Implementation and Performance Evaluation of an NGN prototype using WiMax as an Access Technology

Telecommunications networks have evolved to IP-based networks, commonly known as Next Generation Networks (NGN). The biggest challenge in providing high quality realtime multimedia applications is achieving a Quality of Service (QoS) consistent with user expectations. One of the key additional factors affecting QoS is the existence of different QoS mechanisms on the heterogeneous technologies used on NGN platforms. This research investigates the techniques used to achieve consistent QoS on network technologies that use different QoS techniques. Numerous proposals for solving the end-to-end QoS problem in IP networks have adopted policy-based management, use of signalling protocols for communicating applications QoS requirements across different Network Elements and QoS provisioning in Network Elements. Such solutions are dependent on the use of traffic classification and knowledge of the QoS requirements of applications and services on the networks. This research identifies the practical difficulties involved in meeting the QoS requirements of network traffic between WiMax and an IP core network. In the work, a solution based on the concept of class-of-service mapping is proposed. In the proposed solution, QoS is implemented on the two networks and the concept of class-of-service mapping is used to integrate the two QoS systems. This essentially provides consistent QoS to applications as they traverse the two network domains and hence meet end-user QoS expectations. The work is evaluated through a NGN prototype to determine the capabilities of the networks to deliver real-time media that meets user expectations

A delay-based aggregate rate control for P2P streaming systems

In this paper we consider mesh based P2P streaming systems focusing on the problem of regulating peer transmission rate to match the system demand while not overloading each peer upload link capacity. We propose Hose Rate Control (HRC), a novel scheme to control the speed at which peers offer chunks to other peers, ultimately controlling peer uplink capacity utilization. This is of critical importance for heterogeneous scenarios like the one faced in the Internet, where peer upload capacity is unknown and varies widely. HRC nicely adapts to the actual peer available upload bandwidth and system demand, so that Quality of Experience is greatly enhanced. To support our claims we present both simulations and actual experiments involving more than 1000 peers to assess performance in real scenarios. Results show that HRC consistently outperforms the Quality of Experience achieved by non-adaptive scheme

Resource dimensioning in a mixed traffic environment

An important goal of modern data networks is to support multiple applications over a single network infrastructure. The combination of data, voice, video and conference traffic, each requiring a unique Quality of Service (QoS), makes resource dimensioning a very challenging task. To guarantee QoS by mere over-provisioning of bandwidth is not viable in the long run, as network resources are expensive. The aim of proper resource dimensioning is to provide the required QoS while making optimal use of the allocated bandwidth. Dimensioning parameters used by service providers today are based on best practice recommendations, and are not necessarily optimal. This dissertation focuses on resource dimensioning for the DiffServ network architecture. Four predefined traffic classes, i.e. Real Time (RT), Interactive Business (IB), Bulk Business (BB) and General Data (GD), needed to be dimensioned in terms of bandwidth allocation and traffic regulation. To perform this task, a study was made of the DiffServ mechanism and the QoS requirements of each class. Traffic generators were required for each class to perform simulations. Our investigations show that the dominating Transport Layer protocol for the RT class is UDP, while TCP is mostly used by the other classes. This led to a separate analysis and requirement for traffic models for UDP and TCP traffic. Analysis of real-world data shows that modern network traffic is characterized by long-range dependency, self-similarity and a very bursty nature. Our evaluation of various traffic models indicates that the Multi-fractal Wavelet Model (MWM) is best for TCP due to its ability to capture long-range dependency and self-similarity. The Markov Modulated Poisson Process (MMPP) is able to model occasional long OFF-periods and burstiness present in UDP traffic. Hence, these two models were used in simulations. A test bed was implemented to evaluate performance of the four traffic classes defined in DiffServ. Traffic was sent through the test bed, while delay and loss was measured. For single class simulations, dimensioning values were obtained while conforming to the QoS specifications. Multi-class simulations investigated the effects of statistical multiplexing on the obtained values. Simulation results for various numerical provisioning factors (PF) were obtained. These factors are used to determine the link data rate as a function of the required average bandwidth and QoS. The use of class-based differentiation for QoS showed that strict delay and loss bounds can be guaranteed, even in the presence of very high (up to 90%) bandwidth utilization. Simulation results showed small deviations from best practice recommendation PF values: A value of 4 is currently used for both RT and IB classes, while 2 is used for the BB class. This dissertation indicates that 3.89 for RT, 3.81 for IB and 2.48 for BB achieve the prescribed QoS more accurately. It was concluded that either the bandwidth distribution among classes, or quality guarantees for the BB class should be adjusted since the RT and IB classes over-performed while BB under-performed. The results contribute to the process of resource dimensioning by adding value to dimensioning parameters through simulation rather than mere intuition or educated guessing.Dissertation (MEng (Electronic Engineering))--University of Pretoria, 2007.Electrical, Electronic and Computer Engineeringunrestricte

Packet aggregation for voice over internet protocol on wireless mesh networks

>Magister Scientiae - MScThis thesis validates that packet aggregation is a viable technique to increase call ca-pacity for Voice over Internet Protocol over wireless mesh networks. Wireless mesh networks are attractive ways to provide voice services to rural communities. Due to the ad-hoc routing nature of mesh networks, packet loss and delay can reduce voice quality.Even on non-mesh networks, voice quality is reduced by high overhead, associated with the transmission of multiple small packets. Packet aggregation techniques are proven to increase VoIP performance and thus can be deployed in wireless mesh networks. Kernel level packet aggregation was initially implemented and tested on a small mesh network of PCs running Linux, and standard baseline vs. aggregation tests were conducted with a realistic voice tra c pro le in hop-to-hop mode. Modi cations of the kernel were then transferred to either end of a nine node 'mesh potato' network and those tests were conducted with only the end nodes modi ed to perform aggregation duties. Packet ag- gregation increased call capacity expectedly, while quality of service was maintained in both instances, and hop-to-hop aggregation outperformed the end-to-end con guration. However, implementing hop-to-hop in a scalable fashion is prohibitive, due to the extensive kernel level debugging that must be done to achieve the call capacity increase.Therefore, end-to-end call capacity increase is an acceptable compromise for eventual scalable deployment of voice over wireless mesh networks

Fairness in a data center

Existing data centers utilize several networking technologies in order to handle the performance requirements of different workloads. Maintaining diverse networking technologies increases complexity and is not cost effective. This results in the current trend to converge all traffic into a single networking fabric. Ethernet is both cost-effective and ubiquitous, and as such it has been chosen as the technology of choice for the converged fabric. However, traditional Ethernet does not satisfy the needs of all traffic workloads, for the most part, due to its lossy nature and, therefore, has to be enhanced to allow for full convergence. The resulting technology, Data Center Bridging (DCB), is a new set of standards defined by the IEEE to make Ethernet lossless even in the presence of congestion. As with any new networking technology, it is critical to analyze how the different protocols within DCB interact with each other as well as how each protocol interacts with existing technologies in other layers of the protocol stack. This dissertation presents two novel schemes that address critical issues in DCB networks: fairness with respect to packet lengths and fairness with respect to flow control and bandwidth utilization. The Deficit Round Robin with Adaptive Weight Control (DRR-AWC) algorithm actively monitors the incoming streams and adjusts the scheduling weights of the outbound port. The algorithm was implemented on a real DCB switch and shown to increase fairness for traffic consisting of mixed-length packets. Targeted Priority-based Flow Control (TPFC) provides a hop-by-hop flow control mechanism that restricts the flow of aggressor streams while allowing victim streams to continue unimpeded. Two variants of the targeting mechanism within TPFC are presented and their performance evaluated through simulation

Performance Evaluation of MPLS in a Virtualized Service Provider Core (with/without Class of Service)

The last decade has witnessed a major change in the types of traffic scaling the Internet. With the development of real-time applications several challenges were faced within traditional IP networks. Some of these challenges are delay, increased costs faced by the service provider and customer, limited scalability, separate infrastructure costs and high administrative overheads to manage large networks etc. To combat these challenges, researchers have steered towards finding alternate solutions. Over the recent years, we have seen an introduction of a number of virtualized platforms and solutions being offered in the networking industry. Virtual load balancers, virtual firewalls, virtual routers, virtual intrusion detection and preventions systems are just a few examples within the Network Function Virtualization world! Service Providers are trying to find solutions where they could reduce operational expenses while at the same time meet the growing bandwidth demands of their customers. The main aim of this thesis is to evaluate the performance of voice, data and video traffic in a virtualized service provider core. Observations are made on how these traffic types perform on congested vs uncongested links and how Quality of Service treats traffic in a virtualized Service Provider Core using Round Trip Time as a performance metric. This thesis also tries to find if resiliency features such as Fast Reroute provide an additional advantage in failover scenarios within virtualized service provider cores. Juniper Networks vSRX are used to replicate virtual routers in a virtualized service provider core. Twenty-Four tests are carried out to gain a better understanding of how real-time applications and resiliency methods perform in virtualized networks. It is observed that a trade-off exists when introducing QoS on congested primary and secondary label switched paths. What can be observed thru the graphs is having Quality of Service enabled drops more packets however gives us the advantage of lower Round Trip Time for in-profile traffic. On the hand, having Quality of Service disabled, permits more traffic but leads to bandwidth contention between the three traffic classes leading to higher Round-Trip Times. The true benefit of QoS is seen in traffic congestion scenarios. The test bed built in this thesis, shows us that Fast Reroute does not add a significant benefit to aid in the reduction of packet loss during failover scenarios between primary and secondary paths. However, in certain scenarios fast reroute does seem to reduce packet loss specifically for data traffic

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