232 research outputs found
On bursty packet loss model for TCP performance analysis
In this paper, we study the timeout probability of TCP Reno under the bursty packet loss model, which is widely used to represent the loss characteristics of TCP under drop-tail FIFO queues. With a detailed analysis on the three timeout reasons for TCP Reno, we show that the impact of timeout has been underestimated in the existing literature. Surprisingly, we find that this more precise representation of timeout probability does not match the actual performance of TCP under drop-tail FIFO queues. Therefore we conclude that the bursty loss model is incapable of capturing the behavior of drop-tail FIFO queues, and using bursty loss model to analyze TCP performance is flawed. © 2005 IEEE.published_or_final_versio
Throughput modeling of TCP with slow-start and fast recovery
Despite the rich literature on modeling TCP, we find two common deficiencies with the existing approaches. First, none of the work gives sufficient treatment to slow-start, although almost all of them show that retransmission timeout events are common. Second, the probability that retransmission timeout occurs has been underestimated, because retransmission timeout is coupled with fast recovery but fast recovery has not been properly modeled in the previous work. In this paper, new analytical models for predicting the steady state throughput of TCP flows are proposed. All major TCP mechanisms, including slow-start, congestion avoidance, fast retransmit, and fast recovery, are jointly considered under both bursty and independent loss models. We show that our proposed throughput models capture TCP performance more accurately. © 2005 IEEE.published_or_final_versio
TCP smart framing: a segmentation algorithm to reduce TCP latency
TCP Smart Framing, or TCP-SF for short, enables the Fast Retransmit/Recovery algorithms even when the congestion window is small. Without modifying the TCP congestion control based on the additive-increase/multiplicative-decrease paradigm, TCP-SF adopts a novel segmentation algorithm: while Classic TCP always tries to send full-sized segments, a TCP-SF source adopts a more flexible segmentation algorithm to try and always have a number of in-flight segments larger than 3 so as to enable Fast Recovery. We motivate this choice by real traffic measurements, which indicate that today's traffic is populated by short-lived flows, whose only means to recover from a packet loss is by triggering a Retransmission Timeout. The key idea of TCP-SF can be implemented on top of any TCP flavor, from Tahoe to SACK, and requires modifications to the server TCP stack only, and can be easily coupled with recent TCP enhancements. The performance of the proposed TCP modification were studied by means of simulations, live measurements and an analytical model. In addition, the analytical model we have devised has a general scope, making it a valid tool for TCP performance evaluation in the small window region. Improvements are remarkable under several buffer management schemes, and maximized by byte-oriented schemes
On the impact of link layer retransmission schemes on TCP over 4G satellite links
We study the impact of reliability mechanisms introduced at the link layer on the performance of transport protocols in the context of 4G satellite links. Specifically, we design a software module that performs realistic analysis of the network performance, by utilizing real physical layer traces of a 4G satellite service. Based on these traces, our software module produces equivalent link layer traces, as a function of the chosen link layer reliability mechanism. We further utilize the link layer traces within the ns-2 network simulator to evaluate the impact of link layer schemes on the performance of selected Transmission Control Protocol (TCP) variants. We consider erasure coding, selective-repeat automatic request (ARQ) and hybrid-ARQ link layer mechanisms, and TCP Cubic, Compound, Hybla, New Reno and Westwood. We show that, for all target TCP variants, when the throughput of the transport protocol is close to the channel capacity, using the ARQ mechanism is most beneficial for TCP performance improvement. In conditions where the physical channel error rate is high, hybrid-ARQ results in the best performance for all TCP variants considered, with up to 22% improvements compared to other schemes
Transport Control Protocol (TCP) over Optical Burst Switched Networks
Transport Control Protocol (TCP) is the dominant protocol in modern communication networks, in which the issues of reliability, flow, and congestion control must be handled efficiently. This thesis studies the impact of the next-generation bufferless optical burst-switched (OBS) networks on the performance of TCP congestion-control implementations (i.e., dropping-based, explicit-notification-based, and delay-based).
The burst contention phenomenon caused by the buffer-less nature of OBS occurs randomly and has a negative impact on dropping-based TCP since it causes a false indication of network congestion that leads to improper reaction on a burst drop event. In this thesis we study the impact of these random burst losses on dropping-based TCP throughput. We introduce a novel congestion control scheme for TCP over OBS networks, called Statistical Additive Increase Multiplicative Decrease (SAIMD). SAIMD maintains and analyzes a number of previous round trip times (RTTs) at the TCP senders in order to identify the confidence with which a packet-loss event is due to network congestion. The confidence is derived by positioning short-term RTT in the spectrum of long-term historical RTTs. The derived confidence corresponding to the packet loss is then taken in to account by the policy developed for TCP congestion-window adjustment.
For explicit-notification TCP, we propose a new TCP implementation over OBS networks, called TCP with Explicit Burst Loss Contention Notification (TCP-BCL). We examine the throughput performance of a number of representative TCP implementations over OBS networks, and analyze the TCP performance degradation due to the misinterpretation of timeout and packet-loss events. We also demonstrate that the proposed TCP-BCL scheme can counter the negative effect of OBS burst losses and is superior to conventional TCP architectures in OBS networks.
For delay-based TCP, we observe that this type of TCP implementation cannot detect network congestion when deployed over typical OBS networks since RTT fluctuations are minor. Also, delay-based TCP can suffer from falsely detecting network congestion when the underlying OBS network provides burst retransmission and/or deflection. Due to the fact that burst retransmission and deflection schemes introduce additional delays for bursts that are retransmitted or deflected, TCP cannot determine whether this sudden delay is due to network congestion or simply to burst recovery at the OBS layer. In this thesis we study the behaviour of delay-based TCP Vegas over OBS networks, and propose a version of threshold-based TCP Vegas that is suitable for the characteristics of OBS networks. The threshold-based TCP Vegas is able to distinguish increases in packet delay due to network congestion from burst contention at low traffic loads.
The evolution of OBS technology is highly coupled with its ability to support upper-layer applications. Without fully understanding the burst transmission behaviour and the associated impact on the TCP congestion-control mechanism, it will be difficult to exploit the advantages of OBS networks fully
Integration of Linux TCP and Simulation: Verification, Validation and Application
Network simulator has been acknowledged as one of the most flexible means in studying and developing protocol as it allows virtually endless numbers of simulated network environments to be setup and protocol of interest to be fine-tuned without requiring any real-world complicated and costly network experiment. However, depending on researchers, the same protocol of interest can be developed in different ways and different implementations may yield the outcomes that do not accurately capture the dynamics of the real protocol. In the last decade, TCP, the protocol on which the Internet is based, has been extensively studied in order to study and reevaluate its performance particularly when TCP based applications and services are deployed in an emerging Next Generation Network (NGN) and Next Generation Internet (NGI). As a result, to understand the realistic interaction of TCP with new types of networks and technologies, a combination of a real-world TCP and a network simulator seems very essential. This work presents an integration of real-world TCP implementation of Linux TCP/IP network stack into a network simulator, called INET. Moreover, verification and validation of the integrated Linux TCP are performed within INET framework to ensure the validity of the integration. The results clearly confirm that the integrated Linux TCP displays reasonable and consistent dynamics with respect to the behaviors of the real-world Linux TCP. Finally, to demonstrate the application of the INET with Linux TCP extension, algorithms of other Linux TCP variants and their dynamic over a large-bandwidth long-delay network are briefly presented
On the Modeling of TCP Latency and Throughput
In this thesis, a new model for the slow start phase based on the discrete evolutions of congestion window is developed, and we integrate this part into the improved TCP steady state model for a better prediction performance. Combining these short and steady state models, we propose an extensive stochastic model which can accurately predict the throughput and latency of the TCP connections as functions of loss rate, round-trip time (RTT), and file size. We validate our results through simulation experiments. The results show that our model?s predictions match the simulation results better than the Padhye and Cardwell\u27s stochastic models, about 75% improvement in the accuracy of performance predictions for the steady state and 20% improvement for the short-lived TCP flows
QoS evaluation of different TCPs congestion control algorithm using NS2
The success of the current Internet relies to a large
extent on cooperation between the users and network.
The network signals its current state to the users by
marking or dropping packets. The user then strives to
maximize the sending rate without causing network
congestion. To achieve this, the users implement a flow
control algorithm that controls the rate at which data
packets are sent into the Internet. More specifically, the
Transmission Control Protocol (TCP) is used by the
users to adjust the sending rate in response to changing
network conditions. In this paper, we focus on the
degree of fairness provided to TCP connections by
comparing two packet-scheduling algorithms at the
router. The first one is FIFO (First In First Out, or
Drop-Tail), which is widely used in the current Internet
routers because of its simplicity. The second is RED
(Random Early Detection), which drops incoming
packets at a certain probability
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