Bandwidth-delay Product Research Articles

VFAST TCP: A delay-based enhanced version of FAST TCP

This paper is aimed at describing a delay-based endto-end (e2e) congestion control algorithm, called Very FAST TCP (VFAST), which is an enhanced version of FAST TCP. The main idea behind this enhancement is to smoothly estimate the Round-Trip Time (RTT) based on a nonlinear filter, which eliminates throughput and queue oscillation when RTT fluctuates. In this context, an evaluation of the suggested scheme through simulation is introduced, by comparing our VFAST prototype with FAST in terms of throughput, queue behavior, fairness, stability, RTT and adaptivity to changes in network. The achieved simulation results indicate that the suggested protocol offer better performance than FAST TCP in terms of RTT estimation and throughput. Keywords—Fast tcp, RTT, delay estimation, delay-based congestion control, high speed TCP, large bandwidth delay product.

Loss-resilient window-based congestion control

This paper addresses the problem of fair allocation of bandwidth resources on lossy channels in hybrid heterogeneous networks. It discusses more particularly the ability of window-based congestion control to support non-congestion related losses. We investigate methods for efficient packet loss recovery by retransmission, and build on explicit congestion control mechanisms to decouple the packet loss detection from the congestion feedback signals. For different retransmission strategies that respectively rely on conventional cumulative acknowledgments or accurate loss monitoring, we show how the principles underlying the TCP retransmission mechanisms have to be adapted in order to take advantage of an explicit congestion feedback. A novel retransmission timer is proposed in order to deal with multiple losses of data segments and, in consequence, to allow for aggressive reset of the connection recovery timer. It ensures significant benefit from temporary inflation of the send-out window, and hence the fair share of bottleneck bandwidth between loss-prone and lossy connections. Extensive simulations analyze the performance of the new loss monitoring and recovery strategies, when used with two distinct explicit congestion control mechanisms. The first one relies on a coarse binary congestion notification from the routers. The second one, introduced in [D. Katabi, M. Handley, C. Rohrs, Internet congestion control for high bandwidth-delay product environments, ACM SIGCOMM (2002) 89-102], exploits accurate and finely-tuned router feedbacks to compute a precise congestion window adjustment. For both congestion control mechanisms, we observe that retransmissions triggered based on a precise monitoring of losses lead to efficient utilization of lossy links, and provide a fair share of the bottleneck bandwidth between heterogeneous connections, even for high loss ratios and bursty loss processes. Explicit window-based congestion control, combined with appropriate error control strategies, can therefore provide a valid solution to reliable and controlled connections over lossy network infrastructures.

TCP Performance in IEEE 802.11-Based Ad Hoc Networks with Multiple Wireless Lossy Links

We propose a packet-level model to investigate the impact of channel error on the transmission control protocol (TCP) performance over IEEE-802.11-based multihop wireless networks. A Markov renewal approach is used to analyze the behavior of TCP Reno and TCP Impatient NewReno. Compared to previous work, our main contributions are listed as follows: 1) modeling multiple lossy links, 2) investigating the interactions among TCP, Internet Protocol (IP), and media access control (MAC) protocol layers, specifically the impact of 802.11 MAC protocol and dynamic source routing (DSR) protocol on TCP throughput performance, 3) considering the spatial reuse property of the wireless channel, the model takes into account the different proportions between the interference range and transmission range, and 4) adopting more accurate and realistic analysis to the fast recovery process and showing the dependency of throughput and the risk of experiencing successive fast retransmits and timeouts on the packet error probability. The analytical results are validated against simulation results by using GloMoSim. The results show that the impact of the channel error is reduced significantly due to the packet retransmissions on a per-hop basis and a small bandwidth delay product of ad hoc networks. The TCP throughput always deteriorates less than ~ 10 percent, with a packet error rate ranging from 0 to 0.1. Our model also provides a theoretical basis for designing an optimum long retry limit for IEEE 802.11 in ad hoc networks.

Exploring TCP Parallelisation for performance improvement in heterogeneous networks

TCP Parallelisation uses a set of parallel (modified or standard) TCP connections to transfer data for an application process. With standard TCP connections, TCP Parallelisation has been used to effectively utilise bandwidth for data intensive applications over high bandwidth-delay product (BDP) networks. On the other hand, it has been argued that a single TCP connection with proper modification can emulate and capture the robustness of TCP Parallelisation and thus can well replace TCP Parallelisation. A typical example is HSTCP, which can emulate the behaviour of a set of multiple standard TCP connections. However, through the analysis and the proposed use of TCP Parallelisation in this paper, we found that TCP Parallelisation has its unique merits, which cannot be emulated by a single-connection based approach. As a result, the single-connection based approach (such as HSTCP) may not be able to achieve the same effects as TCP Parallelisation, especially in heterogeneous networks and highly dynamic environments (e.g., high BDP networks combined with wireless access links). In this paper, we demonstrate some of these unique merits of TCP Parallelisation (namely, localisation of disturbances such as Fast Recovery, loss synchronisation between parallel connections and self- healing) over single-connection based approach in the heterogeneous environment.

A Novel Onboard-gateway-based Mechanism to Improve TCP Performance in Aeronautical Satellite Networks

AbstractThe IP-based networks on aircraft serve to support Internet services via satellites. However, in aeronautical satellite hybrid networks, the TCP protocol performance often deteriorates due to improper decreases and slow recovery of the congestion window. This paper proposes a window size determination and notification mechanism, onboard-gateway-based mechanism (OGBM), which is based on the onboard gateway in the networks on aircraft. A cross-layer approach is adopted by the onboard gateway to obtain the satellite link bandwidth information. And then, by the gateway, through changing the receiver's advertised window field in ACK packets, TCP sources are notified of the window size of each TCP source calculated on the ground of bandwidth delay product and flow numbers. The mechanism is able to avoid improper changes of TCP window and serve multiple users. Simulation results show that the mechanism with the fairness index close to 1 improves TCP performance in aeronautical satellite networks.

TCP Performance Dynamics and Link-Layer Adaptation Based Optimization Methods for Wireless Networks

Almost a decade long research on the performance of TCP in wireless networks has resulted in many proposals and solutions to the problem of TCP throughput degradation. Several of these measures, however, have their share of drawbacks. With the continuing emergence of wireless technologies ever since the work on TCP performance over wireless began, smart link-layer mechanisms like adaptive modulation and coding, power control, and incremental redundancy have been designed and deployed. In this work, we outline a cross-layer optimization framework based on the congestion control dynamics of a bulk-transfer TCP flow and demonstrate its application to networks which offer link-layer adaptive measures. We begin by observing that the TCP's congestion window dynamics are comprised of certain recurring patterns which we term as cycles. We then overlay a TCP throughput optimization methodology that selects link-layer transmission modes (e.g. modulation scheme, coding rate, transmission power, or a combination thereof) in accordance with TCP dynamics and wireless channel conditions. We provide insights into the working of the optimization procedure which protects TCP segments against losses on the wireless channel when the TCP congestion window size (in bytes) is below the bandwidth-delay product of the network. The protection against wireless channel losses is rendered by the link-layer by employing robust modulation and coding schemes, high transmission power, etc. We show that TCP dynamics aware link adaptation measures lead to substantial enhancement of TCP throughput in EGPRS and IEEE 802.11a networks

TCP self-clocking and bandwidth sharing

We propose a simple queueing model for TCP transfers sharing a bottleneck link and examine its behavior when the buffer at the bottleneck is large compared to the bandwidth-delay product. This model explains some behaviors of TCP that have already been observed, and predicts other behaviors that are new. We present measurements that demonstrate these behaviors in the current Internet.

TCP Congestion Control with ACK-Pacing for Vertical Handover

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.

TCP-Ho: A Congestion Control Algorithm with Design and Performance Evaluation

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.

Limitations on nonlinear pulse propagation in coupled-resonator slow-light waveguides

Under the constraint of fixed pulse distortion, we study the bandwidth-delay product and nonlinear phase shift performance of coupled-resonator slow-light waveguides with designs that produce maximally-flat transmission and maximally-flat group delay responses. Even though improvement in bandwidth-delay product can be obtained with increasing number of resonators, the nonlinear response fails to improve beyond a certain number of resonators due to the increased filter bandwidth necessary to maintain a fixed pulse distortion. However, as expected, the nonlinear response improves with resonator finesse and degrades with resonator loss.

On window control algorithm for TCP efficiency over wireless networks with high bandwidth-delay product

The existing congestion control algorithms such as TCP Reno are not suitable for efficient utilisation of networks with a high bandwidth-delay product because it takes a very long time to achieve the full link utilisation. Furthermore, it is difficult to guarantee the fairness among the TCP connections with different round-trip times. To overcome these problems, a new window control algorithm using the buffer state and variable gains is proposed. The simulation results show that our algorithm improves the performance of TCP in the wireless networks.

An explicit router feedback framework for high bandwidth-delay product networks

Links with high bandwidth ranging from 1 to 10Gbps are increasingly in use worldwide. Congestion control with the positive use of router feedback that explicitly indicates network conditions is a promising way to address the performance issues of congestion control especially in such high-speed networks. In this paper, we propose SIRENS, a scalable, robust, and flexible fine-grained explicit router feedback framework. SIRENS is a per-hop and in-band notification scheme where each router captures a snapshot of the various kinds of downstream link status along the IP-level path from a sender to a receiver and notifies the receiver of the status. The receiver can find out the overall path status by assembling all the cumulative notifications that indicate the status in each of the single hops and by feedback can share this information with the sender. Such per-hop information is needed by end-hosts if we are to flexibly design novel congestion control mechanisms or to significantly improve the performance of conventional forms of congestion control. We show the feasibility of SIRENS in terms of router processing overhead through the development and evaluation of a network-processor-based high-performance network emulator. As a typical application of SIRENS, we show the configuration of TCP Limited Slow-Start. Through the implementation and evaluation, we have made sure that a sender can shift from the slow-start phase to the congestion avoidance phase without any packet drop, and can achieve more effective bandwidth utilization.

Free-Drop TCP

We introduce a new class of TCP congestion control algorithms that take a non-standard approach: instead of modifying AIMD parameters or exploiting traffic measurements, the new protocols modify the rule for deciding when to cut the congestion window. The class is defined by an additional window with a packet-count parameter w; the congestion window is reduced by half when a packet loss is detected, at time t say, if and only if there has been at least one dropped packet in the last w packet transmissions prior to time t . An algorithm in the class is called Free-Drop TCP , since dropped packets are "free" (they do not cause cuts in the window size) unless they are sufficiently bursty. We propose this new class as a means to achieve high utilizations in high bandwidth-delay product networks with small buffers. We analyze a fluid model which leads to explicit estimates of the average throughput for small loss probabilities. We then give the results of experiments, which show that, relative to TCP, a family of 'shifted' response functions of the form O (1/√ p - ε) can be obtained over a wide range of p by suitably varying w . Potential costs of these increases in throughput are also examined in terms of coefficents of variation and Jain's fairness measure. The costs range from negligible to moderate.

Performance evaluation of high speed TCP over optical burst switching networks

Study of TCP performance over OBS networks has been an important problem of research lately and it was found that due to the congestion control mechanism of TCP and the inherent bursty losses in the Optical Burst Switching (OBS) network, the throughput of TCP connections degrade. On the other hand, High Speed TCP (HSTCP) was proposed as an alternative to the use of TCP in high bandwidth-delay product networks. HSTCP aggressively increases the congestion window used in TCP, when the available bandwidth is high and decreases the window cautiously in response to a congestion event. In this work, we make a thorough simulation study of HSTCP over OBS networks. While the earlier works in the literature used a linear chain of nodes as the network topology for the simulation, we use the popular 14-node NSFNET topology that represents an arbitrary mesh network in our study. We also study the performance of HSTCP over OBS for different bandwidths of access networks. We use two different cases for simulations where in the first HSTCP connections are routed on disjoint paths while in the second they contend for resources in the network links. These cases of simulations along with the mesh topology help us clearly distinguish between the congestion and contention losses in the OBS network and their effect on HSTCP throughput. For completeness of study, we also simulate TCP traffic over OBS networks in all these cases and compare its throughput with that of HSTCP. We observe that irrespective of the access network bandwidth and the burst loss rate in the network, HSTCP outperforms TCP in terms of the throughput and robustness against multiple burst losses up to the expected theoretical burst loss probability of 10 −3.

A receiver-based vertical handover mechanism for TCP congestion control

We propose and evaluate a vertical handover mechanism for TCP, based on receiver bandwidth delay product (BDP) measurement and congestion control using the receiver's advertised window. It addresses the negative impact of abrupt changes in link capacity and latency on TCP performance during make-before-break vertical handover in heterogeneous networks, enabling TCP to seamlessly adapt to new conditions. It resolves the problem of buffer overflow in a low capacity network on handover from a high capacity network, and the under-utilisation problem on handover in the reverse direction. All modifications are restricted to the mobile device, and the proposed technique is fully interoperable with existing TCP and mobile IP infrastructure. It maintains end-to-end semantics, supports IP security, and can be easily deployed

REFWA: An Efficient and Fair Congestion Control Scheme for LEO Satellite Networks

This paper examines some issues that affect the efficiency and fairness of the Transmission Control Protocol (TCP), the backbone of Internet protocol communication, in multi-hops satellite network systems. It proposes a scheme that allows satellite systems to automatically adapt to any change in the number of active TCP flows due to handover occurrence, the free buffer size, and the bandwidth-delay product of the network. The proposed scheme has two major design goals: increasing the system efficiency, and improving its fairness. The system efficiency is controlled by matching the aggregate traffic rate to the sum of the link capacity and total buffer size. On the other hand, the system min-max fairness is achieved by allocating bandwidth among individual flows in proportion with their RTTs. The proposed scheme is dubbed Recursive, Explicit, and Fair Window Adjustment (REFWA). Simulation results elucidate that the REFWA scheme substantially improves the system fairness, reduces the number of packet drops, and makes better utilization of the bottleneck link. The results demonstrate also that the proposed scheme works properly in more complicated environments where connections traverse multiple bottlenecks and the available bandwidth may change over data transmission time

Gallop-Vegas: An enhanced slow-start mechanism for TCP Vegas

In this article, we present a new slow-start variant, which improves the throughput of transmission control protocol (TCP) Vegas. We call this new mechanism Gallop-Vegas because it quickly ramps up to the available bandwidth and reduces the burstiness during the slow-start phase. TCP is known to send bursts of packets during its slow-start phase due to the fast window increase and the ACK-clock based transmission. This phenomenon causes TCP Vegas to change from slow-start phase to congestion-avoidance phase too early in the large bandwidth-delay product (BDP) links. Therefore, in Gallop-Vegas, we increase the congestion window size with a rate between exponential growth and linear growth during slow-start phase. Our analysis, simulation results, and measurements on the Internet show that Gallop-Vegas significantly improves the performance of a connection, especially during the slow-start phase. Furthermore, it is implementation feasible because only sending part needs to be modified.

XCP for shared-access multi-rate media

The eXplicit Control Protocol (XCP) was developed to overcome some of the limitations of TCP, such as low utilization in high bandwidth delay product networks, unstable throughput, large queue build-up, and limited fairness. XCP, however, requires that each queue controller in a path knows the exact capacity of its link. In shared access media, e.g. IEEE 802.11, knowing the actual capacity of the channel is a difficult task.In this paper we propose modifications to the XCP algorithm that enable the utilization of XCP even when the capacity of a link is unknown. These modifications are validated through simulation.We also present the results of a comparison between the performance of the modified XCP and TCP, where XCP controlled flows result more stable, fairness increases, and the network delay becomes lower. In addition, as the bandwidth delay product increases, XCP is able to maintain near-maximum utilization while TCP decreases utilization.

Rate-adaptive snoop: a TCP enhancement scheme over rate-controlled lossy links

We study TCP performance over the wireless links deploying a wireless rate-control technique, whose link characteristics are identified by variable link rate and bursty transmission error. We present a TCP enhancement scheme, called rate-adaptive snoop (RA-Snoop). RA-Snoop caches TCP packets selectively based on the wireless channel condition and the cached packets are retransmitted locally over the wireless link in case corruption loss is detected. In addition, for effective adaptation to variable bandwidth, RA-Snoop calculates the window feedback based on the bandwidth-delay product estimation and the queue level, then conveys this feedback information on the receiver's advertised window field in the acknowledgements returning to TCP sources. We compare the performance of RA-Snoop with that of existing schemes in the aspect of goodput and fairness. Results from simulations reveal that RA-Snoop achieves significant improvements over the existing schemes for various traffic scenarios.

Impact of Bottleneck Queue Size on TCP Protocols and Its Measurement

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.