Satellite Internet is one of the most important networks for emergency communications because of its tolerant of disasters such as earthquake. Therefore, satellite Internet has received considerable attention over recent years. However, most standard implementations of TCP congestion control method perform poorly in satellite Internet due to its high bit error rate and long propagation delay. This paper proposes a new TCP congestion control method called TCP-STAR to improve the throughput over satellite Internet. TCP-STAR has three new mechanisms, namely Congestion Window Setting (CWS) based on available bandwidth, Lift Window Control (LWC), and Acknowledgment Error Notification (AEN). CWS can resist the reduction of the transmission rate when data losses are caused by bit error. LWC is able to increase the congestion window quickly based on the estimated available bandwidth. AEN can avoid the reduction of the throughput by mis-retransmission of data. The mis-retransmission is caused by ack losses or delay. Simulations show that TCP-STAR can obtain the best throughput comparing with other TCP variants (TCP-J and TCP-WestwoodBR). Furthermore, we found that the fairness of TCP-STAR is a little lower than that of TCP-WestwoodBR. However, the fairness of TCP-STAR is equal to TCP-J.

The medium access control (MAC) protocol is the main determiner of the system throughput in Wireless Local Area Networks (WLANs). The MAC technique of the IEEE 802.11 protocol is called Distributed Coordination Function (DCF). DCF is based on a carrier sense multiple access with collision avoidance (CSMA/CA) scheme with binary slotted exponential backoff. Each station generates a random backoff interval before transmitting a packet to minimize the probability of collision with packets being transmitted by other stations. However, when the number of stations increases, the system throughput decreases. This paper proposes a new backoff algorithm that uses finish tags. The proposed algorithm uses the finish tag of each station to control the backoff intervals so as to improve system throughput. The finish tag is updated when a packet reaches the front of its flow, and it is attached to the packet just prior to transmission. When a station receives packets with older finish tags, its backoff time interval is increased. For this reason, the more the stations there are, the larger the backoff time becomes. Simulations confirm that the proposal improves system throughput of a IEEE 802.11 network under saturation conditions.

This letter presents a simple but accurate analytical model to evaluate the throughput of IEEE 802.11 distributed coordination function in non-saturated conditions. The influence of offered load on the throughput of both basic and RTS/CTS access mechanisms are analyzed and compared. It's shown that basic access scheme can achieve the same maximal throughput as that of RTS/CTS mechanism in non-saturated conditions while the latter is robust to the number of contending stations compared to basic mechanism. The analytical results are validated by extensive simulations.

As computer hardware components are achieving greater speeds, network link bandwidths are becoming wider. A number of enhancements to TCP have been developed in order to fully exploit these improvements in network infrastructures, including TCP window scale option, SACK option, and HighSpeed TCP (HSTCP) modifications. However, even with these enhancements, TCP cannot provide satisfactory performance in high-speed long-delay networks. As a means addressing this problem, gentle HighSpeed TCP (gHSTCP) has been proposed in [1]. However, its effectiveness has only been demonstrated in simulation experiments. In the present paper, a refined gHSTCP algorithm is proposed for application to real networks. The performance of the refined gHSTCP algorithm is then assessed experimentally. The refined gHSTCP algorithm is based on the original algorithm, which uses two modes (Reno mode and HSTCP mode) in the congestion avoidance phase and switches modes based on RTT increasing trends. The refined gHSTCP algorithm compares two RTT thresholds and judges which mode will be used. The performance of gHSTCP is compared with TCP Reno/HSTCP and parallel TCP mechanisms. The experimental results demonstrate that gHSTCP can provide a better tradeoff in terms of utilization and fairness against co-existing traditional TCP Reno connections, whereas HSTCP and parallel TCP suffer from the trade-off problem.

We propose a cell library development methodology for throughput enhancement of character projection equipment. First, an ILP (Integer Linear Programming)-based cell selection is proposed for the equipment for which both of the CP (Character Projection) and VSB (Variable Shaped Beam) methods are available, in order to minimize the number of electron beam (EB) shots, that is, time to fabricate chips. Secondly, the influence of cell directions on area and delay time of chips is examined. The examination helps to reduce the number of EB shots with a little deterioration of area and delay time because unnecessary directions of cells can be removed. Finally, a case study is shown in which the numbers of EB shots are shown for several cases.

It is well known that deploying a proxy at the boundary of wireless networks and the Internet is able to improve the performance of transmission control protocol (TCP) over wireless links. Snoop protocol, acting like a transport layer proxy, performs local retransmissions for packets corrupted by wireless channel errors. In this letter, an improvement for the Snoop protocol is proposed to shorten the time spent on local recovery by sending extra copies in every local retransmission attempt. This enables TCP to quickly return to normal, effectively eliminating several of the problems that may cause throughput degradation.

We have proposed Rate-based Multi-path Transmission Control Protocol (R-M/TCP) for improving reliability and performance of data transfer over the Internet by using multiple paths. Congestion control in R-M/TCP is performed in a rate-based and loss-avoidance manner. It attempts to estimate the available bandwidth and the queue length of the used routes in order to fully utilize the bandwidth resources. However, it has been reported that when the used routes' characteristics, i.e. available bandwidth and delay, are much different, R-M/TCP cannot achieve the desired throughput from the routes. This is because R-M/TCP originally transmits data packets in a round-robin manner through the routes. In this paper, therefore, we propose R-M/TCP using Packet Scheduling Algorithm (PSA). Instead of using the round-robin manner, R-M/TCP utilizes PSA that accounts for time-varying bandwidth and delay of each path so that number of data packets arriving in out-of-order at the receiver can be minimized and the desired throughput can be achieved. Quantitative simulations are conducted to show effectiveness of R-M/TCP using PSA.

As an enhancement of the state-of-the-art solutions, a high-throughput architecture of a decoder for structured LDPC codes is presented in this paper. Thanks to the peculiar code definition and to the envisaged architecture featuring memory paging, the decoder is very flexible, and the support of different code rates is achieved with no significant hardware overhead. A top-down design flow of a real decoder is reported, starting from the analysis of the system performance in finite-precision arithmetic, up to the VLSI implementation details of the elementary modules. The synthesis of the whole decoder on 0.18µm standard cells CMOS technology showed remarkable performances: small implementation loss (0.2dB down to BER = 10-8), low latency (less than 6.0µs), high useful throughput (up to 940Mbps) and low complexity (about 375 Kgates).

This paper proposes two different packet schedulers for IEEE 802.16e type time division duplex - orthogonal frequency division multiple access (TDD-OFDMA), which are the weighted fair scheduling (WFS) and the throughput guarantee scheduling (TGS). The performance of proposed schedulers is compared to those of some of conventional schedulers such as round robin (RR), proportional fair (PF), fast fair throughput (FFTH), and fair throughput (FTH) in terms of service coverage, effective throughput and fairness at 64 kbps and 128 kbps minimum user throughput requirements. For a relatively smaller throughput (64 kbps) requirement, the proposed schedulers increase the number of users per sector within 95% service coverage while satisfying the 1xEV-DV fairness criterion. For a relatively larger throughput (128 kbps) requirement, the proposed schedulers provide higher coverage than the PF scheduler while maintaining the same effective aggregate throughput.

In this letter, we propose an enhanced gentle distributed coordination function (GDCF), which is a simple and effective collision resolution mechanism, to improve the performance of IEEE 802.11 DCF. We compare performance of the enhanced GDCF with that of the legacy DCF and the conventional GDCF via analysis and simulations. The enhanced GDCF introduces a new counter to check the number of consecutively successful transmissions, and the maximum permitted values of the counter differ for different backoff stages. The proposed GDCF is shown to have performance superior to that of the conventional GDCF for various combinations of contending stations and frame length.

Recently, multiple-input multiple-output (MIMO) systems that realize high bit rate data transmission with multiple antennas at both transmitter and receiver have drawn much attention for their high spectral efficiency. In MIMO systems, space division multiplexing (SDM) has been researched widely. In SDM, the input data symbols are transmitted from multiple transmit antennas at the transmitter, and the output data symbols are extracted by the signal processing at the receiver. In recent wireless communications, the environments that the number of transmit antennas is larger than that of receive antennas often exist. Under such environments, the MIMO system that transmits independent data streams from each transmit antenna simultaneously cannot separate the received signals, and the signal quality deteriorates largely. Therefore, we need the scheme that attains high quality and high throughput data transmission under such environments. In this paper, we propose a throughput maximization transmission control scheme for MIMO systems. The proposed transmission control scheme selects a transmission scheme (a set of transmit antennas, modulation schemes, and coding rates) with maximum throughput based on output signal to interference and noise ratio (SINR) and output signal to noise ratio (SNR). We show that the proposed transmission control scheme attains high throughput by our computer simulation.

Most of analytical models proposed so far for the IEEE 802.11 distributed coordination function (DCF) focus on saturation performance. In this paper, we develop an analytic model for unsaturation performance evaluation of the IEEE 802.11 DCF with and without slow contention window decrease (SCWD). The model explicitly takes into account the carrier sensing mechanism and an additional backoff interval after successful frame transmission, both of which can be ignored under saturation conditions. Expressions are derived for throughput and delay characteristics by means of the equilibrium point analysis. The accuracy of our model is validated through computer simulation. Numerical results based on the IEEE 802.11b with CCK show that the SCWD can stably achieve approximately 20% performance gain over the normal 802.11 DCF under unsaturation conditions as well as saturation ones.

The issue of importance of multiuser detection for CDMA-based mobile ad hoc networks is addressed in this paper. For conventional scheme, each terminal in the network uses matched filter to receive packets, so the performance (e.g., throughput) of the network suffers from multi-access interference (MAI). Different from above scheme, in this paper, each terminal of the ad hoc network is equipped with an adaptive blind linear multiuser detector, so the ability of MAI-resistance is gained. Employing slotted-ALOHA protocol in MAC layer and using fully-connected network model, the throughput of ad hoc network is studied. Theoretic analysis and simulation results show that multiuser detection can remarkably improve throughput performance of ad hoc networks.

Ad hoc networks are becoming an interesting research area, as they inherently support unique network applications for the wireless communications in a rugged environment, which requires rapid deployment and is difficult to be provided by an infrastructure network. Many issues need to be addressed for the ad hoc networks. In this paper, we propose an efficient distributed coordination function on the media access control protocol to enhance the power conservation of mobile hosts by using a power control algorithm and the network throughput of an ad hoc network by using an algorithm for simultaneous frame transmissions. Extensive simulation is studied to evaluate the improvement of the proposed method. The results of the simulation exhibit significant improvement to the standard access control protocol. With slight improvement of network throughput, up to 85% of the consumed energy was able to be saved in compared to the standard protocol and up to 7 times of the energy efficiency was enhanced with the proposed method.

In this paper, a refined analytic model is presented for the IEEE 802.11 distributed coordination function (DCF) in a time-varying channel environment. In the proposed model, the channel is modelled using a finite-state Markov (FSM) chain. The saturation throughput and average packet delay are analyzed from the proposed model. It is shown using OPNETTM and UltraSANTM simulations that the proposed model accurately predicts the performance of the IEEE 802.11 DCF.

In this paper, we propose a novel array antenna-assisted adaptive modulation scheme for fast fading environments. Although adaptive modulation is an efficient technique capable of establishing high bit-rate digital transmission in a multi-path fading environment, it is sensitive to the fast time variation of the channel because of difficulties in tracking the channel state. To resolve this problem, an array antenna-based Doppler spread compensator was applied to the adaptive modulation scheme. Computer simulation results indicated that the proposed scheme can markedly improve the bit error rate and throughput performance for the region in which the maximum Doppler frequency normalized by the packet length is up to 0.1.

Packet-recognition/code-acquisition (PR/CA) is one of the most important issues in packet communication systems. In a CDMA Unslotted ALOHA system, Multiple Access Interference (MAI) may bring about errors in PR/CA. The MAI mainly stems from already recognized packets and newly arriving packets under the execution of PR/CA. This characteristic of asynchronous transmission in CDMA U-ALOHA systems implies that only one or a few packets arrive at the receiver within a short interval of a execution. Furthermore, newly arriving packets are recognized and code-acquired by using a short preamble part. Consequently, the MAI from the packets under the execution of the PR/CA will be small. Focusing on that point, this paper proposes applying the IC scheme in order to suppress the MAI from the already recognized and code-acquired packets. A performance evaluation demonstrates that such an application is valid due to the small amount of MAI from the packets under the execution of PR/CA. In addition, we demonstrates that the scheme reduces false recognition rather than mis-recognition. Such a scheme improves the performance of not only PR/CA, but also the throughput.

This paper presents an analysis of the Slotted DS-CDMA system with modified node components in order to construct a load control structure in which the service rates of each node can be dynamically adapted without using feedback information. In contrast to the traditional Slotted DS-CDMA which is widely represented with single queue, prior emphasis of the approach is laid on the usage of an additional queue which is applied to manage the collided packet traffic while its queue size is also used as a load control parameter. Semi-Markov process is applied to describe the statistic behavior of the system in steady state. Trade-offs between two major performance parameters, i.e., delay and throughput, are presented and compared with those of the traditional system. Results obtained from the simulation and numerical analysis using queuing concept are compared. With these results, an advantage performance for group packets is shown, and we finally extend the concept based on the obtained results to describe a simple algorithm using one way control message as the tool to alleviate the stability problem.

This letter provides a scalable slotted ring network architecture with nodes using one fixed transmitter, one tunable transmitter and multiple fixed receivers. Furthermore, the novel access protocol with efficient slot reuse is proposed. Theoretical analysis and simulation results show that protocol can achieve high node throughput and low queuing delay.

This paper presents a new asynchronous FIFO design to reduce forward latency in a linear structure. The operation mode for each cell can be reconfigured dynamically as either of the two schemes, wave pipelining or handshaking, according to the data flow in the FIFO. The adoption of wave pipelining to the conventional self-timed FIFO can reduce the overhead of the handshaking as well as latching control in each stage. Initial pre-layout simulations indicate about two times of improvement on latency performance over a state-of-art asynchronous FIFO, while retaining its throughput.