Flow and Congestion Control for Internet Streaming Applications

The emergence of streaming multimedia players provides users with low latency audio and video content over the Internet. Providing high-quality, best-effort, real-time multimedia content requires adaptive delivery schemes that fairly share the available network bandwidth with reliable data protocols such as TCP. This paper proposes a new flow and congestion control scheme, SCP (Streaming Control Protocol) , for real-time streaming of continuous multimedia data across the Internet. The design of SCP arose from several years of experience in building and using adaptive real-time streaming video players. SCP addresses two issues associated with real-time streaming. First, it uses a congestion control policy that allows it to share network bandwidth fairly with both TCP and other SCP streams. Second, it improves smoothness in streaming and ensures low, predictable latency. This distinguishes it from TCP\u27s jittery congestion avoidance policy that is based on linear growth and one-half reduction of its congestion window. In this paper, we present a description of SCP, and an evaluation of it using Internet-based experiments

Design and evaluation of a DASH-compliant second screen video player for live events in mobile scenarios

The huge diffusion of mobile devices is rapidly changing the way multimedia content is consumed. Mobile devices are often used as a second screen, providing complementary information on the content shown on the primary screen, as different camera angles in case of a sport event. The introduction of multiple camera angles poses many challenges with respect to guaranteeing a high Quality of Experience to the end user, especially when the live aspect, different devices and highly variable network conditions typical of mobile environments come into play. Due to the ability of HTTP Adaptive Streaming (HAS) protocols to dynamically adapt to bandwidth fluctuations, they are especially suited for the delivery of multimedia content in mobile environments. In HAS, each video is temporally segmented and stored in different quality levels. Rate adaptation heuristics, deployed at the video player, allow the most appropriate quality level to be dynamically requested, based on the current network conditions. Recently, a standardized solution has been proposed by the MPEG consortium, called Dynamic Adaptive Streaming over HTTP (DASH). We present in this paper a DASH-compliant iOS video player designed to support research on rate adaptation heuristics for live second screen scenarios in mobile environments. The video player allows to monitor the battery consumption and CPU usage of the mobile device and to provide this information to the heuristic. Live and Video-on-Demand streaming scenarios and real-time multi-video switching are supported as well. Quantitative results based on real 3G traces are reported on how the developed prototype has been used to benchmark two existing heuristics and to analyse the main aspects affecting battery lifetime in mobile video streaming

Desain dan Implementasi Live Streaming Televisi Menggunakan Adaptive H264encoding

Teknologi Informasi yang paling luas penyebarannya adalah Televisi, dengan kemajuan teknologi sarana penyiaran Televisi tidak terbatas lagi ke TV broadcast menggunakan teknologi radio di gelombang khusus seperti saat ini, penyiaran TV telah menyebar ke sarana yang lain termasuk internet. Banyak teknologi yang bisa digunakan di internet, tetapi kandidat yang paling kuat adalah video streaming. Untuk aplikasi real-time atau live seperti kampanye atau siaran pengumuman pemerintah dll, teknologi video streaming yang digunakan adalah teknologi video streaming khusus yang disebut dengan live streaming.Teknologi Live Streaming hampir sama dengan video streaming, hanya saja data yang digunakan langsung bersumber dari televisi atau kamera yang bersifat real time. Live Streaming memerlukan proses live encoding dan minimum buffering, sedangkan di sisi lain diharapkan delay seminimal mungkin. Masalah selanjutnya adalah keterbatasan bandwidth. Jaringan komputer yang digunakan untuk melewatkan berbagai aplikasi akan digunakan juga sebagai media streaming yang membutuhkan bitrate cukup tinggi. Proses ini akan menyebabkan beban jaringan bertambah sehingga service yang ada tidak dapat berjalan dengan baik (terganggu). Pada penelitian ini difokuskan pada proses live streaming H264 dengan metode transmisi multicast dengan ditambahkan sebuah program adaptive streaming. Codec H264 dipilih karena performansinya yang cukup baik pada level bitrate yang lebih rendah. Sistem multicast digunakan untuk mengatasi masalah keterbatasan bandwidth yang digunakan dalam streaming. Adaptive streaming digunakan untuk menyesuaikan bitrate dengan kondisi trafik pada jaringan. Didapatkan nilai PSNR 36,58 dB untuk bitrate 500kbps dan 31,42 dB untuk bitrate 200kbps yang masih berada diatas threshold ITU 20dB dengan MOS 3,4 untuk 50 responden, sistem adaptive menyebabkan berkurangnya paket loss dari 1,53% menjadi 0,46%, bandwitdh stream unucast 1698kbps untuk multicast 558kbps

Adaptive delivery of real-time streaming video

Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2001.Includes bibliographical references (p. 87-92).While there is an increasing demand for streaming video applications on the Internet, various network characteristics make the deployment of these applications more challenging than traditional Internet applications like email and the Web. The applications that transmit data over the Internet must cope with the time-varying bandwidth and delay characteristics of the Internet and must be resilient to packet loss. This thesis examines these challenges and presents a system design and implementation that ameliorates some of the important problems with video streaming over the Internet. Video sequences are typically compressed in a format such as MPEG-4 to achieve bandwidth efficiency. Video compression exploits redundancy between frames to achieve higher compression. However, packet loss can be detrimental to compressed video with interdependent frames because errors potentially propagate across many frames. While the need for low latency prevents the retransmission of all lost data, we leverage the characteristics of MPEG-4 to selectively retransmit only the most important data in order to limit the propagation of errors. We quantify the effects of packet loss on the quality of MPEG-4 video, develop an analytical model to explain these effects, and present an RTP-compatible protocol-which we call SR-RTP--to adaptively deliver higher quality video in the face of packet loss. The Internet's variable bandwidth and delay make it difficult to achieve high utilization, Tcp friendliness, and a high-quality constant playout rate; a video streaming system should adapt to these changing conditions and tailor the quality of the transmitted bitstream to available bandwidth. Traditional congestion avoidance schemes such as TCP's additive-increase/multiplicative/decrease (AIMD) cause large oscillations in transmission rates that degrade the perceptual quality of the video stream. To combat bandwidth variation, we design a scheme for performing quality adaptation of layered video for a general family of congestion control algorithms called binomial congestion control and show that a combination of smooth congestion control and clever receiver-buffered quality adaptation can reduce oscillations, increase interactivity, and deliver higher quality video for a given amount of buffering. We have integrated this selective reliability and quality adaptation into a publicly available software library. Using this system as a testbed, we show that the use of selective reliability can greatly increase the quality of received video, and that the use of binomial congestion control and receiver quality adaptation allow for increased user interactivity and better video quality.by Nicholas G. Feamster.M.Eng

Adaptive Video Streaming Testbed Design for Performance Study and Assessment of QoE

[EN] Hypertext Transfer Protocol adaptive streaming switches between different video qualities, adapting to the network conditions, and avoids stalling streamed frames over high¿oscillation client's throughput improving the users' quality of experience (QoE). Quality of experience has become the most important parameter to lead the service providers to know about the end¿user feedback. Implementing Hypertext Transfer Protocol adaptive streaming applications to find out QoE in real¿life scenarios of vast networks becomes more challenging and complex task regarding to cost, agile, time, and decisions. In this paper, a virtualized network testbed to virtualize various machines to support implementing experiments of adaptive video streaming has been developed. Within the test study, the metrics which demonstrate performance of QoE are investigated, respectively, including initial delay (ie, startup delay at the beginning of playback a video), frequency switches (ie, number of times the quality is changed), accumulative video time (ie, number and length of stalls), CPU usage, and battery energy consumption. Furthermore, the relation between effective parameters of QoS on the aforementioned metrics for different segment length is investigated. Experimental results show that the proposed virtualized system is agile, easy to install and use, and costs less than real testbeds. Moreover, the subjective and objective performance studies of QoE evaluation in the system have proven that the segment lengths of 6 to 8 seconds were faired and more efficient than others according to the investigated parameters.Ministerio de Economia y Competitividad, Grant/Award Number: TIN2014-57991-C3-1-PAbdullah, MTA.; Lloret, J.; Ali, A.; García-García, L. (2018). Adaptive Video Streaming Testbed Design for Performance Study and Assessment of QoE. International Journal of Communication Systems. 1-15. https://doi.org/10.1002/dac.3551S11

Seamless Dynamic Adaptive Streaming in LTE/Wi-Fi Integrated Network under Smartphone Resource Constraints

Exploiting both LTE and Wi-Fi links simultaneously enhances the performance of video streaming services in a smartphone. However, it is challenging to achieve seamless and high quality video while saving battery energy and LTE data usage to prolong the usage time of a smartphone. In this paper, we propose REQUEST, a video chunk request policy for Dynamic Adaptive Streaming over HTTP (DASH) in a smartphone, which can utilize both LTE and Wi-Fi. REQUEST enables seamless DASH video streaming with near optimal video quality under given budgets of battery energy and LTE data usage. Through extensive simulation and measurement in a real environment, we demonstrate that REQUEST significantly outperforms other existing schemes in terms of average video bitrate, rebuffering, and resource waste.Peer reviewe

Traffic Profiling for Mobile Video Streaming

This paper describes a novel system that provides key parameters of HTTP Adaptive Streaming (HAS) sessions to the lower layers of the protocol stack. A non-intrusive traffic profiling solution is proposed that observes packet flows at the transmit queue of base stations, edge-routers, or gateways. By analyzing IP flows in real time, the presented scheme identifies different phases of an HAS session and estimates important application-layer parameters, such as play-back buffer state and video encoding rate. The introduced estimators only use IP-layer information, do not require standardization and work even with traffic that is encrypted via Transport Layer Security (TLS). Experimental results for a popular video streaming service clearly verify the high accuracy of the proposed solution. Traffic profiling, thus, provides a valuable alternative to cross-layer signaling and Deep Packet Inspection (DPI) in order to perform efficient network optimization for video streaming.Comment: 7 pages, 11 figures. Accepted for publication in the proceedings of IEEE ICC'1