In ad hoc networks, broadcast forwarding protocols called OR (opportunistic routing) have been proposed to gain path diversity for higher packet delivery rates and shorter end-to-end delays. In general backoff-based OR protocols, each receiver autonomously makes a forwarding decision by using certain metrics to determine if a random backoff time is to be applied. However, each forwarder candidate must wait for the expiration of the backoff timer before forwarding a packet. Moreover, they cannot gain path diversity if the forwarding path includes local sparse areas, and this degrades performance as it strongly depends on the terminal density. In this paper, we propose a novel OR protocol called PRIOR (prioritized forwarding for opportunistic routing). In PRIOR, a terminal, called a prioritized forwarder and which forwards packets without using a backoff time, is selected from among the neighbours. In addition, PRIOR uses lightweight hop-by-hop retransmission control to mitigate the effect of terminal density. Moreover, we introduce an enhancement to PRIOR to reduce unnecessary forwarding by using an explicit acknowledgement. We evaluate PRIOR in comparison with conventional protocols in computer simulations.

Computer networks require sophisticated control mechanisms to realize fair resource allocation among users in conjunction with efficient resource usage. To successfully realize fair resource allocation in a network, someone should control the behavior of each user by considering fairness. To provide efficient resource utilization, someone should control the behavior of all users by considering efficiency. To realize both control goals with different granularities at the same time, a hierarchical network control mechanism that combines microscopic control (i.e., fairness control) and macroscopic control (i.e., efficiency control) is required. In previous works, Aida proposed the concept of chaos-based hierarchical network control. Next, as an application of the chaos-based concept, Aida designed a fundamental framework of hierarchical transmission rate control based on the chaos of coupled relaxation oscillators. To clarify the realization of the chaos-based concept, one should specify the chaos-based hierarchical transmission rate control in enough detail to work in an actual network, and confirm that it works as intended. In this study, we implement the chaos-based hierarchical transmission rate control in a popular network simulator, ns-2, and confirm its operation through our experimentation. Results verify that the chaos-based concept can be successfully realized in TCP/IP networks.

This paper proposes reflection and transmission control panels using artificially designed materials. As the artificially designed material, finite- and infinite-length metal wire array sheets are used here. Laminated structures consisting of the metal wire array sheets and dielectric material are proposed. Reflection and transmission characteristics of these structures can be controlled by changing the metal wire parameters such as wire length, spacing gaps between the wires, and the dielectric material's thickness and relative permittivity. The reflection and transmission characteristics of the laminated structures are evaluated by measurements in free space and by transmission line theory.

To raise the energy efficiency of wireless clients, it is important to sleep in idle periods. When multiple network applications are running concurrently on a single wireless client, packets of each application are sent and received independently, but multiplexed at MAC-level. This uncoordinated behavior makes it difficult to control of sleep timing. In addition, frequent state transitions between active and sleep modes consume non-negligible energy. In this paper, we propose a transport-layer approach that resolves this problem and so reduces energy consumed by multiple TCP flows on a wireless LAN (WLAN) client. The proposed method, called SCTP tunneling, has two key features: flow aggregation and burst transmission. It aggregates multiple TCP flows into a single SCTP association between a wireless client and an access point to control packet transmission and reception timing. Furthermore, to improve the sleep efficiency, SCTP tunneling reduces the number of state transitions by handling multiple packets in a bursty fashion. In this study, we construct a mathematical model of the energy consumed by SCTP tunneling to assess its energy efficiency. Through numerical examples, we show that the proposed method can reduce energy consumption by up to 69%.

Per-flow unfairness of TCP throughput in the IEEE 802.11 wireless LAN (WLAN) environment has been reported in past literature. A number of researchers have proposed various methods for alleviating the unfairness; most require modification of MAC protocols or queue management mechanisms in access points. However, the MAC protocols of access points are generally implemented at hardware level, so changing these protocols is costly. As the first contribution of this paper, we propose a transport-layer solution for alleviating unfairness among TCP flows, requiring a small modification to TCP congestion control mechanisms only on WLAN stations. In the past literature on fairness issues in the Internet flows, the performance of the proposed solutions for alleviating the unfairness has been evaluated separately from the network bandwidth utilization, meaning that they did not consider the trade-off relationships between fairness and bandwidth utilization. Therefore, as the second contribution of this paper, we introduce a novel performance metric for evaluating trade-off relationships between per-flow fairness and bandwidth utilization at the network bottleneck. We confirm the fundamental characteristics of the proposed method through simulation experiments and evaluate the performance of the proposed method through experiments in real WLAN environments. We show that the proposed method can achieve better a trade-off between fairness and bandwidth utilization, regardless of vendor implementations of wireless access points and wireless interface cards.

With the emergence of bandwidth-greedy application services, high-speed transport protocols are expected to effectively and aggressively use large amounts of bandwidth in current broadband and multimedia networks. However, when high-speed transport protocols compete with other standard TCP flows, they can occupy most of the available bandwidth leading to disruption of service. To deploy high-speed transport protocols on the Internet, such unfair situations must be improved. In this paper, therefore, we propose a method to improve fairness, called Kyushu-TCP (KTCP), which introduces a non-aggressive period in the congestion avoidance phase to give other standard TCP flows more chances of increasing their transmission rates. This method improves fairness in terms of the throughput by estimating the stably available bandwidth-delay product and adjusting its transmission rate based on this estimation. We show the effectiveness of the proposed method through simulations.

Traditional TCP has a good congestion control strategy that adapts its sending rate in accordance with network congestion. In addition, a fast recovery algorithm can help TCP achieve better throughput by responding to temporary network congestion well. However, if multiple packet losses occur, the time to enter congestion avoidance phase would be delayed due to the long recovery time. Moreover, during the recovery phase, TCP freezes congestion window size until all lost packets are recovered, and this can make recovery time much longer resulting in performance degradation. To mitigate such recovery overhead, we propose Momentary recovery algorithm that recovers packet loss without an extra recovery phase. As other TCP and variants, our algorithm also halves the congestion window size when packet drop is detected, but it performs congestion avoidance phase immediately as if loss recovery is completed. For lost packets, TCP sender transmits them along with normal packets as long as congestion window permits rather than performs fast retransmission. In this manner, we can eliminate recovery overhead efficiently and reach steady state momentarily after network congestion. Finally, we provide a simulation based study on TCP recovery behaviors and confirm that our Momentary recovery algorithm always shows better performance compared with NewReno, SACK, and FACK.

IEEE 802.15.4 is a new standard, uniquely designed for low rate wireless personal area networks (LR-WPANs). It targets ultra-low complexity, cost, and power, for low-data-rate wireless connectivity. However, one of the main problems of this new standard is its insufficient, and inefficient, media access control (MAC) for priority data. This paper introduces an extended contention access period (XCAP) concept for priority packets, also an traffic adaptive contention differentiation utilizing the XCAP (TACDX). The TACDX determines appropriate transmission policy alternatively according to the traffic conditions and type of packet. TACDX achieves not only enhanced transmission for priority packets but it also has a high energy efficiency for the overall network. The proposed TACDX is verified with simulations to measure the performances.

This paper proposes adaptive transmit window control based on both location of mobile stations and traffic load for channel state based packet transmissions in CDMA cellular downlink communications. The proposed scheme constrains downlink packet transmissions by employing a transmit window individually given to each mobile station. The transmit window size is adjusted by using the optimum threshold value, which is selected with regard to both the mobile locations and the traffic load. The simulation results show that the proposed scheme improved the transmission delay and fairness of service compared with the conventional scheme.

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.

In this paper, we propose new methods which detect tampered-TCP connections at edge routers and protect well-behaved TCP connections from tampered-TCP connections, which results in fairness among TCP connections. The proposed methods monitor the TCP packets at an edge router and estimate the window size or the throughput for each TCP connection. By using estimation results, the proposed methods assess whether each TCP connection is tampered or not and drop packets intentionally if necessary to improve the fairness amongst TCP connections. From the results of simulation experiments, we confirm that the proposed methods can accurately identify tampered-TCP connections and regulate throughput ratio between tampered-TCP connections and competing TCP Reno connections to about 1.

In this paper, we propose a novel analysis method for large-scale networks with consideration of the behavior of the congestion control mechanism of TCP. In the analysis, we model the behavior of TCP at end-host and network link as independent systems, and combine them into a single system in order to analyze the entire network. Using this analysis, we can analyze a large-scale network, i.e. with over 100/1,000/10,000 routers/hosts/links and 100,000 TCP connections very rapidly. Especially, a calculation time of our analysis, it is different from that of ns-2, is independent of a network bandwidth and/or propagation delay. Specifically, we can derive the utilization of the network links, the packet loss ratio of the link buffer, the round-trip time (RTT) and the throughput of TCP connections, and the location and degree of the network congestion. We validate our approximate analysis by comparing analytic results with simulation ones. We also show that our analysis method treats the behavior of TCP connection in a large-scale network appropriately.

In this paper we propose a cross-layer retransmission control for TCP communication over a wireless link. With our proposed control, a retransmission delay for lost packet is reduced, packet losses in the wireless link are eliminated and all packets are delivered in the correct order. No change is required to TCP itself or to the sender. Our proposed method is implemented in a queue between the media access control (MAC) layer and logical link layer in a base station, and is designed to assist local retransmission control in the MAC layer. Computer simulations show that our proposed method can maximally use the bandwidth of the wireless link under high bit error rates conditions with conventional TCP control. The fairness problem of TCP communication between connections with different bit error rates in a wireless link is also improved, and MAC level fairness is also controllable.

Transmission schemes that gain content from multiple servers concurrently have been highlighted due to their ability to provide bandwidth aggregation, stability on dynamic server departure, and load balancing. Previous approaches employ parallel downloading in the transport layer to minimize the receiver buffer size and maximize bandwidth utilization. However, they only focus on the receiver operations and induce considerable overhead at the senders in contradiction to the main goal of a multi-provider environment, offloading popular servers through replication. In the present work, the authors propose MTCP, a novel transport layer protocol that focuses on reduction of the sender overhead through the elimination of unnecessary disk I/Os and efficient buffer cache utilization. MTCP also balances trade-off objectives to minimize buffering at receivers and maximize the request locality at senders.

In this letter we investigated the packet transmission control in downlink CDMA cellular systems. The downlink packet transmission control scheme based on the soft handoff status was proposed to enhance the system performance. The proposed scheme controls the downlink packet transmissions by employing a transmission window which is individually resolved to each mobile station according to its propagation condition and soft handoff status. Computer simulation shows that compared with the conventional scheme the proposed scheme improved the delay performance and fairness of service in packet reception.

In the present paper, ImTCP-bg, a new background TCP data transfer mechanism that uses an inline network measurement technique, is proposed. ImTCP-bg sets the upper limit of the congestion window size of the sender TCP based on the results of the inline network measurement, which measures the available bandwidth of the network path between the sender and receiver hosts. ImTCP-bg can provide background data transfer without affecting the foreground traffic, whereas previous methods cannot avoid network congestion. ImTCP-bg also employs an enhanced RTT-based mechanism so that ImTCP-bg can detect and resolve network congestion, even when reliable measurement results cannot be obtained. The performance of ImTCP-bg is investigated through simulations, and the effectiveness of ImTCP-bg in terms of the degree of interference with foreground traffic and the link bandwidth utilization is also investigated.

This paper focuses on the design and the performance evaluation of a p-persistent transmission control protocol that can enhance the IEEE 802.11 MAC, namely the p-persistent IEEE 802.11 DCF. Unlike the well-known p-persistent CSMA for modeling the legacy IEEE 802.11 MAC, the proposed protocol truly exploits the p-persistent transmission capability for this MAC. Moreover, the protocol is not restricted to IEEE 802.11 and, in fact, it can be executed on the top of a pre-existent access protocol without introducing additional overhead. When considered with WLAN, this protocol can optimize the throughput of the wireless network by setting the optimal transmission probability in the IEEE 802.11 MAC according to the throughput calculation given in this paper. The key characteristics of this protocol are represented by its simplicity, integration with the Standard, complete distribution, absence of modifications to the original IEEE 802.11 MAC frame format, and no requirement of extra messages being shared by the cooperating nodes. Analysis and simulation results confirm the effectiveness of the p-persistent protocol in achieving the optimal throughput and in improving the frame delay. In addition, the protocol can be easily extended to be a distributed priority mechanism, which requires further research.

It has been a very important issue to evaluate the performance of transmission control protocol (TCP), and the importance is still growing up because TCP will be deployed more widely in future wireless as well as wireline networks. It is also the reason why there have been a lot of efforts to analyze TCP performance more accurately. Most of these works are focusing on overall TCP end-to-end throughput that is defined as the number of bytes transmitted for a given time period. Even though each TCP's fast recovery strategy should be considered in computation of the exact time period, it has not been considered sufficiently in the existing models. That is, for more detailed performance analysis of a TCP implementation, the fast recovery latency during which lost packets are retransmitted should be considered with its relevant strategy. In this paper, we extend the existing models in order to capture TCP's loss recovery behaviors in detail. On the basis of the model, the loss recovery latency of three TCP implementations can be derived with considering the number of retransmitted packets. In particular, the proposed model differentiates the loss recovery performance of TCP using selective acknowledgement (SACK) option from TCP NewReno. We also verify that the proposed model reflects the precise latency of each TCP's loss recovery by simulations.

This paper introduces a new approach to realize a multi-state operation on the microwave isolator using ferrite edge-mode. The voltage control of total transmission on the isolator is realized. The operation is based on the unique property of ferrite edge-mode and the variable resistance of PIN diodes. On the isolator, the frequency response is investigated both experimentally and numerically. The numerical analysis is performed by the FDTD method. Both numerical and experimental results have shown that the transmission between two ports can be totally controlled by the applied voltage for the diodes. The experimental results indicate that the transmission direction can be controlled at 11 GHz, and the isolation ratio can be controlled for more than 30 dB.

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.