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In wireless networks, the physical medium is the cause of most of the errors and performance drops. Thus, an efficient predictive estimation of wireless networks performance w.r.t. medium status by the communication peers would be a leap ahead in the improvement of wireless communication. For that purpose, we designed a measurement bench that allows us to accurately control the noise level on an unidirectional WIFI communication link in the protected environment of an anechoic room. This way, we generated different medium conditions and collected several measurements for various PHY layer parameters on that link. Using the collected data, we analyzed the ability to predictively estimate the throughput performance of a noisy wireless link from measured physical medium parameters, using machine learning (ML) algorithms. For this purpose, we chose two different classes of ML algorithms, namely SVR (Support Vector Regression) [1] and k-NN (k-Nearest Neighbors) [2] to study the tradoff between complexity and estimation accuracy. Finally, we ranked the pertinence of the most common physical parameters for estimating or predicting the throughput that can be expected by users on top of the IP layer over a WIFI link. [ABSTRACT FROM AUTHOR]

This paper aims at proposing a methodology for evaluating current IDS capabilities of detecting attacks targeting the networks and their services. This methodology tries to be as realistic as possible and reproducible, i.e. it works with real attacks and real traffic in controlled environments. It especially relies on a database containing attack traces specifically created for that evaluation purpose. By confronting IDS to these attack traces, it is possible to get a statistical evaluation of IDS, and to rank them according to their detection capabilities without false alarms. For illustration purposes, this paper shows the results obtained with 3 public IDS. It also shows how the attack traces database impacts the results got for the same IDS. [ABSTRACT FROM AUTHOR]

This paper deals with a new iterative Network Anomaly Detection Algorithm - NADA, which accomplishes the detection, classification and identification of traffic anomalies. NADA fully provides all information required limiting the extent of anomalies by locating them in time, by classifying them, and identifying their features as, for instance, the source and destination addresses and ports involved. To reach its goal, NADA uses a generic multi-featured algorithm executed at different time scales and at different levels of IP aggregation. Besides that, the NADA approach contributes to the definition of a set of traffic anomaly behavior-based signatures. The use of these signatures makes NADA suitable and efficient to use in a monitoring environment. [ABSTRACT FROM AUTHOR]

Multicast conference recording currently uses a single, arbitrarily placed recorder. The recordings have less than perfect quality because they are often not loss-free, and the recording is as perceived by the recorder (delays from senders to the recorder have been imposed on the streams). An ultimate recording consists of the loss-free data set, extra information about loss patterns received by individual participants, and information about the delays between participants. The delay information is required so that we can re-create the conference from a number of perception points. In this paper, we initially consider off-line recording of material, which produces loss-free recording, can provide timing information, and is simple, but does suffer from deficiencies. Our second (and preferred) solution uses a distributed arrangement of a number of co-operating recording caches, and sender interaction for local repair. [ABSTRACT FROM AUTHOR]

For distributed multimedia applications, adaptive QoS management mechanisms need to be developed to guarantee various end-to-end QoS requirements. In this paper, we propose an adaptive QoS management framework based on multi-agent systems. Adaptation to system resource changes and various user requirements is achieved by the direct or indirect collaboration of agents in the respective phases. In the flow establishment and re-negotiation phase, application agents determine the optimal resource allocation through QoS negotiations to maximize the total users' utility. In the media-transfer phase, stream agents collaborate to adjust each stream QoS reactively. In addition, personal agents help novice users specify stream QoS without any a priori knowledge on QoS. Some multimedia applications have been designed based on the proposed framework. [ABSTRACT FROM AUTHOR]

Video-on-demand servers (VODS) must be based on cost-effective architectures. Therefore architectures based on clusters of PCs will probably be the most suitable to build VODS; they provide the same performance of a large server through the aggregation of many smaller, inexpensive nodes. In this paper, we show that the interconnection network used to build the cluster can be very expensive if it is based on a set of switches. To obtain the most cost-effective architecture, we argue that VODS must be Dual Counter-Rotating Ring-based (DCRR-based). DCRR are very inexpensive and fulfill all the basic criteria needed for VODS architectures except scalability. To address the scalability issue, we propose to enhance the design of DCRR by partitioning it logically in combination with three new policies ("fast stream migration", "stream splitting" and "distributed XORing"). This design brings very cost-effective and scalable VODS able to play up to 13500 MPEG-2 concurrent streams using 252 nodes. [ABSTRACT FROM AUTHOR]

As the Internet evolves into a global commercial infrastructure, there is a growing need to support more enhanced services than the traditional best-effort service. Differentiated services are a suitable solution to quality of service provisioning in the Internet while the number of users keeps growing. However, recent advances in differentiated service models have not spawned much work in pricing yet. This is due to the complexity of economic elements involved and the fact that the shape of the differentiated services Internet is still not fully defined. In this position paper, we briefly review the state of the art in this relatively new area and compare some of the main existing pricing approaches. The impact of pricing schemes on differentiated services is emphasized. Based on a discussion of unsolved issues, possible research directions are identified. [ABSTRACT FROM AUTHOR]

Resource reservation is required to guarantee delivery of continuous media data from a server across a network for continuous playback by a client. This paper addresses the characterization of the network bandwidth requirements of Variable Bit Rate data streams and the corresponding admission control mechanism at the server. We show that a characterization which sends data early, making intelligent use of client buffer space, reduces the amount of network bandwidth reserved per stream without creating any start-up latency. The results of performance experiments in a Continuous Media File Server find that operation with requests arriving over time can deliver up to 90% of the network bandwidth. The experiments also show that a system designer can configure a server so that the network and disk bandwidth can scale together. [ABSTRACT FROM AUTHOR]

Mirroring-based reliability as compared to parity-based reliability significantly simplifies the design and the implementation of video servers, since in case of failure mirroring does not require any synchronization of reads or decoding to reconstruct the lost video data. While mirroring doubles the amount of storage volume required, the steep decrease of the cost of magnetic disk storage makes it more and more attractive as a reliability mechanism. We present in this paper a novel data layout strategy for replicated data on a video server. In contrast to classical replica placement schemes that store original and replicated data separately, our approach stores replicated data adjacent to original data and thus does not require additional seek overhead when operating with disk failure. We show that our approach considerably improves the server performance compared to classical replica placement schemes such as the interleaved declustering scheme and the scheme used by the Microsoft Tiger video server. Our performance metric is the maximum number of users that a video server can simultaneously support (server throughput). [ABSTRACT FROM AUTHOR]

This paper presents a framework for the development of advanced video conferencing systems with very high quality. The design and performance of the developed components like session management, real-time scheduler and a specific transport system are outlined. The core of the framework consists of a toolkit for the processing of audio and video data. It integrates capturing, compression, transmission and presentation of media streams into an object oriented class library. The specifics of audio/video hardware, preferred operation modes, compression algorithms and transport protocol properties are encapsulated. Due to the achieved abstraction level complex application software can easily be developed resulting in very short design cycles. Contrary to existing systems the developed application is regarded in its entirety. The focus on highest possible performance throughout all components is unique to our approach. Due to a number of optimisations, specific compression hardware is not required which leads to low overall system costs. [ABSTRACT FROM AUTHOR]

This paper proposes a scheme for implementing multiparty videoconferencing service on SONET/ATM rings. Different multicasting methods on SONET/ATM rings are discussed and compared. A new multicast VP called "Multi-drop VP" which is particularly suitable for SONET/ATM rings is proposed. An Add-Drop Multiplexer(ADM) structure for rings capable of multi-dropping is also presented. Several VP assignment schemes are proposed and their bandwidth utilizations are compared. [ABSTRACT FROM AUTHOR]

Interactive media streams with real-time characteristics, such as those produced by shared whiteboards, distributed Java applets or shared VRML viewers, are rapidly gaining importance. Current solutions to the recording of interactive media streams are limited to one specific application (e.g. one specific shared whiteboard). In this paper we present a generic recording service that enables the recording and playback of this new class of media. To facilitate the generic recording we have defined a profile for the Real-Time Transport Protocol (RTP) that covers common aspects of the interactive media class in analogy to the profile for audio and video. Based on this profile we introduce a generalized recording service that enables the recording and playback of arbitrary interactive media. [ABSTRACT FROM AUTHOR]

The development of new multimedia services and environments requires new concepts both to support the new working process on distributed computers and to protect the multimedia data during the production and distribution. This article addresses image/video authentication and copyright protection as major security demands in digital marketplaces. First we present a content-based signature technique for image and video authenticity and integrity. Based on this technique, we introduce a tool for interactive video authentication and propose content-fragile watermarking, a concept which combines watermarking and content-based digital signatures to ensure copyright protection and detection of integrity violation. [ABSTRACT FROM AUTHOR]

In this paper 3 is presented an adaptive video compression algorithm designed for the video transmission over best-effort network with services that can adapt to changing network conditions, specifically Available Bit Rate (ABR) services in Asynchronous Transfer Mode (ATM) networks. The proposed algorithm tries to minimise the impact that cell losses could have on the perceived quality, using perceptual bit allocation procedures and Rate-Distortion minimisation techniques. Furthermore the algorithm adaptively estimates the compression ratio by means of a forecast mechanism based on the value of the feedback signal sent by the network. It should be noticed this algorithm is based on a 3D subband coding that uses wavelet filter banks but it does not use motion estimation with block matching procedures to prevent from error propagation effects. [ABSTRACT FROM AUTHOR]

Software composition and frameworks are currently important technologies for multimedia services development, especially on the Web. These technologies can be successfully applied to multimedia collaborative services achieving the derivation of new services with shorter development time. In this paper we present the collaborative aspects of MultiTEL, a compositional framework for developing distributed multimedia applications over the Web, implemented using Java/RMI. MultiTEL is based on a new compositional model, which separate computation and communication aspects in different entities, components and connectors. MultiTEL also includes a middleware platform that supports local resource management, allowing the plug&play of multimedia device components. Currently, our goal is to extend MultiTEL with collaborative issues allowing the development of cooperative services, like application sharing, blackboard, etc. We will address the main collaborative application features of MultiTEL through a case study. [ABSTRACT FROM AUTHOR]

With the advent of multi-channel digital TV, accessing Electronic Programme Guides (EPGs) will become increasingly complex. This paper describes ongoing work in the development of a conversational interface for Interactive TV. We claim that a conversational interface can support incremental refinement of user selections thus assisting user choice without requiring knowledge of editorial categories. We describe the system architecture, which integrates various components dedicated to language processing, EPG access and animation of the conversational character. Finally, we discuss the control strategy for the whole system from the perspective of human-computer dialogue. [ABSTRACT FROM AUTHOR]

Teletext will be one of the first and most prominent multimedia services offered in digital television networks. Teletext is well known and has already been available for many years in analogue television in Europe and elsewhere as an additional service provided by television operators. Digital Broadcast technology allows to enhance this kind of service in quality and functionality. Digital Television technology provides many advantages in comparison to the analogue distribution methods, such as possible reuse of content from Internet-based services, improved navigation, better layout capabilities and support of animated content. These capabilities give more freedom to the application designers. However, it is yet undecided which application format is best suited for Digital Television Networks. In order to develop a large market it is required to consider standardised formats to ensure interoperability between different service providers. Unfortunately, various industrial consortia such as DVB/MHP, ATSC/DASE and DigiTAG are working on different solutions. The formats have to be supported by the designers of the applications, the network operators and the end systems, and specific tools have to be developed to enable application production taking the advantages of the new technology into account. This paper discusses various aspects of the production of Teletext applications for Digital Television Broadcast. The authoring system presented here is based on a generic approach to describe the applications' layout and behaviour utilising XML data representation, templates and server-side scripting. The system is independent of the final interchange format and can generate several different representations such as MHEG-5 and XHTML to meet the requirements of upcoming broadcast environments and to enable publication of applications in the Internet using the same content source. Navigation style-guides are covered and techniques to create applications and the associated content via server-side scripts are discussed. [ABSTRACT FROM AUTHOR]

The retrieval of multimedia objects is influenced by factor such as throughput and maximum delay offered by the network, and has to be carried out in accordance with the specification of object relationships. Many current network architectures address QoS from a provider's point of view and analyze network performance, failing to comprehensively address the quality needs of applications. The work presented in this paper concerns the development of an integrated environment for creation and retrieval of multimedia documents, that intends to preserve the coherence between the different media, even when the process is confronted with a temporary lack of communication resources. This environment implements a communication system that address QoS from the application's point of view and can help in handling variations in network resources availability through a real-time monitoring over these object relationships. [ABSTRACT FROM AUTHOR]

This paper shows the use of protocols dynamically generated for a particular network environment and an application's requirements. We have developed a novel system called PNUT (Protocols configured for Network and User Transmissions), which can be used to dynamically generate protocols. A distributed MPEG player has also been developed using PNUT to demonstrate the need for adaptive protocols that can reconfigure their functionality as the operating environment changes. One of the key features of PNUT is that it allows for the dynamic reconfiguration of the protocol stack, which we believe is necessary when developing protocols for mobile computing environments. The paper also shows how intermediate servers can be used to provide better services to the mobile application. Furthermore, we show how intermediate servers can be used to allow mobile applications using PNUT to interact with servers based around traditional applications [ABSTRACT FROM AUTHOR]

In this paper we propose a new approach to adaptation of content-based processing, media preparation and presentation. We address multimedia consumption terminals having a variable amount of resources. The proposed technique adapts the required service data contained within the multimedia to the abilities of the hybrid network elements and application requirements. In our approach the data representations are transformed and converted efficiently to reduce delivery requirements and to emphasize quality of service ratio. An experimental system is implemented in a distributed environment providing only low processing requirements from the data processing nodes. We perform media profiling based on cross-media translation to achieve an efficient scaling and fit the content with terminal capabilities. Our approach aims to build an intelligent content service infrastructure in a way that the servers, nodes, and terminals are aware of their capabilities along with the capabilities of their surroundings. We demonstrate with an example service with scalable multimedia delivery over hybrid network elements. [ABSTRACT FROM AUTHOR]

This paper aims to present and verify the Extended Package-Segment Model. It provides a more flexible representation of logical structure of multimedia content than the Package-Segment Model (PS-Model) which we have already proposed. The PS-Model structure represents recursively ways to divide content. It is suitable to represent logical structure of content generated by content-based analysis. However, there are also some restrictions in generating the PS-Model structure. To solve these problems, we have reconsidered the generation rules and here propose a new data model called the Extended Package-Segment Model (EPS-Model). The EPS-Model structure can represent scenes and cuts that axe focused. Moreover the EPS-Model inherits the advantages of the PS-Model framework such as indexing, browsing, and retrieval mechanisms. We introduce two experimental systems based on the EPS-Model structure. We verified that this EPS-Model has flexibility and suitability of representation of the structure for multimedia content in any application. [ABSTRACT FROM AUTHOR]

In the context of distributed multimedia applications involving multicast to a large number of users, a single quality of service level may not be appropriate for all participants. It is necessary to distribute part of the QoS management process and allow each user process to make certain QoS decisions based on its local context. In order to allow for different QoS options, we assume that each source provides, for each logical multimedia stream, several different stream variants, representing different choices of user-level QoS parameters. The paper presents the design of a teleteaching system which uses this paradigm for QoS negotiation, and explains how the Delivery Multimedia Integration Framework (DMIF) of MPEG-4 can be adapted as a session protocol for such an application. In fact, it appears that this DMIF protocol, which is now being extended by ISO (DMIF Version 2) to the context of multicasting, provides some general session management functions which are quite useful for many distributed multimedia applications using broadcasting. [ABSTRACT FROM AUTHOR]

The integration of the Internet with the existing PSTN found much interest recently. While in many cases the aim is to use the Internet as a carrier medium for voice phone calls, we will present a proposal to extend classic Internet multimedia services to users on the PSTN. We introduce an architecture and prototype implementation that allow users on the PSTN to actively participate in MBone audio conferences. We present a set of building blocks for applications using voice access via telephones and give an overview of how these can be combined to meet individual requirements. Finally, we discuss application scenarios, usage experiences and potential future enhancements to improve usability and to extend the range of available services. [ABSTRACT FROM AUTHOR]

Conference applications play a key role in workgroup tools, and provide an excellent environment for the development and evaluation of proposals concerning multicasting and reliable communications. A nuclear aspect of conference applications is their ability to use multipoint communication in heterogeneous environments. In the scope this work, a platform that uses several extensions to the CORBA Event Service was developed. This platform is based on ITU T.120 recommendations, and supports reliable multicasting communication for both data and control information. The control information is managed by a service implemented in Java™, according to the CORBA Event Service model. The present paper introduces and discusses the main features of the proposed platform. Also, provides a general description of the conference framework architecture that was built on the platform, with some emphasis on reliable multicasting communication and on system synchronism. In the final part of the paper, the specification and the implementation of the proposed extensions to the CORBA Event Service are presented. [ABSTRACT FROM AUTHOR]

In this paper, we proposed a scheme to support multiple MCSs over a single large cluster in ATM networks, and evaluated its performance by simulation. When an ATM host requests joining into a specific multicast group, the MARS designates a proper MCS among the multiple MCSs for the group member to minimize the average path delay between the sender and the group members. This scheme constructs a 2-phase partial multicast tree based upon the shortest path algorithm. We reduced the average path delay in multicast tree using this scheme with various cluster topologies and MCS distribution scenarios, also distributed load among multiple MCSs [ABSTRACT FROM AUTHOR]