TL;DR: An improvement of the RLS protocol, called CIFL for "Coding-Independent Fair Layered mulaticast", along two axes, which avoids useless join attempts by learning from its unsuccessful previous attempts in the same network conditions.
Abstract: We propose an improvement of our RLS (Receiver-driven Layered multicast with Synchronization points) protocol, called CIFL for "Coding-Independent Fair Layered mulaticast", along two axes. In CIFL, each receiver of a layered multicast transmission will try and find the adequate number of layers to subscribe to, so that the associated throughput is fair towards TCP and stable in steady-state. The first improvement is that CIFL is not specific to any coding scheme. It can work as well with an exponentially distributed set of layers (where the throughput of each layer i equals the sum of the throughputs of all layers below i), or with layers of equal throughputs, or any other scheme. The second improvement is the excellent stability of the protocol which avoids useless join attempts by learning from its unsuccessful previous attempts in the same (or better) network conditions. Moreover, the protocol tries and reaches its ideal TCP-friendly as soon as possible by computing its target throughput in a clever way when an incipient congestion is confirmed.
TL;DR: A tree structured system with uniform cache sizes performs better than the equivalent system with a proxy-like configuration for transmitting video streams through videoon-demand (VoD) way.
Abstract: In this paper, we study how to distribute cache sizes into a tree structured server for transmitting video streams through videoon-demand (VoD) way. We use off-line smoothing for videos and our request rates are distributed according to a 24 hour audience curve. For this purpose we have designed a slotted-time bandwidth reservation algorithm, which has been used to simulate our experiments. Our system tests the quality of service (QoS) in terms of starting delay, and once a transmission has started, the system guarantees that it will be transmitted without any delay or quality loss. We tested it for a wide range of users (from 800 to 240 000) and also for a different number of available videos. We demonstrate that a tree structured system with uniform cache sizes performs better than the equivalent system with a proxy-like configuration. We also study delay distribution and bandwidth usage in our system on a representative case.
TL;DR: A state transmission framework developed for a middleware platform that was constructed by the authors in an earlier project and integrated in the ARMS platform, which already supported several QoS adaptation mechanisms and reliable transmission in multicast environments is presented.
Abstract: Collaborative virtual environments (CVE) require the use of specially designed mechanisms that allow for a consistent sharing of state among involved users. These mechanisms must, somehow, compensate for network latency and losses in such a way that all players have a single, coherent perception of the system state. Common middleware platforms have difficulty in guaranteeing this consistency, and this is the prime reason why the main research topic for CVEs is the efficient, scalable and reliable transmission of state information. This paper presents a state transmission framework developed for a middleware platform that was constructed by the authors in an earlier project [1]. This middleware platform, Augmented Reliable Multicast CORBA Event Service (ARMS), which already supported several QoS adaptation mechanisms and reliable transmission in multicast environments, was extended with CVE-oriented state transmission mechanisms. After an identification of key requirements of collaborative virtual environments, the relevant features of the proposed state transmission framework are presented. This framework has been integrated in the ARMS platform and was subject to a series of performance tests whose results are included and discussed in this paper. The paper ends with a summary of contributions and an identification of guidelines for further work.
TL;DR: With this new coding approach, the MPEG-4 standard opens new frontiers in the way users will play with, create, re-use, access and consume audiovisual content.
Abstract: MPEG [1] has been responsible for the successful MPEG-1 and MPEG-2 standards that have given rise to widely adopted commercial products and services, such as Video-CD, DVD, digital television, digital audio broadcasting (DAB) and MP3 (MPEG-1 Audio layer 3) players and recorders. More recently, the MPEG-4 standard [2] is aimed to define an audiovisual coding standard to address the emerging needs of the communication, interactive and broadcasting service models as well as of the mixed service models resulting from their technological convergence. The MPEG-4 object-based representation approach where a scene is modeled as a composition of objects, both natural and synthetic, with which the user may interact, is at the heart of the MPEG-4 technology. With this new coding approach, the MPEG-4 standard opens new frontiers in the way users will play with, create, re-use, access and consume audiovisual content. Following the same vision underpinning MPEG-4, MPEG initiated after another standardization project addressing the problem of describing multimedia content to allow the quick and efficient searching, processing and filtering of various types of multimedia material: MPEG-7 [3]. The need for a powerful solution for quickly and efficiently identifying, searching, filtering, etc., various types of multimedia content of interest to the user, human or machine, using also non text-based technologies, directly follows from the urge to efficiently use the available multimedia content and the difficulty of doing so.
TL;DR: This paper presents a separable and reusable QoS management service for end-to-end QoS in a distributed environment, and describes the resulting seven-agent QoS manager, a generic management protocol, and defines interfaces between the agents, platform services, and QoS-aware application components.
Abstract: Research in the area of end-to-end Quality of Service (QoS) has produced important results over the last years. However, most solutions are tailored for specific environments, assume layered system architectures, or integrate QoS management within the respective service components, such that the QoS management functionality is not easily reusable. Furthermore, proprietary QoS solutions are not interoperable and QoS management for logical objects is not supported. In this paper, we present a separable and reusable QoS management service for end-to-end QoS in a distributed environment. This QoS middleware extends the classical feedback controller with QoS-aware agents. We describe the resulting seven-agent QoS manager, a generic management protocol, and define interfaces between the agents, platform services, and QoS-aware application components. Wrappers can be used to interface the QoS middleware with all types of legacy distributed service components, both QoS-aware and QoS-unaware.
TL;DR: The identification of new replication requirements in distributed multimedia systems and a multicast-based update propagation mechanism by which not only the update events are signalled, but also the updated data are exchanged between replication managers is identified.
Abstract: This paper presents the design and implementation architecture of a replication mechanism for a distributed multimedia system medianode which is currently developed as an infrastructure to share multimedia-enhanced teaching materials among lecture groups. The proposed replication mechanism supports the quality of service (QoS) characteristics of multimedia data and the availability of system resources. Each type of data handled and replicated are classified according to their QoS characteristics and replication requirements. The main contribution of this paper is the identification of new replication requirements in distributed multimedia systems and a multicast-based update propagation mechanism by which not only the update events are signalled, but also the updated data are exchanged between replication managers. By prototyping the proposed replication mechanism in medianode, we prove the feasibility of our concept for combining the QoS concept with replication mechanisms.
TL;DR: The paper successively presents the design principles of the proposed architecture, the networking platform on which the architecture has been developed and the experimental measurements validating the IP level mechanisms providing the defined services.
Abstract: Research reported in this paper deals with the design of a communication architecture with guaranteed end-to-end quality of service (QoS) in an IPv6 environment providing differentiated services within a single Diff-Serv domain. The paper successively presents the design principles of the proposed architecture, the networking platform on which the architecture has been developed and the experimental measurements validating the IP level mechanisms providing the defined services. Results presented here have been obtained as part of the experiments in the national French project @IRS (Integrated Architecture of Networks and Services).
TL;DR: A Coordination Protocol (CP) is proposed which allows a C-to-C application to coordinate flow behavior in the face of changing network conditions and provides cluster endpoints with a consistent view of network conditions, as well as cluster membership and bandwidth usage information.
Abstract: Future Internet applications will increasingly use multiple communications and computing devices in a distributed fashion. In this paper, we identify an emerging and important application class comprised of a set of processes on a cluster of devices communicating to a remote set of processes on another cluster of devices across a common intermediary Internet path. We call applications of this type cluster-to-cluster (C-to-C) applications. The networking requirements of C-to-C applications present unique challenges that current transport-level protocols fail to address. In particular, these applications require aggregate measurement of network conditions across all associated flows and coordinated transport-level protocol behavior. A Coordination Protocol (CP) is proposed which allows a C-to-C application to coordinate flow behavior in the face of changing network conditions. CP provides cluster endpoints with a consistent view of network conditions, as well as cluster membership and bandwidth usage information. An application may use CP to define and implement a coordination scheme supporting particular flow priorities and other objectives.
TL;DR: In this work every cluster has a clustering agent that is capable of making membership modification decisions, transferring nodes and splitting or merging clusters, and Communication is used only between neighbouring agents to reduce the signalling overhead.
Abstract: A Mobile ad-hoc network is a multihop wireless network, where nodes communicate with each other without any pre-deployed infrastructure. The most important problem on such dynamic networks is to find routing algorithms well performing in most cases. Cluster based algorithms are among the most effective and scaleable approaches. Up till now creation and maintenance clusters were mostly based on basic heuristic methods. Deploying mobile agents has several advantages in the ad-hoc environment due to their flexible, robust and autonomous nature, and their use seems promising for the clustering problem as well. In our proposed architecture every cluster has a clustering agent that is capable of making membership modification decisions, transferring nodes and splitting or merging clusters. Communication is used only between neighbouring agents to reduce the signalling overhead. Clustering decisions can be based on several network parameters modified by an adaptation mechanism to provide adequate performance even under dynamic conditions.
TL;DR: It is demonstrated to what extent this token-based, soft-state access control mechanism improves security and robustness, and offers improved performance over that provided by existing approaches within roaming networks.
Abstract: This paper introduces a novel access control architecture for publicly accessible, wireless networks. The architecture was designed to address the requirements obtained from a case study of ubiquitous Internet service provisioning within the city of Lancaster. The proposed access control mechanism is based on the concepts of secure user authentication, packet marking, and packet filtering at the access routers. The paper demonstrates to what extent this token-based, soft-state access control mechanism improves security and robustness, and offers improved performance over that provided by existing approaches within roaming networks. Early indications show the access control mechanism can better be implemented through the use of active routers, in order to facilitate dynamic rollout and configuration of the system. In addition, extensions to Mobile IPv6 are proposed, which provide support for roaming users at a fundamental level.
TL;DR: Using a quality metric that is based on the properties of the human visual perception process, mechanisms that improve the overall session quality are devised by efficiently apportioning the session bandwidth to the participating flows at appropriate adaptation times are devised.
Abstract: While a considerable amount of research has been conducted to address QoS issues for best-effort Internet multimedia applications by utilising network-centric metrics (loss, delay, RTT, available bandwidth), less attention has been paid to the quality that is perceived by the users of the networked applications. Perceived quality of encoded multimedia is highly dependent on the time-varying characteristics of the content. We describe an approach for content-aware quality adaptation of multimedia sessions consisting of an ensemble of concurrent flows relevant to the presentation scenario. Using a quality metric that is based on the properties of the human visual perception process, we devise mechanisms that improve the overall session quality by efficiently apportioning the session bandwidth to the participating flows at appropriate adaptation times. We discuss the approach, propose suitable adaptation time scales and present results from trace-driven simulations that show the potential of content-aware quality adaptation.
TL;DR: This paper describes mobile 4-in-6, an extension to the Mobile IPv6 protocol that allows mobile nodes to transparently communicate with the IPv4 Internet irrespective of their current location on the IPv6 Internet.
Abstract: Mobility is indisputably one of the major drives promoting IPv6. Making the transition to IPv6 from IPv4, however, is one of the major stumbling blocks. It has recently come to light that Mobile IPv6 (the IETF routing protocol for IPv6 mobile hosts) may hold some of the keys to making this transition. This paper describes mobile 4-in-6, an extension to the Mobile IPv6 protocol that allows mobile nodes to transparently communicate with the IPv4 Internet irrespective of their current location on the IPv6 Internet. Furthermore, the protocol does not require the mobile node to hold an IPv4 address other than on its home network, and does not assume the availability of any Mobile IPv4 services.
TL;DR: This work addresses both issues that it sees as shortcomings of current models, including integrity on state and state transition, as well as group management action and transition policies.
Abstract: Multimedia multicasting brings together two technologies considered to be cornerstones of the future Internet, where rich media content will be distributed to a mass audience. It is obvious that unicasting will not be adequate for such content distribution due to the unacceptable stress imposed on network resources. Multicasting is a solution to this problem, and thus multicast routing and group management are now receiving high attention. However, so far, conditions on the composition of multicast groups have been kept rather simple. Requirements on group membership, member roles and group organization, commonly referred to as group integrity conditions, are rarely addressed or even enforced. Furthermore, the traditional multicasting model has been flat, with no finer granularity than a group, lacking any inter-group relationships. In our work, we address both issues that we see as shortcomings of current models. Our framework allows us to subdivide multicast groups into subgroups (e.g. for high and low quality versions of a media stream) and to form and manage meta groups from groups, integrating inter-related "multi" media groups. On all three levels, we specify a variety of integrity conditions as part of our comprehensive policy framework, including integrity on state and state transition, as well as group management action and transition policies.
TL;DR: The empirical work presented here concerns the effects of very large packet loss and delay conditions upon the performance of an H.323-based VoIP service and audio voice quality under these conditions, and shows that uncontrolled jitter was the most destructive parameter to listening quality.
Abstract: The empirical work presented here concerns the effects of very large packet loss and delay conditions upon the performance of an H.323-based VoIP service. Specifically, it examines call establishment performance and audio voice quality under these conditions. The work was performed as part of an investigation concerning the effects of GEO satellite environments upon H.323- based VoIP, but its findings are relevant to all kinds of networks which may suffer or become subject to highly aberrant transmission conditions. The call establishment tests showed that the H.323 protocols could establish calls successfully under severely impaired conditions. The listening tests showed that uncontrolled jitter was the most destructive parameter to listening quality. Packet loss was also influential, though somewhat less dramatically.
TL;DR: A Quality of Service negotiation scheme for efficient failure recovery in multi-resolution video servers with disk arrays that can increase the number of admitted clients greatly when all disks are operational but also utilize server resources efficiently.
Abstract: In this paper, we present a Quality of Service (QoS) negotiation scheme for efficient failure recovery in multi-resolution video servers with disk arrays. This scheme exploits multi-resolution property of video streams by negotiating service resolutions in order to provide graceful QoS degradation when a disk fails. Using the proposed scheme, not only can we increase the number of admitted clients greatly when all disks are operational but also utilize server resources efficiently. Furthermore, it can provide each client with acceptable QoS even in the presence of disk failure while maximizing server-perceived rewards. The effectiveness of the proposed algorithm is evaluated through simulation-based experiments.
TL;DR: A selection algorithm that is based on the user profiles and the device capabilities is developed and partition the receivers according to the bandwidth limit of their devices into classes and use a virtual representative for each class of receivers.
Abstract: Most adaptive multimedia multicast applications require the source to select the number of streams to transmit as well as the QoS parameters for each stream. If the receivers have different bandwidth limits for their devices and have various preferences for the quality of the data, selecting the QoS parameters that generate the best average satisfaction for all receivers is a challenging problem. In this paper, we developed a selection algorithm that is based on the user profiles and the device capabilities. Receivers are required to send their profiles and the bandwidth limitation on their devices to the source once before the session starts. To avoid the implosion problem and have a constant running time for the selection algorithm, we partition the receivers according to the bandwidth limit of their devices into classes and use a virtual representative for each class of receivers.
TL;DR: The feasibility of performing constraint based configuration of the required proxylets is examined and the approach to incorporate dynamic configuration to cater for load and QoS variations is discussed.
Abstract: Applications such as multimedia streaming for mobile users, or video conferencing, require support within the network for transcoding, compression etc. Proxylets running on servers within the network may be used to transform the media flows in order to meet application or QoS requirements. In this paper we examine the feasibility of performing constraint based configuration of the required proxylets. A set of constraints can be defined to select the required proxylets. A second stage is to define constraints relating to the placement of proxylets on nodes in the network. Eventually we will investigate the use of constraints for dynamic re-configuration to accommodate user mobility, or QoS variation. Some preliminary implementations of the architecture are presented and we discuss our approach to incorporate dynamic configuration to cater for load and QoS variations.
TL;DR: It is claimed that the renting of machine resources on clusters of servers introduces new systems challenges which are different from those hitherto encountered, either in multimedia systems or cluster-based computing.
Abstract: We claim that the renting of machine resources on clusters of servers introduces new systems challenges which are different from those hitherto encountered, either in multimedia systems or cluster-based computing. We characterize the requirements for such "public computing platforms" and discuss both how the scenario differs from more traditional multimedia resource control situations, and how some ideas from multimedia systems work can be reapplied in this new context. Finally, we discuss our ongoing work building a prototype public computing platform.
TL;DR: This paper claims that a single architectural model, taking into account both end- system and network resource management is needed, and proposes a QoS architecture, which extends the concept of service differentiation inside the end-system.
Abstract: A number of research studies show that the operating system has a substantial influence on communication delay in distributed environments. Thus, in order to provide applications with end-to-end QoS guarantees, network resource management alone is not sufficient. To date, several approaches have been proposed, addressing QoS issues from the end-system point of view. However, while network QoS provisioning has achieved a good level of standardization, no standard proposals exist for the end-systems. We claim that a single architectural model, taking into account both end-system and network resource management is needed. In this paper we propose a QoS architecture, which extends the concept of service differentiation inside the end-system. A system prototype has been developed and tested on a Diffserv network scenario. The prototype incorporates a priority-based communication mechanism inside the end-point operating system and a local marker so to implement an appropriate mapping between local classes of services and network QoS levels. Experimental results have also been provided in order to evaluate the impact of the proposed architecture on Diffserv routers.
TL;DR: This paper presents a new video content management that uses logical content, which makes it possible to access physical video files regardless of their formats, and to easily manage video content consisting of multiple physical files.
Abstract: A common video file format is generally used in video content management system composed of modules for archiving, retrieving, and video streaming on a network. However, the file format may not be suitable for these functions. This paper presents a new video content management that uses logical content. Logical content makes it possible to manage a system that may use files of differing formats to store the same content to be used by different modules. Using logical content makes it possible to access physical video files regardless of their formats, and to easily manage video content consisting of multiple physical files. Other modules are able to request the retrieval of a physical file by specifying a logical content ID and its range. Consequently, this system can be designed to be both modularized and extended. We also present an experimental video content management system that verifies the effectiveness of our video content management system.
TL;DR: A middleware system--called Narnia-- is described that supports the development and deployment of these application-aware resource controls and shows how application developers can create flexible, dynamic resource managers as part of multimedia service implementations.
Abstract: This paper highlights some early results from our current research. We are building prototype multimedia services, e.g., digital television, video on demand, and teleconferencing, for metropolitan area networks. Our work illustrates that application-aware resource management plays a critical role in providing such services economically. The paper describes a middleware system--called Narnia--that supports the development and deployment of these application-aware resource controls. Using Narnia programming abstractions, application developers can create flexible, dynamic resource managers as part of multimedia service implementations. The paper also presents a brief summary of our experiences using Narnia to build applications.
TL;DR: This paper gives an overview of MPEG-7 and then presents the design and implementation of a prototype tool, MCAT, that supports annotation of audiovisual objects using standardized, contentdescription metadata so that annotations can be shared between other applications and systems.
Abstract: The XML metadata technology for describing Web objects has emerged as a dominant mode of making information available for both human and machine consumption. Many online Web applications are adopting this concept for their domain-specific applications. For a metadata model to become widely used, a standardization effort is often required so that the metadata content and its structure may be interoperable between applications. The ISO MPEG standards body, with the charter of defining standard metadata to describe audiovisual content, has developed a new metadata technology called MPEG-7 content description. An effective collaboration annotation tool should also use standard metadata structures in order to improve a collaboration environment. This paper gives an overview of MPEG-7 and then presents the design and implementation of a prototype tool, MCAT. The tool supports annotation of audiovisual objects using standardized, contentdescription metadata so that annotations can be shared between other applications and systems.
TL;DR: This paper derives the minimal buffering requirement for this policy assuming that applications use an additive-increase-and-multiplicative-decrease (AIMD) algorithm for congestion control and shows the relationship between parameters of AIMD algorithms and the delay cost.
Abstract: This paper uses analysis and experiments to study the minimal buffering requirements of congestion controlled multimedia applications. Applications in the Internet must use congestion control protocols, which vary transmission rates according to network conditions. To produce a smooth perceptual quality, multimedia applications use buffering and rate adaptations to compensate these rate oscillations. While several adaptation policies are available, they require different amounts of buffering at end-hosts. We study the relationship between buffering requirements and adaptation policies. In particular, we focus on a widely pursued policy that adapts an application's sending rate exactly to the average available bandwidth to maximize throughput. Under this adaptation policy, at least a minimal amount of buffering is required to smooth the rate oscillation inherent in congestion control, and we view this minimal buffering requirement as a cost of maximizing throughput. We derive the minimal buffering requirement for this policy assuming that applications use an additive-increase-and-multiplicative-decrease (AIMD) algorithm for congestion control. The result shows the relationship between parameters of AIMD algorithms and the delay cost. We show that the buffering requirement is proportional to the parameters of the AIMD algorithm and quadratic to the application's sending rate and round-trip-time. We verify this relationship through experiments. Our results indicate that adaptation policies that maximize throughput are not suitable for interactive applications with high bit rates or long round-trip-times.
TL;DR: The TOAST middleware platform which provides component-oriented CORBA support for adaptive distributed multimedia applications is outlined, and how the areas of reflection and open implementation have shaped the approach to adaptation support in TOAST is examined.
Abstract: The rise of mobile computing and wireless network technology means that, increasingly, applications must adapt to their environment, in particular network connectivity and resource availability. This paper outlines the TOAST middleware platform which provides component-oriented CORBA support for adaptive distributed multimedia applications. In particular, the paper examines how the areas of reflection and open implementation have shaped the approach to adaptation support in TOAST. The paper then discusses novel ongoing research which is investigating middleware support for distributed cooperative visualization using TOAST as a base.
TL;DR: This paper revisits and challenges the dogma that TCP is an undesirable choice for streaming multimedia, video in particular, and concerns over the marginal benefit of changing TCP's service model, given the presence of congestion avoidance.
Abstract: In this paper, we revisit and challenge the dogma that TCP is an undesirable choice for streaming multimedia, video in particular. For some time, the common view held that neither TCP nor UDP, the Internet's main transport protocols, are adequate for video applications. UDP's service model doesn't provide enough support to the application while TCP's provides too much. Consequently, numerous research works proposed new transport protocols with alternate service-models as more suitable for video. For example, such service models might provide higher reliability than UDP but not the full-reliability of TCP. More recently, study of Internet dynamics has shown that TCP's stature as the predominant protocol persists. Through some combination of accident and design, TCP's congestion avoidance mechanism seems essential to the Internet's scalability and stability. Research on modeling TCP dynamics in order to effectively define the notion of TCP-friendly congestion avoidance is very active. Meanwhile, proposals for video-oriented transport protocols continue to appear, but they now generally include TCP-friendly congestion avoidance. Our concern is over the marginal benefit of changing TCP's service model, given the presence of congestion avoidance. As a position paper, our contribution will not be in the form of final answers, but our hope is to convince the reader of the merit in reexamining the question: do applications need a replacement for TCP in order to do stream ing video?