Towards scalable end-to-end QoS provision for VoIP applications

The growth of the Internet and the development of its new applications have increased the demand for providing a certain level of resource assurance and service support. The concept of ensuring quality of service (QoS) has been introduced in order to provide the support and assurance for these services. Different QoS mechanisms, such as integrated services (IntServ) and differentiated services (DiffServ), have been developed and introduced to provide different levels of QoS provision. However, IntServ can suffer from scalability issues that make it infeasible for large-scale network implementations. On the other hand, the aggregated-based per-flow technique of DiffServ does not provide such an end-to-end QoS guarantee. Recently, the IETF have proposed a new QoS architecture that implements IntServ over DiffServ in order to provide an end-to-end QoS for scalable networks. Hence, it became possible to provide and support a certain level of QoS for some delay sensitive and bandwidth-demanding applications such as voice over Internet Protocol (VoIP). With regard to VoIP applications, delay, jitter and packet loss are crucial issues that have to be taken into consideration for any VoIP system design and such parameters need a distinct level of QoS support

Performance evaluation of an open distributed platform for realistic traffic generation

Network researchers have dedicated a notable part of their efforts to the area of modeling traffic and to the implementation of efficient traffic generators. We feel that there is a strong demand for traffic generators capable to reproduce realistic traffic patterns according to theoretical models and at the same time with high performance. This work presents an open distributed platform for traffic generation that we called distributed internet traffic generator (D-ITG), capable of producing traffic (network, transport and application layer) at packet level and of accurately replicating appropriate stochastic processes for both inter departure time (IDT) and packet size (PS) random variables. We implemented two different versions of our distributed generator. In the first one, a log server is in charge of recording the information transmitted by senders and receivers and these communications are based either on TCP or UDP. In the other one, senders and receivers make use of the MPI library. In this work a complete performance comparison among the centralized version and the two distributed versions of D-ITG is presented

Heterogeneous LTE/ Wi-Fi architecture for intelligent transportation systems

Intelligent Transportation Systems (ITS) make use of advanced technologies to enhance road safety and improve traffic efficiency. It is anticipated that ITS will play a vital future role in improving traffic efficiency, safety, comfort and emissions. In order to assist the passengers to travel safely, efficiently and conveniently, several application requirements have to be met simultaneously. In addition to the delivery of regular traffic and safety information, vehicular networks have been recently required to support infotainment services. Previous vehicular network designs and architectures do not satisfy this increasing traffic demand as they are setup for either voice or data traffic, which is not suitable for the transfer of vehicular traffic. This new requirement is one of the key drivers behind the need for new mobile wireless broadband architectures and technologies. For this purpose, this thesis proposes and investigates a heterogeneous IEEE 802.11 and LTE vehicular system that supports both infotainment and ITS traffic control data. IEEE 802.11g is used for V2V communications and as an on-board access network while, LTE is used for V2I communications. A performance simulation-based study is conducted to validate the feasibility of the proposed system in an urban vehicular environment. The system performance is evaluated in terms of data loss, data rate, delay and jitter. Several simulation scenarios are performed and evaluated. In the V2I-only scenario, the delay, jitter and data drops for both ITS and video traffic are within the acceptable limits, as defined by vehicular application requirements. Although a tendency of increase in video packet drops during handover from one eNodeB to another is observed yet, the attainable data loss rate is still below the defined benchmarks. In the integrated V2V-V2I scenario, data loss in uplink ITS traffic was initially observed so, Burst communication technique is applied to prevent packet losses in the critical uplink ITS traffic. A quantitative analysis is performed to determine the number of packets per burst, the inter-packet and inter-burst intervals. It is found that a substantial improvement is achieved using a two-packet Burst, where no packets are lost in the uplink direction. The delay, jitter and data drops for both uplink and downlink ITS traffic, and video traffic are below the benchmarks of vehicular applications. Thus, the results indicate that the proposed heterogeneous system offers acceptable performance that meets the requirements of the different vehicular applications. All simulations are conducted on OPNET Network Modeler and results are subjected to a 95% confidence analysis

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated

Resource allocation for multimedia messaging services over EGPRS

The General Packet Radio Service (GPRS) is a new bearer service for GSM that greatly simplifies wireless access to packet data networks, such as the Internet, corporate LANs or to mobile portals. It applies a packet radio standard to transfer user data packets in wellorganized way between Mobile Stations (MS) and external packet data networks. The Enhanced General Packet Radio Service (EGPRS) is an extension of GPRS, offering much greater capacity. These enhancements have allowed the introduction of new services like Multimedia Messaging Services (MMS). MMS enables messaging with full content versatility, including images, audio, video, data and text, from terminal to terminal or from terminal to e-mail. The Wireless Application Protocol (WAP) is the WAP Forum standard for the presentation and delivery of wireless information and telephony services on mobile phones and other wireless terminals. In this thesis it is indicated that efficient radio resource allocation is necessary for managing different types of traffic in order to maintain the quality demands for different types of services. A theoretical model of MMS and WAP traffic is developed, and based on this model a simulator is implemented in Java programming language. This thesis proposes two techniques to improve the radio resource allocation algorithm performance called "radio link condition diversification" and "interactive traffic class prioritization". The radio link condition diversification technique defines minimum radio link quality that allows the user to receive their packets. The interactive traffic class prioritization technique defines different priorities for WAP packets and for MMS packets. Both techniques give good results in increasing user's perception of services and increasing network efficiency. This thesis indicates also that the prioritization mechanism successfully improves the response time of the interactive service by up to 80% with a setting of priority for interactive traffic class and decreasing the performance of the background traffic. This decrease is within a range acceptable by the end-user and that the link conditions limit mechanism has an advantage in terms of resource utilization