This paper presents an integrated network configuration of wired and wireless access systems for nomadic computing and discusses the virtual LAN on a wireless access system. Furthermore, different types of ad hoc networks are summarized to delineate nomadic computing styles. In terms of user mobility, the integrated network provides a seamless connection environment, so a user can move between wireless and wired networks without dropping data communication sessions. This function is critical for nomadic computing users. By defining the integrated network and employing a virtual LAN, a nomadic computing environment can be realized. This paper reviews the key issues to realize integrated networks. They are mobile management including mobile IP, virtual IP and Logical Office, a high performance MAC, and security control.

The Advanced Intelligent Network (Advanced IN) is now commercialized and the Internet is becoming popular all over the world. If these two networks were connected, the potential would exist for new services. This paper surveys and analyzes the possibility of improving both the Internet and Advanced IN with an Advanced IN-the Internet connection. Service customization, which allows customers to define their own service specifications, is one of the most important service applications for the Advanced IN. However, some issues must be resolved before that service can be offered. This paper proposes a solution in which Internet technologies are applied to the IN. We review the system architecture of Service Logic Program (SLP) definition and execution in NTT's IN for service customization. Version management and cost for delivery are the major issue for service customization with the SLP(C) creation tool. We suggest an Internet version of the SLP(C) creation tool to solve these problems. Results of the prototype shows that connecting the IN and the Internet for service customization will benefit both customers and telecommunication operators.

This paper studies the routing algorithms for multi-destination connections where each destination may require different amount of data streams. This asymmetric feature can arise mostly in a large and/or heterogeneous network environment. There are mainly two reasons for this. One is that terminal equipments may have different capabilities. The other is that users may have various interests in the same set of information. We first define the asymmetric multicast problem and describe an original routing method for this type of multicast. The method is then employed in the presented routing algorithms, which can be run in multi-cluster environment. The multi-cluster architecture is considered to be effective for running routing in the networks, where a variety of operating methods might be applied in different clusters but global network performance is required. Our algorithms are designed based on some classical Steiner tree heuristics. The basic goal of our algorithms is to make routing decisions for the asymmetric multicast connections with minimum-cost purpose. In addition, we also consider delay constraint requirements in the multicast connections and propose correspondent algorithms. We compare the performance between SPT (Shortest Path Tree)-based algorithms and the presented algorithms by simulations. We show that performance difference exists among the different types of the algorithms.

In the future broadband networks, multicast services such as video conferencing and distance learning will become increasingly important. To support these multimedia services, one solution is to form an AMT(ATM Multicast Tree)to connect all the conferencing members. In this paper, based on AMT survivability requirements, we investigate the self-healing of an AMT. Self-healing on AMT is a new challenge of survivability of multimedia services. The pre-assign way is a method we usually considered on protection. If we construct a disjoint backup tree, the low building probability and complicated loading on constructing is the first problem. Second, if only one link or node failed on an AMT, we need to reroute links and reserve bandwidth on whole backup tree. Moreover, since the AMT usually transmits video images, the restoration rate will be decreased because even only one branch of backup tree does not endure the required bandwidth. These enhance us to restore the AMT by dynamic restoration scheme. Two proposed dynamic restoration schemes are developed to provide prioritized restoration from a link or node failure. In the first scheme, we apply a link-based restoration scheme on the AMT. The restoration is based on the failed links of network and does not take whole AMT into account. In the second scheme, without changing the multicast services to the members, we allow reconfiguration of the AMT during the restoration phase. Reconfiguration of the AMT is based on a tree-based restoration concept. By computer simulations, we verify the characteristics of the proposed schemes and the results show that the second scheme outperforms the first.

In this paper, we propose flow aggregation algorithms for multicast video transport. Because of heterogeneities of network/client environments and users' preference on the perceived video quality, various QoS requirements must be simultaneously guaranteed even for the single video source in the multicast connection. It is easy but ineffective to provide many video streams according to each user's request. Our flow aggregation algorithm arranges similar QoS requirements of clients into a single QoS requirement, by which the required number of video streams that the video server prepares can be decreased. Then the total amount of the required bandwidth can be reduced by sharing the same video stream among a number of clients. Our flow aggregation algorithm has two variants, which are suitable to sender-initiated and receiver-initiated multicast connections, respectively. Proposed algorithms are evaluated and compared through simulation. Then we show that the server-initiated flow aggregation (an ideal case in our approach) is most effective, but the receiver-initiated flow aggregation can also offer a reasonably effective mechanism.

In this paper, we show the effectiveness of software shaping through evaluation of our extensions to the internet transport protocols, TCP (Transmission Control Protocol) and UDP (User Datagram Protocol). These extensions are aimed at efficient realization of bulk data transfer and continuous media communication. The extensions are to be used with resource reservation, a possible and promising approach to resolve transport issues that the current TCP/IP networks cannot support. Although it seems straightforward to utilize dedicated bandwidth set up via resource reservation, filling up the reserved pipe is not so trivial. Performance analysis shows that, by applying the traffic shaping extensions, not only is the reserved pipe easily filled up, but the timely data delivery required by continuous media communication is also provided. Our experiments with a real system also show that overheads introduced by the new extensions are small enough to permit their practical use. The extensions are implemented in the UNIX system kernel.

On wide area networks (WANs), UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission. Meanwhile, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol (TCP) traffic. As opposed to UDP's simplicity, TCP adopts a unique flow control mechanism with sliding windows. Hence, the quality of service (QoS) of real-time applications using UDP is affected by TCP traffic and its flow control mechanism whenever TCP and UDP share a bottleneck node. In this paper, the characteristics of UDP packet loss are investigated through simulations of WANs conveying UDP and TCP traffic simultaneously. In particular, the effects of TCP flow control on the packet loss of real-time audio are examined to discover how real-time audio should be transmitted with the minimum packet loss, while it is competing with TCP traffic for the bandwidth. The result obtained was that UDP packet loss occurs more often and successively when the congestion windows of TCP connections are synchronized. Especially in this case, the best performance of real-time audio applications can be obtained when they send-small sized packets without reducing their transmission rates.

This paper studies a set of lip synchronization mechanisms for heterogeneous network environments. The set consists of four schemes, types 0 through 3, which are classified into the single-stream approach and the multi-stream approach. Types 0 and 1 belong to the single-stream approach, which interleaves voice and video to form a single transport stream for transmission. On the other hand, types 2 and 3, both of which are the multi-stream approach, set up separate transport streams for the individual media streams. Types 0 and 2 do not exert synchronization control at the destination, while types 1 and 3 do. We first discuss the features of each type in terms of networks intended for use, required synchronization quality of each medium, physical locations of media sources and implementation complexity. Then, a synchronization algorithm, which is referred to as the virtual-time rendering (VTR) algorithm, is specified for stored media; MPEG video and voice are considered in this paper. We implemented the four types on an ATM LAN and on an interconnected ATM-wireless LAN under the TCP protocol. The mean square error of synchronization, total pause time, throughput and total output time were measured in each of the two networks. We compare the measured performance among the four types to find out which one is the most suitable for a given condition of the underlying communication network and traffic.

This paper proposes a bitstream scaling technique for MPEG video for the purpose of media synchronizations. The proposed scaling technique can reduce the frame rate as well as the bit rate of an MPEG data sequence to fit them to the values specified by a synchronization system. The advantage of the proposed technique over existing scaling methods is that it is considering not only the performance of synchronization but also the picture quality of the resulting sequences. To further improve the quality of sequences scaled by the proposed method, this paper also proposes an MPEG encoding technique which sets some of the parameters suitable for the scaling. An experiment using these techniques in an actual media synchronization system has illustrated the usefulness of the proposed approach.

In this paper, we study the optimal allocation of multimedia files in distributed network systems. In these systems, the files are shared by users connected with different servers geographically separated, and each file must be stored in at least one of servers. Users can access any files stored in any servers connected with high-speed communication networks. Copies of the files accessed frequently are to be stored in several servers that have databases. So, it is one of the most important problems how to assign the files to servers in view of costs and delays. Considering these problems in heterogeneous network environments, we present a new system model that covers wide range of multimedia network applications like VOD, CALS, and so on. In these systems, it is obvious that there is trading-off relationship between costs and delays. Our objective is to find the optimal file allocation such that the total cost is minimized subject to the total delay. We introduce a 0-1 integer programming formulation for the optimization problem, and find the optimal file allocation by solving these formulae.

Video-on-Demand (VOD)servers are becoming feasible. These servers are a building component in a heterogeneous multimedia environment but have voluminous data to store and manage. If only disk-based secondary storage systems are used to store and manage this huge amount of data the system cost would be extensively high. A tape-based tertiary storage system seems to be a reasonable solution to lowering the cost of storage and management of this continuous data. However, the usage of a tertiary storage system to store large continuous data introduces several issues. These are mainly the replacement policy on disks, the decomposition and the placement of continuous data chunks on tapes, and the scheduling of multiple requests for materializing objects from tapes to disks. In this paper we address these issues and we propose solutions based on some heuristics we experimented in a simulator.

A password authentication method PERM has been developed for application to e-mail forwarding. This method is suitable for communications in insecure network environments such as the Internet. In particular, it can be adapted to Internet appliances and Java applets which have limited performance. The PERM method does not require password resettings and enables high-speed authentication processing with a small-sized program. Moreover, it does not use facilities or mechanisms for generating random numbers and writing them into and reading them out of an IC card or similar storage medium on the user's side.

An important issue in accelerating the introduction of ATM networks is to offer more convenient access to the customer and a more efficient ATM system architecture. Regarding the first point, ATM network customers are currently inconvenienced by the need to declare traffic parameters, such as peak and average cell rates to the network provider before using the network. However, it is difficult for a customer to predict traffic parameters. This paper proposes a new ATM system with a dynamic bandwidth estimation and allocation scheme. This eliminates the need for traffic parameter declaration, and realizes more convenient ATM service. The proposed ATM system is a ring network. Bandwidth estimation is carried out by the "Network Server" located on the ring network. The estimation is achieved by observing the parameters closely related to media access control (MAC) protocols of LAN/MAN systems. Based on an estimation of customer traffic, the "Network Server" effectively allocates the bandwidth to each customer. This realizes a more efficient ATM network.

By sacrificing approximately ten percent of the transmission speed, ultra-high speed optical time division multiplexed network can be fully operatable by the use of currently available electrical switches. The network utilizes dispersion managed quasi-solitons and transmits TDM packet which comprises of ATM cells that are introduced from a gateway through bit compression to match to the ultra-high speed traffics. The network can provide flexible bandwidth and bit on demand at burst rate of the maximum LAN speed.

This paper develops a prediction model for evaluating the influence of propagation attenuation due to aircraft's flying across the earth-satellite link. This prediction model is based on the Aperture-field method of Huygens-Fresnel wave theory. Considering arriving and taking off course around airport, attenuation impairment is calculated for different types of aircrafts and flight directions. In order to verify this model's accuracy, numerical results are compared with measurement values. The calculations agree well with the measurements. Ground antenna directivity and anticipated impairment to digital broadcasting system such as Perfect TV are also discussed.

Millimeter-wave propagation characteristics are measured in the outdoor environments. Especially, specific features in the urban area and the open meadowland are compared.

The results of experiments on the effect of the height and diameter of the cup on cup microstrip antennas are presented. The results show that the optimum height of the cup for the narrowest beamwidth and the highest gain is about 1/3 λ, and that the beamwidth decreases and the gain increases as the diameter of the cup increases.