Congestion Control for Streaming Media

The Internet has assumed the role of the underlying communication network for applications such as file transfer, electronic mail, Web browsing and multimedia streaming. Multimedia streaming, in particular, is growing with the growth in power and connectivity of today\u27s computers. These Internet applications have a variety of network service requirements and traffic characteristics, which presents new challenges to the single best-effort service of today\u27s Internet. TCP, the de facto Internet transport protocol, has been successful in satisfying the needs of traditional Internet applications, but fails to satisfy the increasingly popular delay sensitive multimedia applications. Streaming applications often use UDP without a proper congestion avoidance mechanisms, threatening the well-being of the Internet. This dissertation presents an IP router traffic management mechanism, referred to as Crimson, that can be seamlessly deployed in the current Internet to protect well-behaving traffic from misbehaving traffic and support Quality of Service (QoS) requirements of delay sensitive multimedia applications as well as traditional Internet applications. In addition, as a means to enhance Internet support for multimedia streaming, this dissertation report presents design and evaluation of a TCP-Friendly and streaming-friendly transport protocol called the Multimedia Transport Protocol (MTP). Through a simulation study this report shows the Crimson network efficiently handles network congestion and minimizes queuing delay while providing affordable fairness protection from misbehaving flows over a wide range of traffic conditions. In addition, our results show that MTP offers streaming performance comparable to that provided by UDP, while doing so under a TCP-Friendly rate

Provider-Controlled Bandwidth Management for HTTP-based Video Delivery

Over the past few years, a revolution in video delivery technology has taken place as mobile viewers and over-the-top (OTT) distribution paradigms have significantly changed the landscape of video delivery services. For decades, high quality video was only available in the home via linear television or physical media. Though Web-based services brought video to desktop and laptop computers, the dominance of proprietary delivery protocols and codecs inhibited research efforts. The recent emergence of HTTP adaptive streaming protocols has prompted a re-evaluation of legacy video delivery paradigms and introduced new questions as to the scalability and manageability of OTT video delivery. This dissertation addresses the question of how to enable for content and network service providers the ability to monitor and manage large numbers of HTTP adaptive streaming clients in an OTT environment. Our early work focused on demonstrating the viability of server-side pacing schemes to produce an HTTP-based streaming server. We also investigated the ability of client-side pacing schemes to work with both commodity HTTP servers and our HTTP streaming server. Continuing our client-side pacing research, we developed our own client-side data proxy architecture which was implemented on a variety of mobile devices and operating systems. We used the portable client architecture as a platform for investigating different rate adaptation schemes and algorithms. We then concentrated on evaluating the network impact of multiple adaptive bitrate clients competing for limited network resources, and developing schemes for enforcing fair access to network resources. The main contribution of this dissertation is the definition of segment-level client and network techniques for enforcing class of service (CoS) differentiation between OTT HTTP adaptive streaming clients. We developed a segment-level network proxy architecture which works transparently with adaptive bitrate clients through the use of segment replacement. We also defined a segment-level rate adaptation algorithm which uses download aborts to enforce CoS differentiation across distributed independent clients. The segment-level abstraction more accurately models application-network interactions and highlights the difference between segment-level and packet-level time scales. Our segment-level CoS enforcement techniques provide a foundation for creating scalable managed OTT video delivery services

Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications

[EN] Newer social multimedia applications, such as Social TV or networked multi-player games, enable independent groups (or clusters) of users to interact among themselves and share services within the context of simultaneous media content consumption. In such scenarios, concurrently synchronized playout points must be ensured so as not to degrade the user experience on such interaction. We refer to this process as Inter-Destination Multimedia Synchronization (IDMS). This paper presents the design, implementation and evaluation of an evolved version of an RTCP-based IDMS approach, including an Adaptive Media Playout (AMP) scheme that aims to dynamically and smoothly adjust the playout timing of each one of the geographically distributed consumers in a specific cluster if an allowable asynchrony threshold between their playout states is exceeded. For that purpose, we previously had also to develop a full implementation of RTP/RTCP protocols for NS-2, in which we included the IDMS approach as an optional functionality. Simulation results prove the feasibility of such IDMS and AMP proposals, by adopting several dynamic master reference selection policies, to maintain an overall synchronization status (within allowable limits) in each cluster of participants, while minimizing the occurrence of long-term playout discontinuities (such as skips/pauses) which are subjectively more annoying and less tolerable to users than small variations in the media playout rate.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-05-11-002-331 Project and in PAID-01-10. Authors also would like to thank the anonymous reviewers that helped to significantly improve the quality of the paper with their constructive comments.Montagud, M.; Boronat, F. (2012). Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications. Computer Networks. 56(12):2912-2933. https://doi.org/10.1016/j.comnet.2012.05.00329122933561

Congestion-Aware Scalable Video Streaming

Postprin

Flow Level QoE of Video Streaming in Wireless Networks

The Quality of Experience (QoE) of streaming service is often degraded by frequent playback interruptions. To mitigate the interruptions, the media player prefetches streaming contents before starting playback, at a cost of delay. We study the QoE of streaming from the perspective of flow dynamics. First, a framework is developed for QoE when streaming users join the network randomly and leave after downloading completion. We compute the distribution of prefetching delay using partial differential equations (PDEs), and the probability generating function of playout buffer starvations using ordinary differential equations (ODEs) for CBR streaming. Second, we extend our framework to characterize the throughput variation caused by opportunistic scheduling at the base station, and the playback variation of VBR streaming. Our study reveals that the flow dynamics is the fundamental reason of playback starvation. The QoE of streaming service is dominated by the first moments such as the average throughput of opportunistic scheduling and the mean playback rate. While the variances of throughput and playback rate have very limited impact on starvation behavior.Comment: 14 page

Design, Development and Assessment of Control Schemes for IDMS in a Standardized RTCP-based Solution

[EN] Currently, several media sharing applications that allow social interactions between distributed users are gaining momentum. In these networked scenarios, synchronized playout between the involved participants must be provided to enable truly interactive and coherent shared media experiences. This research topic is known as Inter-Destination Media Synchronization (IDMS). This paper presents the design and development of an advanced IDMS solution, which is based on extending the capabilities of RTP/RTCP standard protocols. Particularly, novel RTCP extensions, in combination with several control algorithms and adjustment techniques, have been specified to enable an adaptive, highly accurate and standard compliant IDMS solution. Moreover, as different control or architectural schemes for IDMS exist, and each one is best suited for specific use cases, the IDMS solution has been extended to be able to adopt each one of them. Simulation results prove the satisfactory responsiveness of our IDMS solution in a small scale scenario, as well as its consistent behavior, when using each one of the deployed architectural schemes.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-01-10. TNO's work has been partially funded by European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. ICT-2011-8-318343 (STEER Project). CWI's work has been partially funded by the European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. ICT-2011-7-287723 (REVERIE Project).Montagud Aguar, M.; Boronat Segui, F.; Stokking, H.; Cesar, P. (2014). Design, Development and Assessment of Control Schemes for IDMS in a Standardized RTCP-based Solution. Computer Networks. 70:240-259. https://doi.org/10.1016/j.comnet.2014.06.004S2402597

A Semantic-Based Middleware for Multimedia Collaborative Applications

The Internet growth and the performance increase of desktop computers have enabled large-scale distributed multimedia applications. They are expected to grow in demand and services and their traffic volume will dominate. Real-time delivery, scalability, heterogeneity are some requirements of these applications that have motivated a revision of the traditional Internet services, the operating systems structures, and the software systems for supporting application development. This work proposes a Java-based lightweight middleware for the development of large-scale multimedia applications. The middleware offers four services for multimedia applications. First, it provides two scalable lightweight protocols for floor control. One follows a centralized model that easily integrates with centralized resources such as a shared too], and the other is a distributed protocol targeted to distributed resources such as audio. Scalability is achieved by periodically multicasting a heartbeat that conveys state information used by clients to request the resource via temporary TCP connections. Second, it supports intra- and inter-stream synchronization algorithms and policies. We introduce the concept of virtual observer, which perceives the session as being in the same room with a sender. We avoid the need for globally synchronized clocks by introducing the concept of user\u27s multimedia presence, which defines a new manner for combining streams coming from multiple sites. It includes a novel algorithm for estimation and removal of clock skew. In addition, it supports event-driven asynchronous message reception, quality of service measures, and traffic rate control. Finally, the middleware provides support for data sharing via a resilient and scalable protocol for transmission of images that can dynamically change in content and size. The effectiveness of the middleware components is shown with the implementation of Odust, a prototypical sharing tool application built on top of the middleware

MediaPlayer™ versus RealPlayer™ - A Comparison of Network Turbulence

The development of higher speed Internet connections and improvements in streaming media technology promise to increase the volume of streamed media over the Internet. The performance of currently available streaming media products will play an important role in the network impact of streaming media. However, there are few empirical studies that analyze the network traffic characteristics and Internet impact of current streaming media products. This paper presents analysis from an empirical study of the two dominant streaming multimedia products, RealNetworks RealPlayer™ and Microsoft MediaPlayer™. Utilizing two custom media player measurement tools, RealTracker and MediaTracker, we are able to gather application layer and network layer information about RealPlayer and MediaPlayer for the same media under the same network conditions. Our analysis shows that RealPlayer and MediaPlayer have distinctly different behavior characteristics. The packet sizes and rates generated by MediaPlayer are essentially CBR while the packet sizes and rates generated by RealPlayer are more varied. During initial delay buffering, MediaPlayer sends data at the same rate as during playout while RealPlayer can buffer at up to three times the playout rate. For high bandwidth clips, MediaPlayer sends frames that are larger than the network MTU, resulting in multiple IP fragments for each application level frame. From the application perspective, for low bandwidth clips, MediaPlayer has a lower frame rate than RealPlayer. Our work exposes some of the impact of streaming media on the network and provides valuable information for building more realistic streaming media simulations