In this paper, we discuss TCP performance in a wireless overlay network where wireless LANs and cellular networks are integrated. In the overlay network, vertical handover, where a mobile node changes its access link during a session, is one of the most important technologies. When a vertical handover occurs, throughput performance of a TCP flow is degraded due to not only packet losses during the handover, but drastic change of its bandwidth-delay product. In this paper, we propose an ACK-pacing mechanism for TCP congestion control to improve the performance degradation. The proposed system is receiver-driven, so no modification is required to the mechanism of TCP sender. In the proposed system, a TCP receiver adjusts a transmission rate of ACKs according to the relationship between bandwidth-delay products before and after a handover. Since the ACK-clocking mechanism of TCP adjusts the transmission rate of TCP segments, the TCP receiver can seamlessly adjust its congestion window size to the new bandwidth-delay product. Computer simulation results show that the proposed system can improve the TCP performance during the vertical handover.

A critical design issue of Transmission Control Protocol (TCP) is its congestion control that allows the protocol to adjust the end-to-end communication rate based on the detection of packet loss. However, TCP congestion control may function poorly during its slow start and congestion avoidance phases. This is because TCP sends bursts of packets with the fast window increase and the ACK-clock based transmission in slow start, and respond slowly with large congestion windows especially in high bandwidth-delay product (BDP) networks during congestion avoidance. In this article, we propose an improved version of TCP, TCP-Ho, that uses an efficient congestion window control algorithm for a TCP source. According to the estimated available bandwidth and measured round-trip times (RTTs), the proposed algorithm adjusts the congestion window size with a rate between exponential growth and linear growth intelligently. Our extensive simulation results show that TCP-Ho significantly improves the performance of connections as well as remaining fair and stable when the BDP increases. Furthermore, it is feasible to implement because only sending part needs to be modified.

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.

Two congestion control schemes designed specifically to handle changes in the datalink interface of a mobile host are presented. The future mobile environment is expected to involve multimode connectivity to the Internet and dynamic switching of the connection mode depending on network conditions. The conventional Transmission Control Protocol (TCP), however, is unable to maintain stable and efficient throughput across such interface changes. The two main issues are the handling of the change in host Internet Protocol (IP) address, and the reliability and continuity of TCP flow when the datalink interface changes. Although existing architectures addressing the first issue have already been proposed, the problem of congestion control remains. In this paper, considering a large change in bandwidth when the datalink interface changes, two new schemes to address these issues are proposed. The first scheme, Immediate Expiration of Timeout Timer, detects interface changes and begins retransmission immediately without waiting for a retransmission timeout as in existing architectures. The second scheme, Bandwidth-Aware Slow Start Threshold, detects the interface change and estimates the new bandwidth so as to set an appropriate slow start threshold for retransmission. Through simulations, the proposed schemes are demonstrated to provide marked improvements in performance over existing architectures.

In this letter, we suggest a rate-based supervisory congestion control scheme for the ad hoc networks that use TCP as the transport protocol. This scheme makes it possible for the TCP sender to distinguish the causes of packet loss. In addition, this scheme guarantees the fair sharing of the available bandwidth among the connections. We show the reliability of our scheme by using the supervisory control framework and simulations confirm the effectiveness of our scheme.

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.

For backward compatibility, ECN-capable networks should be capable of handling both ECN-capable and ECN-incapable TCP flows. In this letter, we present a backward congestion notification (BCN) scheme that can provide fast congestion indication delivery, while improving fairness between ECN-capable and ECN-incapable flows. Simulation results reveal that the BCN scheme is more effective than the original ECN mechanism in terms of stability, throughput, and fairness.

Since congestion is very likely to happen in the Internet, locating congested areas (path segments) along a congested path is vital to appropriate actions by Internet Service Providers to mitigate or prevent network performance degradation. We propose a practical method to locate congested segments by actively measuring one-way end-to-end packet losses on appropriate paths from multiple origins to multiple destinations, using a network tomographic approach. Then we conduct a long-term experiment measuring packet losses on multiple paths over the Japanese commercial Internet. The experimental results indicate that the proposed method is able to precisely locate congested segments. Some findings on congestion over the Japan Internet are also given based on the experiment.

The most important design goal in Optical Burst Switching (OBS) networks is to reduce burst loss resulting from resource contention. Especially, the higher the congestion degree in the network is, the higher the burst loss rate becomes. The burst loss performance can be improved by employing a judicious congestion control. In this paper, to actively avoid contentions, we propose a peak load-based congestion control scheme that operates based on the highest (called peak load) of the loads of all links over the path between each pair of ingress and egress nodes in an OBS network. Simulation results show that the proposed scheme reduces the burst loss rate significantly, compared to existing OBS protocols, while maintaining reasonable throughput and fairness.

In this paper, based on Equivalent Active Flow, we propose a novel technique called Approximate Fairness Dropping, which is able to approximate fairness by containing misbehaving flows' access queue opportunity with low time/space complexity. Unlike most of the existing Active Queue Management schemes (e.g., RED, BLUE, CHOKE), Approximate Fairness Dropping does not drop the packets whose arriving rate is within the maximum admitted rate, so it protects the well-behaving flows against misbehaving ones, moreover, improves the throughput and decreases the queuing delay. Our simulations and analyses demonstrate that this new technique outperforms the existing schemes and closely approximates the "ideal" case, where full state information is needed.

Network congestion and random errors of wireless link are two well-known noteworthy parameters which degrade the TCP performance over heterogeneous networks. We put forward a novel end-to-end TCP congestion control mechanism, namely TCP BaLDE (Bandwidth and Loss Differentiation Estimate), in which the TCP congestion control categorizes the reason of the packet loss by estimating loss differentiation in order to control the packet transmission rate appropriately. While controlling transmission rate depends on the available bandwidth estimation which is apprehended by the bandwidth estimation algorithm when the sender receives a new ACK with incipient congestion signal, duplicates ACKs or is triggered by retransmission timeout event. Especially, this helps the sender to avoid router queue overflow by opportunely entering the congestion avoidance phase. In simulation, we experimented under numerous different network conditions. The results show that TCP BaLDE can achieve robustness in aspect of stability, accuracy and rapidity of the estimate in comparison with TCP Westwood, and tolerate ACK compression. It can achieve better performance than TCP Reno and TCP Westwood. Moreover, it is fair on bottleneck sharing to multiple TCP flows of the same TCP version, and friendly to existing TCP version.

This paper introduces a network-aware video delivery framework where the quality-of-service (QoS) interaction between prioritized packet video and relative differentiated service (DiffServ) network is taken into account. With this framework, we propose a dynamic class mapping (DCM) scheme to allow video applications to cope with service degradation and class-based resource constraint in a time-varying network environment. In the proposed scheme, an explicit congestion notification (ECN)-based feedback mechanism is utilized to notify the status of network classes and the received service quality assessment to the end-host applications urgently. Based on the feedback information, DCM agent at ingress point can dynamically re-map each packet onto a network class in order to satisfy the desired QoS requirement. Simulation results verify the enhanced QoS performance of the streaming video application by comparing the static class-mapping and the class re-mapping based on loss-driven feedback.

In this letter we examine two transport protocols, HighSpeed TCP [1] and Scalable TCP [2] which are both sender-side varieties of TCP. Based on the fluid flow theory, we develop a general nonlinear model and use gain margin and phase margin to evaluate the stability of a closed-loop system which is composed of a transport protocol and an active queue management scheme. Our results indicate that HSTCP and STCP are stabler than standard TCP when link bandwidth, flow number and round-trip time vary.

During devastating natural disasters, numerous people want to make calls to check on their families and friends in the stricken areas, but many call attempts on mobile cellular systems are blocked due to limited radio frequency resources. To reduce call blocking and enable as many people as possible to access mobile cellular systems, placing a limit on the holding time for each call has been studied [1],[2]. However, during a catastrophe, emergency calls, e.g., calls to fire, ambulance, or police services are also highly likely to increase and it is important that the holding time for these calls is not limited. A method of limiting call holding time to make provision for emergency calls while considering the needs of ordinary callers is proposed. In this method, called the HTL-E method, all calls are classified as emergency calls or other according to the numbers that are dialed or the terminal numbers that are given in advance to the particular terminals making emergency calls, and only the holding time of other calls is limited. The performance characteristics of the HTL-E method were evaluated using computer simulations. The results showed that it reduced the rates of blocking and forced call termination at handover considerably, without reducing the holding time for emergency calls. The blocking rate was almost equal for emergency and other calls. In addition, the HTL-E method handles fluctuations in the demand for emergency calls flexibly. A simple method of estimating the holding-time limit for other calls, which reduces the blocking rate for emergency and other calls to the normal rate for periods of increased call demand is also presented. The calculated results produced by this method agreed well with the simulation results.

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.

The queue size at a bottleneck would impact the performance of TCP protocols, especially when running a single TCP flow in networks with a large bandwidth-delay product. However, queue size has been not well considered in experiments. This paper shows how bottleneck queue size influences TCP protocols performance. Bursityness of advanced TCPs is examined. Ways of estimating queue size are introduced. Sending a UDP packet train until a loss is detected is a method to measure queuing delay to estimate queue size. Watching a loss in a TCP session to measure round trip time and calculate the queue size is also discussed. Results from experiments with a network emulator and a real network are reported. The results indicated that a layer-2 switch at a congestion point would be a major factor of decreasing TCP performance in a fast long distant path.

Congestion control in the Internet consists of two main components: the TCP Additive-Increase Multiplicative-Decrease (AIMD) mechanism on sending windows implemented by end-users, and the Active Queue Management (AQM) scheme implemented in the routers which improves the effectiveness of congestion control. TCP connection is regarded as a feedback control system. Comparably, AQM is classified as a flow controller. There are several kinds of time delays in the network, such as propagation delay, queuing delay in the buffer of the router, etc. The time delays cause degradation of performance and instability of the network. A Smith Predictor is commonly used in feedback control of plants with significant time delays to implement effective compensation. In this paper, a Smith Predictor-based PI-controller for AQM (SPPA) is proposed, which uses a TCP reference model and an average Round-Trip Time (RTT) to reduce unfavorable effects of time delays in TCP networks. The drop probability is calculated by a Proportional-Integral (PI) controller based on the prediction error. When a mismatch exists in between the actual model of the TCP process and the reference model employed by the SPPA, we demonstrate conditions under which the network is stable. The performance, robustness and effectiveness of the proposed SPPA are all evaluated using simulations. The performance of the SPPA is compared with some typical AQMs, such as the Adaptive RED, the PI-controller, and the Proportional-Differential (PD) controller.

In this letter, we propose a protocol sensitive random early detection algorithm for active queue management to improve fairness between TCP and UDP flows and to reduce delay time with small overheads. The algorithm classifies the packets into responsive and unresponsive flows, and applies the RED algorithm individually to each classified group. Using ns-2 simulations, we showed the effectiveness of the proposed PSRED algorithm compared with several well-known AQM schemes, such as RED and RED-PD algorithms.

Adaptive Virtual Queue (AVQ) introduces a novel implementation algorithm for Active Queue Management (AQM). The stability criterion for AVQ was deduced in literature [1], but it lacks practicability due to the difficulty of solving the transcendental equation. In this letter, the AVQ stability is further investigated based on the characteristic roots of delay-differential equation. Another stability criterion explicitly associated with parameters of network configuration is deduced and the upper bound of delay time for stable AVQ algorithm is determined. Finally, the conclusion is validated through simulation experiments.

TCP performance in the IEEE 802.11-based multihop ad hoc networks is extremely poor, because the congestion control mechanism of TCP cannot effectively deal with the problem of packet drops caused by mobility and shared channel contention among wireless nodes. In this paper, we present a cross-layer method, which adaptively adjusts the TCP maximum window size according to the number of RTS (Request To Send) retry counts of the MAC layer at the TCP sender, to control the number of TCP packets in the network and thus decrease the channel contention. Our simulation results show that this method can remarkably improve TCP throughput and its stability.