An analytical model for jitter in IP networks

ABSTRACT: Traditionally, IP network planning and design is mostly based on the average delay or loss constraints which can often be easily calculated. Jitter, on the other hand, is much more difficult to evaluate, but it is particularly important to manage the QoS of real-time and interactive services such as VoIP and streaming video. In this paper, we present simple formulas for the jitter of Poisson traffic in a single queue that can be quickly calculated . It takes into account the packets delay correlation and also the correlation of tandem queues that have a significant impact on the end-to-end jitter. We then extend them to the end-to-end jitter of a tagged stream based on a tandem queueing network. The results given by the model are then compared with event-driven simulations. We find that they are very accurate for Poisson traffic over a wide range of traffic loads and more importantly that they yield conservative values for the jitter so that they can be used in network design procedures. We also find some very counter-intuitive results. We show that jitter actually decreases with increasing load and the total jitter on a path depends on the position of congested links on that path. We finally point out some consequences of these results for network design procedures

An integrated bandwidth allocation and admission control framework for the support of heterogeneous real-time traffic in class-based IP networks

The support of real-time traffic in class-based IP networks requires the reservation of resources in all the links along the end-to-end paths through appropriate queuing and forwarding mechanisms. This resource allocation should be accompanied by appropriate admission control procedures in order to guarantee that newly admitted real-time traffic flows do not cause any violation to the Quality of Service (QoS) experienced by the already established real-time traffic flows. In this paper we initially aim to highlight certain issues with respect to the areas of bandwidth allocation and admission control for the support of real-time traffic in class-based IP networks. We investigate the implications of topological placement of both the bandwidth allocation and admission control schemes. We show that the performance of bandwidth allocation and admission control schemes depends highly on the location of the employed procedures with respect to the end-users requesting the services and the various network boundaries (access, metro, core, etc.). Based on our results we conclude that the strategies for applying these schemes should be location-aware, because the performance of bandwidth allocation and admission control at different points in a class-based IP network, and for the same traffic load, can be quite different and can deviate greatly from the expected performance. Through simulations we also try to provide a quantitative view of the aforementioned deviations. Taking the implications of this “location-awareness” into account, we subsequently present a new Measurement-based Admission Control (MBAC) scheme for real-time traffic that uses measurements of aggregate bandwidth only, without keeping the state of any per-flow information. In this scheme there is no assumption made on the nature of the traffic characteristics of the real-time traffic flows, which can be of heterogeneous nature. Through simulations we show that the admission control scheme is robust with respect to traffic heterogeneity and measurement errors. We also show that our scheme compares favorably against other admission control schemes in the literature

TCP performance enhancement in wireless networks via adaptive congestion control and active queue management

The transmission control protocol (TCP) exhibits poor performance when used in error-prone wireless networks. Remedy to this problem has been an active research area. However, a widely accepted and adopted solution is yet to emerge. Difficulties of an acceptable solution lie in the areas of compatibility, scalability, computational complexity and the involvement of intermediate routers and switches. This dissertation rexriews the current start-of-the-art solutions to TCP performance enhancement, and pursues an end-to-end solution framework to the problem. The most noticeable cause of the performance degradation of TCP in wireless networks is the higher packet loss rate as compared to that in traditional wired networks. Packet loss type differentiation has been the focus of many proposed TCP performance enhancement schemes. Studies conduced by this dissertation research suggest that besides the standard TCP\u27s inability of discriminating congestion packet losses from losses related to wireless link errors, the standard TCP\u27s additive increase and multiplicative decrease (AIMD) congestion control algorithm itself needs to be redesigned to achieve better performance in wireless, and particularly, high-speed wireless networks. This dissertation proposes a simple, efficient, and effective end-to-end solution framework that enhances TCP\u27s performance through techniques of adaptive congestion control and active queue management. By end-to-end, it means a solution with no requirement of routers being wireless-aware or wireless-specific . TCP-Jersey has been introduced as an implementation of the proposed solution framework, and its performance metrics have been evaluated through extensive simulations. TCP-Jersey consists of an adaptive congestion control algorithm at the source by means of the source\u27s achievable rate estimation (ARE) —an adaptive filter of packet inter-arrival times, a congestion indication algorithm at the links (i.e., AQM) by means of packet marking, and a effective loss differentiation algorithm at the source by careful examination of the congestion marks carried by the duplicate acknowledgment packets (DUPACK). Several improvements to the proposed TCP-Jersey have been investigated, including a more robust ARE algorithm, a less computationally intensive threshold marking algorithm as the AQM link algorithm, a more stable congestion indication function based on virtual capacity at the link, and performance results have been presented and analyzed via extensive simulations of various network configurations. Stability analysis of the proposed ARE-based additive increase and adaptive decrease (AJAD) congestion control algorithm has been conducted and the analytical results have been verified by simulations. Performance of TCP-Jersey has been compared to that of a perfect , but not practical, TCP scheme, and encouraging results have been observed. Finally the framework of the TCP-Jersey\u27s source algorithm has been extended and generalized for rate-based congestion control, as opposed to TCP\u27s window-based congestion control, to provide a design platform for applications, such as real-time multimedia, that do not use TCP as transport protocol yet do need to control network congestion as well as combat packet losses in wireless networks. In conclusion, the framework architecture presented in this dissertation that combines the adaptive congestion control and active queue management in solving the TCP performance degradation problem in wireless networks has been shown as a promising answer to the problem due to its simplistic design philosophy complete compatibility with the current TCP/IP and AQM practice, end-to-end architecture for scalability, and the high effectiveness and low computational overhead. The proposed implementation of the solution framework, namely TCP-Jersey is a modification of the standard TCP protocol rather than a completely new design of the transport protocol. It is an end-to-end approach to address the performance degradation problem since it does not require split mode connection establishment and maintenance using special wireless-aware software agents at the routers. The proposed solution also differs from other solutions that rely on the link layer error notifications for packet loss differentiation. The proposed solution is also unique among other proposed end-to-end solutions in that it differentiates packet losses attributed to wireless link errors from congestion induced packet losses directly from the explicit congestion indication marks in the DUPACK packets, rather than inferring the loss type based on packet delay or delay jitter as in many other proposed solutions; nor by undergoing a computationally expensive off-line training of a classification model (e.g., HMM), or a Bayesian estimation/detection process that requires estimations of a priori loss probability distributions of different loss types. The proposed solution is also scalable and fully compatible to the current practice in Internet congestion control and queue management, but with an additional function of loss type differentiation that effectively enhances TCP\u27s performance over error-prone wireless networks. Limitations of the proposed solution architecture and areas for future researches are also addressed

Resource Allocation in Relay-based Satellite and Wireless Communication Networks

A two-level bandwidth allocation scheme is proposed for a slotted Time-Division Multiple Access high data rate relay satellite communication link to provide efficient and fair channel utilization. The long-term allocation is implemented to provide per-flow/per-user Quality-of-Service guarantees and shape the average behavior. The time-varying short-term allocation is determined by solving an optimal timeslot scheduling problem based on the requests and other parameters. Through extensive simulations, the performance of a suitable MAC protocol with two-level bandwidth allocation is analyzed and compared with that of the existing static fixed-assignment scheme in terms of end-to-end delay and successful throughput. It is also shown that pseudo-proportional fairness is achieved for our hybrid protocol. We study rate control systems with heterogeneous time-varying propagation delays, based on analytic fluid flow models composed of first-order delay-differential equations. Both single-flow and multi-flow system models are analyzed, with special attention paid to the Mitra-Seery algorithm. The stationary solutions are investigated. For the fluctuating solutions, their dynamic behavior is analyzed in detail, analytically and numerically, in terms of amplitude, transient behavior, fairness and adaptability, etc.. Especially the effects of heterogeneous time-varying delays are investigated. It is shown that with proper parameter design the system can achieve stable behavior with close to pointwise proportional fairness among flows. Finally we investigate the resource allocation in 802.16j multi-hop relay systems with rate fairness constraints for two mutually exclusive options: transparent and non-transparent relay systems (T-RS and NT-RS). Single-Input Single-Output and Multi-Input Multi-Output antenna systems are considered in the links between the Base Station (BS) and Relay Stations (RS). 1 and 3 RSs per sector are considered. The Mobile Station (MS) association rule, which determines the access station (BS or RS) for each MS, is also studied. Two rules: Highest MCS scheme with the highest modulation and coding rate, and Highest (Mod) ESE scheme with the highest (modified) effective spectrum efficiency, are studied along with the optimal rule that maximizes system capacity with rate fairness constraints. Our simulation results show that the highest capacity is always achieved by NT-RS with 3 RSs per sector in distributed scheduling mode, and that the Highest (Mod) ESE scheme performs closely to the optimal rule in terms of system capacity

Analyse mathématique, méthode de calcul de la gigue et applications aux réseaux Internet

RÉSUMÉ Internet, ces dernières années, sert de support de communication à un grand nombre d’applications. L’évolution des réseaux à haut débit ont facilité le progrès des applications multimédia comme la voix sur IP, la vidéo streaming ou la vidéo interactive en temps réel... La variation de la disponibilité des ressources du réseau ne peut pas garantir une bonne qualité à tout moment pour ces services. C’est dans ce contexte que les travaux de ce projet de doctorat s’inscrivent et précisément dans le cadre de l’optimisation de la qualité de service (QoS). Les mécanismes de contrôle de QoS sont variés. On retrouve le contrôle de délai, assuré par la stratégie d’ordonnancement des paquets. Le contrôle de débit, quant à lui, fait en sorte que le débit de la source soit égal à la bande passante disponible dans le réseau. Excepté que les applications vidéo, surtout en temps réel, sont très sensibles à la variation du délai, appelée la gigue. En effet, la qualité perçue par les clients des vidéos en ligne dépend étroitement de la gigue. Une augmentation de la gigue engendre principalement des problèmes de démarrage retardé de la vidéo, des interruptions au cours de la vidéo et des distorsions de la résolution. L’objectif de cette thèse est d’étudier le paramètre de la gigue, qui demeure peu étudiée dans la littérature sur les réseaux IP, ainsi que d’envisager l’impact de l’augmentation de ce paramètre sur la vidéo transmise sur IP, l’une des applications les plus populaires de nos jours. Toutefois, au-delà des difficultés de la modélisation du trafic et du réseau, cet objectif majeur pose de nombreuses problématiques. Comment calculer la gigue analytiquement pour un trafic modélisé par des distributions généralisées au niveau paquet ? Est-ce que les modèles proposés sont suffisamment simples et faciles à calculer ? Comment intégrer ces nouvelles formalisations pour le contrôle des performances ? Comment l’estimation analytique peut- elle minimiser le trafic des paquets de contrôle des connexions vidéo? Nous explorons tout d’abord le calcul de la gigue dans des files d’attente avec des trafics autres que le trafic Poisson. Ce dernier est largement utilisé pour modéliser le trafic sur Internet étant donnée sa simplicité en échange de la imprécision. L’idée pour le calcul de la gigue est d’utiliser, d’une part la même formule que le cas du Poisson mais en intégrant d’autres distributions, et d’autre part des approximations et des hypothèses quand la caractérisation analytique du temps de transit n’est pas possible. Nous adoptons la simulation pour valider les modèles approximatifs. L’ensemble de simulations montre que la gigue moyenne calculée par notre modèle et celle obtenue par simulation coïncident avec des intervalles de confiance adéquats. De plus, le temps de calcul estimé pour évaluer la gigue est minime, ce qui facilite l’utilisation des formules proposées dans des outils de contrôle et en optimisation.-----------ABSTRACT In recent years, we have witnessed the huge use of the Internet Protocol for delivering multimedia trafic. Developments in broadband networks led the progress in multimedia applications such as voice over IP, video streaming or real-time videos. However, the stochastic nature of the networks, in particular mobile networks, make it difficult to maintain a good quality at all times. The research of this PhD thesis deals with the improvement of the quality of service (QoS) for this kind of applications. Current network protocols provide multiple QoS control mechanism. Congestion control and transmission delay optimization are provided by packet scheduling strategies and bandwidth planning. Moreover, flow control adjusts the mismatch between the video server rate and the receiver available bandwidth. Nevertheless, video applications, in particular interactive videos, are very sensitive to delay variation, commonly called jitter. Indeed, the customers’ perceived video quality depends on it. A jitter increase may cause a large video start-up delay, video interruptions and a decrease of image quality. The main objective of this thesis is the study of jitter, which is not much studied in the IP literature. We also examine the impact of the increase of this parameter on video transmission. However, beyond the difficulties of modeling traffic and network, this major objective raises many other issues. How to calculate jitter analytically for traffic models with general distributions? Are the proposed models sufficiently simple and easy to calculate? How to integrate these new formalizations into performance monitoring? How can the analytical estimate minimize the traffic control packets exchange for each video connection? We first explore the jitter calculation in queues with traffic other than Poisson traffic, that was widely used to model Internet traffic because of its simplicity. The idea is to compute jitter with the same formula for the Poisson traffic case, but with other distributions. For this, we need some approximations and assumptions when the analytical characterization of the transit time is not possible. We adopt simulations to validate the approximate models. The set of simulations shows that the average jitter calculated by our model and by simulation coincide within an appropriate confidence intervals. Moreover, the execution time to evaluate jitter is small, which facilitates the use of the proposed formulas in control tools and in optimization models. We then study the possibility of exploiting this analytical results to control jitter buffers, an important component in the video transmission. We find that it is possible to evaluate its performances analytically by estimating jitter inside this type of buffer

Quality-of-service management in IP networks

Quality of Service (QoS) in Internet Protocol (IF) Networks has been the subject of active research over the past two decades. Integrated Services (IntServ) and Differentiated Services (DiffServ) QoS architectures have emerged as proposed standards for resource allocation in IF Networks. These two QoS architectures support the need for multiple traffic queuing systems to allow for resource partitioning for heterogeneous applications making use of the networks. There have been a number of specifications or proposals for the number of traffic queuing classes (Class of Service (CoS)) that will support integrated services in IF Networks, but none has provided verification in the form of analytical or empirical investigation to prove that its specification or proposal will be optimum. Despite the existence of the two standard QoS architectures and the large volume of research work that has been carried out on IF QoS, its deployment still remains elusive in the Internet. This is not unconnected with the complexities associated with some aspects of the standard QoS architectures. [Continues.

A flexible, abstract network optimisation framework and its application to telecommunications network design and configuration problems

A flexible, generic network optimisation framework is described. The purpose of this framework is to reduce the effort required to solve particular network optimisation problems. The essential idea behind the framework is to develop a generic network optimisation problem to which many network optimisation problems can be mapped. A number of approaches to solve this generic problem can then be developed. To solve some specific network design or configuration problem the specific problem is mapped to the generic problem and one of the problem solvers is used to obtain a solution. This solution is then mapped back to the specific problem domain. Using the framework in this way, a network optimisation problem can be solved using less effort than modelling the problem and developing some algorithm to solve the model. The use of the framework is illustrated in two separate problems: design of an enterprise network to accommodate voice and data traffic and configuration of a core diffserv/MPLS network. In both cases, the framework enabled solutions to be found with less effort than would be required if a more direct approach was used

Telecommunications Networks

This book guides readers through the basics of rapidly emerging networks to more advanced concepts and future expectations of Telecommunications Networks. It identifies and examines the most pressing research issues in Telecommunications and it contains chapters written by leading researchers, academics and industry professionals. Telecommunications Networks - Current Status and Future Trends covers surveys of recent publications that investigate key areas of interest such as: IMS, eTOM, 3G/4G, optimization problems, modeling, simulation, quality of service, etc. This book, that is suitable for both PhD and master students, is organized into six sections: New Generation Networks, Quality of Services, Sensor Networks, Telecommunications, Traffic Engineering and Routing

Mobile Ad-Hoc Networks

Being infrastructure-less and without central administration control, wireless ad-hoc networking is playing a more and more important role in extending the coverage of traditional wireless infrastructure (cellular networks, wireless LAN, etc). This book includes state-of-the-art techniques and solutions for wireless ad-hoc networks. It focuses on the following topics in ad-hoc networks: quality-of-service and video communication, routing protocol and cross-layer design. A few interesting problems about security and delay-tolerant networks are also discussed. This book is targeted to provide network engineers and researchers with design guidelines for large scale wireless ad hoc networks

New quality of service routing algorithms based on local state information. The development and performance evaluation of new bandwidth-constrained and delay-constrained quality of service routing algorithms based on localized routing strategies.

The exponential growth of Internet applications has created new challenges for the control and administration of large-scale networks, which consist of heterogeneous elements under dynamically changing traffic conditions. These emerging applications need guaranteed service levels, beyond those supported by best-effort networks, to deliver the intended services to the end user. Several models have been proposed for a Quality of Service (QoS) framework that can provide the means to transport these services. It is desirable to find efficient routing strategies that can meet the strict routing requirements of these applications. QoS routing is considered as one of the major components of the QoS framework in communication networks. In QoS routing, paths are selected based upon the knowledge of resource availability at network nodes and the QoS requirements of traffic. Several QoS routing schemes have been proposed that differ in the way they gather information about the network state and the way they select paths based on this information. The biggest downside of current QoS routing schemes is the frequent maintenance and distribution of global state information across the network, which imposes huge communication and processing overheads. Consequently, scalability is a major issue in designing efficient QoS routing algorithms, due to the high costs of the associated overheads. Moreover, inaccuracy and staleness of global state information is another problem that is caused by relatively long update intervals, which can significantly deteriorate routing performance. Localized QoS routing, where source nodes take routing decisions based solely on statistics collected locally, was proposed relatively recently as a viable alternative to global QoS routing. It has shown promising results in achieving good routing performance, while at the same time eliminating many scalability related problems. In localized QoS routing each source¿destination pair needs to determine a set of candidate paths from which a path will be selected to route incoming flows. The goal of this thesis is to enhance the scalability of QoS routing by investigating and developing new models and algorithms based on the localized QoS routing approach. For this thesis, we have extensively studied the localized QoS routing approach and demonstrated that it can achieve a higher routing performance with lower overheads than global QoS routing schemes. Existing localized routing algorithms, Proportional Sticky Routing (PSR) and Credit-Based Routing (CBR), use the blocking probability of candidate paths as the criterion for selecting routing paths based on either flow proportions or a crediting mechanism, respectively. Routing based on the blocking probability of candidate paths may not always reflect the most accurate state of the network. This has motivated the search for alternative localized routing algorithms and to this end we have made the following contributions. First, three localized bandwidth-constrained QoS routing algorithms have been proposed, two are based on a source routing strategy and the third is based on a distributed routing strategy. All algorithms utilize the quality of links rather than the quality of paths in order to make routing decisions. Second, a dynamic precautionary mechanism was used with the proposed algorithms to prevent candidate paths from reaching critical quality levels. Third, a localized delay-constrained QoS routing algorithm was proposed to provide routing with an end-to-end delay guarantee. We compared the performance of the proposed localized QoS routing algorithms with other localized and global QoS routing algorithms under different network topologies and different traffic conditions. Simulation results show that the proposed algorithms outperform the other algorithms in terms of routing performance, resource balancing and have superior computational complexity and scalability features.Umm AlQura University, Saudi Arabi