Traffic Management and Traffic Engineering for the Future Internet

Mobile TV is growing beyond the stage of experimentation and evaluation and is (about) to become part of our daily lives. Additionally, it is being delivered through heterogeneous networks and to a variety of receiving devices, which implies different versions of one and the same video content must be transported. We propose two (approximate) analytic methods for capacity demand estimation in a (mobile) TV broadcast system. In particular, the methods estimate the required transport capacity for a bouquet of channels offered on request and in different versions (video formats or in different quality) over a multicast-enabled network, encoded in non-constant bit rate targeting constant quality. We compare a transport strategy where the different versions (of one channel) are simulcast to a scalable video encoding (SVC) transport strategy, where all resolutions (of one channel) are embedded in one flow. In addition, we validate the proposed analytic methods with simulations. A realistic mobile TV example is considered with two transported resolutions of the channels: QVGA and VGA. We demonstrate that not always capacity gain is achieved with SVC as compared to simulcast since the former comes with some penalty rate and the gain depends on the system parameters.

This paper presents a scheme for reallocating bandwidth in path-oriented transport networks. At random time points, bandwidth is allocated to those paths that (possibly temporarily) value it most highly. The scheme acts according to local rules and without centralised control. The proposed scheme is thus distributed and scalable. Previous studies have shown that bandwidth reallocation, together with the provision of appropriate amounts of spare capacity on certain links, can be used to rapidly deploy and capacitate equivalent recovery routes in the event of network equipment failure. The purpose of this study is to determine if the same reallocation mechanism can also deal effectively with repeated, small scale random traffic fluctuations and with time varying traffics. We present a simulation model of a 30-node 46-link network which we use to evaluate the efficacy of the bandwidth reallocation scheme. The simulation study shows that bandwidth reconfiguration can substantially reduce the connection loss probabilities.

The current data network scenario makes Traffic Engineering (TE) a very challenging task. The ever growing access rates and new applications running on end-hosts result in more variable and unpredictable traffic patterns. By providing origin-destination (OD) pairs with several possible paths, load-balancing has proven itself an excellent tool to face this uncertainty. Most previous proposals defined the load-balancing problem as minimizing a certain network cost function of the link’s usage, assuming users would obtain a good performance as a consequence. Since the network operator is interested in the communication between the OD nodes, we propose instead to state the load-balancing problem in their terms. We define a certain utility function of the OD’s perceived performance and maximize the sum over all OD pairs. The solution to the resulting optimization problem can be obtained by a distributed algorithm, whose design we outline. By means of extensive simulations with real networks and traffic matrices, we show that our approach results in more available bandwidth for OD pairs and a similar or decreased maximum link utilization than previously proposed load-balancing schemes. Packet-level simulations verify the algorithm’s good performance in the presence of delayed and inexact measurements.

This paper presents the impact of congestion control mechanisms proposed for Flow-Aware Networks on packet transmission in the overloaded network after a link failure. The results of simulation based analysis show how to set the values of the congestion control parameters in order to decrease the acceptance time of the interrupted streaming flows in the backup link. The research was performed for three congestion control mechanisms, the Enhanced Flushing Mechanism (EFM), the Remove Active Elastic Flows (RAEF), and Remove and Block Active Elastic Flows (RBAEF) in two different cross-protect router architectures, with the PFQ (Priority Fair Queuing) and with the PDRR (Priority Deficit Round Robin) scheduling algorithms. Moreover, the advantages and weaknesses of using the proposed solutions in FAN, considering the effects of a network element failure, are described and analyzed.

The development of realistic Internet traffic models of applications and services calls for a good understanding of the nature of Internet flows, which can be affected by many factors. Especially relevant among these are the limitations imposed by link capacities and router algorithms that control bandwidth on a per-flow basis. In the paper, we perform a statistical analysis of an Internet traffic trace that specifically takes into account the upper-bounds on the duration and rate of measured flows. In particular, we propose a new model for studying the dependencies between the logarithm of the size, the logarithm of the duration, and the logarithm of the transmission rate of an Internet flow. We consider a bivariate lognormal distribution for the flow size and flow duration, and derive estimators for the mean, the variance and the correlation, based on a truncated domain that reflects the upper-bounds on the duration and rate of measured flows. Moreover, we obtain regression equations that describe the expected value of one characteristic (size, duration or rate) given other (size or duration), thus providing further insights on the dependencies between Internet flow characteristics. In particular, for flows with large sizes we are able to predict durations and rates that are coherent with the upper-bound on transmission rates imposed by the network.

In the last few years, peer-to-peer (P2P) file-sharing applications have become very popular: more users are continuously joining such systems and more objects are being made available, seducing even more users to join. Today, the traffic generated by P2P systems accounts for a major fraction of the Internet traffic and is bound to increase. P2P networks are currently evolving towards a real time multimedia content distribution able to provide reliable IPTV and Video on Demand (VoD) services. These new services create several challenges, the most important one being how to efficiently share resources in an heterogeneous, dynamic and continuously evolving group of peers. A detailed peer-level characterization is therefore fundamental for the technical and marketing design, deployment and management of these new services.

This paper will be focused on the characterization of BitTorrent as one of the most important P2P applications and, from that analysis, will try to extrapolate the main characteristics and behaviours of the potential P2P real-time multimedia content distribution clients. Specifically, the paper analyses the geographical localization of the involved peers, the peers’ availability and the Round Trip Time (RTT) between peers as a function of the time and for different categories of shared files.

The crucial future role of Internet in society makes of network monitoring a critical issue for network operators in future network scenarios. The Future Internet will have to cope with new and different anomalies, motivating the development of accurate detection algorithms. This paper presents a novel approach to detect unexpected and large traffic variations in data networks. We introduce an optimal volume anomaly detection algorithm in which the anomaly-free traffic is treated as a nuisance parameter. The algorithm relies on an original parsimonious model for traffic demands which allows detecting anomalies from link traffic measurements, reducing the overhead of data collection. The performance of the method is compared to that obtained with the Principal Components Analysis (PCA) approach. We choose this method as benchmark given its relevance in the anomaly detection literature. Our proposal is validated using data from an operational network, showing how the method outperforms the PCA approach.

This paper focuses on traffic engineering of telecommunication networks, which arises in the context of switched Ethernet networks. It addresses the minimization of the maximum network link load. With the IEEE 802.1s Multiple Spanning Tree Protocol, it is possible to define multiple routing spanning trees to provide multiple alternatives to route VLAN traffic demands. Two compact mixed integer linear programming models defining the optimization problem and several models based on the Dantzig-Wolfe decomposition principle, which are solved by branch-and-price, are proposed and compared. The different decompositions result from defining as subproblems either the supporting spanning trees and/or the demand routing paths, which can be solved by well known efficient algorithms.

We show that an undirected network has the local tree-restoration property if and only if it is 2-connected. In particular, we provide a quite simple procedure for assigning costs to the arcs of a 2-connected network so that the property holds. For the directed case, we show that deciding whether a network has the local tree-restoration property is NP-hard, even in some “simple” cases.

A method is proposed to estimate traffic matrices in the presence of long-range dependent traffic, while the methods proposed so far for that task have been designed for short-range dependent traffic. The method employs the traffic measurements on links and provides the maximum likelihood estimate of both the traffic matrix and the Hurst parameter. It is “blind”, i.e. it does not exploit any model neither for the traffic intensity values (e.g. the gravity model) nor for the mean-variance relationship (e.g. the power-law model). In the application to a sample network the error on traffic intensities decays rapidly with the traffic intensity down to below 30%. The estimation error of the Hurst parameter can be reduced to a few percentage points with a proper choice of the measurement interval.

Future Internet will be based in heterogeneous infrastructures with a large landscape of access technologies. Moreover, the terminals are being equipped with multiple interfaces, enabling the simultaneous connection to the different available technologies. In such multihoming environments, the traffic control mechanisms will require complex operations, as it is now possible to implement the concept of “always best connected” (ABC), using the devices and access technologies that best suit communications needs, users and networks. This paper presents a performance study of a traffic control mechanism that is able to perform context-aware and personalized network selection, using context and preferences, network and terminal characteristics, to determine the best access connection for each terminal and service. The results show the benefits of using such an algorithm in the network, mainly in terms of QoS in these multiservice technologies, and address the influence of specific criteria and constraints considered in the decision process.

The rapid growth in client application demands, in terms of bandwidth and Quality of Service (QoS), has motivated the deployment of the optical technology at Metro Access and Metro Core Networks. A new Ring-based Optical Network Architecture is being designed by Alcatel-Lucent with a view to offer a cost effective solution that provides also some added values in terms of reliability and performance. In this paper, we propose a packet size-based scheduling algorithm and a priority-based Packet Erasing Mechanism (PEM). These two mechanisms improve the performance (in terms of packet loss ratios and mean access delays) of the proposed optical ring architecture. Finally, through simulation works we present a performance analysis and comparison of three different optical ring architectures in competition. Obtained results provide interesting conclusions on the behavior and performance tendencies for each one of the studied architectures.

This work focuses on improving the performance of video communications based on the recently developed H.264 Scalable Video Coding (SVC) standard over 802.11e wireless networks. The H.264/SVC standard is particularly suitable for wireless communications because of its compression efficiency and the ability to adapt to different network scenarios. The adoption of the H.264/SVC codec in 802.11 networks poses however some issues. In particular, since pictures are encoded into several small video units, the overhead imposed by the 802.11 contention-based channel access mechanism might be large. Thus, the strategy employed for the packetization of the video data plays a key role in determining the performance of the network. This work proposes two network adaptation strategies for H.264/SVC video to efficiently use the QoS-enabled 802.11e extension of the 802.11 standard by designing a scheme for joint optimization of video data aggregation and unequal error protection. Simulations of video transmissions in a realistic home networking scenario characterized by direct communications between devices in ad hoc mode show that the proposed strategies reduce the packet loss rate and significantly improve the quality of the communication with PSNR gains up to about 2 dB. Moreover, the performance of the proposed low-complexity strategy is close to that of the optimal, high-complexity, strategy.

This paper proposes a study of the IPTV world, focusing on network and system architectures, video codecs, network protocols, services and quality assurance. Based on this study, a new framework for Personal TV was developed. This new system is designed primarily to provide new personalized services to the user. In the architecture of this framework there are three main elements: Clients, Aggregators and Producers. The most important element is the aggregator that provides all video contents from the producers to its clients. The client has the possibility of creating his own channel that is sent to the network through its aggregator, of creating customized channels that can be viewed by other clients, among other features. The architecture designed and developed is based on new and studied concepts. It was tested to prove its viability, to assess its performance and to draw conclusions about its scalability, based on functional tests, compatibility tests and performance tests.