User-Centric Quality of Service Provisioning in IP Networks

The Internet has become the preferred transport medium for almost every type of communication, continuing to grow, both in terms of the number of users and delivered services. Efforts have been made to ensure that time sensitive applications receive sufficient resources and subsequently receive an acceptable Quality of Service (QoS). However, typical Internet users no longer use a single service at a given point in time, as they are instead engaged in a multimedia-rich experience, comprising of many different concurrent services. Given the scalability problems raised by the diversity of the users and traffic, in conjunction with their increasing expectations, the task of QoS provisioning can no longer be approached from the perspective of providing priority to specific traffic types over coexisting services; either through explicit resource reservation, or traffic classification using static policies, as is the case with the current approach to QoS provisioning, Differentiated Services (Diffserv). This current use of static resource allocation and traffic shaping methods reveals a distinct lack of synergy between current QoS practices and user activities, thus highlighting a need for a QoS solution reflecting the user services. The aim of this thesis is to investigate and propose a novel QoS architecture, which considers the activities of the user and manages resources from a user-centric perspective. The research begins with a comprehensive examination of existing QoS technologies and mechanisms, arguing that current QoS practises are too static in their configuration and typically give priority to specific individual services rather than considering the user experience. The analysis also reveals the potential threat that unresponsive application traffic presents to coexisting Internet services and QoS efforts, and introduces the requirement for a balance between application QoS and fairness. This thesis proposes a novel architecture, the Congestion Aware Packet Scheduler (CAPS), which manages and controls traffic at the point of service aggregation, in order to optimise the overall QoS of the user experience. The CAPS architecture, in contrast to traditional QoS alternatives, places no predetermined precedence on a specific traffic; instead, it adapts QoS policies to each individual’s Internet traffic profile and dynamically controls the ratio of user services to maintain an optimised QoS experience. The rationale behind this approach was to enable a QoS optimised experience to each Internet user and not just those using preferred services. Furthermore, unresponsive bandwidth intensive applications, such as Peer-to-Peer, are managed fairly while minimising their impact on coexisting services. The CAPS architecture has been validated through extensive simulations with the topologies used replicating the complexity and scale of real-network ISP infrastructures. The results show that for a number of different user-traffic profiles, the proposed approach achieves an improved aggregate QoS for each user when compared with Best effort Internet, Traditional Diffserv and Weighted-RED configurations. Furthermore, the results demonstrate that the proposed architecture not only provides an optimised QoS to the user, irrespective of their traffic profile, but through the avoidance of static resource allocation, can adapt with the Internet user as their use of services change.France Teleco

Machine learning for Quality of Experience in real-time applications

L'abstract è presente nell'allegato / the abstract is in the attachmen

Secure VoIP Performance Measurement

This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality

LTE Optimization and Resource Management in Wireless Heterogeneous Networks

Mobile communication technology is evolving with a great pace. The development of the Long Term Evolution (LTE) mobile system by 3GPP is one of the milestones in this direction. This work highlights a few areas in the LTE radio access network where the proposed innovative mechanisms can substantially improve overall LTE system performance. In order to further extend the capacity of LTE networks, an integration with the non-3GPP networks (e.g., WLAN, WiMAX etc.) is also proposed in this work. Moreover, it is discussed how bandwidth resources should be managed in such heterogeneous networks. The work has purposed a comprehensive system architecture as an overlay of the 3GPP defined SAE architecture, effective resource management mechanisms as well as a Linear Programming based analytical solution for the optimal network resource allocation problem. In addition, alternative computationally efficient heuristic based algorithms have also been designed to achieve near-optimal performance

Profiling Skype video calls: Rate control and video quality

Video telephony has recently gained its momentum and is widely adopted by end-consumers. But there have been very few studies on the network impacts of video calls and the user Quality-of-Experience (QoE) under different network conditions. In this paper, we study the rate control and video quality of Skype video calls. We first measure the behaviors of Skype video calls on a controlled network testbed. By varying packet loss rate, propagation delay and bandwidth, we observe how Skype adjusts its rates, FEC redundancy and video quality. We find that Skype is robust against mild packet losses and propagation delays, and can efficiently utilize the available network bandwidth. We also find that Skype employs an overly aggressive FEC protection strategy. Based on the measurement results, we develop rate control model, FEC model, and video quality model for Skype. Extrapolating from the models, we conduct numerical analysis to study the network impacts of Skype. We demonstrate that user back-offs upon quality degradation serve as an effective user-level rate control scheme. We also show that Skype video calls are indeed TCP-friendly and respond to congestion quickly when the network is overloaded.Engineering, Electrical & ElectronicTelecommunicationsEICPCI-S(ISTP)

On the processing time for detection of Skype traffic

Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works. [P. M. Santiago del Río. J. Ramos, J. L. García-Dorado, J. Aracil, A. Cuadra-Sánchez, and M. Cutanda-Rodríguez, "On the processing time for detection of Skype traffic", in 7th International Wireless Communications and Mobile Computing Conference, IWCMC 2011, p. 1784 - 1788The last few years have witnessed VoIP applications gaining a tremendous popularity and Skype, in particular, is leading this continuous expansion. Unfortunately, Skype follows a closed source and proprietary design, and typically uses encryption mechanisms, making it very difficult to identify its presence from a traffic aggregate. Several algorithms and approaches have been proposed to perform such task with promising results in terms of accuracy. However, such approaches typically require significant computation resources and it is unlikely that they can be deployed in nowadays high-speed networks. In this light, this paper focuses on cutting the processing cost of algorithms to detect Skype traffic. We have conveniently tuned a previous well-validated algorithm and we have assessed its performance. To this end, we have used real traces from public repositories, from a Spanish 3G operator, and synthetic traces. Our results show that a single process can detect Skype traffic at 1 Gbps rates reading replayed real traces directly from a NIC. Even more, 3.7 Gbps are achieved reading from traces previously allocated in memory using a single process and 45 Gbps using 16 concurrent processes. This fact paves the way for 10 Gbps processing in commodity hardware

Call admission control for adaptive bit-rate VoIP over 802.11 WLAN.

Cui, Yuanyuan.Thesis (M.Phil.)--Chinese University of Hong Kong, 2009.Includes bibliographical references (p. 64-68).Abstract also in Chinese.Chapter Chapter 1 --- Introduction --- p.1Chapter 1 .1 --- Motivations and Contributions --- p.1Chapter 1.2 --- Related Works --- p.3Chapter 1.3 --- Organization of the Thesis --- p.4Chapter Chapter 2 --- Background --- p.5Chapter 2.1 --- IEEE 802.11 --- p.5Chapter 2.1.1 --- IEEE 802.11 Topologies --- p.5Chapter 2.1.2 --- IEEE 802.11 MAC --- p.8Chapter 2.2 --- Voice over Internet Protocol (VoIP) --- p.11Chapter 2.2.1 --- A VoIP system --- p.11Chapter 2.2.2 --- QoS requirements for VoIP --- p.11Chapter 2.2.3 --- VoIP speech codecs --- p.12Chapter 2.3 --- VoIP over WLAN --- p.13Chapter 2.3.1 --- System Architecture of VoIP over WLAN --- p.14Chapter 2.3.2 --- VoIP Capacity over WLAN --- p.15Chapter 2.4 --- Skype --- p.16Chapter Chapter 3 --- Skype Rate Adaptation Mechanism --- p.17Chapter 3.1 --- Experimental Setting --- p.17Chapter 3.2 --- Overview --- p.19Chapter 3.3 --- Flow Rate Region --- p.20Chapter 3.4 --- Feedback: Receiver Report (RR) --- p.21Chapter 3.5 --- Bandwidth Usage Target (BM) --- p.24Chapter 3.6 --- Summary of Skype Rate Adaptation Mechanism --- p.28Chapter 3.7 --- Skype-emulating Traffic Generator --- p.28Chapter Chapter 4 --- "Call Admission, Fairness and Stability Control" --- p.32Chapter 4.1 --- Unfair and Instability problems for AVoIP --- p.32Chapter 4.1.1 --- Analysis --- p.32Chapter 4.1.2 --- Simulation Evaluation --- p.34Chapter 4.2 --- CFSC scheme --- p.37Chapter 4.2.1 --- Pre-admission Bandwidth-reallocation Call Admission Control (PBCAC) --- p.39Chapter 4.2.2 --- Fairness Control --- p.42Chapter 4.2.3 --- Stability Control --- p.43Chapter Chapter 5 --- Performance Evaluation of CFSC --- p.44Chapter 5.1 --- Evaluation of Fairness Control --- p.44Chapter 5.2 --- Evaluation of Stability Control --- p.46Chapter 5.3 --- Evaluation of PBCAC --- p.46Chapter 5.4 --- Evaluation of complete CFSC --- p.49Chapter Chapter 6 --- Conclusion --- p.51Appendices --- p.53References --- p.6