A Survey of Approaches and Challenges in the Real-time Multimedia Streaming
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paper_title: Development and Coverage Evaluation of ZigBee-Based Wireless Network Applications

paper_content:

Network coverage is one of the basic issues for information collection and data processing in ZigBee-based wireless sensor networks. Each node may be randomly distributed in a monitoring area, reflecting the network event of tracking in ZigBee network applications. This paper presents the development and coverage evaluation of a ZigBee-based wireless network application. A stack structure node available for home service integration is proposed, and all data of sensing nodes with an adaptive weighted fusion (AWF) processing are passed to the gateway and through the gateway to reexecute packet processing and then reported to the monitoring center, which effectively optimize the wireless network to the scale of the data processing efficiency. The linear interpolation theory is used for background graphical user interface so as to evaluate the working status of each node and the whole network coverage case. A testbed has been created for validating the basic functions of the proposed ZigBee-based home network system. Network coverage capabilities were tested, and packet loss and energy saving of the proposed system in longtime wireless network monitoring tasks were also verified.
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paper_title: Scalable Video Multicast Using Expanding Window Fountain Codes

paper_content:

Fountain codes were introduced as an efficient and universal forward error correction (FEC) solution for data multicast over lossy packet networks. They have recently been proposed for large scale multimedia content delivery in practical multimedia distribution systems. However, standard fountain codes, such as LT or Raptor codes, are not designed to meet unequal error protection (UEP) requirements typical in real-time scalable video multicast applications. In this paper, we propose recently introduced UEP expanding window fountain (EWF) codes as a flexible and efficient solution for real-time scalable video multicast. We demonstrate that the design flexibility and UEP performance make EWF codes ideally suited for this scenario, i.e., EWF codes offer a number of design parameters to be ldquotunedrdquo at the server side to meet the different reception criteria of heterogeneous receivers. The performance analysis using both analytical results and simulation experiments of H.264 scalable video coding (SVC) multicast to heterogeneous receiver classes confirms the flexibility and efficiency of the proposed EWF-based FEC solution.
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paper_title: Tuning Skype's Redundancy Control Algorithm for User Satisfaction

paper_content:

Determining how to transport delay-sensitive voice data has long been a problem in multimedia networking. The difficulty arises because voice and best-effort data are different by nature. It would not be fair to give priority to voice traffic and starve its best-effort counterpart; however, the voice data delivered might not be perceptible if each voice call is limited to the rate of an average TCP flow. To address the problem, we approach it from a user-centric perspective by tuning the voice data rate based on user satisfaction. Our contribution in this work is threefold. First, we investigate how Skype, the largest and fastest growing VoIP service on the Internet, adapts its voice data rate (i.e., the redundancy ratio) to network conditions. Second, by exploiting implementations of public domain codecs, we discover that Skype's mechanism is not really geared to user satisfaction. Third, based on a set of systematic experiments that quantify user satisfaction under different levels of packet loss and burstiness, we derive a concise model that allows user-centric redundancy control. The model can be easily incorporated into general VoIP services (not only Skype) to ensure consistent user satisfaction. Index Terms—MOS, PESQ, Piggyback, QoE (Quality of Ex- perience), QoS (Quality of Service), VoIP
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paper_title: OneClick: A Framework for Measuring Network Quality of Experience

paper_content:

As the service requirements of network applications shift from high throughput to high media quality, interactivity, and responsiveness, the definition of QoE (Quality of Experience) has become multidimensional. Although it may not be difficult to measure individual dimensions of the QoE, how to capture users' overall perceptions when they are using network applications remains an open question. In this paper, we propose a framework called OneClick to capture users' perceptions when they are using network applications. The framework only requires a subject to click a dedicated key whenever he/she feels dissatisfied with the quality of the application in use. OneClick is particularly effective because it is intuitive, lightweight, efficient, time-aware, and application-independent. We use two objective quality assessment methods, PESQ and VQM, to validate OneClick's ability to evaluate the quality of audio and video clips. To demonstrate the proposed framework's efficiency and effectiveness in assessing user experiences, we implement it on two applications, one for instant messaging applications, and the other for first- person shooter games. A Flash implementation of the proposed framework is also presented.
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paper_title: Impact of Network Performance on Cloud Speech Recognition

paper_content:

Interactive real-time communication between people and machine enables innovations in transportation, health care, etc. Using voice or gesture commands improves usability and broad public appeal of such systems. In this paper we experimentally evaluate Google speech recognition and Apple Siri - two of the most popular cloud-based speech recognition systems. Our goal is to evaluate the performance of these systems under different network conditions in terms of command recognition accuracy and round trip delay - two metrics that affect interactive application usability. Our results show that speech recognition systems are affected by loss and jitter, commonly present in cellular and WiFi networks. Finally, we propose and evaluate a network coding transport solution to improve the quality of voice transmission to cloud-based speech recognition systems. Experiments show that our approach improves the accuracy and delay of cloud speech recognizers under different loss and jitter values.
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paper_title: Video telephony for end-consumers: measurement study of Google+, iChat, and Skype

paper_content:

Video telephony requires high-bandwidth and low-delay voice and video transmissions between geographically distributed users. It is challenging to deliver high-quality video telephony to end-consumers through the best-effort Internet. In this paper, we present our measurement study on three popular video telephony systems on the Internet: Google+, iChat, and Skype. Through a series of carefully designed active and passive measurements, we are able to unveil important information about their key design choices and performance, including application architecture, video generation and adaptation schemes, loss recovery strategies, end-to-end voice and video delays, resilience against random and bursty losses, etc. Obtained insights can be used to guide the design of applications that call for high-bandwidth and low-delay data transmissions under a wide range of "best-effort" network conditions.
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paper_title: Video quality evaluation for Internet streaming applications

paper_content:

We carried out a number of subjective experiments using typical streaming content, codecs, bitrates and network conditions. In an attempt to review subjective testing procedures for video streaming applications, we used both Single Stimulus Continuous Quality Evaluation (SSCQE) and Double Stimulus Impairment Scale (DSIS) methods on the same test material. We thus compare these testing methods and present an analysis of the experimental results in view of codec performance. Finally, we use the subjective data to corroborate the prediction accuracy of a real-time non-reference quality metric.
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paper_title: Video quality evaluation for mobile streaming applications

paper_content:

This paper presents the results of our quality evaluations of video sequences encoded for and transmitted over a wireless channel. We selected content, codecs, bitrates and bit error patterns representative of mobile applications and we concentrated on the MPEG-4 and Motion JPEG2000 coding standards. We carried out subjective experiments using the Single Stimulus Continuous Quality Evaluation (SSCQE) method on this test material. We analyze the results of the subjective data and use them to compare codec performance and resilience to transmission errors. Finally, we use the subjective data to validate the prediction performance of a real-time non-reference quality metric.
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paper_title: Mesh-Pull-Based P2P Video Streaming System Using Fountain Codes

paper_content:

In this work, we propose a simple but effective mesh-pull-based P2P video streaming system using Fountain codes with variable symbol sizes for video-on-demand services. The goal of the proposed system is to provide a stable video streaming service of high quality with a low computational complexity and short initial latency. Fountain codes are adopted in the proposed system to simplify the handshaking procedure which causes a large initial latency, and to support a robust video streaming service. The proposed system works by using feedback information to reduce unnecessary encoded symbol generation. In addition, the Fountain code symbol size is continuously adjusted to minimize additional computational overhead required for Fountain encoding/decoding.
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paper_title: Improved coexistence and loss tolerance for delay based TCP congestion control

paper_content:

Loss based TCP congestion control has been shown to not perform well in environments were there is non-congestion related packet losses. Delay based TCP congestion control algorithms provide a low latency connection with no congestion related packet losses, and have the potential for being tolerant to non-congestion related losses. Unfortunately, delay based TCP does not compete well with loss based TCP, currently limiting its deployment. We propose a delay based algorithm which extends work by Budzisz et al. [1] to provide tolerance to non-congestion related losses, and better coexistence with loss based TCP in lightly multiplexed environments. We demonstrate that our algorithm improves the throughput when there are 1% packet losses by about 150%, and gives more than 50% improvement in the ability to share capacity with NewReno in lightly multiplexed environments.
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paper_title: On the limit of fountain MDC codes for video Peer-To-Peer networks

paper_content:

Video streaming for heterogeneous types of devices, where nodes have different devices characteristics in terms of computational capacity and display, is usually handled by encoding the video with different qualities. This is not well suited for Peer-To-Peer (P2P) systems, as a single peer group can only share content of the same quality, thus limiting the peer group size and efficiency. To address this problem, several existing works propose the use of Multiple Descriptions Coding (MDC). The concept of this type of video codec is to split a video in a number of descriptions which can be used on their own, or aggregated to improve the global quality of the video. Unfortunately existing MDC codes are not flexible, as the video is split in a defined number of descriptions. In this paper, we focus on the practical feasibility of using a Fountain MDC code with properties similar to existing Fountain erasure codes, including the ability to create any number of descriptions when needed (on the fly). We perform simulations using selected pictures to assess the feasibility of using these codes, knowing that they should improve the availability of the video pieces in a P2P system and hence the video streaming quality. We observe that, although this idea seems promising, the evaluated benefits, demonstrated by the PSNR values, are limited when used in a real P2P video streaming system.
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paper_title: A crowdsourceable QoE evaluation framework for multimedia content

paper_content:

Until recently, QoE (Quality of Experience) experiments had to be conducted in academic laboratories; however, with the advent of ubiquitous Internet access, it is now possible to ask an Internet crowd to conduct experiments on their personal computers. Since such a crowd can be quite large, crowdsourcing enables researchers to conduct experiments with a more diverse set of participants at a lower economic cost than would be possible under laboratory conditions. However, because participants carry out experiments without supervision, they may give erroneous feedback perfunctorily, carelessly, or dishonestly, even if they receive a reward for each experiment. In this paper, we propose a crowdsourceable framework to quantify the QoE of multimedia content. The advantages of our framework over traditional MOS ratings are: 1) it enables crowdsourcing because it supports systematic verification of participants' inputs; 2) the rating procedure is simpler than that of MOS, so there is less burden on participants; and 3) it derives interval-scale scores that enable subsequent quantitative analysis and QoE provisioning. We conducted four case studies, which demonstrated that, with our framework, researchers can outsource their QoE evaluation experiments to an Internet crowd without risking the quality of the results; and at the same time, obtain a higher level of participant diversity at a lower monetary cost.
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paper_title: A measurement-based study of MultiPath TCP performance over wireless networks

paper_content:

With the popularity of mobile devices and the pervasive use of cellular technology, there is widespread interest in hybrid networks and on how to achieve robustness and good performance from them. As most smart phones and mobile devices are equipped with dual interfaces (WiFi and 3G/4G), a promising approach is through the use of multi-path TCP, which leverages path diversity to improve performance and provide robust data transfers. In this paper we explore the performance of multi-path TCP in the wild, focusing on simple 2-path multi-path TCP scenarios. We seek to answer the following questions: How much can a user benefit from using multi-path TCP over cellular and WiFi relative to using the either interface alone? What is the impact of flow size on average latency? What is the effect of the rate/route control algorithm on performance? We are especially interested in understanding how application level performance is affected when path characteristics (e.g., round trip times and loss rates) are diverse. We address these questions by conducting measurements using one commercial Internet service provider and three major cellular carriers in the US.
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<format>
Title: A Survey of Approaches and Challenges in the Real-time Multimedia Streaming

Section 1: Introduction
Description 1: Introduce the significance of multimedia streaming, the motivation for the study, and the scope of the survey.

Section 2: Related Work
Description 2: Discuss various experimental evaluations and studies conducted on multimedia streaming applications, focusing on their findings and implications.

Section 3: Conclusion
Description 3: Summarize the general trends, challenges, and potential solutions in streaming applications, and highlight open research areas and advancements in the field.
</format>
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