285 research outputs found

    Network-supported layered multicast transport control for streaming media

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    Multicast is very efficient in distributing large volume of data to multiple receivers over the Internet. Layered multicast helps solve the heterogeneity problem in multicast delivery. Extensive work has been done in the area of layered multicast, for both congestion control and error control. In this paper, we focus on network-supported protocols for streaming media. Most of the existing work solves the congestion control and error control problems separately, and do not give an integrated, efficient solution. In this paper, after reviewing related work, we introduce our proposed protocols, RALM and RALF. The former is a congestion control protocol and the latter is an error control protocol. They work under the same framework and provide an integrated solution. We also extend RALM to RALM-II, which is compatible with TCP traffic. We analyze the complexity of the proposed protocols in the network and investigated their performance through simulations. We show that our solution achieves significant performance gains with reasonable additional complexity. © 2007 IEEE.published_or_final_versio

    System Support for Bandwidth Management and Content Adaptation in Internet Applications

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    This paper describes the implementation and evaluation of an operating system module, the Congestion Manager (CM), which provides integrated network flow management and exports a convenient programming interface that allows applications to be notified of, and adapt to, changing network conditions. We describe the API by which applications interface with the CM, and the architectural considerations that factored into the design. To evaluate the architecture and API, we describe our implementations of TCP; a streaming layered audio/video application; and an interactive audio application using the CM, and show that they achieve adaptive behavior without incurring much end-system overhead. All flows including TCP benefit from the sharing of congestion information, and applications are able to incorporate new functionality such as congestion control and adaptive behavior.Comment: 14 pages, appeared in OSDI 200

    A Framework for Controlling Quality of Sessions in Multimedia Systems

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    Collaborative multimedia systems demand overall session quality control beyond the level of quality of service (QoS) pertaining to individual connections in isolation of others. At every instant in time, the quality of the session depends on the actual QoS offered by the system to each of the application streams, as well as on the relative priorities of these streams according to the application semantics. We introduce a framework for achieving QoSess control and address the architectural issues involved in designing a QoSess control laver that realizes the proposed framework. In addition, we detail our contributions for two main components of the QoSess control layer. The first component is a scalable and robust feedback protocol, which allows for determining the worst case state among a group of receivers of a stream. This mechanism is used for controlling the transmission rates of multimedia sources in both cases of layered and single-rate multicast streams. The second component is a set of inter-stream adaptation algorithms that dynamically control the bandwidth shares of the streams belonging to a session. Additionally, in order to ensure stability and responsiveness in the inter-stream adaptation process, several measures are taken, including devising a domain rate control protocol. The performance of the proposed mechanisms is analyzed and their advantages are demonstrated by simulation and experimental results

    Scaleable audio for collaborative environments

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    This thesis is concerned with supporting natural audio communication in collaborative environments across the Internet. Recent experience with Collaborative Virtual Environments, for example, to support large on-line communities and highly interactive social events, suggest that in the future there will be applications in which many users speak at the same time. Such applications will generate large and dynamically changing volumes of audio traffic that can cause congestion and hence packet loss in the network and so seriously impair audio quality. This thesis reveals that no current approach to audio distribution can combine support for large number of simultaneous speakers with TCP-fair responsiveness to congestion. A model for audio distribution called Distributed Partial Mixing (DPM) is proposed that dynamically adapts both to varying numbers of active audio streams in collaborative environments and to congestion in the network. Each DPM component adaptively mixes subsets of its input audio streams into one or more mixed streams, which it then forwards to the other components along with any unmixed streams. DPM minimises the amount of mixing performed so that end users receive as many separate audio streams as possible within prevailing network resource constraints. This is important in order to allow maximum flexibility of audio presentation (especially spatialisation) to the end user. A distributed partial mixing prototype is realised as part of the audio service in MASSIVE-3. A series of experiments over a single network link demonstrate that DPM gracefully manages the tradeoff between preserving stable audio quality and being responsive to congestion and achieving fairness towards competing TCP traffic. The problem of large scale deployment of DPM over heterogeneous networks is also addressed. The thesis proposes that a shared tree of DPM servers and clients, where the nodes of the tree can perform distributed partial mixing, is an effective basis for wide area deployment. Two models for realising this in two contrasting situations are then explored in more detail: a static, centralised, subscription-based DPM service suitable for fully managed networks, and a fully distributed self-organising DPM service suitable for unmanaged networks (such as the current Internet)

    Scalable reliable on-demand media streaming protocols

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    This thesis considers the problem of delivering streaming media, on-demand, to potentially large numbers of concurrent clients. The problem has motivated the development in prior work of scalable protocols based on multicast or broadcast. However, previous protocols do not allow clients to efficiently: 1) recover from packet loss; 2) share bandwidth fairly with competing flows; or 3) maximize the playback quality at the client for any given client reception rate characteristics. In this work, new protocols, namely Reliable Periodic Broadcast (RPB) and Reliable Bandwidth Skimming (RBS), are developed that efficiently recover from packet loss and achieve close to the best possible server bandwidth scalability for a given set of client characteristics. To share bandwidth fairly with competing traffic such as TCP, these protocols can employ the Vegas Multicast Rate Control (VMRC) protocol proposed in this work. The VMRC protocol exhibits TCP Vegas-like behavior. In comparison to prior rate control protocols, VMRC provides less oscillatory reception rates to clients, and operates without inducing packet loss when the bottleneck link is lightly loaded. The VMRC protocol incorporates a new technique for dynamically adjusting the TCP Vegas threshold parameters based on measured characteristics of the network. This technique implements fair sharing of network resources with other types of competing flows, including widely deployed versions of TCP such as TCP Reno. This fair sharing is not possible with the previously defined static Vegas threshold parameters. The RPB protocol is extended to efficiently support quality adaptation. The Optimized Heterogeneous Periodic Broadcast (HPB) is designed to support a range of client reception rates and efficiently support static quality adaptation by allowing clients to work-ahead before beginning playback to receive a media file of the desired quality. A dynamic quality adaptation technique is developed and evaluated which allows clients to achieve more uniform playback quality given time-varying client reception rates

    Optimization-based rate control in overlay multicast.

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    Zhang Lin.Thesis (M.Phil.)--Chinese University of Hong Kong, 2004.Includes bibliographical references (leaves 74-78).Abstracts in English and Chinese.Chapter Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Why use economic models? --- p.1Chapter 1.2 --- Why Overlay? --- p.2Chapter 1.3 --- Our Contribution --- p.3Chapter 1.4 --- Thesis Organization --- p.5Chapter Chapter 2 --- Related Works --- p.7Chapter 2.1 --- Overlay Multicast --- p.7Chapter 2.2 --- IP Multicast Congestion Control --- p.11Chapter 2.2.1 --- Architecture Elements of IP Multicast Congestion Control --- p.11Chapter 2.2.2 --- Evaluation of Multicast Video --- p.13Chapter 2.2.3 --- End-to-End Schemes --- p.14Chapter 2.2.4 --- Router-supported Schemes --- p.16Chapter 2.2.5 --- Conclusion --- p.19Chapter 2.3 --- Optimization-based Rate Control in IP unicast and multicast --- p.20Chapter 2.3.1 --- Optimization-based Rate Control for Unicast Sessions --- p.21Chapter 2.3.2 --- Optimization-based Rate Control for Multi-rate Multicast Sessions --- p.24Chapter Chapter 3 --- Overlay Multicast Rate Control Algorithms --- p.27Chapter 3.1 --- Motivations --- p.27Chapter 3.2 --- Problem Statement --- p.28Chapter 3.2.1 --- Network Model --- p.28Chapter 3.2.2 --- Problem Formulation --- p.29Chapter 3.2.3 --- Algorithm Requirement --- p.33Chapter 3.3 --- Primal-based Algorithm --- p.34Chapter 3.3.1 --- Notations --- p.34Chapter 3.3.2 --- An Iterative Algorithm --- p.36Chapter 3.3.3 --- Convergence Analysis --- p.37Chapter 3.3.3.1 --- Assumptions --- p.37Chapter 3.3.3.2 --- Convergence with various step-sizes --- p.39Chapter 3.3.3.3 --- Theorem Explanations --- p.39Chapter 3.4 --- Dual-based Algorithm --- p.40Chapter 3.4.1 --- The Dual Problem --- p.41Chapter 3.4.2 --- Subgradient Algorithm --- p.43Chapter 3.4.3 --- Interpretation of the Prices --- p.44Chapter 3.4.4 --- Convergence Analysis --- p.45Chapter Chapter 4 --- Protocol Description and Performance Evaluation --- p.47Chapter 4.1 --- Motivations --- p.47Chapter 4.2 --- Protocols --- p.47Chapter 4.2.1 --- Notations --- p.48Chapter 4.2.2 --- Protocol for primal-based algorithm --- p.48Chapter 4.2.3 --- Protocol for dual-based algorithm --- p.53Chapter 4.3 --- Performance Evaluation --- p.57Chapter 4.3.1 --- Simulation Setup --- p.57Chapter 4.3.2 --- Rate Convergence Properties --- p.59Chapter 4.3.3 --- Data Rate Constraint --- p.67Chapter 4.3.4 --- Link Measurement Overhead --- p.68Chapter 4.3.5 --- Communication Overhead --- p.70Chapter Chapter 5 --- Conclusion Remarks and Future Work --- p.73References --- p.7

    Equation-Based Congestion Control for Unicast and Multicast Data Streams

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    We believe that the emergence of congestion control mechanisms for relatively-smooth congestion control for unicast and multicast traffic can play a key role in preventing the degradation of end-to-end congestion control in the public Internet, by providing a viable alternative for multimedia flows that would otherwise be tempted to avoid end-to-end congestion control altogether. The design of good congestion control mechanisms is a hard problem, even more so for multicast environments where scalability issues are much more of a concern than for unicast. In this dissertation, equation-based congestion control is presented as an alternative form of congestion control to the well-known TCP protocol. We focus on areas of equation-based congestion control which were not yet well understood and for which no adequate solutions existed. Starting from a unicast congestion control mechanism which in contrast to TCP provides smooth rate changes, we extend equation-based congestion control in several ways. We investigate how it can work together with applications which can only operate in a very limited region of available bandwidth and whose rate can thus not be adapted to the network conditions in the usual way. Such a congestion control mechanism can also complement conventional equation-based congestion control in regimes where available bandwidth is too low for further rate reduction. When extending unicast congestion control to multicast, it is of paramount importance to ensure that changes in the network conditions anywhere in the multicast tree are reported back to the sender as quickly as possible to allow the sender to adjust the rate accordingly. A scalable feedback mechanism that allows timely congestion feedback in the face of potentially very large receiver sets is one of the contributions of this dissertation. But also other components of a congestion control protocol, such as the rate increase/decrease policy or the slow-start mechanism, need to be adjusted to be able to use them in a multicast environment. Our resulting multicast congestion control protocol was implemented in a simulation environment for extensive protocol testing and turned into a library for the use in real-world applications. In addition, a simple video transmission tool was built for test purposes that uses this congestion control library

    Scaleable audio for collaborative environments

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    This thesis is concerned with supporting natural audio communication in collaborative environments across the Internet. Recent experience with Collaborative Virtual Environments, for example, to support large on-line communities and highly interactive social events, suggest that in the future there will be applications in which many users speak at the same time. Such applications will generate large and dynamically changing volumes of audio traffic that can cause congestion and hence packet loss in the network and so seriously impair audio quality. This thesis reveals that no current approach to audio distribution can combine support for large number of simultaneous speakers with TCP-fair responsiveness to congestion. A model for audio distribution called Distributed Partial Mixing (DPM) is proposed that dynamically adapts both to varying numbers of active audio streams in collaborative environments and to congestion in the network. Each DPM component adaptively mixes subsets of its input audio streams into one or more mixed streams, which it then forwards to the other components along with any unmixed streams. DPM minimises the amount of mixing performed so that end users receive as many separate audio streams as possible within prevailing network resource constraints. This is important in order to allow maximum flexibility of audio presentation (especially spatialisation) to the end user. A distributed partial mixing prototype is realised as part of the audio service in MASSIVE-3. A series of experiments over a single network link demonstrate that DPM gracefully manages the tradeoff between preserving stable audio quality and being responsive to congestion and achieving fairness towards competing TCP traffic. The problem of large scale deployment of DPM over heterogeneous networks is also addressed. The thesis proposes that a shared tree of DPM servers and clients, where the nodes of the tree can perform distributed partial mixing, is an effective basis for wide area deployment. Two models for realising this in two contrasting situations are then explored in more detail: a static, centralised, subscription-based DPM service suitable for fully managed networks, and a fully distributed self-organising DPM service suitable for unmanaged networks (such as the current Internet)
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