Modeling and estimation techniques for understanding heterogeneous traffic behavior

The majority of current internet traffic is based on TCP. With the emergence of new applications, especially new multimedia applications, however, UDP-based traffic is expected to increase. Furthermore, multimedia applications have sparkled the development of protocols responding to congestion while behaving differently from TCP. As a result, network traffc is expected to become more and more diverse. The increasing link capacity further stimulates new applications utilizing higher bandwidths of future. Besides the traffic diversity, the network is also evolving around new technologies. These trends in the Internet motivate our research work. In this dissertation, modeling and estimation techniques of heterogeneous traffic at a router are presented. The idea of the presented techniques is that if the observed queue length and packet drop probability do not match the predictions from a model of responsive (TCP) traffic, then the error must come from non-responsive traffic; it can then be used for estimating the proportion of non-responsive traffic. The proposed scheme is based on the queue length history, packet drop history, expected TCP and queue dynamics. The effectiveness of the proposed techniques over a wide range of traffic scenarios is corroborated using NS-2 based simulations. Possible applications based on the estimation technique are discussed. The implementation of the estimation technique in the Linux kernel is presented in order to validate our estimation technique in a realistic network environment

Performance Analysis of Transactional Traffic in Mobile Ad-hoc Networks

Mobile Ad Hoc networks (MANETs) present unique challenge to new protocol design, especially in scenarios where nodes are highly mobile. Routing protocols performance is essential to the performance of wireless networks especially in mobile ad-hoc scenarios. The development of new routing protocols requires com- paring them against well-known protocols in various simulation environments. The protocols should be analysed under realistic conditions including, but not limited to, representative data transmission models, limited buffer space for data transmission, sensible simulation area and transmission range combination, and realistic moving patterns of the mobiles nodes. Furthermore, application traffic like transactional application traffic has not been investigated for domain-specific MANETs scenarios. Overall, there are not enough performance comparison work in the past literatures. This thesis presents extensive performance comparison among MANETs comparing transactional traffic including both highly-dynamic environment as well as low-mobility cases

Transport Layer Optimizations for Heterogeneous Wireless Multimedia Networks

The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.Ph.D.Committee Chair: Vijay Madisetti; Committee Member: Raghupathy Sivakumar; Committee Member: Sudhakar Yalamanchili; Committee Member: Umakishore Ramachandran; Committee Member: Yucel Altunbasa

Modeling and estimation techniques for understanding heterogeneous traffic behavior

The majority of current internet traffic is based on TCP. With the emergence of new applications, especially new multimedia applications, however, UDP-based traffic is expected to increase. Furthermore, multimedia applications have sparkled the development of protocols responding to congestion while behaving differently from TCP. As a result, network traffc is expected to become more and more diverse. The increasing link capacity further stimulates new applications utilizing higher bandwidths of future. Besides the traffic diversity, the network is also evolving around new technologies. These trends in the Internet motivate our research work. In this dissertation, modeling and estimation techniques of heterogeneous traffic at a router are presented. The idea of the presented techniques is that if the observed queue length and packet drop probability do not match the predictions from a model of responsive (TCP) traffic, then the error must come from non-responsive traffic; it can then be used for estimating the proportion of non-responsive traffic. The proposed scheme is based on the queue length history, packet drop history, expected TCP and queue dynamics. The effectiveness of the proposed techniques over a wide range of traffic scenarios is corroborated using NS-2 based simulations. Possible applications based on the estimation technique are discussed. The implementation of the estimation technique in the Linux kernel is presented in order to validate our estimation technique in a realistic network environment

Attaining Realistic Simulations of Mobile Ad-hoc NETworks

Mobile Ad-hoc Networks (MANET) are comprised of wireless systems that communicate without the assistance of centrally managed routers or base stations. MANET research and development has increased due to computing technologies offering smaller, faster, smarter, and more power efficient platforms to operate on. Largely the testing and evaluation of new and existing MANET protocols has resided in simulation environments. This is due in part to the complexities and expenses incurred when conducting real world tests. Many researchers have come to recognize that these current simulations tend to assume away critical components of the MANET domain. These assumptions are made either to simplify the physical layer of the simulation so that the protocol can be tested or out of necessity because the current simulation platforms are not capable of providing a more realistic physical layer simulation environment. This thesis is focused on addressing these assumptions that affect the physical layer of the MANET protocol by gathering data in the real world and then modifying the simulation environment to model as closely as possible to the gathered results. This modified environment is then compared to the basic MANET simulation environment by analyzing packet delivery and propagation effects of both models

Downstream Bandwidth Management for Emerging DOCSIS-based Networks

In this dissertation, we consider the downstream bandwidth management in the context of emerging DOCSIS-based cable networks. The latest DOCSIS 3.1 standard for cable access networks represents a significant change to cable networks. For downstream, the current 6 MHz channel size is replaced by a much larger 192 MHz channel which potentially can provide data rates up to 10 Gbps. Further, the current standard requires equipment to support a relatively new form of active queue management (AQM) referred to as delay-based AQM. Given that more than 50 million households (and climbing) use cable for Internet access, a clear understanding of the impacts of bandwidth management strategies used in these emerging networks is crucial. Further, given the scope of the change provided by emerging cable systems, now is the time to develop and introduce innovative new methods for managing bandwidth. With this motivation, we address research questions pertaining to next generation of cable access networks. The cable industry has had to deal with the problem of a small number of subscribers who utilize the majority of network resources. This problem will grow as access rates increase to gigabits per second. Fundamentally this is a problem on how to manage data flows in a fair manner and provide protection. A well known performance issue in the Internet, referred to as bufferbloat, has received significant attention recently. High throughput network flows need sufficiently large buffer to keep the pipe full and absorb occasional burstiness. Standard practice however has led to equipment offering very large unmanaged buffers that can result in sustained queue levels increasing packet latency. One reason why these problems continue to plague cable access networks is the desire for low complexity and easily explainable (to access network subscribers and to the Federal Communications Commission) bandwidth management. This research begins by evaluating modern delay-based AQM algorithms in downstream DOCSIS 3.0 environments with a focus on fairness and application performance capabilities of single queue AQMs. We are especially interested in delay-based AQM schemes that have been proposed to combat the bufferbloat problem. Our evaluation involves a variety of scenarios that include tiered services and application workloads. Based on our results, we show that in scenarios involving realistic workloads, modern delay-based AQMs can effectively mitigate bufferbloat. However they do not address the other problem related to managing the fairness. To address the combined problem of fairness and bufferbloat, we propose a novel approach to bandwidth management that provides a compromise among the conflicting requirements. We introduce a flow quantization method referred to as adaptive bandwidth binning where flows that are observed to consume similar levels of bandwidth are grouped together with the system managed through a hierarchical scheduler designed to approximate weighted fairness while addressing bufferbloat. Based on a simulation study that considers many system experimental parameters including workloads and network configurations, we provide evidence of the efficacy of the idea. Our results suggest that the scheme is able to provide long term fairness and low delay with a performance close to that of a reference approach based on fair queueing. A further contribution is our idea for replacing `tiered\u27 levels of service based on service rates with tiering based on weights. The application of our bandwidth binning scheme offers a timely and innovative alternative to broadband service that leverages the potential offered by emerging DOCSIS-based cable systems

Evaluation of Probabilistic Early Response TCP (PERT) for Video Delivery and Extension with ACK Coalescing

This thesis demonstrates the performance of Probabilistic Early Response TCP (PERT), a new TCP congestion control, for video streaming. As a delay based protocol, it measures the delay at the end host and adjusts the congestion window accordingly. Our experiments show that PERT improves video delivery performance by decreasing the fraction of packets delivered late. Furthermore, our Linux live streaming test indicates that PERT is able to reduce the playback glitches, when high resolution video is delivered over a link with non-zero packet loss. In order to operate PERT at higher thoughputs, we design PERT to work with Acknowledgement (ACK) coalescing at the receiver. ACK coalescing makes data transfers burstier and makes it hard to estimate delays accurately. We apply TCP pacing to fix this issue, and validate its effectiveness in the aspects of throughput, packet loss and fairness. Our experiment results also show that PERT with Delayed ACK and Pacing is more friendly, and therefore more suitable when multiple traffic flows are competing for limited bottleneck bandwidth or sharing the same router buffer

Enterprise networks (modern techniques for analysis, measurement and performance improvement)

Dans l'évaluation d'Internet au cours des années, un grand nombre d'applications apparaissent, avec différentes exigences de service en termes de bande passante, délai et ainsi de suite. Pourtant, le trafic Internet présente encore une propriété de haute variabilité. Plusieurs études révèlent que les flux court sont en général liés à des applications interactives-pour ceux-ci, on s'attend à obtenir de bonne performance que l'utilisateur perçoit, le plus souvent en termes de temps de réponse court. Cependant, le schéma classique FIFO/drop-tail déployé des routeurs/commutateurs d'aujourd'hui est bien connu de parti pris contre les flux courts. Pour résoudre ce problème sur un réseau best-effort, nous avons proposé un nouveau et simple algorithme d'ordonnancement appelé EFD (Early Flow Discard). Dans ce manuscrit, nous avons d'abord évaluer la performance d'EFD dans un réseau câblé avec un seul goulot d'étranglement au moyen d'étendu simulations. Nous discutons aussi des variantes possibles de EFD et les adaptations de EFD à 802.11 WLAN - se réfèrent principalement à EFDACK et PEFD, qui enregistre les volumes échangés dans deux directions ou compte simplement les paquets dans une direction, visant à améliorer l'équité à niveau flot et l'interactivité dans les WLANs. Enfin, nous nous consacrons à profiler le trafic de l'entreprise, en plus de elaborer deux modèles de trafic-l'une qui considère la structure topologique de l'entreprise et l'autre qui intègre l'impact des applications au-dessus de TCP - pour aider à évaluer et à comparer les performances des politiques d'ordonnancement dans les réseaux d'entreprise classiques.As the Internet evolves over the years, a large number of applications emerge with varying service requirements in terms of bandwidth, delay, loss rate and so on. Still, the Internet traffic exhibits a high variability property the majority of the flows are of small sizes while a small percentage of very long flows contribute to a large portion of the traffic volume. Several studies reveal that small flows are in general related to interactive applications for which one expects to obtain good user perceived performance, most often in terms of short response time. However, the classical FIFO/drop-tail scheme deployed in today s routers/switches is well known to bias against short flows over long ones. To tackle this issue over a best-effort network, we have proposed a novel and simple scheduling algorithm named EFD (Early Flow Discard). In this manuscript, we first evaluate the performance of EFD in a single-bottleneck wired network through extensive simulations. We then discuss the possible variants of EFD and EFD s adaptations to 802.11 WLANs mainly refer to EFDACK and PEFD. Finally, we devote ourselves to profiling enterprise traffic, and further devise two workload models - one that takes into account the enterprise topological structure and the other that incorporates the impact of the applications on top of TCP - to help to evaluate and compare the performance of scheduling policies in typical enterprise networks.PARIS-Télécom ParisTech (751132302) / SudocSudocFranceF