This paper deals with the challenge of modeling the performance of planned ultrabroadband access networks while maintaining technological neutrality and accuracy in measurable quality. We highlight the importance of such modeling also for addressing public funding policies compared to models mainly based on the maximum nominal speed of the access networks, taking also into account the widespread use of measurement tools like "speed test" that have influenced the perceived quality by end-users. We present a performance modelling approach based on the extension of well-known traffic models that accurately characterizes the performance of broadband access networks. We also show how the presented model has been validated with data from two network operators and has been applied to address the recent Italian public interventions for the development of ultrabroadband access networks in market failure areas.

In this paper, we investigate possible time correlation of answered call arrivals in sets of real GSM telephone traffic data. Instead of attempting to model the empirical distribution of the call interarrival time, as done in several previous works in literature, we emphasize results obtained by the modified Allan variance (MAVAR), a widely used time-domain quantity with excellent capability of discriminating power-law noise types. The call arrival rate exhibits a diurnal trend, with peak hours in the morning and late afternoon. Besides this diurnal rate change, the number of call arrivals in a second is found perfectly uncorrelated to the number of arrivals in other seconds and Poisson distributed, with good consistency by X/sup 2/- test evaluation. Uniform and accurate whiteness of call arrivals per second is verified in all hours, regardless the time of the day. In all series analyzed, the empirical statistics of both originated and terminated call arrivals proves excellent consistency with the ideal Poisson model with diurnal variable rate lambda(t). This study may be valuable to researchers concerned about realistic traffic modelling in planning and performance evaluation of cellular networks.

This paper outlines trends reshaping the business of International Long Distance (ILD) voice minutes trading. Our analysis is centered on Arbinet's business model: an electronic market-place for trading ILD voice minutes. Arbinet, who adopted a minute-based termination model, claims to have switched about 12 x 109minutes in 2005 with 13% growth. Since 2003, other competitors, like Stealth's VPF (Voice Peering Fabric), entered the long distance voice market by introducing different flavors of VoIP (Voice over IP) peering. VPF, who adopted a flat pricing scheme, claims to have routed traffic for an equivalent 18 x 109minutes in 2005 with 750% annual growth. VPF installed its first switch in London late 2005, thus stepping into the ILD arena. The huge growth of flat peering exchanges does not necessarily imply that Arbinet has to give up its minute-based termination model in favor of flat or free peering arrangements. As a matter of fact, the way termination model will change in the future is far to be clear. Drivers that determine who will be fit to survive in this thinly profitable business are thus emphasized.

We investigate the statistical properties of both originated and terminated call arrivals in sets of real GSM telephone traffic data (TIM, Italy), emphasizing results obtained by the modified Allan variance (MAVAR), a widely used time-domain quantity with excellent capability of discriminating power-law noise. The call arrival rate exhibits a diurnal trend, with peak hours in the morning and late afternoon. Besides this diurnal change, the number of call arrivals in a second is found perfectly uncorrelated to the number of arrivals in other seconds and Poisson distributed, with good consistency by chi2-test evaluation. Uniform and accurate whiteness of call arrivals per second is verified in all hours, regardless the time of the day. In all series analyzed, the empirical statistics of both originated and terminated call arrivals proved excellent consistency with the ideal Poisson model with variable rate lambda(t)- This study may be valuable to researchers concerned about realistic modelling of traffic in planning and performance evaluation of cellular networks.

This paper gives a snapshot of the convergence of the telecommunications and computer industries to a common vision of networking models for an integrated services information infrastructure meeting the stringent requirements of multimedia applications. First we summarize the requirements and elements of such an infrastructure and introduce a networking model that is framed into four layers: interconnection (bearer), internetworking, interoperability (middleware), and application. Then, we offer a review of the classical telecommunications networking model, the current status of asynchronous transfer mode (ATM) multimedia networking features, and the ongoing activities for fostering evolution of telecommunications networks to enhance control, management, and service delivery. An overview of the current Internet network architecture follows, presenting the recent achievements and the evolving networking protocols to cope with real-time multimedia applications. As described in this paper, the schemes for integrating Internet protocols (IPs) and ATM resource reservation protocols, which address an efficient resource utilization and quality of service guarantee in broadband networks, fall into the first two lower architectural layers, which are the foundation of network computing and communications services. In particular, we present a summary description of the comparative features of both IP and ATM protocol approaches, and we report on current proposals for provisioning IP on ATM networks. Last, we discuss trends in middleware services, including the emerging technology of transportable computation and intelligent agents. Interoperability of heterogeneous networked systems is based on an agreed distributed application encoding and computation environment, thus pushing the key capability of the layered networking architecture up into the middleware layer.

A new class of switching architectures for broadband packet networks, called shuffleout, is described and analyzed in the paper. Shuffleout is basically an output-queued architecture with a multistage interconnection network built out of unbuffered b/spl times/2b switching elements. Its structure is such that the number of cells that can be concurrently switched from the inlets to each output queue equals the number of stages in the interconnection network. The switching element operates the cell self-routing adopting a shortest path algorithm which, in case of conflict for interstage links, is coupled with deflection routing. The paper presents the basic shuffleout architecture, called open-loop shuffleout, in which the cells that cross the whole interconnection network without entering the addressed output queues are lost. The key target of the proposed architecture is coupling the implementation feasibility of a self-routing switch with the desirable traffic performance typical of output queueing.>

A new class of switching architectures for broadband packet networks, called shuffleout, is described and analyzed. Shuffleout is basically an output-queued architecture with a multistage interconnection network built out of unbuffered b/spl times/2b switching elements. Its structure is such that the number of cells that can be concurrently switched from the inlets to each output queue equals the number of stages in the interconnection network. The switching element operates the cell self-routing adopting a shortest path algorithm which, in case of conflict for interstage links, is coupled with deflection routing. The basic version of this architecture is called open-loop shuffleout. This paper describes the closed-loop shuffleout architecture with 2/spl times/4 switching elements in which cells that have crossed the whole interconnection network re-enter the network as long as they are not successfully routed to the addressed switch outlet. This result is accomplished by adding to the basic open-loop structure recirculation paths so that each packet can cross several times the interconnection network. Two different solutions are proposed to implement such functionality, the buffered closed-loop shuffleout and the expanded closed-loop shuffleout architecture. Both these solutions aim at reducing the number of stages in the network, compared to the open-loop structure, so as to reduce the complexity of the switch internal wiring and to simplify the output queue interface.>

The bidirectional Manhattan street network (BMSN) and the ShuffleNet (SN), four-connected topologies implementing the logical layout of so-called multichannel metropolitan area networks (MANs), are considered. An open issue in multichannel MANs, namely, broadcast service, is addressed. Two different strategies are proposed to deliver broadcast traffic in BMSN and SN, and they are compared on the basis of throughput and delivery delay results, as obtained by developing an analytic model. The availability of an analytic model allows the effectiveness of the proposed algorithms in large size networks to be investigated.>

The authors describe a model for the performance evaluation of the Shuffleout asynchronous transfer mode (ATM) switch under arbitrary nonuniform traffic patterns. Shuffleout is a blocking multistage structure using shortest path routing with deflection, in which output queues are connected to all the stages. The analytical model developed computes the load on each interstage link by tracing all the paths that a packet can follow to reach the addressed switch outlet, that is, by taking into account all the deflections from its shortest path it can receive. Such a model allows the computation not only of the average load per stage but also its distribution over the different links belonging to the interstage pattern. Thus fairness in packet switching by Shuffleout is provided by the model.>

The authors evaluate the performance of two different network architectures, implementing multichannel metropolitan area networks (M-MANs). Deflection routing, no internal storage, and a high connectivity degree are the main characteristics of regular networks built by interconnecting small-size multichannel nodes. Analytical models have been developed for network architectures with a connectivity degree equal to four: the bidirectional Manhattan street network (BMSN) and the shuffle net (SN). The authors evaluate the performance under uniform traffic distribution and also under the realistic assumption that the user's attachments to network nodes are equipped with input and output buffers. The performances under investigation are throughput, average delay, and packet loss. The availability of an analytical model makes it possible to investigate the performance of networks with a large number of nodes. The strong and weak points of the two M-MAN architectures are pointed out in the discussion of the results.>

The challenge of the 1990s to the telecommunications industry will be to make ''Personal Communications become a reality. In North America the acronym used is PCS/PCN (Personal Communication Services/Networks), while in Europe and in the CCITT the term used is UPT (Universal Personal Telecommunications). Both terms, though, aim at the same objective. UPT enables access to telecommunication services by allowing personal mobility. It enables each UPT user to participate in a user-defined (personalized) set of subscribed services and to initiate and receive calls on the basis of a personal number across multiple networks, at any terminal, fixed or mobile, irrespective of the geographical location. This talk presents an overview of the major challenges for UPT evolution and relates this to existing and evolving network capabilities. UFT arise from the merging of two enabling network capabilities: wireless access and intelligent network services, Both capabilities require complex network operations to manage, call-by-call, subscriber mobility and service profile, as well as access security and confidentiality of user information. A flexible and robust support capability for service/network OAM&P (operation, administration, maintenance and provisioning) is the key catalyst to make UPT happen during the 1990s.

Congestion control mechanisms are required in asynchronous transfer mode (ATM) networks to fairly and efficiently allocate network resources to the various service types. Bandwidth assignment is a preventive control mechanism. A bandwidth called equivalent bandwidth is allocated in order to ensure the cell transport objectives. A definition is proposed of equivalent bandwidth for stream, bursty data and variable bit rate video sources. The equivalent bandwidth values are evaluated by simulation considering both homogeneous and heterogeneous sources. In the heterogeneous mixes with different cell loss probability requirements the selective discarding mechanism permits a more efficient utilization of the network resources.>

A new class of asynchronous transfer mode (ATM) switching architectures for broadband packet networks, called Shuffleout, is introduced. Its interconnection network is a multistage structure built out of unbuffered 2*4 switching elements. Shuffleout is basically an output-queued architecture in which the number of cells that can be concurrently switched from the inlets to each output queue equals the number of stages in the interconnection network. The switching element operates the cell self-routing adopting a shortest path algorithm which, in case of conflict for interstage links, is coupled with deflection routing. The specific architecture presented is called Closed-Loop Shuffleout, since the cells that cross the whole interconnection network without entering the addressed output queues are lost. The key target of the proposed architecture is coupling the implementation feasibility of a self-routing switch with the desirable traffic performance typical of output queueing.>

The suggested class-related rule (CRR) was tested for mixes of two and three different bursty service classes. Performance was evaluated at the ATM cell level in terms of effective assigned bandwidth with respect to the peak bandwidth allocation. The gains obtained using the CRR are significant for the various bursty classes investigated with burstiness values b>or=5. It was found that the CRR is a conservative approach, based on the extensive simulation of a wide range of cases and of the assumed traffic and multiplexing models. Performance of the CRR was then evaluated at the virtual call level in terms of call blocking probability in a group of ATM trunks with respect to the peak bandwidth allocation. It was found that the CRR gain at the virtual call level is greater than the CRR gain at the ATM cell level, thus increasing the attractiveness of using statistical bandwidth assignment. All the results reported require further careful verifications, in particular to test other types of service classes and cases of multiple streams mixed with bursty and variable-bit-rate traffic sources.>

An overview is given of CCITT activities on speech-processing standards for evolving digital communications networks. Such standards apply to worldwide interconnected digital networks and provide for adequate voice and nonvoice quality as assessed by the leading laboratories and experts in the world telecommunications community. Current status of standards is briefly discussed, and perspectives and given on results to be achieved by the end of CCITT study period (1984-8).>

This paper gives a brief overview of the I-Series of Recommendations on the integrated services digital network (ISDN) developed by the International Telephone and Telegraph Consultative Committee (CCITT). This set of Recommendations is printed in the CCITT "Red Book." The I-Series Recommendations represent the first major step towards worldwide harmonization of the fast-growing digital network capabilities in support of multiservice (voice, data, and image) user applications. Over 25 different Recommendations set up standard guidelines and features for service capability, network architecture, and user-network interfaces in the ISDN.

This paper gives a snapshot of the international activity on standards to be set for speech, sound and picture signals processing for application in the emerging Integrated Services Digital Networks (ISDN). The ISDN is a focal point for the ongoing activity of the CCITT (International Telegraph and Telephone Consultative Committee). ISDNs are conceived as networks which have evolved from the basic digital telephone networks and they will provide end-to-end digital connectivity to support a wide range of services including voice, data, sound and video applications. The CCITT forum has represented, in past years, a driving force for digital equipment and network development. Today, the CCITT standardization role is of increasing importance in view of the rapid advances in digital communication technology, and the expected world-wide development of new communication services. After giving a general outline of the emerging network scenarios for digital communication, the paper concentrates on two aspects: (i) current status of standards for digital speech, sound and video communication; (ii) current activity plans for the evolution of such standards and perspectives for results achievable during the present CCITT study period.

This paper deals with arrangements for local access to emerging Integrated Services Digital Networks (ISDN's). The ISDN concept is here introduced as the focal point for the ongoing international activity on multiservice digital networks. ISDN's are conceived as networks which have evolved from the basic digital telephone networks and they will provide end-to-end digital connectivity to support a wide range of services including voice, data, sound, and video applications. After giving a general outline of the evoving ISDN network scenarios for digital communications, the paper reviews the international trends on user access arrangements in terms of 1) user equipment and interface configurations, 2) access channel structures and information transfer capabilities, 3) local distribution plant configurations (with emphasis on the copper plant), and 4) access protocols.

Alphabetic pseudoternary codes, which are being used largely for high-capacity pulse-code modulation (PCM) transmission, require character alignment of the received ternary sequence for correct conversion from pseudoternary to straight binary. In this paper a method and a strategy of character alignment for MS43 line code are proposed. The method is based on the recognition of the constraints and correlations involved by the conversion rules of MS43; the strategy, which is similar to that already used for frame alignment in PCM time-division-multiplex (PCM-TDM) systems, allows limiting of the effects of simulations and line errors on the processes of loss and recovery of character alignment. Statistical properties of these processes are analyzed both theoretically and by computer simulation. The results are used to design the character alignment strategy for a 34.368 Mbit/s PCM system using MS43 as line code.

The choice of the fundamental parameters of a pulsecode modulation (PCM) system hierarchy depends on many factors, such as transmission media characteristics, time division switching and multiplexing plant requirements, traffic demand and network topology, etc. This paper deals with the problem of clarifying these interdependences and gives the general design criteria for planning a digital hierarchy. In accordance with these criteria, the main characteristics of a digital hierarchy, starting from the 2.048-Mbit/s primary PCM system, are derived and proposed. At present, systems in accordance with this proposal are being developed in Italy.