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

ECN verbose mode: a statistical method for network path congestion estimation

This article introduces a simple and effective methodology to determine the level of congestion in a network with an ECN-like marking scheme. The purpose of the ECN bit is to notify TCP sources of an imminent congestion in order to react before losses occur. However, ECN is a binary indicator which does not reflect the congestion level (i.e. the percentage of queued packets) of the bottleneck, thus preventing any adapted reaction. In this study, we use a counter in place of the traditional ECN marking scheme to assess the number of times a packet has crossed a congested router. Thanks to this simple counter, we drive a statistical analysis to accurately estimate the congestion level of each router on a network path. We detail in this paper an analytical method validated by some preliminary simulations which demonstrate the feasibility and the accuracy of the concept proposed. We conclude this paper with possible applications and expected future work

Performance of active multicast congestion control

This paper aims to provide insight into the behavior of congestion control mechanisms for reliable multicast protocols. A multicast congestion control based on active networks has been proposed and simulated using ns-2 over a network topology obtained using the Tiers tool. The congestion control mechanism has been simulated under different network conditions and with different settings of its configuration parameters. The objective is to analyze its performance and the impact of the different configuration parameters on its behavior. The simulation results show that the performance of the protocol is good in terms of delay and bandwidth utilization. The compatibility of the protocol with TCP flows has not been demonstrated, but the simulations performed show that by altering the parameter settings, the proportion of total bandwidth taken up by the two types of flow, multicast and TCP, may be modified.Publicad

Multimedia congestion control: circuit breakers for unicast RTP sessions

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms

A quantitative analysis and performance study of fast congestion notification (FN) mechanism

Congestion in computer network happens when the number of transmission requests exceeds the transmission capacity at a certain network point (called a bottle-neck resource) at a specific time. Congestion usually causes buffers overflow and packets loss. The purpose of congestion management is to maintain a balance between the transmission requests and the transmission capacity so that the bottle-neck resources operate on an optimal level, and the sources are offered service in a way that assures fairness. Fast Congestion Notification (FN) is one of the proactive queue management mechanisms that limits the queuing delay and achieves the maximum link utilization possible with minimum packet drops. In this paper we present a detailed performance comparison of the Linear FN algorithm to RED based on the results obtained through simulations. The paper shows how FN can be tuned for different window size (Ws) and periods of time constant (T) to achieve higher link utilization; reduce the queuing delay, and lower packet drop ratio

RED behavior with different packet sizes

We consider the adaptation of random early detection (RED) as a buffer management algorithm for TCP traffic in Internet gateways where different maximum transfer units (MTUs) are used. We studied the two RED variants described in [4] and point out a weakness in both. The first variant where drop probability is independent from the packet size discriminates connections with smaller MTUs. The second variant results in a very high packet loss ratio (PLR), and as a consequence low goodput, for connections with higher MTUs. We show that fairness in terms of loss and goodput can be supplied through an appropriate setting of the RED algorithm.Comment: 13 pages, submitted to IEEE symposium on computer communications (ISCC2000