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.

For successful data collection in wireless sensor networks, it is important to ensure that the required delivery ratio is maintained while keeping a fair rate for every sensor. Furthermore, emerging high-rate applications might require complete reliability and the transfer of large volume of data, where persistent congestion might occur. These requirements demand a complete but efficient solution for data transport in sensor networks which reliably transports data from many sources to one or more sinks, avoids congestion and maintains fairness. In this paper, we propose congestion-aware and rate-controlled reliable transport (CRRT), an efficient and low-overhead data transport mechanism for sensor networks. CRRT uses efficient MAC retransmission to increase one-hop reliability and end-to-end retransmission for loss recovery. It also controls the total rate of the sources centrally, avoids the congestion in the bottleneck based on congestion notifications from intermediate nodes and centrally assigns the rate to the sources based on rate assignment policy of the applications. Performance of CRRT is evaluated in NS-2 and simulation results demonstrate the effectiveness of CRRT.

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.

Wireless technology has become widely popular and an important means of communication. A key issue in delivering wireless services is the problem of congestion which has an adverse impact on the Quality of Service (QoS), especially timeliness. Although a lot of work has been done in the context of RRM (Radio Resource Management), the deliverance of quality service to the end user still remains a challenge. Therefore there is need for a system that provides real-time services to the users through high assurance. We propose an intelligent agent-based approach to guarantee a predefined Service Level Agreement (SLA) with heterogeneous user requirements for appropriate bandwidth allocation in QoS sensitive cellular networks. The proposed system architecture exploits Case Based Reasoning (CBR) technique to handle RRM process of congestion management. The system accomplishes predefined SLA through the use of Retrieval and Adaptation Algorithm based on CBR case library. The proposed intelligent agent architecture gives autonomy to Radio Network Controller (RNC) or Base Station (BS) in accepting, rejecting or buffering a connection request to manage system bandwidth. Instead of simply blocking the connection request as congestion hits the system, different buffering durations are allocated to diverse classes of users based on their SLA. This increases the opportunity of connection establishment and reduces the call blocking rate extensively in changing environment. We carry out simulation of the proposed system that verifies efficient performance for congestion handling. The results also show built-in dynamism of our system to cater for variety of SLA requirements.

It is known that TCP data transfer in a wireless multihop network experiences a degradation in inter-connection fairness and throughput. This is because TCP is designed for use in wired networks, and the wireless multihop network has characteristics of sharing of the medium resources among nodes, which wired networks do not have. In particular, in wireless multihop networks where wireless nodes widely exist, hidden/exposed terminal problems are caused even if an RTS/CTS handshake is used. In this paper, two methods are proposed to improve fairness and throughput, without any feedback information from the intermediate nodes or cross-layer information. One method restricts the transfer period, while the other restrains the TCP congestion window. We evaluated these methods using computer simulations.

Vertical handoff is a new type of handoff that is triggered when a mobile node moves over heterogeneous wireless networks with each proving different access bandwidth, transmission latency, and coverage. A mobile node can achieve higher throughput by accessing a higher bandwidth providing wireless network. However, TCP has to experience drastic changes of the bandwidth and the latency due to the vertical handoff which must be recognized as a network congestion, and this degrades end-to-end performance. In this paper, we propose a TCP context switching scheme, named Context-Switching TCP, that maintains TCP variables separately for different types of wireless networks. Through simulations, Context-Switching TCP shows higher performance than TCP SACK for vertical handoff. Especially, it shows much higher performance gain when vertical handoff occurs frequently.

An important issue in designing a TCP congestion control algorithm is that it should allow the protocol to quickly adjust the end-to-end communication rate to the bandwidth on the bottleneck link. However, the TCP congestion control may function poorly in high bandwidth-delay product networks because of its slow response with large congestion windows. In this paper, we propose an enhanced version of TCP Vegas called Quick Vegas, in which we present an efficient congestion window control algorithm for a TCP source. Our algorithm improves the slow-start and congestion avoidance techniques of original Vegas. Simulation results show that Quick Vegas significantly improves the performance of connections as well as remaining fair when the bandwidth-delay product increases.

This proposes a scalable QoS control scheme, called Elephant Flow Control Scheme (EFCS) for high-speed large-capacity networks; it controls congestion and provides appropriate bandwidth to normal users' flows by controlling just the elephant flows. EFCS introduces a sampling packet threshold and drops packets considering flow size. EFCS also adopts a compensation parameter to control elephant flows to an appropriate level. Numerical results show that the sampling threshold increases control accuracy by 20% while reducing the amount of memory needed for packet sampling by 60% amount of memory by packet sampling; the elephant flows are controlled as intended by the compensation parameter. As a result, EFCS provides sufficient bandwidth to normal TCP flows in a scalable manner.

With the rapid development of wireless network technologies, heterogeneous networks with wired and wireless links are becoming common. However, the performance of TCP data transmission deteriorates significantly when a TCP connection traverses such networks, mainly because of packet losses caused by the high bit error rate of wireless links. Many solutions for this problem have been proposed in the past literature. However, most of them have various drawbacks, such as difficulties in their deployment by the wireless access network provider and end users, violation of TCP's end-to-end principle by splitting the TCP connection, or inapplicability to IP-level encrypted traffic because the base station needs to access the TCP header. In this paper, we propose a new mechanism without such drawbacks to improve the performance of TCP over wired and wireless heterogeneous networks. Our mechanism employs a receiver-based approach, which does not need modifications to be made to the sender TCP or the base station. It uses the ACK-splitting method for increasing the congestion window size quickly in order to restrain the throughput degradation caused by packet losses due to the high bit error rate of wireless links. We evaluate the performance of our mechanism and show that our mechanism can increase throughput by up to 94% in a UMTS network. The simulation results also show that our mechanism does not significantly deteriorate even when the receiver cannot perfectly distinguish whether packet losses are due to network congestion or bit errors on the wireless links.

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.

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.

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.

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.

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.

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.

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.

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.

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.