Quality of experience-centric management of adaptive video streaming services : status and challenges

Video streaming applications currently dominate Internet traffic. Particularly, HTTP Adaptive Streaming ( HAS) has emerged as the dominant standard for streaming videos over the best-effort Internet, thanks to its capability of matching the video quality to the available network resources. In HAS, the video client is equipped with a heuristic that dynamically decides the most suitable quality to stream the content, based on information such as the perceived network bandwidth or the video player buffer status. The goal of this heuristic is to optimize the quality as perceived by the user, the so-called Quality of Experience (QoE). Despite the many advantages brought by the adaptive streaming principle, optimizing users' QoE is far from trivial. Current heuristics are still suboptimal when sudden bandwidth drops occur, especially in wireless environments, thus leading to freezes in the video playout, the main factor influencing users' QoE. This issue is aggravated in case of live events, where the player buffer has to be kept as small as possible in order to reduce the playout delay between the user and the live signal. In light of the above, in recent years, several works have been proposed with the aim of extending the classical purely client-based structure of adaptive video streaming, in order to fully optimize users' QoE. In this article, a survey is presented of research works on this topic together with a classification based on where the optimization takes place. This classification goes beyond client-based heuristics to investigate the usage of server-and network-assisted architectures and of new application and transport layer protocols. In addition, we outline the major challenges currently arising in the field of multimedia delivery, which are going to be of extreme relevance in future years

LTE 네트워크에서 비디오 전달 서비스의 성능 향상

학위논문 (박사)-- 서울대학교 대학원 : 전기·컴퓨터공학부, 2015. 2. 권태경.LTE includes an enhanced multimedia broadcast/multicast service(eMBMS)but delay-sensitive real-time video streaming requires the combination of efficient handling of wireless link bandwidth and reduced handover delays, which remains a challenge. The 3GPP standard introduces a Multimedia Broadcast and multicast service over a Single Frequency Network (MBSFN) area which is a group of base stations broadcasting the same multicast packets. It can reduce the handover delay within MBSFN areas, but raises the traffic load on LTE networks. In this dissertation, we first presents an MBSFN architecture based on location management areas (LMAs) which can increase the sizes of MBSFN areas to reduce the average handover delay without too much bandwidth waste. An analytical model is developed to quantify service disruption time, bandwidth usage, and blocking probability for different sizes of MBSFN areas and LMAs while considering user mobility, user distribution, and eMBMS session popularity. Using this model, we also propose how to determine the best sizes of MBSFN areas and LMAs along with performance guarantees. Analytical and simulation results demonstrate that our LMA-based MBSFN scheme can achieve bandwidth-efficient multicast delivery while retaining an acceptable service disruption time. We next propose to transmit the real-time video streaming packets of eMBMSs proactively and probabilistically, so that the average handover delay perceived by a user is stochastically guaranteed. To quantify the tradeoff between the perceived handover delay and the bandwidth overhead of proactive transmissions, we develop an analytical model considering user mobility, user distribution, and session popularity. Comprehensive simulation is carried out to verify the analysis. On the other hand, hypertext transfer protocol (HTTP) based adaptive streaming (HAS) is expected to be a dominant technique for non-real-time video delivery in LTE networks. In this dissertation, we first analyze the root causes of the problems of the existing HAS techniques. Based on the insights gained from our analysis, we propose a network-side HAS solution to provide a fair, efficient, and stable video streaming service. The key characteristics of our solution are: (i) unification of video- and data-users into a single utility framework, (ii) direct rate control conveying the assigned rates to the video client through overwritten HTTP Response messages, and (iii) rate allocation for stability by a stateful approach. By the experiments conducted in a real LTE femtocell network, we compare the proposed solution with state-of-the-art HAS solutions. We reveal that our solution (i) enhances the average video bitrates, (ii) achieves the stability of video quality, and (iii) supports the control of the balance between video- and data-users.Abstract i I. Introduction 1 II. Performance Improvements on Real-time Multicast Video Delivery 4 2.1 Introduction 4 2.2 Related Work 7 2.3 Location Management Area Based MBSFN 9 2.3.1 Location Management Area (LMA) 10 2.3.2 Handover Delays 12 2.3.3 LMA-based MBSFN Area Planning 12 2.4 Performance Analysis 14 2.4.1 Disruption Time 17 2.4.2 Bandwidth Usage 20 2.4.3 Blocking Probability 21 2.5 Numerical Results 23 2.5.1 Effect of NZ and NL 24 2.5.2 Deciding NZ and NL 27 2.5.3 Effects of v and rho* 31 2.5.4 Effect of alpha 32 2.6 Simulation Results 35 2.7 Conclusion 37 III. Proactive Approach for LMA-based MBSFN 39 3.1 Introduction 39 3.2 Network and MBSFN Modeling 41 3.3 Proactive LMA-based MBSFN 44 3.3.1 Problem Formulation 45 3.3.2 Overall procedure 47 3.4 Performance Evaluation 48 3.4.1 Simulation Setup 48 3.4.2 Computation of pi 50 3.4.3 Simulation Results 51 3.5 Conclusions 53 IV. Performance Improvements on HTTP Adaptive Video Streaming 55 4.1 Introduction 55 4.2 Related Work 57 4.3 Problem Definition 59 4.4 Utility-aware Network-side Streaming Approach 62 4.4.1 Streaming Proxy (SP) 63 4.4.2 Message Flows 65 4.4.3 Characteristics 67 4.5 Bitrate Assignment 68 4.5.1 Bitrate Calculation 69 4.5.2 Enhancing Stability 70 4.5.3 Algorithm for Continuous Bitrates 71 4.5.4 Handling the Bottleneck of Wired Networks 71 4.6 Simulation 73 4.6.1 Static Scenario 73 4.6.2 Mobile Scenarios 75 4.6.3 Algorithm for Continuous Bitrates 77 4.7 Experiments 78 4.7.1 Implementation of DASH Player 79 4.7.2 Implementation of eNB 80 4.7.3 Implementation of Streaming Proxy 83 4.7.4 Experimental Results 83 4.8 Conclusion 87 V. Summary & FutureWork 89 Bibliography 92Docto

QoE Evaluation Across a Range of User Age Groups in Video Applications

PhDQuality of Service (QoS) measures are the network parameters; delay, jitter, and loss and they do not reflect the actual quality of the service received by the end user. To get an actual view of the performance from a user’s perspective, the Quality of the Experience (QoE) measure is now used. Traditionally, QoS network measurements are carried on actual network components, such as the routers and switches since these are the key network components. In this thesis, however, the experimentation has been done on real video traffic. The experimental setup made use of a very popular network tool, Network Emulator (NetEm) created by the Linux Foundation. NetEm allows network emulation without using the actual network devices such as the routers and traffic generator. The common NetEm offered features are those that have been used by the researchers in the past. These have the same limitation as a traditional simulator, which is the inability of NetEm delay jitter model to represent realistic network traffic models, such to reflect the behaviour of real world networks. The NetEm default method of inputting delay and jitter adds or subtracts a fixed amount of delay on the outgoing traffic. NetEm also allows the user to add this variation in a correlated fashion. However, using this technique the outputted packet delays are generated in such a way as to be very limited and hence not much like real internet traffic which has a vast range of delays. The standard alternative that NetEm allows is generate the delays from either a Normal (Gaussian) or Pareto distribution. This research, however, has shown that using a Gaussian or Pareto distribution also has very severe limitations, and these are fully discussed and described in Chapter 5 on page 68 of this thesis. This research adopts another approach that is also allowed (with more difficulty) by NetEm: by measuring a very large number of packet delays generated from a double exponential distribution a packet delay profile is created that far better imitates the actual delays seen in Internet traffic. In this thesis a large set of statistical delay values were gathered and used to create delay distribution tables. Additionally, to overcome another default behaviour of NetEm of re-ordering packets once jitter is implemented, PFIFO queuing discipline has been deployed to retain the original packet order regardless of the highest levels of implemented jitter. Furthermore, this advancement in NetEm’s functionality also incorporates the ability to combine delay, jitter, and loss, which is not allowed on NetEm by default. In the literature, no work has been found to have utilised NetEm previously with such an advancement. Focusing on Video On Demand (VOD) it was discovered that the reported QoE may differ widely for users of different age groups, and that the most demanding age group (the youngest) can require an order of magnitude lower PLP to achieve the same QoE than is required by the most widely studied age group of users. A bottleneck TCP model was then used to evaluate the capacity cost of achieving an order of magnitude decrease in PLP, and found it be (almost always) a 3-fold increase in link capacity that was required. The results are potentially very useful to service providers and network designers to be able to provide a satisfactory service to their customers, and in return, maintaining a prosperous business.EPSRC (1589943)