study considering the interference from other power sources (APs and TEs) are required. However, since it is difficult to understand the detailed relationships between the QoS factors, the analytical calculations were also performed. With the experimental and analytical results, we demonstrated that the auto rate fallback algorithm of WLAN causes degradation of the QoS performance. We propose two transmission rates controlling schemes to improve the QoS performance.

Improving Quality of Service (QoS) in wireless networks is important and necessary for mobile users. We have previously proposed Comfort Route (CR) Navigation, which navigates users to their destinations using high QoS communication areas, such as Wi-Fi APs, rather than the geographical Shortest Route (SR). In this paper, we employ an analytical model to estimate the CR gain in a theoretical manner which assumes that available cellular and Wi-Fi throughputs are uniform within their coverage. The CR gain is computed by using basic parameters, including wireless network bandwidth and transmission time. To validate our model, we compare simulation results and real observation. These results conclude that the CR gain could estimate by using our analytical model.

A GPS system selects routes between two points with minimum physical distance or minimum driving time. Here we address a different type of route selection problem. Given a road map with driving distance and wireless connectivity for each road segment, find a driving route that maximizes total wireless connectivity while its length is bounded by a predetermined value. In this paper, we present three heuristic-algorithms. Initially they compute maximum connection-capacity shortest path for determining a bound for route length. The first algorithm (i) augments the road map by replacing each road segment with ratio of the distance of the road segment and its wireless communication capacity, and (ii) selects a route on the augmented map that satisfies route-length bound. The second algorithm assigns a penalty value to intersections based on their distance from a shortest path -- closer the intersection, higher the penalty. The algorithm selects among unexplored intersections one that has the minimum penalty value. The final algorithm utilize the first algorithm twice for selecting a route -- once to find distance and communication capacity of each intersection from the origin and then to find the same from the destination. Through extensive simulation of grid road networks it was found that on an average all three algorithms select routes that have higher communication capacity than any shortest paths. The most interesting observation is that the communication capacity gain is higher than the route length increase. For instance, when distance increase was bounded by 20%, on an average path selected by one algorithm was 11.4% longer than the length of the shortest path but connection capacity was about 32.5% higher than that of all shortest paths.

End-to-end TCP throughput on best-effort networks fluctuates widely depending on cross traffic and radio interference, which degrades the quality of public cloud services. However, if throughput variation is predictable on the application layer, service providers can be proactive in solving the problem of quality deterioration. We argue that TCP throughput fluctuation occurs in two states, stationary and non-stationary, and developed a method for characterizing these states. We also constructed a stochastic model of TCP throughput, which is a mixture of a stationary process model and a non-stationary one based on stationarity analysis. An evaluation experiment showed that our proposed model could predict the next 200 seconds of TCP throughput fluctuation through the Internet on three types of access networks (HSDPA/LTE/Wi-Fi) with an average accuracy of about 90%.

This paper presents a study of mobile wireless LAN (WLAN) hotspots which are used to provide cellular-WiFi tethering service to personal devices. A dense deployment scenario for fixed and mobile WLAN is described and potential performance problems due to interference are identified. An analytical model for coexisting fixed and mobile WLAN hotspots with heterogeneous traffic is presented. The model is used to evaluate the performance of a mobile WLAN as it transits through a set of densely deployed fixed access points (APs), and performance problems due to lack of frequency coordination are identified. An adaptive channel assignment (ACA) scheme for improving mobile AP performance is proposed and evaluated. It is shown that significant performance gains can be achieved with ACA with maximum absolute and percentage throughput gains up to 1.24 Mbps and 42.8% respectively. We also show that setting the scanning interval in ACA requires consideration of the speed at which the mobile WLAN is moving in order to compensate for the throughput losses during channel scanning.

We propose an optimal access-point (AP) selection algorithm for maximizing system throughput while preserving newly arrived-user throughput. In our algorithm, newly arrived users cooperate with a WLAN system they are trying to use, i.e., they are willing to move toward an appropriate AP. We discuss the effectiveness of our proposed AP selection algorithm by using numerical analysis. We also clarify and analyze the characteristics of system throughput.

IEEE 802.11e Enhanced Distributed Channel Access (EDCA) is a MAC protocol for WLAN to realize Quality of Service (QoS) control. EDCA offers the possibility of defining four different classes of service at the MAC layer so that QoS requirements can be supported. Service between classes is differentiated by different sets of channel contention parameters. However, it is difficult to guarantee QoS of the high-priority stations in EDCA where both EDCA stations and Distributed Coordination Function (DCF) stations are connected to an access point (AP), because quality control is not considered in DCF. This paper proposes a novel scheme to guarantee the QoS requirement of the high-priority stations in EDCA in such a mixed environment. In this scheme, Receiving-Opportunity Control in MAC Frame (ROC) proposed by the authors is employed. ROC probabilistically refuses data frames from low-priority stations. This paper analyzes the appropriate probability which satisfies the requirement of high-priority stations in EDCA. The effectiveness of the proposed scheme is validated through simulation.

We propose to use a buffer management scheme, called adaptive pushout, in the transmission buffer of the access point (AP) in order to solve unfairness problems between competing TCP flows over wireless LANs. The adaptive pushout comes from the observation that multiple ACK segments in a TCP connection are redundant because of the cumulative acknowledgment mechanism of the TCP. Under the adaptive pushout, an arriving packet encountering the full buffer removes one of redundant ACKs from the transmission buffer of the AP to enter there. When the buffer is full and has no redundant ACK, the arriving packet adaptively chooses one of packets in the buffer for pushing it out from the buffer. The choice of the packet to be pushed out depends on the number of uplink and downlink TCP flows, which are continuously monitored in the proposal. We have conducted simulation experiments to show that the adaptive pushout greatly improves the fairness compared with existing proposals.

In general, automobiles travel from the origin to the destination using a shortest route. However, the shortest route may not be a highest wireless connection-capacity route, because of availability of wireless services (base station and access points etc.) along the route. To the best of our knowledge, currently no algorithm exists for selecting a route that maximizes wireless connection-capacity, while keeping route length shortest and close to shortest. In this paper, we propose two modified version of Dijkstra route selection algorithms: one for selecting a maximum connection capacity shortest route, and the other is for discovering higher wireless connection-capacity routes; the length of the route could be larger than a shortest route, but no larger than predetermined bound. The second proposed algorithm exploits the state change of the intersection to broaden the search range of possible routes. Results from our extensive simulation for a Manhattan-street type grid network with the heterogeneous IEEE 802.11a wireless access, show that for a 50% increase in route length and 15 Access Points (APs), the proposed algorithm can increases wireless connection-capacity by 35.67% and 31.27% compared to the shortest and random route selection algorithms, respectively.

The technology of flow-based QoS control on wireless LANs is receiving a lot of attention. There are several QoS technologies that can provide bandwidth guarantees by using TCP congestion control (QoS-TCP) and MAC control (QoS-MAC). Unfortunately, since QoS-TCP runs on the end terminal, it is possible that QoS-TCP will attempt to guarantee bandwidth without the permission of the network administrator (e.g. a hotspot service provider). In such a case, it is possible that these flows compress the throughout of coexisting flows unfairly. That is, it is possible for a non-priority flow to claim to be a priority flow by using QoS-TCP without the permission of the network administrator. We call such flows greedy flows. The purpose of this study is restraining such greedy TCP flows by using QoS-MAC over the wireless LAN network. This study evaluates the performance of our proposal by using ns-2. Simulations show that the proposal can control the greedy flows by using a QoS-MAC.

Mobile users want to be connected to a network at any time for mobile computing applications, such as thin-client communications and network games, even while they are moving toward their destinations. They could take an alternate route (called a “longcut” route) if the route is geographically longer than a shortcut route to their destinations but gives more wireless network resources, such as bandwidth and throughput. In the case of a single user, the longcut routes are shown to be effective when they are optimally selected within a specified cost restriction. In the case of multiple users, users should be diversified in their route to avoid competing with each other. Optimization should be achieved by the appropriate arrangement of routes an d, in addition, by incentives to relax users' cost restrictions.

The purpose of our study is to guarantee the required throughput of high-priority (UDP) flows and to achieve the stable throughput for each flow in a wireless LAN in which both high-priority and low-priority (TCP) flows exist. The authors previously proposed a QoS control scheme called Receiving Opportunity Control in MAC Frame (ROC), in which an Access Point (AP) intentionally does not return an ACK for low-priority data frames. In this paper, we propose a novel QoS control scheme using Token Bucket Filter (TBF) based ROC, which achieves the above function. The TBF mechanism is known as a basic token based traffic control. The simulation results show the effectiveness of the proposed scheme.

In the future mobile network era and even now, we are faced with more diverse user and social needs for the network. These needs are changing the priorities of necessities on network and also making more practical evaluations indispensable. In general, people take a "shortcut" route when they move from one location to another. However, this will not necessarily be true for future mobile users. A "longcut" route might be highly preferable, depending on their applications that requires longlasting network connectivity or high data rate. Here, the longcut route is the optimal route for maximizing a user's satisfaction, e.g., by considering tradeoffs between the gain in transmission performance and degradation in trip time. This paper proposes a longcut route concept and evaluates its effectiveness in realistic environments by computer simulation using the network simulator ns-2, from the viewpoints of start/goal node locations, the speed of mobile nodes, the number of base stations, and the density of base stations. The results show that even in practical environments the longcut route can provide us with much capacity gain in return for a slightly longer trip time.

TCP uplink flow over wireless LANs is well known to cause two serious throughput-unfairness issues. The first is TCP throughput unfairness among uplink flows. The second is unfairness between uplink and downlink flows. Several approaches addressing these serious issues have been proposed. While they improve performance, there are difficulties in cases of legacy wireless LAN MAC stations, large numbers of stations, and encrypted packets. In this paper, we propose a simple and scalable traffic control method that can easily be applied to all of the above cases. The main idea is for access points to discard TCP uplink packets, with the discard probability determined on the basis of downlink buffer occupancy. Our proposal has the advantage that only access points require modification, not wireless stations. Simulation results show that proposed method can improve two kinds of fairness with small degradation of aggregated throughput due to the intentional uplink packet discard.

Video streaming, including VOD (Video on Demand) services, over the Internet has been rapidly becoming popular. Although RTP/UDP has been considered as a standard transport protocol for video streaming, TCP is already widely used for VOD services because of its flexible accessibility to user clients beyond firewalls or NATs. However, poor video quality, for example, frequent pause of playback, due to TCP congestion control has been pointed out. In this paper, we propose an enhancement for TCP congestion control algorithm, which we call TCP-AV, to realize stable video streaming using TCP. TCP-AV incorporates two principal mechanisms, (i) dynamic TCP parameter tuning to stabilize TCP throughput around the target rate, and (ii) temporal target rate reduction to avoid severe congestion. Simulation results show that TCP-AV provides better rate control for maintaining target rate, and thus better video quality, even when the network is shared by many co- existing flows. In our scenario for a metropolitan VOD service, TCP-AV accommodates roughly 3 times larger number of VOD flows compared to TCP-Reno.

Multi-homed environments are increasingly common, especially for mobile users. To efficiently utilize multiple access lines for single file transfer, multipath TCP communication methods have been proposed. A multipath TCP enables simultaneous distributed data transfer between two end-points on multiple TCP connections. However, these methods cannot fully utilize the available bandwidth of multiple paths because they do not properly consider the end-to-end delay of packet transmission, so out-of-order data arrival at a receiver causes a bottleneck in data sort operations. This problem is more severe in environments where the quality of each path is different or unstable, such as in wireless environments. To solve this problem, we propose a multipath TCP communication method that includes a data distribution method to enable in-order delivery at a receiver. We call this arrival-time matching load-balancing (ATLB). ATLB continuously calculates the delay of each path, including the TCP queuing delay at a sender and the network delay, and then sends a data segment through the TCP connection with the lowest end-to-end delay. Simulation results show that ATLB improves end-to-end throughput, especially in heterogeneous environments where the quality of paths differs. For example, ATLB enabled twice the throughput with the conventional multipath TCP. We also report performance evaluation results from our ATLB test bed system in a wireless network environment. Our ATLB test bed system was able to fully utilize the aggregate available bandwidth of unstable multiple wireless links.

It has been recognized that current TCP (mostly TCP-Reno) throughput deteriorates in high-speed networks with large bandwidth-delay-product. A number of protocols, such as high speed TCP and scalable TCP, have been proposed to address this problem. However, their lack of friendliness to existing protocols has hampered their wide deployment in public networks. In this paper, we propose TCP-AR (adaptive Reno) to ensure friendliness to TCP-Reno, as well as efficiency in high-speed networks. A key feature of TCP-AR is that it dynamically adjusts the TCP response function based on congestion level estimation via RTT measurement. Namely, it increases congestion window faster and decreases the window less than TCP-Reno when it recognizes no congestion. As the congestion level increases, it tunes the response function so that it behaves like TCP-Reno. Simulation results show that TCP-AR maintains friendliness to TCP-Reno in networks with varying buffer capacities with or without RED, and flows with varying RTTs, while it achieves much higher throughput than TCP-Reno or even high speed TCP.

The appearance of various quality-sensitive applications has greatly changed the requirements for network management. To manage the quality of these applications, monitoring of individual traffic flows, as well as aggregated traffic statistics, has become more important. Since per-flow monitoring involves a high processing cost, especially in emerging high-speed links, packet sampling techniques have been attracting considerable attention. However, existing sampling techniques, such as NetFlow and sFlow, have mainly targeted traffic volume monitoring and there has been little discussion on the monitoring of quality indexes including packet loss rate. In this paper, we propose a method to estimate the TCP packet loss rate from sampled packets. The proposed method detects packet loss events by monitoring duplicate ACK events induced by a TCP receiver indicating the loss events. Since only a portion of packet loss events can be detected from the sampled packets, the correct packet loss rate is estimated by means of statistical approximation. Simulation results show that the proposed method accurately estimates the TCP packet loss rate from 10% of sampled packets.

TCP overlay networks that control data transmission quality at the transport layer are being paid a lot of attention as users' demands for diversified Internet services increase. They are expected to enhance the end-to-end throughput of the TCP connection essentially because the round trip times and the packet loss ratios of each split TCP connection are reduced. However, performance degradation may occur due to undesired interactions among the split TCP connections. We introduce an analysis approach to estimate end-to-end throughput of data transmission with a TCP proxy mechanism considering performance degradation. Our analysis results reveal that we confirm the effect of the TCP proxy mechanism. We also find that we cannot ignore performance degradations due to interactions among split TCP connections, especially when the congestion level of the network they traverse is small. Further, we clarify that we should take into account the packet loss ratios, performance degradations and propagation delays of the network when we consider issues relating to the design of TCP overlay networks.

Because of the development of recent broadband access technologies, fair services among users are becoming more important criteria. The most promising scheme of router mechanisms for providing fair service is per-flow traffic management. However, it is difficult to be implemented in high speed core routers because per-flow state management is prohibitive; thus, a large number of flows are aggregated into a small number of queues. This is not a preferable situation because the more number of flows aggregated into a queue increases, the worse fairness tends to become. In this paper, we propose a new traffic management scheme called hierarchically aggregated fair queueing (HAFQ) to provide per-flow fair service. Our proposed scheme can adjust flow aggregation levels according to the queue handling capability of various routers. That means the proposed scheme is scalably used in high-speed networks. HAFQ improves the fairness among aggregated flows by estimating the number of flows aggregated in a queue and allocating bandwidth to the queue proportionally. In addition, since HAFQ can identify flows having higher arrival rates simultaneously in estimating the number of flows, it enhances the fairness by preferentially dropping their packets. We show that our proposed scheme can provide per-flow fair service through extensive simulation and experimental studies using a network processor. Since the currently available network processors (Intel IXP1200 in our case) are not high capacity, we also give extensive discussions on the applicability of our scheme to the high-speed core routers.

A content-based switch makes forwarding decisions (server selections), based on application layer information, and forwards data in the application layer. After making forwarding decisions, existing content-based switches increase their forwarding performance by TCP splicing, which releases them from maintaining TCP endpoints and allows them to forward data by packet forwarding. However, TCP splicing prevents the switches from using the application layer information for forwarding decisions. Thus existing content-based switches cannot hand off pipelined HTTP transactions, which can greatly reduce client perceived latencies. This paper proposes asymmetric TCP splicing and a method to hand off HTTP transactions between servers. Asymmetric TCP splicing allows the content-based switches to use all the application layer information in the TCP data stream from clients to servers, although it allows the switches to forward the TCP data stream from servers to clients by packet forwarding. The proposed handoff method, which uses a TCP half-close to detect the boundary of server responses, enables content-based switches to support pipelined HTTP transactions in combination with asymmetric TCP splicing. Our evaluation shows that if a content-based switch supports pipelined HTTP transactions by asymmetric TCP splicing and our proposed handoff method, client-perceived latencies can be reduced substantially.

The assured forwarding per hop behavior standardized by the IETF Differentiated Services working group provides four class-based differentiated IP services. In this service, however, unexpected service degradation may occur and differentiation among classes may be disordered if the network is designed to minimize over-provisioning or is under-provisioned. We therefore developed a packet scheduling scheme that dynamically allocates bandwidth to each class queue to guarantee the differentiation among classes under any traffic conditions. The scheme estimates the sum of CIRs (committed information rates), i.e. rate of the packets having lowest drop preference, of active flows in each class and initially allocates the link bandwidth according to the sum of CIRs. It allocates the excess bandwidth by using a combination of CIR-proportional allocation and equal-share allocation. The equal share part enables that the flows in best effort class or the flows having zero CIRs can utilize minimum share of the bandwidth. Our scheme also introduces a scalable scheduling technique to improve fairness among flows in the same class. We evaluate the proposed scheme and show that it makes DiffServ operations fairer under any traffic conditions.

We propose an IP forwarding table search engine architecture, VLMP (vertical logical operation with mask-encoded prefix-length), for routers with multi-gigabit/sec speed links. We discuss the existing approaches and the requirements for search engines, and go on to propose VLMP search engine architecture that expands upon a content addressable memory (CAM) and can perform wire-speed packet processing of an OC-192 (9.6 Gb/s) link. In this architecture, prefixes can be stored in arbitrary order, while existing ternary CAMs require prefixes to be stored in the order of their lengths. Also presented is a newly developed search LSI in which the architecture is implemented.

The authors propose a call admission control scheme based on a method of estimating cell loss quality for individual bursty traffic sources. The estimate is expressed in terms of virtual cell loss probability, which may be defined by two traffic characteristic parameters alone: peak and mean rate. The approach is suitable for the estimation of real cell loss probability in heterogeneous and homogeneous traffic models when burst length is larger than buffer capacity. The concept of virtual cell loss probability is extended to the individual call level so as to be able to estimate the quality of service (QOS) provided to individual calls. A virtual bandwidth method is used to develop a practical call admission control system. Quality is ensured by combining a traffic clustering scheme, with a scheme for assigning individual clusters to subcapacities of a link. Priority levels are presented in terms of the class of QOS required, i.e., deterministic or statistical, and the allocation of virtual bandwidth is discussed in terms of both QOS class and traffic characteristics.>