The Burst ACK scheme enhances effective throughput by reducing ACK overhead when a transmitter sends sequentially multiple data frames to a destination. IEEE 802.11e is one such example. The size of the data frame body and the number of burst data frames are important burst transmission parameters that affect throughput. The larger the burst transmission parameters are, the better the throughput under error-free conditions becomes. However, large data frame could reduce throughput under error-prone conditions caused by signal-to-noise ratio (SNR) deterioration. If the throughput can be calculated from the burst transmission parameters and error rate, the appropriate ranges of the burst transmission parameters could be narrowed down, and the necessary buffer size for storing transmit data or received data temporarily could be estimated. In this paper, we present a method that features a simple algorithm for estimating the effective throughput from the burst transmission parameters and error rate. The calculated throughput values agree well with the measured ones for actual wireless boards based on the IEEE 802.11-based original MAC protocol. We also calculate throughput values for larger values of the burst transmission parameters outside the assignable values of the wireless boards and find the appropriate values of the burst transmission parameters.

The scalability of three-stage Clos-network packet switches makes them an attractive approach in implementing large-size packet switches. However, the configuration time of Clos-network switches depends on both the buffering strategy used and the adopted configuration process. To reduce configuration time, this paper focuses on the so-called Memory-Space-Memory (MSM) Clos-network packet switch, where the switch modules in the first and third stages use memory to support resolution of output port contention. The configuration of these switches is then based on a process to dispatch cells from the first-stage modules to the third-stage modules. Therefore, the throughput of an MSM Clos-network switch depends on the dispatching scheme used. This paper introduces a cell dispatching scheme, called maximum weight matching dispatching (MWMD) scheme, for MSM Clos-network switches and a request queue structure in the first-stage modules. The MWMD scheme performs maximum weight matching, similar to that used for input-queued single-stage packet switches, that in combination with the request queues can achieve 100% throughput under independent and identical admissible traffic. This high throughput can be achieved without allocating buffers in the second stage and without expanding the second stage of this three-stage packet switch. A low-complexity dispatching scheme, the maximal oldest-cell-first matching dispatching (MOMD) scheme, is also introduced as an alternative to MWMD. The performance evaluation in this paper shows that MOMD achieves high throughput under unbalanced traffic through the execution of a finite number of iterations.

Routing in wireless ad hoc networks is a challenging issue because it dynamically controls the network topology and determines the network performance. Most of the available protocols are based on single-rate radio networks and they use hop-count as the routing metric. There have been some efforts for multirate radios as well that use transmission-time of a packet as the routing metric. However, neither the hop-count nor the transmission-time may be a sufficient criterion for discovering a high-throughput path in a multirate wireless ad hoc network. Hop-count based routing metrics usually select a low-rate bound path whereas the transmission-time based metrics may select a path with a comparatively large number of hops. The trade-off between transmission time and effective transmission range of a data rate can be another key criterion for finding a high-throughput path in such environments. In this paper, we introduce a novel routing metric based on the efficiency of a data rate that balances the required time and covering distance by a transmission and results in increased throughput. Using the new metric, we propose an on-demand routing protocol for multirate wireless environment, dubbed MR-AODV, to discover high-throughput paths in the network. A key feature of MR-AODV is that it controls the data rate in transmitting both the data and control packets. Rate control during the route discovery phase minimizes the route request (RREQ) avalanche. We use simulations to evaluate the performance of the proposed MR-AODV protocol and results reveal significant improvements in end-to-end throughput and minimization of routing overhead.

Recent research on overlay networks has revealed that user-perceived network performance could be improved by an overlay routing mechanism. The effectiveness of overlay routing is mainly a result of the policy mismatch between the overlay routing and the underlay IP routing operated by ISPs. However, this policy mismatch causes a "free-riding" traffic problem, which may become harmful to the cost structure of Internet Service Providers. In the present paper, we define the free-riding problem in the overlay routing and evaluate the degree of free-riding traffic to reveal the effect of the problem on ISPs. We introduce a numerical metric to evaluate the degree of the free-riding problem and confirm that most multihop overlay paths that have better performance than the direct path brings the free-riding problem. We also discuss the guidelines for selecting paths that are more effective than the direct path and that mitigate the free-riding problem.

Overlay Access Technology can compensate for the spectrum underutilization problem by exploiting Cognitive Radios capabilities. MAC design is an important aspect of Overlay Access research. In this paper we present the overlay access environment and the challenges it poses to MAC design. Then, we propose the use of a modified Distributed Coordination Function as the MAC protocol for distributed Overlay Access networks. The resulted Intermittent DCF performs with robustness in the demanding overlay access environment, which is characterized by frequent spectrum scan procedure interruptions and low achievable transmission rates. The most recent DCF Markov Chain Model is extended in order to include the overlay operation modifications. Our extension concerns the slot duration expectations calculation which, in the overlay environment, have not constant values but depend on overlay operation parameters. Using the analytical model we evaluate the performance of the DCF under the effect of certain overlay access parameters. The new analytical model predictions are validated with simulations, and are found to accurately capture many interesting features of the overlay operation. Our model can be used in feasibility studies of realistic overlay scenarios and in admission control algorithms of QoS enabled distributed overlay access networks that engage the Intermittent DCF.

This paper describes a proposed propagation estimation method and TCP/IP-based evaluations for mobile communications employing a stratospheric platform. To estimate a wireless channel, a realistic and detailed description of its physical environment must be accurately defined. Therefore, a building distribution model characterizing the physical environment in areas in Japan is presented. The analyses of the propagation estimation method are based on the "ray-tracing" model. The results from the proposed method are derived depending on elevation and azimuth angles. In order to validate our results, comparisons between the proposed method and our previous measurement are made for a typical semi-urban area in Japan. The comparisons show close agreement between the estimation results and the measurement results. Finally and interestingly, we present communication performance evaluations based on TCP/IP protocol by using the results achieved from our channel estimation with semi-analytical and simulation approach.

This paper quantitatively analyzes the maximum UDP (User Datagram Protocol) throughput for two-way flows in wireless string multi-hop networks. The validity of the analysis is shown by the comparison with the simulation and the experiment results. The authors also clarify the difference fundamental characteristics between a one-way flow and a two-way flow in detail based on the simulation results. The result shows that collisions at both ends' nodes are decisive in determining the throughput for two-way flows. The analyses are applicable to the estimation of VoIP (Voice over Internet Protocol) capacity for string multi-hop networks represented by WLAN (Wireless Local Area Network) mesh networks.

A multiuser beamforming algorithm using codebook is proposed in this letter to reduce inter-user interference and improve the throughput of the downlink of multiuser MIMO systems. Basing on the minimal leakage criterion, we combine the codebook selection, limited feedforward and MMSE detection method in the new proposed algorithm. It not only improves the performance of systems, but also reduces the computational complexity at transmitter. Simulation results shows that the proposed algorithm has better performance than conventional beamforming methods.

A parallel multiplexing scheduling (PMS) scheme is proposed for distributed antenna systems (DAS), which greatly improves average system throughput due to multi-user diversity and multi-user multiplexing. However, PMS has poor fairness because of the use of the "best channel selection" criteria in the scheduler. Thus we present a parallel proportional fair scheduling (PPFS) scheme, which combines PMS with proportional fair scheduling (PFS) to achieve a tradeoff between average throughput and fairness. In PPFS, the "relative signal to noise ratio (SNR)" is employed as a metric to select the user instead of the "relative throughput" in the original PFS. And only partial channel state information (CSI) is fed back to the base station (BS) in PPFS. Moreover, there are multiple users selected to transmit simultaneously at each slot in PPFS, while only one user occupies all channel resources at each slot in PFS. Consequently, PPFS improves fairness performance of PMS greatly with a relatively small loss of average throughput compared to PFS.

The IEEE 802.11 standard has been extensively deployed all over the world. Many studies have been put on its performance, especially throughput. Most research focused on the analysis of saturated throughput, but non-saturated situation is more usual in practice. By extending a saturation throughput model, a concise and novel model is proposed in this paper, which can be used to analyze both saturated and non-saturated conditions. Moreover, the model can also deal with the heterogeneous condition, which allows stations to have different traffic. Different access mechanisms and packet payloads are used in simulation to validate it, and the results show that the model is accurate.

Context-based Adaptive Binary Arithmetic Coding(CABAC) is one of the algorithmic improvements that the H.264/AVC standard provides to enhance the compression ratio of video sequences. Compared with the context-based adaptive variable length coding (CAVLC), CABAC can obtain a better compression ratio at the price of higher computation complexity. In particular, the inherent data dependency and various types of syntax elements in CABAC results in a dramatically increased complexity if two bins obtained from binarized syntax elements are handled at a time. By analyzing the distribution of binarized bins in different video sequences, this work shows how to effectively improve the encoding rate with limited hardware overhead by allowing only a certain type of syntax element to be processed two bins at a time. Together with the proposed context memory management scheme and range renovation method, experimental results reveal that an encoding rate of up to 410 M-bin/s can be obtained with a limited increase in hardware requirement. Compared with related works that do not support multi-symbol encoding, our development can achieve nearly twice their throughput rates with less than 25 % hardware overhead.

Recent research on overlay networks has revealed that user-perceived network performance, such as end-to-end delay performance, could be improved by an overlay routing mechanism. However, these studies consider only end-to-end delay, and few studies have focused on bandwidth-related information, such as available bandwidth and TCP throughput, which are important performance metrics especially for long-lived data transmission. In the present paper, we investigate the effect of overlay routing both delay and bandwidth-related information, based on the measurement results of network paths between PlanetLab nodes. We consider three metrics for selecting the overlay route: end-to-end delay, available bandwidth, and TCP throughput. We then show that the available bandwidth-based overlay routing provides significant gain, as compared with delay-based routing. We further reveal the correlation between the latency and available bandwidth of the overlay paths and propose several guidelines for selecting an overlay path.

The integration of multihop relays with orthogonal frequency-division multiple access (OFDMA) cellular infrastructures can meet the growing demands for better coverage and higher throughput. Resource allocation in the OFDMA two-hop relay system is more complex than that in the conventional single-hop OFDMA system. With time division between transmissions from the base station (BS) and those from relay stations (RSs), fixed partitioning of the BS subframe and RS subframes can not adapt to various traffic demands. Moreover, single-hop scheduling algorithms can not be used directly in the two-hop system. Therefore, we propose a semi-distributed algorithm called ASP to adjust the length of every subframe adaptively, and suggest two ways to extend single-hop scheduling algorithms into multihop scenarios: link-based and end-to-end approaches. Simulation results indicate that the ASP algorithm increases system utilization and fairness. The max carrier-to-interference ratio (Max C/I) and proportional fairness (PF) scheduling algorithms extended using the end-to-end approach obtain higher throughput than those using the link-based approach, but at the expense of more overhead for information exchange between the BS and RSs. The resource allocation scheme using ASP and end-to-end PF scheduling achieves a tradeoff between system throughput maximization and fairness.

IEEE 802.11 MAC is the most commonly used protocol in multi-hop ad-hoc networks which need no infrastructure. In multi-hop ad-hoc networks based on IEEE 802.11, since nodes try transmitting their packets at every possible slot, there is a high packet loss probability. This degrades network performance attributes such as throughput and fairness. In this paper, we focus on achieving the maximum throughput of the given network topology by adding extra backoff time to random backoff time based on IEEE 802.11. Specifically, the optimal load rate is obtained with constraints of Carrier Sensing Multiple Access/Collision Avoidance (CSMA/CA) mechanism.

QoS (Quality of Service) control in WLAN (IEEE802.11 Wireless LAN) is becoming increasingly important because WLAN is widely deployed as an access network and also plays a key role in providing seamless QoS communication between wired networks and wireless terminals. Although previous research has attempted to increase total throughput or available bandwidth in WLAN, few studies have treated individual TCP/UDP flow QoS. EDCA in IEEE802.11e might provide prioritized QoS functions that would partially address this problem. However, in uplink flow, which is defined as data moving from a terminal toward an Access Point, EDCA has limitations. These manifest themselves both across classes and in differentiated QoS control between terminals in the same class. Furthermore, 802.11e requires modification of terminals as well as other alterations proposed by other researchers. Instead of 802.11e or other modifications of 802.11, we propose an approach to controlling QoS that requires no terminal modifications or installation of additional software/hardware. The proposed idea is MAC-frame Receiving-Opportunity Control (ROC), in which a MAC (Media Access Control) frame is completely delivered only if it has sufficiently high priority; otherwise either the MAC frame is discarded or an Acknowledge (ACK) to the frame is not sent. The frame that was discarded is forced to accept a back-off waiting time for retransmission, consistent with 802.11DCF. This results in QoS degradation for low priority flows and eventually results in QoS improvements for high priority flows. Performance evaluation shows that the ROC causes some performance degradation in total WLAN throughput but can achieve not only QoS priority control but also arbitrary throughput performance. In particular, the ROC (in the MAC layer) can also permit different throughputs for high priority and low priority flows, conditioned on control processes in other layers. These may include rate adaptation (in the MAC layer) and TCP congestion control (in the TCP layer).

Recently, real-time media delivery services such as video streaming and VoIP have rapidly become popular. For these applications requiring high-level QoS guarantee, our research group has proposed a transport-layer approach to provide predictable throughput for upper-layer applications. In the present paper, we propose a congestion control mechanism of TCP for achieving predictable throughput. It does not mean we can guarantee the throughput, while we can provide the throughput required by an upper-layer application at high probability when network congestion level is not so high by using the inline network measurement technique for available bandwidth of the network path. We present the evaluation results for the proposed mechanism obtained in simulation and implementation experiments, and confirm that the proposed mechanism can assure a TCP throughput if the required bandwidth is not so high compared to the physical bandwidth, even when other ordinary TCP (e.g., TCP Reno) connections occupy the link.

We propose Adaptive Resource Allocation for the Partial Block MC-CDMA (ARA-PB/MC-CDMA) system. The ARA-PB/MC-CDMA system aims to improve total throughput performance and frequency efficiency across various channel conditions. It adaptively changes the number of blocks to improve the throughput performance and frequency efficiency according to the Signal to Interference Ratio (SIR). Therefore, the proposed system supports various Quality of Service (QoS) requirements for various SIR values.

In this paper, we investigate the throughput efficiency of the Go-Back-N ARQ protocol on parallel multiple channels with burst errors. We assume that packet errors occur according to a two-state Markov chain on each channel. The effect of the decay factor of the Markov chain on throughput efficiency is evaluated based on the results of numerical analysis.

Character projection (CP) lithography is utilized for maskless lithography and is a potential for the future photomask manufacture because it can project ICs much faster than point beam projection or variable-shaped beam (VSB) projection. In this paper, we first present a projection mask set development methodology for multi-column-cell (MCC) systems, in which column-cells can project patterns in parallel with the CP and VSB lithographies. Next, we present an INLP (integer nonlinear programming) model as well as an ILP (integer linear programming) model for optimizing a CP mask set of an MCC projection system so that projection time is reduced. The experimental results show that our optimization has achieved 33.4% less projection time in the best case than a naive CP mask development approach. The experimental results indicate that our CP mask set optimization method has virtually increased cell pattern objects on CP masks and has decreased VSB projection so that it has achieved higher projection throughput than just parallelizing two column-cells with conventional CP masks.

In this paper, we study the power control for the adaptive rate system that adapts the data rate to the channel and interference condition by adjusting the SINR requirement. The power control technique aims to decrease the interference to obtain the satisfactory throughput. We first find the optimal CINR that allows as many users as possible to transmit their data with the satisfied throughput requirements. Then we use the optimal CINR as the target CINR in the proposed distributed power control algorithm. During the operation of this algorithm, we let each user adjust the transmit power for several iterations to obtain the satisfactory CINR, which in turn leads to the satisfactory throughput. We show that the user can meet its throughput requirement by increasing the number of iterations, and once the user meets its throughput requirement, this condition will hold from then on. Furthermore, it is proven that the throughput can converge to the throughput requirement for each user.