This paper summarizes recent reports on the internet's energy consumption and the internet's benefits on climate actions. It discusses energy-efficiency and the need for a common standard for evaluating the climate impact of future communication technologies and suggests a model that can be adapted to different internet applications such as streaming, online reading and downloading. The two main approaches today are based on how much data is transmitted or how much time the data is under way. The paper concludes that there is a need for a standardized method to estimate energy consumption and CO2 emission related to internet services. This standard should include a method for energy-optimizing future networks, where every Wh will be scrutinized.

This paper reports the results of a new test on what types of failure cause falls in the stock prices of telecommunication service providers. This analysis of stock price is complementary to our previous one on market share. A clear result of our new test is that the type of failure causing falls in stock price is different from the type causing decline in market share. Specifically, the previous study identified frequent failures as causes of decline in market share, while the current study indicates large failures affecting many users as causes of falls in stock price. Together, these analyses give important information for reliability designs of telecommunications networks.

In emergency situations, telecommunication networks become congested due to large numbers of call requests. Also, some infrastructure breaks down, so undamaged communication resources must be utilized more efficiently. Therefore, several lines in telephone exchanges are generally reserved for emergency calls whose users communicate crucial information. The number of lines reserved for emergency calls is determined by a threshold, on a trunk reservation control method. To accept both required emergency calls and more general calls, the traffic intensity of arriving emergency calls should be estimated in advance, and a threshold should be configured so that the number of reserved lines becomes lower than the estimation. Moreover, we propose that the holding time for general calls should be positively limited. By guaranteeing the holding time sufficient for communicating essential information, holding time limitation reduces long-period calls so more general calls are accepted. In this paper, we propose a new CAC method to utilize undamaged communication resources more efficiently during emergencies. Our proposed method accepts more general calls by collaboratively relaxing the threshold of trunk reservation and limiting holding time of general calls. This method is targeted at not only the telephone exchange but also various systems on networks, e.g. base stations of the wireless network or SIP servers. With our method, the threshold is configured in consideration of the ratio of traffic intensities estimated in advance. We modeled the telephone exchange as a queueing loss system and calculated call-blocking rates of both emergency and general calls by using computer simulation. The comparison with the conventional holding time limitation method showed that our proposed method accepts the required number of emergency calls by appropriately relaxing the threshold, while suppressing the increase in call-blocking of general calls.

Due to the discontinuity of packet based traffic, the user terminals in next generation mobile telecommunications systems will be equipped with sleep mode operation functions for power saving purpose. The sleep mode parameters should be appropriately configured so that power consumption can be sufficiently decreased while packet queue length and packet delay are restricted within a demanded level. This paper proposes an adaptive sleep mode parameter configuration scheme which is able to jointly optimize the inactivity timer and sleep period in response to the variation of user traffic arrival pattern. The optimization target of this scheme is to minimize mobile terminal power consumption while ensuring that the mean downlink packet queue length do not exceed a certain threshold. Results of computer simulations prove that, the presented approach perfectly manages packet queue length restriction, packet delay control and power saving in a wide range of user packet inter-arrival rates both in single- and dual-service scenarios.

This paper presents an intelligent routing algorithm, called Q-Agents, which bases its actions only on the agent-environment interaction. This algorithm combines properties of three learning strategies (Q-learning, dual reinforcement learning and learning based on ant colony behavior), adding to them two further mechanisms to improve its adaptability. Hence, the proposed algorithm is composed of a set of agents, moving through the network independently and concurrently, searching for the best routes. The agents share knowledge about the quality of the paths traversed through indirect communication. Information about the network and traffic status is updated by using Q-learning and dual reinforcement updating rules. Q-Agents were applied to a model of an AT&T circuit-switched network. Experiments were carried out on the performance of the algorithm under variations of traffic patterns, load level and topology, and with addition of noise in the information used to route calls. Q-Agents suffered a lower number of lost calls than two algorithms based entirely on ant colony behavior.

A timer-based scheme is proposed to manage information within terminal and service profiles for both incall registration/deregistration of UPT users and incall registration resets of terminal owners. In the timer-based scheme, information related to incall registration for a UPT user in a terminal profile is deleted due to a timer expiration without accessing the terminal profile. The performance of the timer-based scheme is compared with the previously proposed request-based scheme in terms of; 1) total cost and, 2) the number of terminal profile accesses per unit time for a terminal. Even though provision of the timer-based scheme requires the modification of incoming call delivery procedure, the timer-based scheme can reduce both the total cost and the number of terminal profile accesses compared to the previously proposed request-based scheme.

The intermediate language (IL) modularizes a compiler into target processor independent and dependent parts, called the front-end and the back-end. By adding a new back-end, it is possible to port existing software from one processor to another. This paper presents a new efficient approach to achieve multiple targeting to quite different architectures using different processors as well, by translating from one IL into other existing ILs. This approach makes it possible to reuse existing back-ends. It has been successfully applied to a commercial-scale project for porting public switching system software. Since the target ILs were not predictable in advance, we provided an abstract syntax tree (AST) with attributes accessible by abstract data type (ADT) interface to convey the source language information from our front-end to back-ends. It was translated into several ILs that were developed independently. These translations made the compiler available in a very short time for different cross-target platforms and on several workstations we needed. The structure of this AST and the mapping to these ILs are presented, and retargeting cost is evaluated.

In universal personal telecommunication (UPT) environments, UPT networks retain information related to incall/outcall registration in UPT user service profiles in order to provide incoming UPT calls for UPT users in any location who have registered at a terminal. As UPT networks support incall registration, terminal users can be different from terminal owners, and several UPT users can register for incoming calls on a single terminal. Therefore, appropriate third-party protection procedures are needed to protect the rights of terminal owners. A terminal profile database can be used to store information regarding terminal states and incall UPT users registered on a terminal in order to enable third-party protection procedures. In order to manage information within both the terminal profile and the service profile, we propose a request-based scheme for incall registration/deregistration of UPT users and incall registration resets of terminal owners. We evaluate the performance of the scheme in terms of; 1) total cost and, 2) the number of terminal profile accesses per unit time for a terminal.

Exception handling is not only useful for increasing program readability, but also provides an effective means to check and locate errors, so it increases productivity in large-scale program development. Some typical and frequent program errors, such as out-of-range indexing, null dereferencing, and narrowing violations, cause exceptions that are otherwise unlikely to be caught. Moreover, the absence of a catcher for exceptions thrown by API procedures also causes uncaught exceptions. This paper discusses how the exception handling mechanism should be supported by the compiler together with the operating system and debugging facilities. This mechanism is implemented in the compiler by inserting inline check code and accompanying propagation code. One drawback to this approach is the runtime overhead imposed by the inline check code, which should therefore be optimized. However, there has been little discussion of appropriate optimization techniques and efficiency in the literature. Therefore, a new solution is proposed that formulates the optimization problem as a common assertion elimination (CAE). Assertions consist of check code and useful branch conditions. The latter are effective to remove redundant check code. The redundancy can be checked and removed precisely with a forward iterative data flow analysis. Even in performance-sensitive applications such as telecommunications software, figures obtained by a CHILL optimizing compiler indicate that CAE optimizes the code well enough to be competitive with check suppressed code.

This paper presents an FPGA architecture for high-speed systems, such as next-generation B-ISDN telecommunications systems. Such a system requires an LSI in which an over-10K-gate circuit can be implemented and that has a clock cycle rate of 80MHz. So far, the FPGA architecture has only been discussed in terms of its circuit structure. In contrast we consider the circuit structure of the FPGA along with the performance of its dedicated CAD system. We evaluate several FPGA logic-element structures with a technology mapping method. From these experiments, a multiplexor-based logic-element is found to be suitable for implementing such a high-speed circuit using the BDD-based technology mapping method. In addition, we examine how to best utilize the characteristics of the selected logic-cell structure in designing the wiring structure. It is found that the multiplexor-based cell can be connected efficiently in a clustered wiring structure.

With the dramatic rise in the number of Internet users, the expansion of international corporate activity and the arrival of multimedia services, highly-competitive information and communication services are entering a period of severe global change. There is widespread recognition that as these changes take place, it is important to establish operation and management (O&M) that will enable "seamless" end-to-end communication services between any points on the earth. With worldwide trends moving toward automation, the study of automation of O&M is being energetically pursued. This paper describes the requirements for service O&M and gives an overview of the technology believed to be essential to achieve them. The progress of developments is also reported.

Telecommunications management activities have mostly been supported by operators; however, machines are gradually playing more important roles in the management arena by utilizing computing technology. Additionally, management systems can now be networked by using standard interface specifications. The study of human and machine integration is thus essential for achieving the sophisticated management objectives of telecommunications. This paper proposes the principles for a telecommunications management integration network (TMIN), which integrates human and machine management networks, and proposes a source text description method for transferring management communication knowledge from human to machine. First, reference models are proposed for the management process and management communication. These models cover network management activities of both humans and machines. Second, the contents of the source text are clarified. Source text presents human management knowledge in a form suitable for machine-machine communication. Third, an efficient source text description method is proposed that reduces redundancy and proliferation. Finally, a means of harmonizing management information definitions with TMIN is suggested to facilitate human-machine cooperation.

Telecommunications management is essentially an aggregation of a wide range of activities, including operations and management (O & M) of telecommunications services and customers as well as the network and network elements. During the period of rapid growth in telecommunications, the highest priority was to meet increasing market demand and to construct a telecommunications network infrastructure. Therefore, research and development in telecommunications management were subordinated to the evolution of telecommunications services and systems. Recently, customers have been demanding higher quality services, as a variety of new, advanced services have been introduced. This has led to the need to integrate telecommunications services and O & M services. This paper first reviews the history of the development of telecommunications O & M in parallel with the progress of telecommunications in Japan and clarifies specific features in each step of this progress. This analysis identifies urgent problems and their solutions. The results suggests that telecommunications O & M and O & M services should be considered as a key to making future services possible and to the value of those services in a multi-media telecommunications services environment. Based on these studies, the future direction of O & M is then shown, focusing on cooperative O & M involving the customer in the multi-media, multidomain telecommunications environment.

A highly reliable, operated efficiently, and scalable switching node is required for multivendor system management networks. This paper presents techniques for applying the Telecommunications Management Network (TMN) concept to a distributed switching node. First, we select the TMN protocol structures so as to minimize the command response time. To ensure efficient operation and flexible design of the operation, administration and maintenance (OA & M) software, we propose a command-message handling function for mapping managed objects (MOs) to the OA & M software. We have designed MO classes for switching-system-specific devices with a redundant configuration that ensures highly reliable system operation. The feasibility of these techniques have been confirmed on a prototype system.

Multimedia services based on broadband ISDN (B-ISDN) technology need a network architecture that satisfies the requirements of users, carriers, and vendors. This paper describes a new network architecture for B-ISDN service control and management based on INA. We list general requirements, and present implementation issues of INA. A network architecture and main components, which resolve implementation issues, is then proposed. We also describe a video-on-demand service based on our proposed architecture.

Development of a large-scale mobile communications network (IMN: Intelligent Mobile communications Network), as an infrastructure integrating multimedia functions, is indispensable for the support of future mobile communication services aiming toward "personalization," "intelligence," and "multimedia services." This paper discusses the aims of mobile communications and the outline of network technology aspects of PDC (Personal Digital Cellular) network which is currently in service. In addition, the future prospect of mobile communication technologies is discussed with special focuses on the support of universal mobility, network architecture including mobile communications platform, and multimedia technologies in the transport and access systems.

Linear lightwave networks (LLNs) are optical networks in which network nodes perform only linear operations on optical signals: power splitting, combining, and non-regenerative amplification. While previous efforts on LLNs assume only one fiber per link, we consider a multi-fiber linear lightwave network (M-LLN) architecture for telecommunications where switching exchanges are normally connected by multi-fiber cables. We propose a class of linear path (LP) allocation schemes for establishing optical paths in M-LLNs, and show that they have a better performance than those proposed for single-fiber LLNs. We show that M-LLNs can be implemented with commercially available components, and discuss the implementation issues in detail.

The performance of various ARQ protocols has recently been analyzed for multidestination environments. In all previous work, the round-trip delays between the transmitter and each of the receivers are assumed (or forced) to be equal to the maximum one, to simplify the analysis and/or the operation. This assumption obviously will sacrifice the system performance. In this paper, we evaluate the throughput efficiencies of three multidestination GBN ARQ protocols under unequal round-trip delays. In the investigated protocols, multiple copies of each data block are (re)transmitted contiguously to the receivers. Tight lower bounds are obtained for the throughput efficiencies of the schemes in which each data block is transmitted with the optimum number of copies. Results show that assuming all the round-trip delays to be equal to the maximum one may sacrifice the performance significantly. We also compare the performances of the three investigated protocols. In general, the performance becomes better as the transmitter utilizes more of the outcomes of previous transmission attempts.

The term telecommunications is derived from "tele", meaning at a distance, and "communications", meaning exchanging of information. The history of electronic communications has thus far been applied to the exchange of spoken, visual, and or textual information between pairs of people, pairs of machines, and people and machines. The role of telecommunications has been to provide a medium for the exchange of the information, with the burden placed on the communicating people or machines to initiate the communication and to interpret or process the information being exchanged. In this paper we attempt to predict some future trends in telecommunications, reaching into the next century. Such predictions are inevitably incomplete, inaccurate, or both. Nevertheless, it is a useful exercise to try to anticipate these trends, and more importanly the issues and problems that will arise in the future, as a way of focusing near-term research efforts and suggesting opportunities. One of our hypotheses about the future is that telecommunications networks will become much more active in initiating, controlling, and participating in the exchange of information. Our approach will be to first review some particularly important past developments, and then to try to predict the future in two ways: First, by extrapolating present trends and activities, and second, by criticizing current trends and anticipating problems looming on the horizon.

Information technology equipment connected to telecommunications line can be a source of electromagnetic interference. Two sources of interference have been under evaluation. One is the digital pulses in the switching regulator and the clock oscillator, and the other is the signal's common mode voltage. In this paper, the interference-inducing mechanism for the signal's common mode voltage and a method for measuring the interference are described. An equivalent circuit representing both the equipment and the line is derived on the basis of the interference-inducing model. A method for estimating the signal's common mode voltage from the differential mode voltage and the line unbalance is obtained using the equivalent circuit. It is confirmed that the level difference between the estimated and the measured common mode level is less than 3dB.