Congestion control is an important aspect of network technology, where traffic load is balanced to not cause an overflow in the system. Google has proposed its own protocol, QUIC, which is described as being set to supersede the TCP protocol. QUIC has several advantages, namely having high efficiency and low latency, but also a more flexible congestion control due to it being situated in the user space. To be used in tandem with QUIC, Google developed a new congestion control algorithm called BBR meant to fully exploit these advantages, by reducing latency and increasing throughput. However, as BBR is still a rather new algorithm, there are many different improvements possible to make it more efficient. In this paper, a modified BBR algorithm (mBBR) is proposed, which is comprised of three other algorithms meant to improve BBR by adjusting the otherwise static congestion window and pacing rate gain values based on the round-trip time flow, and is compared to the CUBIC, NewReno, and QUIC/TCP BBR algorithms. mBBR has a greatly lower RTT over CUBIC and NewReno, and reduces it by as much as 20% over the default QUIC BBR algorithm, while maintaining the same level of throughput. This improvement makes mBBR more suitable for usage in RAN-applications and other areas where a lower delay is crucial, without sacrificing network speeds.

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Yes / This paper presents a new analytical framework for the congestion control of Internet traffic using a queue threshold scheme. This framework includes two discrete-time analytical models for the performance evaluation of a threshold based congestion control mechanism and compares performance measurements through typical numerical results. To satisfy the low delay along with high throughput, model-I incorporates one threshold to make the arrival process step reduce from arrival rate ¿1 directly to ¿2 once the number of packets in the system has reached the threshold value L1. The source operates normally, otherwise. Model-II incorporates two thresholds to make the arrival rate linearly reduce from ¿1 to ¿2 with system contents when the number of packets in the system is between two thresholds L1 and L2. The source operates normally with arrival rate ¿1 before threshold L1, and with arrival rate ¿2 after the threshold L2. In both performance models, the mean packet delay W, probability of packet loss PL and throughput S have been found as functions of the thresholds and maximum drop probability. The performance comparison results for the two models have also been made through typical numerical results. The results clearly demonstrate how different load settings can provide different tradeoffs between throughput, loss probability and delay to suit different service requirements.

No / Traffic congestion arising from the shared nature of uplink channels in wireless networks can cause serious problems for the provision of QoS to various services. One approach to overcome these problems is to implement some effective congestion control mechanisms at the downlink buffer at the mobile network link layer or at gateways on the behalf of wireless network access points. Active queue management (AQM) is an effective mechanism to support end-to-end traffic congestion control in modern high-speed networks. Initially developed for Internet routers, AQM is now being also considered as an effective congestion control mechanism to enhance TCP performance over 3G links. This paper proposes an analytical performance model for AQM using various dropping functions. The selection of different dropping functions and threshold values required for this scheme plays a critical role on its effectiveness. The model uses a well-known Markov-modulated Poisson process (MMPP) to capture traffic burstiness and correlations. The validity of the model has been demonstrated through simulation experiments. Extensive analytical results have indicated that exponential dropping function is a good choice for AQM to support efficient congestion control.

In recent years, Internet and IP technologies have made inroads into almost every commu- nication market ranging from best-effort services such as email and Web, to soft real-time applications such as VoIP, IPTV, and video. However, providing a transport service over IP that meets the timeliness and availability requirements of soft real-time applications has turned out to be a complex task. Although network solutions such as IntServ, DiffServ, MPLS, and VRRP have been suggested, these solutions many times fail to provide a trans- port service for soft real-time applications end to end. Additionally, they have so far only been modestly deployed. In light of this, this thesis considers transport protocols for soft real-time applications. Part I of the thesis focuses on the design and analysis of transport protocols for soft real- time multimedia applications with lax deadlines such as image-intensive Web applications. Many of these applications do not need a completely reliable transport service, and to this end Part I studies so-called partially reliable transport protocols, i.e., transport protocols that enable applications to explicitly trade reliability for improved timeliness. Specifically, Part I investigates the feasibility of designing retransmission-based, partially reliable transport protocols that are congestion aware and fair to competing traffic. Two transport protocols are presented in Part I, PRTP and PRTP-ECN, which are both extensions to TCP for partial reliability. Simulations and theoretical analysis suggest that these transport protocols could give a substantial improvement in throughput and jitter as compared to TCP. Additionally, the simulations indicate that PRTP-ECN is TCP friendly and fair against competing congestion- aware traffic such as TCP flows. Part I also presents a taxonomy for retransmission-based, partially reliable transport protocols. Part II of the thesis considers the Stream Control Transmission Protocol (SCTP), which was developed by the IETF to transfer telephony signaling traffic over IP. The main focus of Part II is on evaluating the SCTP failover mechanism. Through extensive experiments, it is suggested that in order to meet the availability requirements of telephony signaling, SCTP has to be configured much more aggressively than is currently recommended by IETF. Fur- thermore, ways to improve the transport service provided by SCTP, especially with regards to the failover mechanism, are suggested. Part II also studies the effects of Head-of-Line Blocking (HoLB) on SCTP transmission delays. HoLB occurs when packets in one flow block packets in another, independent, flow. The study suggests that the short-term effects of HoLB could be substantial, but that the long-term effects are marginal.

Active Queue Management (AQM) aims to achieve high link utilization, low queuing delay and low loss rate in routers. However, it is difficult to adapt AQM parameters to constantly provide desirable transient and steady-state performance under highly dynamic network scenarios. They need to be a trade-off made between queuing delay and utilization. The queue size would become unstable when round-trip time or link capacity increases, or would be unnecessarily large when round-trip time or link capacity decreases. Effective ways of adapting AQM parameters to obtain good performance have remained a critical unsolved problem during the last fifteen years. This thesis firstly investigates existing AQM algorithms and their performance. Based on a previously developed dynamic model of TCP behaviour and a linear feedback model of TCP/RED, Auto-Parameterization RED (AP-RED) is proposed which unveils the mechanism of adapting RED parameters according to measurable network conditions. Another algorithm of Statistical Tuning RED (ST-RED) is developed for systematically tuning four key RED parameters to control the local stability in response to the detected change in the variance of the queue size. Under variable network scenarios like round-trip time, link capacity and traffic load, no manual parameter configuration is needed. The proposed ST-RED can adjust corresponding parameters rapidly to maintain stable performance and keep queuing delay as low as possible. Thus the sensitivity of RED's performance to different network scenarios is removed. This Statistical Tuning algorithm can be applied to a PI controller for AQM and a Statistical Tuning PI (ST-PI) controller is also developed. The implementation of ST-RED and ST-PI is relatively straightforward. Simulation results demonstrate the feasibility of ST-RED and ST-PI and their capabilities to provide desirable transient and steady-state performance under extensively varying network conditions.

In computer networks, congestion is a condition in which one or more egressinterfaces are offered more packets than are forwarded at any given instant [1]. In wireless sensor networks, congestion can cause a number of problems including packet loss, lower throughput and poor energy efficiency. These problems can potentially result in a reduced deployment lifetime and underperforming applications. Moreover, idle radio listening is a major source of energy consumption therefore low-power wireless devices must keep their radio transceivers off to maximise their battery lifetime. In order to minimise energy consumption and thus maximise the lifetime of wireless sensor networks, the research community has made significant efforts towards power saving medium access control protocols with Radio Duty Cycling. However, careful study of previous work reveals that radio duty cycle schemes are often neglected during the design and evaluation of congestion control algorithms. This thesis argues that the presence (or lack) of radio duty cycle can drastically influence the performance of congestion control mechanisms. To investigate if previous findings regarding congestion control are still applicable in IPv6 over low power wireless personal area and duty cycling networks; some of the most commonly used congestion detection algorithms are evaluated through simulations. The research aims to develop duty cycle aware congestion control schemes for IPv6 over low power wireless personal area networks. The proposed schemes must be able to maximise the networks goodput, while minimising packet loss, energy consumption and packet delay. Two congestion control schemes, namely DCCC6 (Duty Cycle-Aware Congestion Control for 6LoWPAN Networks) and CADC (Congestion Aware Duty Cycle MAC) are proposed to realise this claim. DCCC6 performs congestion detection based on a dynamic buffer. When congestion occurs, parent nodes will inform the nodes contributing to congestion and rates will be readjusted based on a new rate adaptation scheme aiming for local fairness. The child notification procedure is decided by DCCC6 and will be different when the network is duty cycling. When the network is duty cycling the child notification will be made through unicast frames. On the contrary broadcast frames will be used for congestion notification when the network is not duty cycling. Simulation and test-bed experiments have shown that DCCC6 achieved higher goodput and lower packet loss than previous works. Moreover, simulations show that DCCC6 maintained low energy consumption, with average delay times while it achieved a high degree of fairness. CADC, uses a new mechanism for duty cycle adaptation that reacts quickly to changing traffic loads and patterns. CADC is the first dynamic duty cycle pro- tocol implemented in Contiki Operating system (OS) as well as one of the first schemes designed based on the arbitrary traffic characteristics of IPv6 wireless sensor networks. Furthermore, CADC is designed as a stand alone medium access control scheme and thus it can easily be transfered to any wireless sensor network architecture. Additionally, CADC does not require any time synchronisation algorithms to operate at the nodes and does not use any additional packets for the exchange of information between the nodes (For example no overhead). In this research, 10000 simulation experiments and 700 test-bed experiments have been conducted for the evaluation of CADC. These experiments demonstrate that CADC can successfully adapt its cycle based on traffic patterns in every traffic scenario. Moreover, CADC consistently achieved the lowest energy consumption, very low packet delay times and packet loss, while its goodput performance was better than other dynamic duty cycle protocols and similar to the highest goodput observed among static duty cycle configurations.

In our work, we have proposed a new framework for the classification and comparative study of ATM congestion control schemes. The different aspects on which we have classified the algorithms are control theoretic approach, action and congestion notification. These three aspects present of the classification present a coherent framework on which congestion control algorithms are to be classified. Such a classification will also help in developing new algorithms.

Esta dissertação apresenta a construção de um protótipo de servidor Web distribuído, baseado no modelo de servidor Web com diferenciação de serviços (SWDS) e a implementação e avaliação de algoritmos de seleção, utilizando o conceito de controle de congestionamento para requisições HTTP. Com isso, além de implementar uma plataforma de testes, este trabalho também avalia o comportamento de dois algoritmos de controle de congestionamento. Os dois algoritmos estudados são chamados de Drop Tail e RED (Random Early Detection), no qual são bastante difundidos na literatura científica e aplicados em redes de computadores. Os resultados obtidos demostram que, apesar das particularidades de cada algoritmo, existe uma grande relação entre tempo de resposta e a quantidade de requisições aceitas / This MSc dissertation presents the implementation of a prototype for a distributed web server based on the SWDS, a model for a web server with service differentiation, and the implementation and evaluation of selection algorithms adopting the concept of congestion control for HTTP requests. Thus, besides implementing a test platform this work also evaluates the behavior of two congestion control algorithms. The two algorithms studied are the Drop Tail and the RED (Random Early Detection), which are frequently discussed in the scientific literature and widely applied in computer networks. The results obtained show that, although the particularities of each algorithm, there is a strong relation between the response times and the amount of requests accepted in the server

Made available in DSpace on 2015-03-05T13:53:45Z (GMT). No. of bitstreams: 0 Previous issue date: 3 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Aplicações como a World-Wide Web, correio eletrônico e transferência de arquivos são a principal fonte de tráfego em redes de comunicação entre computadores hoje em dia. A tecnologia IP, base da Internet, não dispõe de reservas de recursos, e a divisão justa da capacidade da rede entre os fluxos que competem se dá através de mecanismos que devem estar presentes nos protocolos executados nas estações fim (hosts). Tal é denominado controle de congestionamento fim a fim. O tráfego na Internet é predominantemente composto por fluxos TCP, que é dotado de controle de congestionamento. Para o bom funcionamento da Internet, novos protocolos de transporte ou de aplicação devem incluir mecanismos de controle de congestionamento que sejam "amigáveis" ao TCP, ou seja, que não ocupem mais recursos do que deveriam. O surgimento de IP multicast viabilizou, no final nos anos 90, novas aplicações na Internet, como aplicações de trabalho em grupo, bancos de dados distribuídos, vídeo-conferência, etc. Muitas dessas aplicações