Model and performance of a no-reference quality assessment metric for video streaming

Video streaming via Transmission Control Protocol (TCP) networks has become a popular and highly demanded service, but its quality assessment in both objective and subjective terms has not been properly addressed. In this paper, based on statistical analysis a full analytic model of a no-reference objective metric, namely pause intensity (PI), for video quality assessment is presented. The model characterizes the video playout buffer behavior in connection with the network performance (throughput) and the video playout rate. This allows for instant quality measurement and control without requiring a reference video. PI specifically addresses the need for assessing the quality issue in terms of the continuity in the playout of TCP streaming videos, which cannot be properly measured by other objective metrics such as peak signal-to-noise-ratio, structural similarity, and buffer underrun or pause frequency. The performance of the analytical model is rigidly verified by simulation results and subjective tests using a range of video clips. It is demonstrated that PI is closely correlated with viewers' opinion scores regardless of the vastly different composition of individual elements, such as pause duration and pause frequency which jointly constitute this new quality metric. It is also shown that the correlation performance of PI is consistent and content independent

Development of a multi-mode self-adaptive algorithm to create an efficient wireless network on a university campus

The expanding use of ubiquitous computing has created a significant demand on existing network infrastructures. The demands of voice, video, and data on the same medium require a quality of service (QoS) at a level acceptable to users. Many network providers simply scale their networks to increase bandwidth and hardware to deal with the increasing demands. However, a network may still reach its design limits with peak traffic or malicious overuse of resources. In addition, with technology changing at a rapid pace, it is difficult to provide sufficient staffing to monitor and adjust the network settings to avoid issues during periods of network saturation. One of the common method to address these issues involves implementing a traffic shaper. A traffic shaper is a computer network management technique by which data sent across the network is delayed or routed in a way to accommodate a specific level of traffic to reach a desired QoS. There are many existing traffic shaping algorithms, each performing well under specific circumstances improving some QoS measures. The algorithms make use of queuing schemes to sort and send traffic based on the parameters provided to the system. To determine the need for this research, a survey was administered which revealed dissatisfaction with QoS of the wireless network. The purpose of this study focused on the development of a traffic shaping algorithm that would improve the QoS on a local area network on a university campus. The goal of the research was to create a new architecture that would allow a router to dynamically shift between different queuing mechanisms to improve network delay and packet loss without negatively impacting data throughput. The Multi-Mode Self-Adaptive (MMSA) algorithm was proposed to define a mechanism for this architecture. The MMSA was implemented within the code of a Cisco® router in the OPNET Modeler software and tested in a simulated university network environment. The results of the simulation revealed an improvement in end to end delay and packet loss rate with an insignificant change in average transmit rate between the router and the external server. The results of this research can be used as a basis for future research to create a new QoS framework. The new framework could be implemented in a router to allow configurations tailored to the network requirements of a service provider

Eş düğümler arası ağlarda çoklu ortam verilerinin gerçek zamanlı iletimi için yeni bir yöntem

06.03.2018 tarihli ve 30352 sayılı Resmi Gazetede yayımlanan “Yükseköğretim Kanunu İle Bazı Kanun Ve Kanun Hükmünde Kararnamelerde Değişiklik Yapılması Hakkında Kanun” ile 18.06.2018 tarihli “Lisansüstü Tezlerin Elektronik Ortamda Toplanması, Düzenlenmesi ve Erişime Açılmasına İlişkin Yönerge” gereğince tam metin erişime açılmıştır.İnternet kullanıcılarının gerçek zamanlı ortam verilerini paylaşma ihtiyacı her geçen gün artmaktadır. Artan bu ihtiyacın karşılanmasında, klasik istemci-sunucu mimarisi pek çok parametreden dolayı istenilen verimi sağlayamamaktadır. Bu nedenle kullanıcılar bulundukları ağ yapılarından bağımsız olarak birbirleri ile doğrudan iletişim kurabilmelidirler. Bu nedenle eş düğümler arası iletişim için sorun oluşturan durumların belirlenmesi ve araştırmacılar tarafından uygun yöntemlerin ortaya konulması gerekmektedir. Eş düğümler arası iletişimde karşımıza çıkan temel sorunların başında ağ adres dönüştürücü ve güvenlik duvarı gibi özel ağ oluşturan cihazların arkasındaki istemcilere kamusal ağdan erişilememesi gelmektedir. Bu sorunun çözümüne yönelik literatürde öne sürülen çözüm önerilerinin değişik avantaj ve dezavantajları bulunmaktadır. Bu çözüm önerilerinden İnteraktif Bağlantı Kurulumu ve Gerçek Zamanlı Medya Akış Protokolü, gerek internet altyapısından bağımsız oluşları, gerekse de dinamik yapılar için uygunlukları ile öne çıkmaktadırlar. Yapılan çalışma ile ağ adres dönüştürücü geçişi için tüm adımların tanımlandığı bir yöntem geliştirilerek eş düğümler arası ağlarda çoklu ortam verilerinin iletiminde uçtan uca tam bir model ortaya konulmuş ve "Durum Tabanlı Ağ Adres Dönüştürücü Geçişi" olarak isimlendirilmiştir. Geliştirilen model ile İnteraktif Bağlantı Kurulumu protokolünün bağlantı kurulum süresi, band genişliği ve paket kullanımı parametreleri iyileştirilmiştir.The usage of peer-to-peer (P2P) networks that provide sharing of real-time environmental data by internet users is becoming more and more popular. As a result, it is necessary to identify the problems during P2P communication and to develop proper solutions. One of the major problems of P2P communication is that it is not possible to reach the clients behind devices that create private networks like network address translation (NAT) and firewalls from the public network. Among the solutions proposed for this problem, Interactivity Connectivity Establishment (ICE) and Real Time Media Flow Protocol (RTMFP) are the methods most preferred in the literature. These methods seem more attractive than other NAT traversal mechanisms since they are independent from internet infrastructure and are also appropriate for dynamic structures. However, they do have some disadvantages. With this thesis work, a new state-based end-to-end communication technique (SBN) for NAT traversal has been designed and realized. The performance of the designed method was evaluated against three criteria connectivity check delay, connection packet count and bandwidth and compared to the ICE method

Quality-driven resource utilization methods for video streaming in wireless communication networks

This research is focused on the optimisation of resource utilisation in wireless mobile networks with the consideration of the users’ experienced quality of video streaming services. The study specifically considers the new generation of mobile communication networks, i.e. 4G-LTE, as the main research context. The background study provides an overview of the main properties of the relevant technologies investigated. These include video streaming protocols and networks, video service quality assessment methods, the infrastructure and related functionalities of LTE, and resource allocation algorithms in mobile communication systems. A mathematical model based on an objective and no-reference quality assessment metric for video streaming, namely Pause Intensity, is developed in this work for the evaluation of the continuity of streaming services. The analytical model is verified by extensive simulation and subjective testing on the joint impairment effects of the pause duration and pause frequency. Various types of the video contents and different levels of the impairments have been used in the process of validation tests. It has been shown that Pause Intensity is closely correlated with the subjective quality measurement in terms of the Mean Opinion Score and this correlation property is content independent. Based on the Pause Intensity metric, an optimised resource allocation approach is proposed for the given user requirements, communication system specifications and network performances. This approach concerns both system efficiency and fairness when establishing appropriate resource allocation algorithms, together with the consideration of the correlation between the required and allocated data rates per user. Pause Intensity plays a key role here, representing the required level of Quality of Experience (QoE) to ensure the best balance between system efficiency and fairness. The 3GPP Long Term Evolution (LTE) system is used as the main application environment where the proposed research framework is examined and the results are compared with existing scheduling methods on the achievable fairness, efficiency and correlation. Adaptive video streaming technologies are also investigated and combined with our initiatives on determining the distribution of QoE performance across the network. The resulting scheduling process is controlled through the prioritization of users by considering their perceived quality for the services received. Meanwhile, a trade-off between fairness and efficiency is maintained through an online adjustment of the scheduler’s parameters. Furthermore, Pause Intensity is applied to act as a regulator to realise the rate adaptation function during the end user’s playback of the adaptive streaming service. The adaptive rates under various channel conditions and the shape of the QoE distribution amongst the users for different scheduling policies have been demonstrated in the context of LTE. Finally, the work for interworking between mobile communication system at the macro-cell level and the different deployments of WiFi technologies throughout the macro-cell is presented. A QoEdriven approach is proposed to analyse the offloading mechanism of the user’s data (e.g. video traffic) while the new rate distribution algorithm reshapes the network capacity across the macrocell. The scheduling policy derived is used to regulate the performance of the resource allocation across the fair-efficient spectrum. The associated offloading mechanism can properly control the number of the users within the coverages of the macro-cell base station and each of the WiFi access points involved. The performance of the non-seamless and user-controlled mobile traffic offloading (through the mobile WiFi devices) has been evaluated and compared with that of the standard operator-controlled WiFi hotspots

Dual-Mode Congestion Control Mechanism for Video Services

Recent studies have shown that video services represent over half of Internet traffic, with a growing trend. Therefore, video traffic plays a major role in network congestion. Currently on the Internet, congestion control is mainly implemented through overprovisioning and TCP congestion control. Although some video services use TCP to implement their transport services in a manner that actually works, TCP is not an ideal protocol for use by all video applications. For example, UDP is often considered to be more suitable for use by real-time video applications. Unfortunately, UDP does not implement congestion control. Therefore, these UDP-based video services operate without any kind of congestion control support unless congestion control is implemented on the application layer. There are also arguments against massive overprovisioning. Due to these factors, there is still a need to equip video services with proper congestion control.Most of the congestion control mechanisms developed for the use of video services can only offer either low priority or TCP-friendly real-time services. There is no single congestion control mechanism currently that is suitable and can be widely used for all kinds of video services. This thesis provides a study in which a new dual-mode congestion control mechanism is proposed. This mechanism can offer congestion control services for both service types. The mechanism includes two modes, a backward-loading mode and a real-time mode. The backward-loading mode works like a low-priority service where the bandwidth is given away to other connections once the load level of a network is high enough. In contrast, the real-time mode always demands its fair share of the bandwidth.The behavior of the new mechanism and its friendliness toward itself, and the TCP protocol, have been investigated by means of simulations and real network tests. It was found that this kind of congestion control approach could be suitable for video services. The new mechanism worked acceptably. In particular, the mechanism behaved toward itself in a very friendly way in most cases. The averaged TCP fairness was at a good level. In the worst cases, the faster connections received about 1.6 times as much bandwidth as the slower connections