According to the Ethicnet paradigm, which has been recently proposed, communication networks are called to consider Key Value Indicators (KVI)s besides the traditional Key Performance Indicators (KPI)s. The actual realization of the Ethicnet, however, requires several issues to be addressed. One of the most important is the identification of the KVIs to be considered and the way in which they can be measured, controlled, and monitored. This paper contributes to the discussion regarding such topic by answering fundamental questions regarding the identification of the KVIs, the measurement of the extent to which a given network solution contributes to the KVIs, and the definition of requirements in terms KVIs.

The introduction of the Software Defined Networking (SDN) paradigm is radically changing the way in which networks are thought, designed, and managed. The reasons of such huge success are manifold; one of the most relevant is that SDN enables an agile management of the network that can quickly adapt its behavior to the dynamics of the user needs and context conditions. In this sense, SDN represents a fundamental step towards the vision of an autonomous network. Nevertheless, network adaptation still requires the intervention of the network manager that installs and configures the network applications executed by the network operating system. In this context, the application of the Digital Twin (DT) technology can trigger a significant leap forward. In fact, the DT of the network can interact with the other relevant DTs and, applying artificial intelligence techniques, interact with the network operating system and take care of the network application installation and configuration. In this paper we motivate and introduce a platform, implemented exploiting microservices, for the realization of such a vision. The proposed platform is applied in a simple use case to demonstrate the proposed workflow and the resulting impact on network operations.

Industrial wireless sensor networks (IWSN) are expected to support in the near future a wide range of applications in which mobile nodes, for example, robots or humans, interoperate between them and with machinery. However, the real-time constraints imposed by the industrial processing make mobility support quite challenging. This article proposes an approach based on software-defined networking (SDN), herein named forwarding and time-slotted channel hopping scheduling over SDN (FTS-SDN), that handles transmission scheduling and node mobility in IWSN providing bounded end-to-end delays. The article presents the detailed design of the FTS-SDN and the implementation on commercial-off-the-shelf devices. In addition, the FTS-SDN performance are extensively discussed, providing simulation results and experimental measurements obtained in a real scenario.

One of the main features of future factories envisaged by Industry 4.0 is the integration of humans, machinery and sensors, which will be able to exchange information, trigger actions and collaborate while on the move. The management of Industrial Wireless Sensor Networks (IWSNs) including mobile nodes is a challenging target, as mobility support has a significant impact on the network complexity and management.Recently, Software Defined Networking (SDN) has been proposed as a paradigm for controlling IWSN, with the purpose of providing a smart management of mobile nodes in industrial scenarios. However, resource management, that is, slot and frequency assignment, needs ad hoc clustering strategies to avoid the waste of resources that would impair IWSN scalability. Network scalability is a crucial requisite for the successful development of both IWSN and the Industry 4.0 vision as a whole. For this reason, clustering policies are today a cutting-edge research topic.This paper exploits the SDN approach to manage IWSN and proposes a platform for the evaluation of clustering strategies which can be profitably applied to scalable IWSN that support mobility.

Nowadays the role of mobile nodes, either human workers or robots, in industrial production is becoming increasingly important. Consequently, in Industrial Wireless Sensor Networks (IWSN), alongside with reliability and realtime constraints, mobility support is becoming a compelling requirement. However, providing mobility support in IWSN is quite challenging, as the uncertainty and dynamics of industrial environments, combined with the need for supporting real-time constraints of the data and the mobility of the nodes, complicate routing. Software-Defined Networking (SDN) is considered a promising approach for reducing the complexity of network management in many contexts, including industrial automation networks. This paper therefore presents a new SDN approach to handle node mobility and scheduling in IWSN based on the Time Slotted Channel Hopping (TSCH) protocol of the IEEE 802.15.4 standard. The paper describes the proposed approach, that is called the Forwarding and TSCH Scheduling over SDN (FTS-SDN), and provides simulative performance assessments.

Flow-induced molecular communications are a promising communication paradigm based on the exchange of molecule concentrations in a continuous fluid, guided and directed through microfluidic pipes. An emerging approach to implement this type of communications is based on discrete (or droplet-based) microfluidics. It exploits the idea to represent binary information by encapsulating the signaling molecules inside droplets, which are dispersed in an immiscible fluid and delivered, through micro pipes to the destination. The objective of this paper is to study a binary droplet-based microfluidic system for flow-induced molecular communications from an information theoretical perspective. To this end, we first show that the binary discrete microfluidic channel is characterized by memory and anticipation. Accordingly, we provide a mathematical model based on a finite-state Markov chain. Finally, we evaluate the capacity of the binary droplet-based microfluidic channel.

Application of Software Defined Network (SDN) and Network Function Virtualization (NFV) paradigms to telecommunications networks are determining an evolution towards a so-called Software Defined Infrastructure (SDI), where network and service functions are developed and controlled as software applications. Management and orchestration of resources are key elements in this softwarization process. To support it, a theoretical activity of modeling network nodes and networks of nodes is needed in both design and operational phases. To this purpose, this paper defines an analytical model of an SDN/NFV node that can work as a building block for a more complex model of a whole softwarized network.

Network "softwarization" is gaining increasing popularity to achieve dynamicity and flexibility. Cloud computing, together with the new paradigms of SDN and NFV are supporting this evolution. However, the need to move services closer to users to guarantee low latency in the service fruition on one hand, and the trend to support personalization of services on the other hand, are stimulating a migration of the services toward edge nodes. This is the target of the platform proposed in the INPUT project to support Future Internet personal cloud services in a more scalable and sustainable way. The INPUT platform enables next-generation cloud applications to go beyond classical service models replacing physical smart home devices, with virtual entities, providing them to users "as a Service." In this paper, we discuss the design and open-source implementation of a virtual Set-Top-Box that is compliant to the INPUT platform.

In the last years, the two networking paradigms Software Defined Networks (SDN) and Network Functions Virtualization (NFV), are driving the Internet evolution from a pure transport network, to a user-centric platform providing customers with different levels of quality according to their needs. By leveraging on these technologies, the European project INPUT has defined the concept of Personal Networks, an extension of the user's home Local Area Network (LAN), also including virtual images of physical devices. In this context, for many reasons that will be presented in the paper, it could be very useful to export a real hardware or software resource, that is physically or virtually located in the same Personal Network of a given user (e.g. a smart TV or a virtual set-top-box), to another Personal Network. With this in mind, this paper proposes the architecture that allows to instantiate a virtual presence in a Personal Network, in such a way that it allows using it as present locally in the visited personal network. The presented results are compliant with the SDN/NFV platform developed within the EU project INPUT.

In the last few years, network “softwarization” is gaining increasing popularity to achieve dynamicity and flexibility. Cloud computing, as well as the new paradigms of software defined networking (SDN) and network functions virtualization (NFV), are supporting this evolution. However, the need to move services closer to users to guarantee low latency in the service fruition on one hand, and the trend to support personalization of services on the other, are stimulating the migration of services toward edge nodes (in the so-called “fog computing” fashion). This is the target of the INPUT platform, proposed in the INPUT project to support future Internet personal cloud services in a more scalable and sustainable way, and with innovative added-value capabilities. The INPUT platform enables next-generation cloud applications to go beyond classical service models, and even replaces physical smart devices, usually placed in users' homes (e.g., set-top-boxes, etc.), with virtual entities, providing them to users “as a Service.” In this paper, we present the INPUT paradigm and discuss a relevant use case-namely, the virtual set-top-box-adopted to prove the feasibility of the softwarized SDN/NFV paradigm jointly with the fog-computing approach for the support of personal cloud services. The INPUT platform is also compared with a legacy approach to evaluate the gain in terms of quality of experience for both static and mobile users.

This paper introduces μ-NET, a microfluidic LAN that supports the exchange of both digital information and biochemical information carried by droplets moving across molecular processors in a microfluidic chip. The μ-NET can be used to support molecular biology applications like DNA, RNA, and protein biosynthesis. The μ-NET is the first realization of a microfluidic networking paradigm that controls movements of droplets in microfluidic chips by exploiting hydrodynamic phenomena only and builds on recent solutions to achieve communications in the microfluidic domain. The μ-NET integrates techniques to represent addressing information, as well as switching and medium access control solutions. In fact, in μ-NET, the address of the molecular processor where a droplet should be sent to is encoded into the distance between droplets; switching is executed to steer the droplets inside the microfluidic device; medium access control is applied to avoid collisions between droplets that may result in their fusion and, thus, loss of the biochemical information. In this paper, the design of μ-NET is presented in detail, and simulation results validating μ-NET operations are shown.

In the last few years, Software Defined Networks (SDN) and Network Functions Virtualization (NFV) have been introduced in the Internet as a new way to design, deploy and manage networking services. Working together, they are able to consolidate and deliver the networking components using standard IT virtualization technologies not only on high-volume servers, but even in the end user premises. In this context, this paper presents the NetFATE architecture, a platform aimed at putting virtual network functions (VNF) at the edge of the network. This platform is based on free and open source software on Provider Equipment (PE) and Customer Premise Equipment (CPE) nodes, so allowing function deployment simplification and management cost reduction. Finally, the paper proposes a case study, consisting in the implementation of two virtual personal firewalls used by two clients moving between two access points located at the edge of the core network.

The rapid evolution of low-cost standard hardware coupled with the diffusion and the wide spread of new paradigms are stimulating Telco providers to explore the possibility of breaking down the ossification of the current Internet. This paves the way for the introduction of new approaches like Software Defined Network (SDN) and Network Function Virtualization (NFV) into the next generation networks to provide their users with software defined networking services. This context paves the way to software defined services, dynamically allocating the network functions over the network nodes. In this paper we propose an analytical approach to support designers in the allocation of virtualized function over the network, taking into account both the functions requested by the traffic flows and the quality of service (QoS) that the network provides to them. The proposed analytical approach is able to account the routing, the processing capacities of the network nodes and the statistical characterization of the flows requesting each virtualization function. This paper follows the direction of analytically modeling a network with NFV capabilities for performance evaluation and design support.

In the last decade, one of the main goals in wireless telecommunications has been to reduce energy consumption of mobile devices. However, making a network device green can cause performance deterioration. The target of this paper is to propose a cross-layer approach for the design of a mobile video cloud for the uplink transmission towards the Internet. The proposed approach is adaptive in both the video sources and the wireless transmitter. A source Rate Controller is applied to compensate transmission bandwidth reduction due to the energy saving policies. Energy saving in wireless transmission on the mobile cloud cellular channel is achieved by introducing an energy-efficient ARQ protocol. This protocol can apply different transmission laws, in order to exploit the correlation of the cellular channel behavior. An analytical model of the system is defined to compare the transmission laws, and provide some design guidelines to choose one of them and design its parameters.

In the last few years, the ever increasing attention to the eco-sustainability of Internet has pointed out the need of providing advanced power management schemes in network devices, in order to be able to reduce energy consumption according to the traffic load. In this context, starting from the observation that most of the traffic of today's Internet is carried by TCP, we focused on the impact that the dynamics of the TCP congestion control may have on energy efficiency of end hosts. In this paper we demonstrate that great energy consumption is related to network congestion conditions. Moreover, we propose to use a technique, called Active Window Management (AWM), with the aim of driving TCP sending rate according to the network available bandwidth, so improving congestion control and providing high TCP performance. AWM requires no changes to any TCP version. We demonstrate that the TCP traffic shaping performed by AWM allows a great reduction of the energy consumption in end hosts, with respect to the case where only TCP is used as congestion control.

The feasibility of using microfluidic technologies for communication and networking among Labs-On-a-Chip (LoCs), in order to increase reusability, their effectiveness and flexibility, has been already demonstrated in recent works. LoCs are devices, in the size of few centimeters, characterized by μ-scale channels in which small volumes of fluids, either in the form of a continuous stream or of sequences of discrete droplets, move in order to deliver chemical or biological samples to a given destination entity in the microfluidic circuit. They are usually realized through monolithic devices. Samples are processed by passing them through a preset sequence of fixed microfluidic channels. We propose and assess a pure-hydrodynamic switching element in which the path followed by a sample can be controlled through a series of junctions simply modulating the distance between droplets flowing in the microfluidic channel.

Microfluidics deals with manipulation and control of fluids which flow in constrained sub-millimetric media. In this paper communication concepts and networking approaches, typical of telecommunications, are extended to the microfluidic domain. The work illustrated in this paper investigates on possible approaches to information encoding and evaluation of the corresponding channel capacity as well as design of switching solutions. Based on the results of this study, the Hydrodynamic Controlled Microfluidic Network (HCN) paradigm is proposed, which is based on a pure hydrodynamic microfluidic switching function. The HCN paradigm can be applied to interconnect Labs-on-a-Chip (LoCs) in a microfluidic network in which chemical/biological samples are routed between the LoCs by exploiting hydrodynamic effects only. The resulting LoCs are expected to be highly flexible and inexpensive, and thus to become extremely useful in chemical/biological analysis and synthesis.

One of the most weighty matter in the Internet today is the waste of energy due to the fact that consumption of network nodes is not tuned with the input traffic. For this reason, the implementation of rate adaptation facilities in the routers constitutes a challenging problem to make the network energy efficient. Rate adaptation in the green routers is usually achieved by scaling the processing power according to the data rate the router has to manage; at this purpose the clock frequency driving the router processes can be modified according to the input data rate. In this context this paper, starting from a measurement study of the Reference Router implemented on the NetFPGA platform, defines a model of the consumed power as a function of both the managed input traffic and the available clock frequencies. As demonstrated in the paper, the model can be applied by router designers to choose the main router platform parameters, i.e. the number of clock frequencies and the clock frequency switching time, while respecting a given tradeoff between the percentage of energy saved and the maximum tolerated loss probability due to frequency switch.

Microfluidics refers to manipulation and control of little amount of fluids flowing into channels of micro-scale size and is the basic technology utilized by Labs-on-a-Chip (LoC) systems. In the recent past researchers have analyzed the possibility to support information exchange between different elements of a LoC. However, support of communication in microfluidic systems relies on the use of appropriate information coding strategies and deep knowledge of channel propagation characteristics. Accordingly, in this paper we compare two different schemes for information coding and evaluate the corresponding capacities depending on the characteristics of the fluids utilized. Furthermore, we discuss how such encoding schemes can be integrated in networking solutions for LoCs.

The reference router implementation on the Net FPGA platform has been changed in order to hijack the incoming packets according to rules specified by the user through Net FPGA registers. This means that we are able to change any field of any incoming packets, of course, depending on whether we are changing TCP or IP header fields, we need to recompute the TCP or IP checksum and store them back into the packets. It might be useful to change information of the IP or TCP header or the data itself. For example, we could crypto the data, change the URL address in order to point to the desired web site or for URL filtering, change the priority information of a bunch of data, and much more. Our implementation works at user data path level and modifies packet fields if certain conditions defined by the user through Net FPGA registers are satisfied. The project has been implemented as a fully open-source project and serves as an exemplar project on how to build and distribute Net FPGA applications. All the code (Verilog, system software, graphical user interface, verification scripts, make files, and support tools) can be freely downloaded from the Net FPGA.org website.

A traffic monitor system has been implemented on the NetFPGA. The NetFPGA is an open networking platform accelerator that enables researchers and instructors to build working prototypes of high-speed, hardware-accelerated networking systems. The traffic monitor application allows network packets to be captured and analyzed from up to all four of the Gigabit Ethernet ports. A graphical user interface showing the traffic of any port has been implemented on top of it. The project has been implemented as a fully open-source project and serves as an exemplar project on how to build and distribute NetFPGA applications. All the code (Verilog, hardware, system software, verification scripts, makefiles, and support tools) can be freely downloaded from the NetFPGA.org website. System performance has been compared with other two implementations: one using the same NetFPGA architecture but implementing the reference router with port mirroring, and the other being a software implementation built on top of the Click Modular Router.

Improving video streaming performance is a main target today for the development of services and related business opportunities in the Internet. For this reason the definition of an efficient video transport paradigm is yet a challenging task. The most common protocol for data transmission in the Internet is TCP but it has been widely believed not suitable for multimedia streaming because of bandwidth instability and retransmissions. However, the dominance of TCP makes desirable the possibility to use it without any modifications. The main idea of this paper is to improve the performance of video streaming by using TCP, by combining it with Active Window Management (AWM), an algorithm to be implemented by video content Providers in their Internet access router. Through an extensive analysis the paper demonstrates that thanks to AWM, packet losses are avoided while per-packet delivery delay is maintained almost constant, strongly decreasing transmission jitter; consequently the required video playout buffer can be substantially reduced with respect to the classical approaches.

Addressing performance degradations in end-to-end congestion control has been one of the most active research areas in the last decade. A huge number of techniques have been proposed in the past, but some of them miss the challenging target of both minimizing network delay and keeping goodput close to the network capacity (e.g., active queue management (AQM) techniques); others require changes in both network routers and hosts that make them difficult to deploy [e.g., eXplicit Control Protocol (XCP)]. In a previous paper, a novel mechanism, called Active Window Management (AWM), has been proposed by the same authors with the aim of controlling the queue length in network routers, maintaining it almost constant to achieve no loss, while maximizing network utilization. The target of this paper is twofold: 1) carrying out a deep analysis of the AWM algorithm stability to evaluate the impact of some system parameters on the performance of the whole system; 2) demonstrating that AWM can be gradually introduced, maintaining the compatibility with the current Internet. An extensive numerical analysis is proposed to demonstrate that AWM stabilizes the queue around a target value with small oscillations.

Performance analysis frameworks for P2P networks are in great demand by Internet service providers (ISP) and telecommunications companies in order to re-design their systems so as to support the new paradigm. The target of this paper is to propose an analytical model of a real-time video transmission system using P2P to achieve multipoint communication on the current Internet. The system considered is defined in such a way as to maximize the number of users, while guaranteeing a given quality of service level in terms of the maximum number of hops connecting them with the video source.

Peer-to-peer (P2P) networks are emerging as an interesting alternative to media streaming delivery with respect to content delivery networks (CDN). Although several P2P protocols have been proposed to support video streaming with an adequate degree of quality of service (QoS), a critical point today is how to evaluate them and optimize their parameters. The difficulty lies in the fact that the behavior of the P2P overlay network and the one of the underlying packet network are closely related and need to be analyzed simultaneously. Ad-hoc simulators for each P2P architecture have been proposed but they neglect the interactions between peers and the underlying network. This paper proposes a new approach to P2P video streaming performance analysis and system design which provides a very accurate cross-layer simulation of the real behavior of P2P clients over TCP/IP networks. A fluid-flow approach has been adopted to simulate the behavior of the network elements supporting an emulated overlay network consisting of real implementations of P2P clients. The P2P SplitStream video streaming protocol is considered as a case study to demonstrate that the tool is able to capture performance parameters at both the overlay (e.g. inter description synchronization) and packet network levels (e.g. traffic source round-trip time and router queue behavior).

Many techniques have been proposed in the last few years to address performance degradations in end-to-end congestion control. Although these techniques require parameter tuning to operate in different congestion scenarios, they miss the challenging target of both minimizing network delay and keeping goodput close to the network capacity. In this paper we propose a new mechanism, called Active Window Management (AWM), which addresses these targets by stabilizing the queue length in the network gateways. AWM acts on the Advertised Window parameter in the TCP segment carrying the acknowledge, but it does not affect the TCP protocol. The proposed technique is implemented in the network access gateways, that is, in the gateways through which both the incoming and outgoing packets related to a given TCP connection are forced to go, whatever the routing strategy used in the network. We show that when the access gateways implementing AWM are the bottleneck in the networks, TCP performance is very close to that of a pseudo constant bit rate protocol providing no loss, while network utilization is close to one.

In this paper we propose a Markovian model of BitTorrent. Unlike already developed works which capture demographic dynamics, it focuses on the behavior of individual peers. To this end, we center our attention on a generic peer, called tagged peer (TP); for each possible logical state of a BT peer-to-peer connection maintained by the TP, we consider a stochastic process which counts the number of such links, and characterize them according to their state. Validation is carried out and steady-state analysis is performed in order to illustrate how performance evaluation can be extracted from our model

In the current Internet scenario increasing interest is being shown in quality of service (QoS) issues, so simulation and parameter tuning are becoming very hot topics. In this paper we present a user-friendly, easily expandable and time-saving tool, which is able to perform network simulation through fluid-flow models and parameter tuning with parallel genetic algorithms. All the network elements involved (TCP sources, links, routers and so on) are modeled with the fluid-flow approach as independent systems: their outputs only depend on their inputs and internal state variables. Network elements can be connected to each other by a powerful graphical interface. Subsequently the user can select network parameters to be optimized in order to achieve prearranged QoS guarantees. During this phase the tool adopts a parallel implementation of a genetic algorithm (GA) which reduces the execution time required for optimization. A case study is also presented.

With the advent of multimedia traffic over the Internet several TCP-friendly protocols have been developed to improve UDP with congestion control mechanisms. The most promising TCP-friendly protocol is the TFRC. This paper addresses a novel analytical model for TFRC traffic sources. The model is able to capture TFRC behavior and its dependence on the network congestion state. Moreover, the model includes the TFRC loss probability estimation algorithm performed by the receiver. In order to keep the model manageable, whatever the number of sources and network routers, a fluid-flow approach is used.

BitTorrent is today one of the largest P2P systems which allows file sharing for Internet users. Very little effort has been dedicated to this target up to now. The goal of this paper is to develop an analytical model of a free-rider in a BitTorrent network. Unlike previous analytical models which capture the behavior of the network as a whole, the proposed model is able to analyze the performance from the user perspective. The model is applied to a case study to evaluate performance in a real case, and to obtain some insights into the influence of BitTorrent parameters on system performance.

In this paper, a hierarchical paradigm modeling both the intramedia and intermedia time relationships of multimedia sources is derived. Each multimedia source is considered as being made up of a master monomedia source, which is independent, and slave monomedia sources, which are dependent on the master. The paradigm takes different timescale statistics into account and is used to characterize intermedia relationships in terms of the second-order statistics of the multimedia source. Finally, the impact of intermedia relationships on the performance of each monomedia source making up a multimedia source is analyzed when a number of multimedia sources are multiplexed together. The error introduced in both loss and jitter performance analysis when intermedia relationships are not modeled is evaluated through a case study. Numerical results show that this error may be dramatically significant in many common scenarios.

This paper demonstrates that higher network resource efficiency can be achieved by using resource management protocols which consider service disciplines based on service curves together with statistical traffic modeling. To this end, an appropriate analytical framework is introduced which allows calculation of the performance statistically guaranteed to any flow out of an aggregate. This feature enables the analytical framework to be applied to the elements of the core network where aggregates of traffic are considered instead of single flows in order to avoid scalability problems. Given that flows are modeled in the analytical framework through switched batch Bernoulli processes (SBBPs), the whole queueing system is denoted as SBBP/Sc/1/K. The performance is calculated in terms of loss probability and delay distribution. The proposed framework is applied in a significant multinode case study.

Promoting the evolution of the Internet from a simple data network to a true multiservice network constitutes a challenging task. To this end, the Internet Engineering Task Force (IETF) has set up the Integrated Services (IntServ) and Differentiated Services (DiffServ) Working Groups, with the goal of defining a next-generation Internet, in which traditional best-effort datagram delivery and additional enhanced quality of service delivery classes coexist. The IntServ framework, in particular, is designed to be used in the access network, and requires a traffic source with the capability of matching the traffic characteristics declared to the network, TSpec. We propose to use the rate control facility, usually implemented in current MPEG encoders to provide a constant bit rate, to shape the output traffic according to the declared TSpec while maintaining an acceptable perceived image quality. In order to assess this scenario, we introduce an SBBP/D/N/K queueing system, where the SBBP (switched batch Bernoulli process) emission is varied according to the quantizer scale parameter chosen by the addressed rate control mechanism. The analytical framework allows us: 1) to evaluate system performance in terms of both the marking probability of nonconforming output traffic and the quantization distortion introduced by the encoder; 2) to choose the TSpec parameters to be declared such that given performance parameters are respected.

The MPEG standard is the most popular technique in video and audio encoding. However, when MPEG encoding is used to transmit video information, one intrinsic drawback is its high burstiness. A large number of proposed solutions to this problem implement encoding control schemes based mainly on on-line adjusting of the encoder quantizer level through the quantization parameter. However, the choice of the feedback law is still a challenge to make MPEG encoding suitable for transmission over the forthcoming guaranteed service scenarios, where a negotiated bandwidth profile is available for video transmission. Up to now, this problem has only been approached by using simulation; no analytical framework exists yet which allows designers to study the most effective feedback law for transmitting video in the network environment to be used. This paper introduces an analytical paradigm modeling a rate-controlled MPEG encoding system in which a feedback makes the quantizer scale parameter dependent on both the state of the output counter/buffer in the controller and the activity level of the frame to be encoded. To this end, an SBBP/D/N/K queuing system is used, where the SBBP (switched batch Bernoulli process) arrival process is varied according to the quantizer scale parameter chosen by the addressed feedback law at the frame level. The analytical framework allows us to evaluate system performance in terms of both the distortion introduced by the encoding system and the resulting output process statistics; it is general enough to be applied whatever the feedback law is, provided that the parameter to be varied is the quantizer scale parameter. A case study is introduced to assess the proposed framework.

Design of network devices requires accurate models for network traffic. The main difficulty lies in determination of the traffic statistics which effectively influence the network performance, and in construction of a tractable and easy analytical model capturing these statistics. This paper demonstrates that, in order to evaluate performance of a multiplexer in a discrete-time environment, it is necessary to define a model which is able to capture the probability density function, autocovariance function, and the overload probability of the input traffic, simultaneously. The paper also provide an algorithm for the construction of an SBBP model matching these statistics.

The "new Internet" will allow individual application sessions to request different QoS parameters to the network. Two QoS classes have been specified by the IETF: guaranteed services and controlled-load services. For both of them, it is necessary to provide users with the capability of calculating the traffic characteristics, Tspec, to be declared to the network, on the basis of a limited set of parameters statistically characterizing the data source and the required level of QoS. The target of this paper is to realize an analytical framework modeling the source and the access network nodes, with the purpose of calculating the Tspec parameters for MPEG video traffic. To this end the video source is modeled with a switched batch Bernoulli process (SBBP), and performance in the source smoother is analytically evaluated. Then the token bucket at the network access point, loaded by the output traffic of the smoother, is modeled to calculate the probability of marking nonconforming data packets.

The enormous increase in wireless communications has led to the need to configure wireless access networks with cellular structures based on the use of micro- and picocells. A key issue in connection with this is the rerouting of the data flow directed to a user making a handover. The paper considers one of the most widely-used connectionless protocols, the signaling network layer (SNL), which is based on a tree-structured routing architecture. Through an analytical model based on a queueing network, its performance is analyzed in terms of the distribution of the rerouting delay. The proposed model is then applied to a case study to demonstrate its versatility and manageability and to get important insights to the dimensioning of the routing architecture.

Multimode coders are able to exploit the different characteristics of the speech waveform and to take into account the different peculiarities of background noise, thus allowing improvements in both signal reconstruction and network-offered load. In this context the variable rate code excited linear prediction (VR-CELP) coding, that is, a multimode variable bit rate (VBR) coding based on the CELP technique, has been introduced in the literature and is currently being considered for use in various applications, especially in the third-generation UMTS cellular systems. The target of the paper is to introduce an efficient and accurate framework allowing a network designer to analyze the impact of multimode VBR speech coding on the quality of service (QoS) provided by a wireless/wired ATM network. In order to capture the coder output characteristics, we propose to model a VR-CELP voice source by using a switched batch Bernoulli process (SBBP). More specifically, three models are introduced and compared in terms of accuracy and simplicity in determining network performance. As a result of the comparison, a four-state model has been chosen as the best tradeoff. The model is then used to analytically derive the loss probability and the jitter probability density function of an ATM multiplexer loaded by a number of VR-CELP sources. Finally, the proposed paradigm has been assessed in a case study where we demonstrate that, for a given output ATM link capacity and for a number of telecommunication services involving voice transmission, VR-CELP coding performs better than traditional on-off coding.

One of the most important quality-of-service parameters in a multimedia environment is skew, defined as the difference between the delays suffered by the monomedia flows belonging to the same multimedia stream. An analytical paradigm is proposed to evaluate the skew affecting a multimedia traffic stream in an asynchronous transfer mode multiplexer. For this purpose, the emission process of each multimedia source loading the multiplexer is defined as the superposition of heterogeneous correlated emission processes, each of which models one monomedia source as a switched batch Bernoulli process (SBBP). In order to model the intermedia relationships, the transition probabilities in the Markov chain underlying each SBBP are functions of the state of the other monomedia sources. The model is applied to a case study and the dependence of skew performance on some of the source characteristics, such as intermedia correlation, and some of the environment characteristics, such as buffer size, output capacity, and buffer utilization is analyzed and discussed.

Thanks to the ever growing diffusion of distributed multimedia applications, VBR compressed video traffic is expected to constitute the majority of the B-ISDN load in the immediate future. At the moment, the MPEG coding scheme is widely used for many types of applications. Therefore, it is crucial to obtain a model for the generation of sequences with the same statistical characteristics of real MPEG traces. In this paper, after demonstrating that intra-GoP (group of pictures) correlation has to be taken into account, an accurate model allowing the generation of an MPEG-like trace is proposed. Moreover, as the MPEG traffic has been demonstrated to be self-similar, the proposed algorithm allows one to generate a trace, fitting the short-range dependence (SRD) and long-range dependence (LRD) behavior and the probability density function of any given MPEG sequence.

Flow theory is a rich, effective theory introduced in order to study real-time network protocols. It is based on discrete mathematics in which a flow of data is represented by an infinite sequence of numbers and is characterized by its upper bounds. Unfortunately up to now flow theory has used a traffic characterization that in some cases can be too loose, thus causing an inefficient network utilization. In this paper we enhance flow theory by providing it with a more accurate traffic characterization and with all the tools necessary to deal with it. As we will see, this enhancement provide a more accurate performance evaluation and thus better network utilization.

An accurate and treatable Markov-based model for MPEG video traffic is proposed. Before reaching a definition of the model, a large number of MPEG video sequences are analyzed and their statistical characteristics are highlighted. Besides the well-known autocorrelation function, testifying the pseudo-periodicity of the sequence in the short term, and the gamma-shaped probability density function the correlation between different frames belonging to the same group of pictures (GOP) is also accurately studied. Then a structured model is proposed. Thanks to its two-level structure it is able to capture both the inter-GOP and the intra-GOP correlation. In order to obtain the first level of the model, the well-known inverse eigenvalue problem is solved in the discrete-time domain. Finally, the model is used to evaluate the loss probability in an ATM multiplexer loaded by an MPEG video source and an aggregate of external traffic. The accuracy of the model is evaluated by comparing the analytically obtained loss probability with the loss probability calculated by simulating the system using real traffic sequences as driven traces.

Various tracking strategies for the location of a mobile user in a cellular system have been introduced in literature. Up to now they have been evaluated using simple mobility models that do not take into account the various directions in which users often move. We present a comprehensive study of the distance-based, movement-based and multilayer strategies, from which it emerges that the effectiveness of these techniques depends not only on the call-to-mobility rate, a parameter typically referred to in literature, but also on the directional characteristics of the user mobility model.

We propose an analytical model for the study of two different routing strategies in an ATM network supporting multimedia traffic flow: the multimedia virtual circuit (MVC) and the independent virtual circuit (IVC). The first strategy consists of assigning a single virtual channel to all of the monomedia streams which make up the multimedia flows; the second strategy consists of multiplexing homogeneous monomedia streams belonging to different multimedia sources on the same virtual channel. With this aim, a multimedia source is modeled as an arrival process defined as the superposition of heterogeneous correlated arrival processes, each of which models one monomedia source. In order to take into account the intermedia relationships which exist in a multimedia stream, each monomedia source is modeled as an interrupted Bernoulli process in which the transition and the arrival probabilities are functions of the states of the other monomedia sources. A finite-buffer discrete-time approach is used in order to compare MVC and IVC performance when an aggregate of N heterogeneous multimedia sources loads the network. Performance is evaluated for each monomedia source in terms of loss probability and jitter probability density function. To assess the proposed paradigm, a case study is shown.

The skew affecting a multimedia traffic stream when N multimedia streams are mixed in an ATM multiplexer is evaluated. To this purpose a multimedia source is modeled by an arrival process defined as the superposition of heterogeneous correlated arrival processes, each of which models one monomedia source by means of an Markov modulated Bernoulli process (MMBP) process whose transition probabilities are functions of the state of the other monomedia sources.

The anticipated emergence of third-generation mobile systems, referred to as universal mobile telecommunication systems (UMTS), raises the problem of reconsidering the design of the databases destined to contain the user information. In particular, it is expected that the key concepts of the new database architectures will be high distribution and fast updating of information. So far, the problem of determining the most appropriate distributed database (DDB) architectures for third-generation mobile systems has not been widely dealt with in literature. This paper presents a methodology for evaluating hierarchical DDB architectures by means of an analytical model of the data querying operation. This methodology allows for structural alternatives, differing on account of the number of levels and branches in the hierarchy, to be evaluated in terms of query loads and mean response times, according to a given user mobility characterization and a given search protocol operation. By way of illustration, the paper discusses a case study, concerning a query operation arising from a location updating procedure and applied to a hierarchical tree-like DDB in which some structural alternatives are considered.>

The fair queue dual bus (FQDB) medium-access control (MAC) protocol for dual bus networks, which is intrinsically fair in bandwidth sharing and retains all the positive features of distributed-queue dual-bus (DQDB) systems including simplicity, robustness, flexibility, 100% bandwidth utilization, and zero access delay at very low load, is presented. Features which make this protocol appealing with respect to others DQDB alternatives are represented by the fact that FQDB is compatible with the DQDB standard with respect to the physical arrangement and attachment to the bus, the XOR write tap, the access control field (ACF) length, and the three priority levels. Nevertheless, its properties are attractive also for protocols operating at 1 Gb/s and up. Fairness is formally proved, and it is shown that a bandwidth sharing pattern different from the uniform one can easily be enforced, as may be the case if bridges or other large users are presented. Queueing disciplines, priorities, and implementation issues are discussed, and comparisons obtained by simulation are given.>

The authors introduce a control strategy for bandwidth management in asynchronous transfer mode (ATM) networks. A two-level hierarchy is defined, where one level performs fixed class-selective call admission control strategies that are periodically dynamically coordinated by a higher-level bandwidth allocation controller. Each admission controller decides to accept or refuse an incoming call on the basis of a class-selective rule designed to maintain a certain quality of service. A decision is taken according to the virtual capacity share that is assigned to the various service classes by the allocation controller. Bandwidth shares are periodically recomputed online by means of the constrained minimization of a cost function that takes cell loss probability and refused traffic into account. The control structure, the strategies, and the optimization algorithm used are described as well as the assumptions underlying the choices made. Initial simulation results are reported.>

The authors present the delay-throughput performance of a new MAC protocol, the fair queue dual bus (FQDB), which relieves dual bus networks from the unfairness problems exhibited by the basic version of the distributed queue dual bus (DQDB) standard. The protocol considered is absolutely fair in bandwidth sharing and does not imply any change in the DQDB transmission system. It retains all the positive features of DQDB including simplicity, robustness, flexibility, 100% bandwidth utilization, and little slot access delay at light load. Comparisons to DQDB with the bandwidth balancing mechanism show that the FQDB delay-throughput performance was always superior while its implementation complexity was comparable.>

An architecture and a media access control (MAC) protocol aimed at improving fairness in dual-bus networks are introduced. The major feature of this proposal is that the distributed operation of updating the queue is made atomic, so that the average bus access delay is equalized over the network. An equalized access gain is provided by mandating that a station waiting for transmission consider itself as positioned in the distributed queue only after a constant delay, which is needed for all the other stations to register the request of the first one. This delay is equal to the propagation time of the request along the entire bus. It is shown that under certain conditions the average access delay results in the one experienced in the distributed queue dual bus (DQDB) by those stations which are placed at the middle of the bus. Consequently, the global end-to-end delay remains much below the limits fixed by the CCITT for delay sensitive services. The actual average delay is even smaller than the one estimated due to the presence in the protocol of the stand-by mechanism which is the same as in the DQDB version. In comparison with the traditional DQDB, the increase in access-delay values is rewarded by the significant improvement in terms of fairness.>

The design of a gateway that can interconnect teletex and OSI (Open Systems Interconnection) networks is shown. This goal is achieved by defining a functional block, the gateway mapping function (GMF), that provides a bridge between the procedures defined in CCITT Recommendation T.62 and the ISO OSI VT (virtual terminal) specification. The general architecture of the gateway is provided and the specification of its functions by means of SDL (specification and description language) diagrams are described.>