Existing filtering methods of TCP ACK packets are known to be effective in reducing the required bandwidth, resulting in the improvement of TCP throughput. However, the methods cannot handle the filtering of piggyback ACK packets. Considering that most TCP applications require bidirectional data exchange, the lack of the functionality to deal with the piggyback ACK packets should be addressed. This paper proposes a novel filtering scheme for WiMAX systems that can handle the piggyback ACK packets. The novelty comes from the fact that the proposed method overlaps the processing time of packet merging with the round trip delay of the bandwidth request-and-grant procedure. It is advantageous because it does not require extra time for the merging. The results from an analytical model and simulations show that the required uplink bandwidth is decreased while the downlink throughput is increased.

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.

TCP's performance significantly degrades in multi-hop wireless networks because TCP's retransmission timeouts (RTOs) are frequently triggered regardless of congestion due to sudden delay and wireless transmission errors. Such RTOs non-related to congestions lead to TCP's unnecessary behaviors such as retransmitting all the outstanding packets which might be located in the bottleneck queue or reducing sharply its sending rate and increasing exponentially its back-off value even when the network is not congested. Since traditional TCP has no ability to identify if a RTO is triggered by congestion or not, it is unavoidable for TCP to underutilize available bandwidth by blindly reducing its sending rate for all the RTOs. In this paper, we propose an algorithm to detect the RTOs non-related to congestion in order to let TCP respond to the RTOs differently according to the cause. When a RTO is triggered, our algorithm estimates the queue usage in the network path during the go-back-N retransmissions, and decides if the RTO is triggered by congestion or not when the retransmissions end. If any RTO non-related to congestion is detected, our algorithm prevents TCP from increasing unnecessarily its back-off value as well as reducing needlessly its sending rate. Throughout the extensive simulation scenarios, we observed how frequently RTOs are triggered regardless of congestion, and evaluated our algorithm in terms of accuracy and goodput. The experiment results show that our algorithm has the highest accuracy among the previous works and the performance enhancement reaches up to 70% when our algorithm is applied to TCP.

The distribution of streaming multicast and real time audio/video applications in the Internet has been quickly increased in the Internet. Commonly, these applications rarely use congestion control and do not fairly share provided network capacity with TCP-based applications such as HTTP, FTP and emails. Therefore, Internet communities will be threatened by the increase of non-TCP-based applications that likely cause a significant increase of traffics congestion and starvation. This paper proposes a set of mechanisms, such as providing various data rates, background traffics, and various scenarios, to act friendly with TCP when sending multicast traffics. By using 8 scenarios of simulations, we use 6 layered multicast transmissions with background traffic Pareto with the shape factor 1.5 to evaluate performance metrics such as throughput, delay/latency, jitter, TCP friendliness, packet loss ratio, and convergence time. Our study shows that non TCP traffics behave fairly and respectful of the co-existent TCP-based applications that run on shared link transmissions even with background traffic. Another result shows that the simulation has low values on throughput, vary in jitter (0-10 ms), and packet loss ratio > 3%. It was also difficult to reach convergence time quickly when involving only non TCP traffics.

This letter investigates the space complexity of the sender buffer in a TCP variant, TCP-PR, to deal with packet reordering. Our finding is that with the SACK option used, TCP-PR requires the sender buffer of (β+1) pipesize where β indicates the number of RTTs that must pass before packet loss is detected.

Previous research shows that the IEEE 802.11 DCF channel contention mechanism is not capable of providing throughput fairness among nodes in different locations of the wireless mesh network. The node nearest the gateway will always strive for the chance to transmit data, causing fewer transmission opportunities for the nodes farther from the gateway, resulting in starvation. Prior studies modify the DCF mechanism to address the fairness problem. This paper focuses on the fairness study when TCP flows are carried over wireless mesh networks. By not modifying lower layer protocols, the current work identifies TCP parameters that impact throughput fairness and proposes adjusting those parameters to reduce frame collisions and improve throughput fairness. With the aid of mathematical formulation and ns2 simulations, this study finds that frame transmission from each node can be effectively controlled by properly controlling the delayed ACK timer and using a suitable advertised window. The proposed method reduces frame collisions and greatly improves TCP throughput fairness.

In multi-hop ad hoc wireless networks, it is well known that TCP suffers severe performance degradation. This is due to its window-based approach to transmission control, which injects traffic bursts into the network. These bursts increase the frequency of contention in the MAC layer which forces the dropping of some packets. This paper proposes an efficient TCP with pacing, Paced TCP, to alleviate MAC contention and thus achieve better performance than the traditional TCP variants. Our design approach is a TCP that probe the available bandwidth of the network without affecting the stability of the network. Simulations show that Paced TCP not only achieves better performance but is also friendly to UDP traffic.

We consider a single-link multi-source network with FAST TCP sources. We adopt a continuous-time dynamic model for FAST TCP sources, and propose a static model to adequately describe the queuing delay dynamics at the link. The proposed model turns out to have a structure that reveals the time-varying network feedback delay, which allows us to analyze FAST TCP with due consideration of the time-varying network feedback delay. Based on the proposed model, we establish sufficient conditions for the boundedness of congestion window of each source and for the global asymptotic stability. The asymptotic stability condition shows that the stability property of each source is affected by all other sources sharing the link. Simulation results illustrate the validity of the sufficient condition for the global asymptotic stability.

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.

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 letter proposes a new retransmission persistence management scheme for selective repeat automatic repeat request (SR-ARQ). By considering the overall traffic load that has to be managed by SR-ARQ, the proposed scheme arbitrates the retransmission persistence to prevent an abrupt delay increment due to excessive link-level local retransmissions. OPNET simulations show that SR-ARQ performs better with the proposed scheme than with a fixed value of retransmission persistence in terms of the throughput of transmission control protocol (TCP).

When TCP operates in multi-hop wireless networks, it suffers from severe performance degradation. This is because TCP reacts to wireless packet losses by unnecessarily decreasing its sending rate. Although previous loss differentiation algorithms (LDAs) can identify some of the packet losses due to wireless transmission errors as wireless losses, their accuracy is not high as much as we expect, and these schemes cannot avoid sacrificing the accuracy of congestion loss discrimination by misclassifying congestion losses as wireless losses. In this paper, we suggest a new end-to-end loss differentiation scheme which has high accuracy in both wireless loss discrimination and congestion loss discrimination. Our scheme estimates the rate of queue usage using information available to TCP. If the estimated queue usage is larger than 50% when a packet is lost, our scheme diagnoses the packet loss as congestion losses. Otherwise, it diagnoses the packet loss as wireless losses. Because the estimated queue usage is highly correlated to congestion, our scheme has an advantage to more exactly identify packet losses related to congestion and those unrelated to congestion. Through extensive simulations, we compare and evaluate our scheme with previous LDAs in terms of correlation, accuracy, and stability. And the results show that our scheme has the highest accuracy as well as its accuracy is more reliable than the other LDAs.

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.

This paper proposes an adaptive transmission control method for massive and intensive telecommunication traffic generated by communication-broadcasting integrated services. The proposed method adaptively controls data transmissions from viewers depending on the congestion states, so that severe congestion can be effectively avoided. Furthermore, it utilizes the broadcasting channel which is not only scalable, but also reliable for controlling the responses from vast numbers of viewers. The performance of the proposed method is evaluated through experiments on a test bed where approximately one million viewers are emulated. The obtained results quantitatively demonstrate the performance of the proposed method and its effectiveness under massive and intensive traffic conditions.

In this paper, we study the end-to-end TCP performance over a path deploying a High-Availability cluster, whose characteristics are highlighted by the failover procedure to remove single-point failure. This paper proposes an approach, called High-Availability Local Recovery (HALR), to enhance TCP performance in the face of a cluster failover. To minimize the latency of retransmission, HALR saves TCP packets selectively and resends them locally after the failover is finished. For better understanding, we further develop simple analytic models to predict the TCP performance in the aspect of flow latency under a range of failover times and the effects of HALR. Using simulation results, we validate our models and show that HALR improves the TCP performance significantly over a failover event as compared with the original TCP. Typically, HALR reduces the flow latency from 4.1 sec to less than 1.9 sec when the failover time equals to 500 ms. The simulation by real packet trace further demonstrates that the memory requirement of the proposed solution is not a concern for modern network equipments.

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 an extension to GridFTP that optimizes its performance by dynamically adjusting the number of parallel TCP connections. GridFTP has been used as a data transfer protocol to effectively transfer a large volume of data in Grid computing. GridFTP supports a feature called parallel data transfer that improves throughput by establishing multiple TCP connections in parallel. However, for achieving high GridFTP throughput, the number of TCP connections should be optimized based on the network status. In this paper, we propose an automatic parallelism tuning mechanism called GridFTP-APT (GridFTP with Automatic Parallelism Tuning) that adjusts the number of parallel TCP connections according to information available to the Grid middleware. Through simulations, we demonstrate that GridFTP-APT significantly improves the performance of GridFTP in various network environments.

In video streaming applications over the Internet, TCP-friendly rate control schemes are useful for improving network stability and inter-protocol fairness. However, they do not always guarantee a smooth video streaming. To simultaneously satisfy both the network and user requirements, video streaming applications should be quality-adaptive. In this paper, we propose a new quality adaptation mechanism to adjust the quality of congestion-controlled video stream by controlling the frame rate. Based on the current network condition, it controls the frame rate of video stream and the sending rate in a TCP-friendly manner. Through a simulation, we prove that our adaptation mechanism appropriately adjusts the quality of video stream while improving network stability.

A feature of services provided in a Next Generation Network (NGN) is that the end-to-end quality is guaranteed. This is quite a challenging issue, given the considerable fluctuation in network conditions within a Fixed Mobile Convergence (FMC) network. Therefore, a novel approach, whereby a network node and a mobile terminal such as a cellular phone cooperate with each other to control service quality is essential. In order to achieve such cooperation, the mobile terminal needs to become more intelligent so it can estimate the service quality, including the user's perceptual quality, and notify the measurement result to the network node. Subsequently, the network node implements some kind of service control function, such as a resource and admission control function, based on the notification from the mobile terminal. In this paper, the role of the mobile terminal in such collaborative system is focused on. As a part of a QoS/QoE measurement system, we describe an objective speech quality assessment with payload discrimination of lost packets to measure the user's perceptual quality of VoIP. The proposed assessment is so simple that it can be implemented on a cellular phone. We therefore did this as part of the QoS/QoE measurement system. By using the implemented system, we can measure the user's perceptual quality of VoIP as well as the network QoS metrics, in terms of criteria such as packet loss rate, jitter and burstiness in real time.

In this letter, we emphasize the performance associated problem of the TCP protocol in the wired-wireless networks. It is shown that the increase of TCP congestion window is strongly influenced by the wireless link. To accelerate the increase of TCP congestion window regardless of wireless link conditions we adopt a fast snoop agent that sends indirect acknowledgement to the sender. Simulation results show that the proposed scheme achieves higher throughput with small data size.