1,797 research outputs found

    On the quality of VoIP with DCCP for satellite communications

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    We present experimental results for the performance of selected voice codecs using DCCP with CCID4 congestion control over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs for a number of simultaneous calls using the ITU E-model. We analyse the sources of packet losses and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4. We also demonstrate the fairness of the proposed modifications to other flows. Although the recently adopted changes to TFRC specification alleviate some of the performance issues for VoIP on satellite links, we argue that the characteristics of commercial satellite links necessitate consideration of further improvements. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications

    Predicting expected TCP throughput using genetic algorithm

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    Predicting the expected throughput of TCP is important for several aspects such as e.g. determining handover criteria for future multihomed mobile nodes or determining the expected throughput of a given MPTCP subflow for load-balancing reasons. However, this is challenging due to time varying behavior of the underlying network characteristics. In this paper, we present a genetic-algorithm-based prediction model for estimating TCP throughput values. Our approach tries to find the best matching combination of mathematical functions that approximate a given time series that accounts for the TCP throughput samples using genetic algorithm. Based on collected historical datapoints about measured TCP throughput samples, our algorithm estimates expected throughput over time. We evaluate the quality of the prediction using different selection and diversity strategies for creating new chromosomes. Also, we explore the use of different fitness functions in order to evaluate the goodness of a chromosome. The goal is to show how different tuning on the genetic algorithm may have an impact on the prediction. Using extensive simulations over several TCP throughput traces, we find that the genetic algorithm successfully finds reasonable matching mathematical functions that allow to describe the TCP sampled throughput values with good fidelity. We also explore the effectiveness of predicting time series throughput samples for a given prediction horizon and estimate the prediction error and confidence.Peer ReviewedPostprint (author's final draft

    Optical Space Switches in Data Centers: Issues with Transport Protocols

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    A number of new architectures for data centre networks employing reconfigurable, SDN controlled, all-optical networks have been reported in recent years. In most cases, additional capacity was added to the system which unsurprisingly improved performance. In this study, a generalised network model that emulates the behaviour of these types of network was developed but where the total capacity is maintained constant so that system behaviour can be understood. An extensive emulated study is presented which indicates that the reconfiguration of such a network can have a detrimental impact on Transmission Control Protocol (TCP) congestion control mechanisms that can degrade the performance of the system. A number of simple scheduling mechanisms were investigated and the results show that an on-demand scheduling mechanism could deliver a throughput increase of more than ∼50% without any increase in total installed network capacity. These results, therefore, indicate the need to link the network resource management with new datacentre network architectures

    An Analysis of the Impact of Out-Of-Order Recovery Algorithms on MPTCP Throughput

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    Investigating TCP performance in mobile ad hoc networks

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    Mobile ad hoc networks (MANETs) have become increasingly important in view of their promise of ubiquitous connectivity beyond traditional fixed infrastructure networks. Such networks, consisting of potentially highly mobile nodes, have provided new challenges by introducing special consideration stemming from the unique characteristics of the wireless medium and the dynamic nature of the network topology. The TCP protocol, which has been widely deployed on a multitude of internetworks including the Internet, is naturally viewed as the de facto reliable transport protocol for use in MANETs. However, assumptions made at TCP’s inception reflected characteristics of the prevalent wired infrastructure of networks at the time and could subsequently lead to sub-optimal performance when used in wireless ad hoc environments. The basic presupposition underlying TCP congestion control is that packet losses are predominantly an indication of congestion in the network. The detrimental effect of such an assumption on TCP’s performance in MANET environments has been a long-standing research problem. Hence, previous work has focused on addressing the ambiguity behind the cause of packet loss as perceived by TCP by proposing changes at various levels across the network protocol stack, such as at the MAC mechanism of the transceiver or via coupling with the routing protocol at the network layer. The main challenge addressed by the current work is to propose new methods to ameliorate the illness-effects of TCP’s misinterpretation of the causes of packet loss in MANETs. An assumed restriction on any proposed modifications is that resulting performance increases should be achievable by introducing limited changes confined to the transport layer. Such a restriction aids incremental adoption and ease of deployment by requiring minimal implementation effort. Further, the issue of packet loss ambiguity, from a transport layer perspective, has, by definition, to be dealt with in an end-to-end fashion. As such, a proposed solution may involve implementation at the sender, the receiver or both to address TCP shortcomings. Some attempts at describing TCP behaviour in MANETs have been previously reported in the literature. However, a thorough enquiry into the performance of those TCP agents popular in terms of research and adoption has been lacking. Specifically, very little work has been performed on an exhaustive analysis of TCP variants across different MANET routing protocols and under various mobility conditions. The first part of the dissertation addresses this shortcoming through extensive simulation evaluation in order to ascertain the relative performance merits of each TCP variant in terms of achieved goodput over dynamic topologies. Careful examination reveals sub-par performance of TCP Reno, the largely equivalent performance of NewReno and SACK, whilst the effectiveness of a proactive TCP variant (Vegas) is explicitly stated and justified for the first time in a dynamic MANET environment. Examination of the literature reveals that in addition to losses caused by route breakages, the hidden terminal effect contributes significantly to non-congestion induced packet losses in MANETs, which in turn has noticeably negative impact on TCP goodput. By adapting the conservative slow start mechanism of TCP Vegas into a form suitable for reactive TCP agents, like Reno, NewReno and SACK, the second part of the dissertation proposes a new Reno-based congestion avoidance mechanism which increases TCP goodput considerably across long paths by mitigating the negative effects of hidden terminals and alleviating some of the ambiguity of non-congestion related packet loss in MANETs. The proposed changes maintain intact the end-to-end semantics of TCP and are solely applicable to the sender. The new mechanism is further contrasted with an existing transport layer-focused solution and is shown to perform significantly better in a range of dynamic scenarios. As solution from an end-to-end perspective may be applicable to either or both communicating ends, the idea of implementing receiver-side alterations is also explored. Previous work has been primarily concerned with reducing receiver-generated cumulative ACK responses by “bundling” them into as few packets as possible thereby reducing misinterpretations of packet loss due to hidden terminals. However, a thorough evaluation of such receiver-side solutions reveals limitations in common evaluation practices and the solutions themselves. In an effort to address this shortcoming, the third part of this research work first specifies a tighter problem domain, identifying the circumstances under which the problem may be tackled by an end-to-end solution. Subsequent original analysis reveals that by taking into account optimisations possible in wireless communications, namely the partial or complete omission of the RTS/CTS handshake, noticeable improvements in TCP goodput are achievable especially over long paths. This novel modification is activated in a variety of topologies and is assessed using new metrics to more accurately gauge its effectiveness in a wireless multihop environment

    Network Simulation Cradle

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    This thesis proposes the use of real world network stacks instead of protocol abstractions in a network simulator, bringing the actual code used in computer systems inside the simulator and allowing for greater simulation accuracy. Specifically, a framework called the Network Simulation Cradle is created that supports the kernel source code from FreeBSD, OpenBSD and Linux to make the network stacks from these systems available to the popular network simulator ns-2. Simulating with these real world network stacks reveals situations where the result differs significantly from ns-2's TCP models. The simulated network stacks are able to be directly compared to the same operating system running on an actual machine, making validation simple. When measuring the packet traces produced on a test network and in simulation the results are nearly identical, a level of accuracy previously unavailable using traditional TCP simulation models. The results of simulations run comparing ns-2 TCP models and our framework are presented in this dissertation along with validation studies of our framework showing how closely simulation resembles real world computers. Using real world stacks to simulate TCP is a complementary approach to using the existing TCP models and provides an extra level of validation. This way of simulating TCP and other protocols provides the network researcher or engineer new possibilities. One example is using the framework as a protocol development environment, which allows user-level development of protocols with a standard set of reproducible tests, the ability to test scenarios which are costly or impossible to build physically, and being able to trace and debug the protocol code without affecting results

    An Efficient Framework of Congestion Control for Next-Generation Networks

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    The success of the Internet can partly be attributed to the congestion control algorithm in the Transmission Control Protocol (TCP). However, with the tremendous increase in the diversity of networked systems and applications, TCP performance limitations are becoming increasingly problematic and the need for new transport protocol designs has become increasingly important.Prior research has focused on the design of either end-to-end protocols (e.g., CUBIC) that rely on implicit congestion signals such as loss and/or delay or network-based protocols (e.g., XCP) that use precise per-flow feedback from the network. While the former category of schemes haveperformance limitations, the latter are hard to deploy, can introduce high per-packet overhead, and open up new security challenges. This dissertation explores the middle ground between these designs and makes four contributions. First, we study the interplay between performance and feedback in congestion control protocols. We argue that congestion feedback in the form of aggregate load can provide the richness needed to meet the challenges of next-generation networks and applications. Second, we present the design, analysis, and evaluation of an efficient framework for congestion control called Binary Marking Congestion Control (BMCC). BMCC uses aggregate load feedback to achieve efficient and fair bandwidth allocations on high bandwidth-delaynetworks while minimizing packet loss rates and average queue length. BMCC reduces flow completiontimes by up to 4x over TCP and uses only the existing Explicit Congestion Notification bits.Next, we consider the incremental deployment of BMCC. We study the bandwidth sharing properties of BMCC and TCP over different partial deployment scenarios. We then present algorithms for ensuring safe co-existence of BMCC and TCP on the Internet. Finally, we consider the performance of BMCC over Wireless LANs. We show that the time-varying nature of the capacity of a WLAN can lead to significant performance issues for protocols that require capacity estimates for feedback computation. Using a simple model we characterize the capacity of a WLAN and propose the usage of the average service rate experienced by network layer packets as an estimate for capacity. Through extensive evaluation, we show that the resulting estimates provide good performance

    MMPTCP: a multipath transport protocol for data centers

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    Modern data centres provide large aggregate network capacity and multiple paths among servers. Traffic is very diverse; most of the data is produced by long, bandwidth hungry flows but the large majority of flows, which commonly come with strict deadlines regarding their completion time, are short. It has been shown that TCP is not efficient for any of these types of traffic in modern data centres. More recent protocols such MultiPath TCP (MPTCP) are very efficient for long flows, but are ill-suited for short flows. In this paper, we present MMPTCP, a novel transport protocol which, compared to TCP and MPTCP, reduces short flows' completion times, while providing excellent goodput to long flows. MMPTCP runs in two phases; initially, it randomly scatters packets in the network under a single congestion window exploiting all available paths. This is beneficial to latency-sensitive flows. After a specific amount of data is sent, MMPTCP switches to a regular MPTCP mode. MMPTCP is incrementally deployable in existing data centres as it does not require any modifications outside the transport layer and behaves well when competing with legacy TCP and MPTCP flows. Our extensive experimental evaluation shows that all design objectives for MMPTCP are met
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