Recently, through high speed computer networks, multifarious information such as text, moving and still images, video, voice and control data is available. There is a natural demand to store such multimedia data in databases to facilitate their reuse in a wide variety of applications. Therefore, important research issues pertain to the investigation of appropriate database systems in multimedia computer network environments. In this paper, we first discuss the required technologies for multimedia information systems. Then we look at many multimedia information services through computer networks, and consider the importance of storing and effectively reusing such available multimedia data. To facilitate developing databases for use in these environments, we discuss the evolution of the notion of database systems. Finally, we demonstrate, as practical examples of such database systems, two prototype systems that we are currently implementing, i.e., the campus wide news on demand system and the ASN. 1 database system.

New multimedia applications have emerged on top of information infrastructures, such as on-demand services, digital libraries and museums, online shopping, information Q & A, concurrent engineering, document management, and desktop program production, which require new databases. That is, next-generation database systems must enable users to efficiently and flexibly develop and execute such advanced multimedia applications. Moreover, in some applications, there is no existent data; in others, there are databases or files to be integrated. This requires both top-down and bottom-up database development. To this end, we focus on development of a database system which enables flexible and efficient acquisition, storage, access and retrieval, and distribution and presentation of large amounts of heterogeneous media data. In particular, we propose a multimedia data model as an integration of structural, temporal, spatial, and control functionality. That is, we take an approach based on an object-oriented database, which is more suitable for description of media structures and operations than a traditional relational database. And we extend the object-oriented approach by providing temporal and spatial operators, and control of distributing computing and QOS (quality of service). In this paper, we describe a multimedia data model and its efficient implementation.

In the course of the development of the "Hyper Hospital," a novel medical care system constructed in the computerized information network using virtual reality as its human interface, we devised a virtual reality world creating system which allows users to con figure the world interactively. The re-configuration of the virtual world was designed to be carried out without interruptions of activity and the world can continue to exist during the reconfiguration process. This facility comprises an important part of our Hyper Hospital system because one of our major goals of this proposal of the Hyper Hospital is to restore maximum freedom for patients in the medical care system. Discussion was given in the present study with respect to the basic requirements of the system to be realized, including discussions on the permission given to the participants of different levels, and means by which to modify the structure of the virtual world. A preliminary implementation was described following this general consideration. The developed prototype was shown to be practically suitable to the test of our virtual environment applied to realistic medical scenes.

This paper discussed a distributed processing architecture for a virtual space teleconferencing system. Virtual space teleconferencing is a promising application field for networked virtual environments. People in different places will be able to meet each other in a virtual teleconferencing room and proceed with various cooperative tasks. When such a system creates a realistic virtual environment, it can be referred to as a "Teleconferencing with realistic sensations" system. Further more, as the conference environment can be shared by a number of users, it is possible to perform various kinds of cooperative work using the system. In this paper, the architecture for networked multi-user virtual space systems are classified, and then a case study is described for building a proposed teleconferencing system. The system reproduces a 3D image of each conference participant in a virtual meeting room. Compared with the former system, the new system can deal with more than three participants at the same time and can connect them through commercial telephone lines. Based on the virtual world database management structure, the system was classified as a central server system. However, a central server architecture limits the number of conference sites. We confirmed that the system can serve up to 14 sites using multi-modal interaction without significant latency in operation from summational experiments. Then, introducing some assumptions to the results, we have proposed processing model of the system. The results of model could describe the experimental results and we could indicate roughy estimated system capacity to realize a requaied system performance.

Synchronization of media streams is recognized as an important requirement not only in media retrieval such as a video on-demand service but also in groupware such as a remote conferencing system. In a remote conferencing system, synchronization is more complicated because Live Media Streams (LMS) such as the live raw voice of some participants must be taken into consideration as well as Retrieved Media Streams (RMS) such as media streams retrieved from video equipment. In this paper, we propose a mechanism to synchronize RMSs and LMSs in a remote conferencing system DMSIC (Distributed Multimedia System with Interactive Control) which has been implemented on UNIX workstations connected by Ethernet. In this mechanism, synchronization among RMSs (we call it R & R synchronization) is kept by maintaining the Current Presentation Positions (CPP) on Media Buffers (MB) close to the Ideal Presentation Position (IPP). Synchronization among RMSs and LMSs (we call it R & L synchronization) is kept by adjusting the IPPs among multiple nodes. We have implemented the synchronization mechanism in DMSIC to confirm the effectiveness of it.

Multimedia communication services are being made available with the advent of broadband optical fiber networks. As many different services will be accommodated in such networks, network survivability has been recognized to be a crucial concern. In this paper, we propose a new restoration algorithm for ATM networks providing multimedia services. Our proposed restoration algorithm adopts the message bundling scheme of the Multi-Destination Flooding (MDF) algorithm which was previously proposed for STM-based networks to handle catastrophic failures such as multiple link and node failures. Virtual Paths (VP) with the same communication speed are bundled and Operation Administration and Maintenance (OAM) cells are used for communication of restoration messages. In addition, the following modifications are made on the original MDF to improve restoration performance. The pre-cancellation scheme is adopted to arbitrate reservation contention to realize high restoration ratio. The dual queue scheme is applied to avoid congestion of restoration messages. Moreover, the connection control scheme for VPI connections is proposed to prevent alternative routes from being misconnected. This paper describes the design concept of our restoration algorithm, processes in each restoration phase, and the performance evaluation by computer simulation.

This paper describes a flexible point-to-multipoint access protocol for the fiber-optic passive double star (PDS) system. To provide various types of services, and permit flexibility in changing transport capacity, a time division multiple access (TDMA) scheme for the PDS system is considered. Dynamic time slot multiplexing based on TDMA is proposed to provide required time slots efficiently according to service changes. The effectiveness of dynamic time slot multiplexing is calculated and compared to fixed time slot multiplexing for telephony services. A TCM/TDMA frame structure and an access protocol enabling dynamic time slot multiplexing are proposed. ONU bandwidth is dynamically assigned by using a set of pointers. The ONU access protocol causes no interruption to operating ONUs on the same PDS system during the configuration or reconfiguration of an ONU. The access time is analyzed to estimate the performance of the access protocol. The probability density of access time is calculated for the number of ONUs connected. The calculation results indicate that a PDS system can accommodate up to around 60 ONUs within the maximum access time specified by ITU-T. The experimental results also agree fairly well with the theoretical values.

A novel wireless multimedia network employing a time division duplex CDMA/TDMA scheme is proposed. The network connects mobile multimedia terminals to an ATM based LAN through a radio central unit, and provides both uplink and downlink with unbalanced data rates in the same frequency band. The uplink (from a mobile terminal to a radio central unit) employs a CDMA scheme to transmit low speed human interface signals (-2.4kbit/s), and the downlink employs a TDMA scheme to transmit high speed video signals (-24Mbit/s). The data rates of both links are independent from that of the LAN. The uplink also employs a RAKE receiver and a forward error correction (FEC) scheme using a BCH code in order to reduce bit errors caused by multipath fading. To mitigate channel degradation caused by the near-far problem and multipath fading, a transmission power control (TPC) method for both links and a channel equalizer (CEQ) for the downlink are proposed. The control signals for the TPC and the CEQ are estimated from the impulse response of the channel which is extracted as the output of the matched filters in the CDMA receiver. Theoretical analyses are performed to evaluate the bit error rate (BER) characteristics of the proposed network. The BER performance is derived for a general multipath fading condition modeled by the Nakagami-m distribution and a typical delay profile. Numerical calculation using recent propagation measurements shows the bit error rates of both uplink and downlink to be less than 10^-6 when both of the TPC and the CEQ are employed if there are some specular components in the received signals. This excellent performance can cut a way to realize a mobile multimedia terminal for customer premises. Furthermore, the configuration of the mobile terminal is quite simple even if the high speed TDMA signals are received over a multipath fading channel.

An optimum design scheme for logical network topologies on a Flexible Multi-QoS Logical ATM Network, named Full-Net, is proposed. Full-Net offers high-quality Virtual-Path (VP) networks and controls end-to-end QoS only at the VP-network's access points. To develop the optimum network topology for multimedia traffic in a single ATM network, a logically con figured Virtual Channel Handler (VCH) interconnection network is associated with each QoS class. Many logical networks can be mapped at the same time on the same network, because mapping is independent of the network's physical implementation. To achieve an optimum design scheme for logical networks, the number of disjoint routes is introduced as the parameter used to optimize logical network topology. The number of disjoint routes is chosen so as to maximize total network efficiency. The optimum number of disjoint routes depends on the required QoS, VC-traffic characteristics, and traffic demand. By choosing the relevant cost characteristics, the network operator can easily maximize network efficiency and provide customers with the QoS they request at minimum cost. The proposed optimum multi-QoS network design scheme on a Full-Net architecture is an efficient solution to implementing multi-QoS control in an ATM network.

This paper studies a video storage problem that occurs in Video-on-Demand (VOD) networks and in other distributed database systems. Videos should be stored in order to respect various constraints, especially available storage and transmission capacities. We show there exists an algorithm to solve this combinatorial problem through a pricing mechanism and that it converges to a solution under some general conditions. Simulation results with up to 43-node networks and up to 300 videos show that the algorithm is fast.

Currently there is considerable world-wide speculation regarding the introduction of optical fiber cable into access networks, because optical fiber has a big potential for providing attractive multimedia services. Since optical fiber cable can provide a variety of grade of services, high-reliability of cable networks would be required compared with the conventional copper cable networks. To develop multimedia telecommunication networks as an infrastructure, it is urgent to clarify the highly reliable optical fiber cable network architecture. Since cable network architecture deeply depends on regional conditions such as demand, area size, duct layer networks (consisting of ducts, manholes, tunnels, feeder points etc.), it is necessary to develop a cable network designing tool with user-friendly interfaces for efficiently evaluating cable network architectures. This paper firstly proposes the heuristic algorithms enhanced by the disjoint-shortest-path and the depth-first-search methods that would be applicable for real access networks. Secondly, the design method of highly reliable optical fiber cable network based on the heuristic algorithms in terms of network cost and unavailability caused by cable breakdown is proposed. It can design the combination of star- and loop-shaped (where two diversified routes exist between a feeder point and central office) cable network. Furthermore, comparison with the conventional design method which simply applies star- or loop-shaped cable network is done in terms of economy and reliability on real access networks in the Tokyo metropolitan area. It is concluded that the proposed method can reduce the network cost further and realize a short unavailability value compared with the conventional method.

This paper presents a layered hierarchical switching-software technology, which is based on an object-oriented design approach, that improves software reusability and productivity. This technology enables a non-stop, service-enhanceable software environment (called NOSES), which satisfies customer demands for quick provisioning of new service features without interrupting service, and which improves software reliability. This technology was developed as part of our overall plan to establish a communications software platform that can be customized for use by various communications systems, such as STM, ATM, and IN. The developed non-stop service enhanceable software techniques are call-recovery restart, system file update, and on-line partial file modification; they were achieved by using dynamic program modification. A system file update inevitably affects calls in service, despite efforts to save in-service calls by copying the call data from the old file to the new one. We have therefore developed a different approach: on-line partial-file modification. Our prototype switching system has proven the effectiveness of this modification method and has shown that it can cover a limited range of service feature additions (which meets customer demands for quick service provisioning), as well as all bug fixes (which can lead to higher software reliability due to not using conventional machine code for software patching), without interrupting service. This paper describes on-line partial-file modification, which can be applied to communications systems that require resident program modification or initialization without program loading; that is, the program exists permanently in main memory. An evaluation of this approach also showed that the productivity of service-layer software increases about two times and that the total increase in systems development productivity is about 25%.

In this paper, we develop parallel scrambling techniques for the distributed sample scrambling (DSS), which are directly applicable to the bit- and multibit-interleaved multiplexing environments. We first consider how to realize PSRGs, parallel samplings and parallel corrections for the multibit-parallel DSS (MPDSS), which are the fundamental problems in realizing the MPDSS scramblers and descramblers. The results are summarized in three sets of theorems, and a corollary is attached to each theorem to specifically handle the case of the parallel DSS (PDSS). The theorems and corollaries are supported by examples that demonstrate the relevant MPDSS scramblers and descramblers.

A new concept for a navigation and communication satellite system has been proposed. The navigation satellite system that forms the basis of the proposed system has been studied by one of the authors and extended to add a mobile communication function to the system. The satellite system consists of 15 satellites in quasi-geostationary orbit (QGEO) that have a geostationary altitude and high inclination and provide global coverage and positioning capability to the observer through only reception of the range measurement signals generated at the satellites, which are in the same configuration as the satellites in Global Positioning System (GPS), Three satellites out of the 15 satellite are designated to install a subsystem for mobile satellite communication in order to satisfy mobile communication convenience as required in a Future Air Navigation System's (FANS) concept of International Civil Aviation Organization (ICAO). The case studies of 15-satellites constellations demonstrate not only an acceptable positioning accuracy over the whole globe, but also an accuracy distribution weighted on the north pole region as an example of a weighted accuracy distribution. The addition of a mobile communication function suggests a unified system of satellite navigation and communication, which might provide convenience for the civil aviation industry, because the two functions currently depend on different systems.

This paper gives several explicit formulas for the waiting times in each stage in an alternating-service, two-stage tandem queue (M/G/1 type queue) with a gate in the first stage and server vacations (or setup time). These formula are obtained by using simple conversion relationships between the performance measures. This study has been motivated by the performance evaluation of call (packet or message) processing in telecommunication switching systems.

The word error probability of linear block codes is computed for diversity systems with maximal ratio combining in mobile communications with three decoding algorithms: error correction (EC), error/erasure correction (EEC), and maximum likelihood (ML) soft decoding algorithm. Ideal interleaving is assumed. EEC gives 0.1-1.5dB gain over EC. The gain of EEC over EC decreases as the number of diversity channels increases. ML soft gives 1.8-5.5dB gain over EC.