Surging capacity demand triggered by the increasingly mobile-oriented and exponentially growing Internet has accelerated convergence of networking technologies. In the core network side, IP and photonics have been the two key driving factors of technical innovations. Amid this technical turmoil, Generalized MPLS (GMPLS) in IETF has recently attracted sizable attentions, as it offers potential for "Grand Unification Theory" for network technology convergence. Despite its prospects, however, the proposal is still missing comparable structures in management plane, which is in dire need for carrier-class, reliable operations. Among many industry proposals and standards, TINA vision on connection management architecture (CMA) is the one offering practical and deployable architecture for the converged photonic IP network. TINA IP Control and Management (IPCM) WG was established during TINA phase II (1998-2000), to study IP control and management issues using the architecture basis of TINA-CMA. Latest activities in TINA IPCM WG, compiling experience at Sprint, Telia, Telecom Italia Lab., and Fujitsu, have resulted in a specification for connectivity provider reference points, namely ConS, ConC, and FCon. Use of TINA CMA as building blocks for the IP photonic network convergence is illustrated. An overview of a ConS reference point specification for managed IP connectivity service, named ConS-IPCM, is explained.

This paper describes the JAIN^TM Java^TM Call Control (JCC) Application Programming Interface (API), and its relationship to network protocols, in particular the Session Initiation Protocol (SIP). JCC is a high-level object-oriented open, standard API for Next Generation Network (NGN) softswitches that enables rapid creation, by third parties, of services that can run independently of the underlying network technology (e.g. wireless, wired, packet, IP, PSTN) and protocols. SIP is a protocol that has been proposed for a wide variety of uses in IP networks, including call control. We argue that instead of being competitors, JCC and SIP are complementary, with JCC offering higher-layer programming abstractions and protocol-independence, and demonstrate by examples how to map JCC version 1.0 to a SIP environment. We thus show that for common call control applications using JCC is simpler, faster and less maintenance intensive than using SIP directly.

The number of computer break-ins from the outside of an organization has increased with the rapid growth of the Internet. Since many intruders from the outside of an organization employ stepping stones, it is difficult to trace back where the real origin of the attack is. Some research projects have proposed tracing methods for DoS attacks and detecting method of stepping stones. It is still difficult to locate the origin of an attack that uses stepping stones. We have developed IDA (Intrusion Detection Agent system), which has an intrusion tracing mechanism in a LAN environment. In this paper, we improve the tracing mechanism so that it can trace back stepping stones attack in the Internet. In our method, the information about tracing stepping stone is collected from hosts in a LAN effectively, and the information is made available at the public information server. A pursuer of stepping stone attack can trace back the intrusion based on the information available at the public information server on an intrusion route.

In recent years, the concept of Quality of Service (QoS) has gained attention in the network research community and its extension towards delivering QoS level assurance to sensitive applications has generally been agreed to be crucial for the longevity and usability of the Internet. On the Internet, the Differentiated Services (DiffServ) architecture is a general framework for the differentiated treatment of traffic aggregates in the core network. However, DiffServ does not extend to end-to-end QoS assurance. In this paper, we propose a simple resource-brokering scheme operating over a hierarchically connected internetwork of administratively-independent DiffServ-capable domains that is a simplified model of the Internet. The proposed asynchronous, multi-agent resource-brokering scheme operates locally but avoids conflicts globally. In this way, the traffic control reflects the underlying structure of the internetwork, introduces only a localized complexity thus scaling up well, and permits independent policies between the interconnected domains.

Recently, demand on class-of-service (CoS) has known a great increase thanks to a set of real-time applications such as Internet Telephony service. Class-Based Queuing (CBQ) is considered as an efficient queuing mechanism to guarantee CoS. ALTQ is a widely used platform for realizing CBQ. In this paper, we verify through experiments that bandwidth control of CBQ/ALTQ contains overhead for fluctuating traffic. To avoid such an overhead, we introduce dynamic bandwidth allocation scheme for real-time traffic fluctuating within fixed ranges. In the light of the limited network resources, it quickly becomes obvious that when the traffic rate exceeds the maximum available bandwidth, arriving packets will be accumulated in the router queue. As a result, the traffic delay increases and the quality of real-time applications is degraded. To cope with such a problem, we revise the RED algorithm for a large amount of traffic and propose a new packet discard algorithm that uses bandwidth as a trigger. Experiment results show that our proposal outperforms the already existing packet discard algorithms (RED, DropTail) in providing lower delay/jitter services. We show the efficiency of our proposal using a real system.

Layered multicast approach enables IP multicast to adapt to heterogeneous networks. In layered multicast, each layer of a session is sent to separate multicast groups. These layers will be transmitted on the same route, or on different routes. However, traditional congestion control schemes of layered multicast do not consider the case when layers of a session are transmitted on different routes. In this paper, at first we show that in sparse-mode routing protocols like PIM-SM and CBT, layers of a session can be mapped to different Rendezvous Points or cores due to the bootstrap mechanism. It means that layers of a session can be transmitted on different routes. We then show that traditional congestion control schemes of layered multicast do not work properly in sparse-mode routing regions. At last we introduce Rendezvous Point based Layered Multicast (RPLM), a novel congestion control scheme suitable for sparse-mode routing regions, and show that RPLM works efficiently in regions using sparse mode routing protocols. RPLM uses per-RP packet loss rate instead of the overall one to detect congestion on each route, and can react to congestion quickly by dropping the highest layer on the congested route. In addition, RPLM simultaneously drops all the layers those are useless in quality's improvement to prevent bandwidth waste.

As the network speed becomes faster and requirements about various services are increased, a number of groups are currently developing technologies aimed at evolving and enhancing the capabilities of existing network. A Next-Generation Network (NGN) is defined as a hybrid telecommunications network that employs new distributed processing techniques to provide all types of services. By integrating the Intelligent Network (IN) technology and the Mobile Agent (MA) technology we can support service flexibility and service portability in NGN. In this paper, we propose a caching-based mobile agent model for NGN and analyze the performance of this model. The mobile agent technology increases the service portability and the caching strategy does the service reusability. Each Physical Entity (PE) has MAs within their repository through the caching strategy and processes service requests from users without the control of the central system such as Service Control Point (SCP). Therefore, we can decrease the total network load and the response time for user requests.

It is widely recognized that IP-based mobile network will be a dominant trend. For mobile IP networks, the address starvation problem and scalable mobility management for mobile nodes are important issues. In order to cope with these issues, we propose an approach to realize mobile IP network supporting private addresses for mobile nodes. Our approach introduces regional registration of mobile nodes (Hierarchical Mobile IPv4) and coordinates NAT and DNS functions with the Mobile IP protocol. It enables a mobile node to be assigned a global address temporally in a visited network and to accept a call initiated by a correspondent node connected to the global IP network. This paper describes the detailed design of our approach and the implementation of proposed procedures based on the Mobile IPv4 software developed by the CMU Monarch project.

Internet access via mobile communications networks is growing rapidly; NTT DoCoMo's Internet access service using cellular phones, known as i-mode and started in February 1999, is no exception. The i-mode service enables the user to send e-mail and access Web sites for a variety of information through simple operation of a mobile terminal equipped with a browser. As a result, the traffic to be carried by the PDC (Personal Digital Cellular)-- Packet mobile communication network, which is used to provide the i-mode service, is also increasing rapidly. To meet this growing demand, the switching systems in place are being either increased in capacity or replaced by more powerful ones. To plan this effectively, it is necessary to make an accurate evaluation of the i-mode processing capacity. We have developed a new method of evaluating processing capacity, which is based on the conventional method but takes account of the characteristics specific to the PDC-Packet network. This paper discusses the method of evaluating the processing capacity of switching systems used in the PDC-Packet mobile network.

Reliable and scalable network technologies are desired to meet the emerging demand for multimedia communication. Asynchronous Transfer Mode (ATM) is a key technology and its importance is widely recognized. An ATM layer service category, Available Bit Rate (ABR), was specified at the ATM Forum in 1996. ABR is intended to meet the requirements of non-real-time applications that assume best effort data transportation. It has distinctive features compared to other ATM layer categories. We proposed Application Program Interfaces (APIs) for ABR that allow applications to use ABR capability directly. The API is now a part of the ATM Forum specification. In this paper, we describe the background and necessity of ABR APIs and explain the primitives for them in detail. In addition to having common API primitives for requesting bandwidth or delay requirements during connection setup, ABR APIs can exchange feedback information during communication. Applications for such APIs are addressed and their effectiveness is shown by demonstrating simulation for the TCP-ABR interworking for a backbone ABR network. Finally, a migration scenario for utilizing such APIs is proposed.

DANSE (Dynamically Adaptive Networking Service Environment) is a new architecture for adaptive network service systems. In this paper, a framework for context-aware service construction based on DANSE architecture is presented. In DANSE, any hardware, software, information, and services that are available on a network are treated as network resources. DANSE coordinates the construction of an end user's service based on the user's requests and situation or context (i.e., user's location, schedule, co-workers, etc.). To provide users with satisfactory services, it monitors user context continuously and searches for network resources that are convenient for a target user. Moreover, it detects changes in user context and invokes service construction if needed. If the desired service is not available, alternative services are automatically constructed. With those capabilities, DANSE enables ubiquitous provision of services any time, anywhere.

This paper proposes an Active Networks architecture for VoIP gateway. In the proposed architecture, instead of procedural language, declarative language is used to describe the up-loaded program. This allows for a reduction in size of the up-loaded program, and increases the flexibility for describing the up-loaded program. VoIP gateway can then provide highly flexible services. An experimental system was implemented for feasibility studies. Specification of the declarative language used for describing the up-loaded program, basic functionalities of an interpreter for the language, and execution control program which executes components programs stored the node beforehand were confirmed.

Telepresentations will be popular in the future of ubiquitous digital media. To realize a telepresentation easily over the Internet, we design a communication protocol to control a remote material (digital media) used in a telepresentation. We describe our proposed protocol; RMOP (Remote Material Operation Protocol) in this paper. This protocol specifies commands for material operations such as synchronization of slides, drawing, and pointing. Since this protocol just specifies the common formats through IP networks independent of special functions of presentation tools, it can be applied to various presentation tools. To design the protocol, we consider the trade-off between reliability of IP multicast and practical availability in the actual Internet. We adopt the reliable multicast mechanism to improve reliability but not to lose practicality in the protocol. Also, we describe an implementation of our prototype system using RMOP for a telepresentation. Then we show some evaluations such as the protocol overhead and comparisons with the other existing systems. Last, we show a case study of a telepresentation over the Internet using our system.

Transmission and distribution systems are investigated for application in future fiber optic super-broadband and super-multi-channel subscriber loops. Gradual upgrading is considered so that future systems can keep compatibility with existing systems. First, time frame and strategies for subscriber loop upgrade are overviewed and assumptions for evolution of broadband multimedia distribution systems are discussed. It is suggested that upgrading to super-high definition (SHD) quality multimedia is desirable. Next, some examples of extra-auxiliary picture (EAP) formats are discussed to show the possibility of improving upgradability and compatibility by using extra-channels. Then multiplexing and channel selecting systems are investigated for economical realization of super-multi-channel distribution and flexible channel selection, and hybrid multiplexing (HMUX) and a trans-selector (T-SEL) are proposed. Finally, the efficiencies of HMUX and T-SEL are discussed by using numerical examples. Although broadband down streams are mainly considered, other streams such as IP traffics can be accommodated in the distribution systems investigated in this paper.

In this paper a different view on Viterbi's method for the estimation of the reverse link capacity of a single cell of CDMA based mobile communications systems is given. Viterbi's approach is well-known and of great importance for the capacity estimation. However, the interpretation of Viterbi's result on the system capacity is not that clear. Thus, we introduce a new approach giving accurate reasons for Viterbi's capacity estimation. When neglecting the noise power, both methods provide nearly the same result. We conclude that Viterbi's finding relates to the average capacity, which is an important statistical parameter. However, we should note that this average capacity will be not available all the time. The improvements discussed in this paper focus on the specification of a certain reliability about the availability of the average capacity.

Reverse link performance analyses of single-code (SC) and multi-code (MC) CDMA systems in multipath fading environments are presented. The degree of orthogonality loss among multiple spreading code channels is characterized by introducing the orthogonality factor. This factor depends on the multipath delay power profiles of the propagation channel and the number of paths resolved at a Rake receiver. The link capacity and the signal power of both CDMA systems are then analyzed according to varying system parameters, including spreading bandwidth, traffic load, the orthogonality factor, and the number of spreading codes assigned to a user. Analytical results show that the SC-CDMA system provides a larger link capacity and MC users require more power than SC users. The dominant parameters affecting the performance difference are the spreading bandwidth and multipath delay power profiles.

A digital noncoherent demodulation scheme is presented for reception of Gaussian frequency shift keying (GFSK) signals with small modulation index. The proposed differential demodulator utilizes oversampled signals to estimate the symbol timing and to compensate the frequency offset. The performance of the proposed receiver is evaluated in terms of the bit-error rate (BER). Numerical results show that the proposed demodulator provides performance comparable to that of conventional baseband differential demodulator, while significantly reducing the implementation complexity suitable for single chip integration with direct conversion radio frequency module. Finally the performance of the proposed receiver is improved by adding a simple decision feedback module.

CATV networks are considered as promising transmission channels in that they permit wide bandwidth and high quality data communications. In apartment houses, however, the ingress noise in the up-links due to the tree and branch structure of a network deeply degrades the transmission performance of data communication channels. This is a serious problem especially in apartment houses which are often equipped with old coaxial cables. It permits the noise generated from electrical appliances to disturb up-link data communications. In this paper, we propose a wireless CATV system. In the proposed system, the noise generated in the room of a subscriber does not intrude into a trunk line. We analyze the upstream channels of this system. Based on the results of numerical analyses, we found that the proposed system is suitable and practical for up-link operation in CATV networks for apartment houses.

Network bandwidth has rapidly increased, and high-speed networks have come into wide use, but overheads in legacy network protocols prevent the bandwidth of networks from being fully utilized. Even UDP, which is far lighter than TCP, has been a bottleneck on high-speed networks due to its overhead. This overhead mainly occurs from per-byte overhead such as data copy and checksum. Previous works have tried to minimize the per-byte overhead but are not easily applicable because of their constraints. The goal of this paper is to investigate how to fully utilize the bandwidth of high-speed networks. We focus on eliminating data copy because other major per-byte overhead, such as checksum, can be minimized through hardware. This paper introduces a new concept called Asynchronous UDP and shows that it eliminates data copy completely. We implement Asynchronous UDP on Linux with ATM and present the experiment results. The experiments show that Asynchronous UDP is much faster than the existing highly optimized UDP by 133% over ATM. In addition to the performance improvement, additional advantages of Asynchronous UDP include: (1) It does not have constraints that previous attempts had, such as copy-on-write and page-alignment; (2) It uses much less CPU cycles (up to 1/3) so that the resources are available for more connections and/or other useful computations; (3) It gives more flexibility and parallelism to applications because applications do not have to wait for the completion of network I/O but can decide when to check the completion.

We propose an efficient, low cost, multicast ATM switch which is fair to all inputs. The switch consists of a novel copy network which creates unicast packets in a fair manner, followed by a network that routes packets to their correct Address Translation Tables (ATT's) and ultimately a unicast routing network which ensures sequencing. The copy and routing networks are based on deflection routing. We show that our switch requires O(log N) stages and can be designed for any arbitrarily low level of packet loss. The theoretical results are backed up by simulations. Switching elements in both the copying and routing networks have O(1) bit complexity, making the overall bit level hardware complexity of the network O(N log N). The latency of the switch is proportional to the number of stages O(log N). Unlike other existing copy networks, our copy network drops packets in a fair manner and hence can provide quality of service (QoS) support. The switch is output queued and allows the delivery of multiple packets to the same destination during a time slot.

In this paper, the statistical characteristics of rain attenuation in the equatorial zone are investigated. A more reasonable LMS channel model incorporating weather impairments is proposed and compared to the weather-affected Ka-band land mobile satellite (LMS) channel model suggested by Loo. The proposed LMS model uses Lutz's LMS channel model as its basis. The PDF of the received signal and BER performance derived from Loo's model and the proposed channel model are quantified and compared to verify the effectiveness of the proposed model. Finally, the influence of weather impairments on the BER performance is evaluated under various weather conditions, which clearly shows the superiority of the proposed model.

The international telecommunication standards bodies such as ITU-T, ATM Forum, and IETF recommend the dual leaky bucket for the traffic specifications for VBR service. On the other hand, recent studies have demonstrated multiple time-scale burstiness in compressed video traffic. In order to fill this gap between the current standards and real traffic characteristics, we present a standard-compatible traffic parameter selection method based on the notion of a critical time scale (CTS). The defined algorithm is optimal in the sense that it minimizes the required amount of link capacity for a traffic flow under a maximum delay constraint. Simulation results with compressed video traces demonstrate the efficiency of the defined traffic parameter selection algorithm in resource allocation.

In this letter, we describe the conditions for measuring the nonlinear refractive index n₂ when using the self-phase modulation-based cw dual-frequency method. We clarify the appropriate measurement conditions for dispersion-shifted and conventional single-mode fibers both numerically and experimentally. We also show experimentally that the evaluated n₂ values for conventional single-mode fiber depend on the signal wavelength separation.

This letter describes a new parallel combinatorial delayed multiplexing CDMA system for high-bit-rates mobile communications. It combines delayed multiplexing and parallel combinatory methods with the CDMA system to provide higher bit rates without the use of complex receivers. The results of computer simulations using the double-spike Rayleigh fading channel model in a multiple-user environment show that its down-link BER performance is the same as that of the conventional multicode system.

It is well known that interpath interference (IPI) is a major factor that limits the performance of high data rate transmissions over a variable spreading factor wideband-CDMA (W-CDMA) link since the spreading factor is in general small. An optimum combining scheme suppressing IPI was recently proposed for RAKE reception in [1]. The main contribution of this letter is to present a theoretical model for the outage probability and bit error probability of a RAKE receiver utilizing the optimum combining scheme. Analytical and simulation results are closely matched and show that the optimum scheme provides significant performance improvement compared to the conventional maximum ratio combining (MRC) scheme.

We present iterative round-robin matching for an input and output buffered switch with multiple switching planes. The suggested algorithm is based on iSLIP and consists of request, grant and accept steps. The pointer update scheme of iSLIP is altered in the suggested algorithm to enhance the switch performance. Simulation results under Bernoulli traffic show the suggested algorithm is more appropriate than iSLIP for cell scheduling of input and output buffered switches.