The telecommunications industry is undergoing a major transformation driven by new technologies alongside economic challenges that are reshaping the entire ecosystem. The increasing digitalization of industries and sectors demands more advanced network infrastructures tailored to diverse market verticals and supporting businesses in their digital transformation. However, the ecosystem’s future remains uncertain, influenced by operators’ strategic choices, business model innovation, national and industrial policies, regulations, and cross-industry investments. This paper extends previous work by not only outlining possible evolutionary scenarios of the telecommunications ecosystem but also analyzing the reciprocal relationship between technological advances and these scenarios. Specifically, it investigates how emerging market, ecosystem, and economic conditions influence the evolution and application of technology, while technology itself drives changes in these domains. Focusing on the European ecosystem, with implications for global contexts, this integrated perspective aims to support market players and policymakers in navigating transformation and making informed decisions. This work responds to the Grand Challenge “Creating a vision of the future evolution of telecommunications” within the RESTART research program, the most important public R&D initiative ever launched in the Italian telecommunications sector, funded by the European Union / Italian Ministry of University and Research, with an investment of 116 million euros. Consistently with the elements outlined above, it focuses on the trajectory of the telecommunications ecosystem, providing a comprehensive and forward looking assessment of the key business, technology and regulatory drivers at both national and European level.

Integrated Sensing and Communication (ISAC) systems promise unprecedented capabilities by merging connectivity and situational awareness, but also expose new attack surfaces at the physical layer. In this work, we demonstrate a blind full-frame OFDM replay attack that manipulates sensing outputs by injecting false targets and concealing real ones, without disrupting communication. The blind nature of our attack lies in the fact that it requires neither synchronization nor any knowledge of the signal structure, reference signals, or sensing parameters, making it not only practically viable, but even (somewhat) straightforward to execute. By replaying entire OFDM frames with a controlled delay and a frequency shift, the attacker can distort range estimations and induce Doppler shifts, mimicking the presence of moving targets. We present a general analytical framework to characterize the attack’s impact on range-Doppler processing and validate it through both system-level simulations with 5G NR parameters and real-world experiments. Experimental results build directly on a working 5G testbed with software-defined radios and commercial off-the-shelf hardware, which we extend with sensing capabilities, thereby demonstrating the attack’s feasibility and impact in a realistic ISAC scenario.

The integration of localization capabilities within the cellular architecture through dedicated 5G network functions has notably enhanced cellular positioning accuracy and enabled new location-based services. However, this architectural shift requires placing measurement acquisition and computation at the network edge and core, resulting in distributed computational resources and increased latency and security risks. As a result, minimizing latency and ensuring resilience against security threats, in addition to achieving high accuracy, become critical performance indicators in location-based services. This paper examines both 3GPP-standardized and O-RAN-based 5G architectures, detailing the key functions, interfaces, and parameters influencing the localization process, from measurement acquisition to position estimation. We define performance indicators for evaluating localization services and develop a system model that quantifies costs related to latency, accuracy, computation, and resilience against security threats. By jointly considering these factors, we formulate a multi-objective optimization problem that guides the selection of an optimal system configuration to simultaneously satisfy multiple localization requirements. We validate our approach through a case study of an end-to-end 5G system using both simulations and experimental data. Specifically, we evaluate various algorithms and implementations across standardized channels and scenarios. Furthermore, we conduct experimental measurements using Software-Defined Radios (SDRs) and open-source 5G platforms to assess operational latency with commercial-off-the-shelf (COTS) devices.

Positioning is a key focus in 5G standardization, starting with 3GPP Release 16. However, most of the effort from the research community and work presented in the technical standardization has been limited mainly to simulation studies. This paper explores the use of software-defined radios (SDRs) platforms for 5G positioning, presenting an overview of the current state-of-the-art and available open-source platforms. Utilizing an advanced SDR-based multi-gNodeBs (gNBs) synchronized testbed, the paper conducts a series of time-based, over-the-air measurements. Results offer insights into the impact of various real-world system parameters, such as the number of gNBs, transmission bandwidth, signal processing techniques, and localization algorithms on positioning accuracy and Time to First Fix (TTFF). These findings provide a pathway for the cost-effective and efficient implementation of high-precision 5G localization systems. The paper contributes to advancing both theoretical understanding and practical applications, serving as a guide for the development of 5G positioning technology.

Modern wireless communication networks are threatened by new generations of radio hackers. These are skilled attackers equipped with low-cost software radios, suitably instrumented so as to monitor, degrade, or even alter the radio signals. The aim of this paper is to devise innovative detection architectures against the most common classes of threats: broadband noise jammers, whose goal is to reduce the signal-to-noise ratio, and spoofing/meaconing attacks, which aim to inject false or incorrect information into the receiver. To this end, we resort to the hypothesis testing theory and solve the associated problems by means of the GLRT possibly accounting for penalty terms. The resulting decision schemes represent the main technical novelty of this work. The analysis of their performance focuses on a location security case study for 4G/5G cellular networks. To this end, we leverage measurement models from the cellular localization literature and generate data according to these models. The numerical results show the effectiveness of the proposed approaches in comparison with suitable counterparts.

While GPS has traditionally been the primary positioning technology, 3GPP has more recently begun to include positioning services as native, built-in features of future-generation cellular networks. With Release 16 of the 3GPP, finalized in 2021, a significant standardization effort has taken place for positioning in 5G networks, especially in terms of physical layer signals, measurements, schemes, and architecture to meet the requirements of a wide range of regulatory, commercial and industrial use cases. However, experimentally-driven research aiming to assess the real-world performance of 5G positioning is still lagging behind, root causes being i) the slow integration of positioning technologies in open-source 5G frameworks, ii) the complexity in setting up and properly configuring a 5G positioning testbed and iii) the cost of a multi-BS deployment. This paper sheds some light on all such aspects. After a brief overview of state of the art in 5G positioning and its support in open-source platforms based on software-defined radios, we provide advice on how to set-up positioning testbeds, and we demonstrate, via a set of real-world measurements, how to assess aspects such as reference signal configurations, localization algorithms, and network deployments, even with a cost-constrained limited-size testbed.

We target the planning of a 5G cellular network under 5G service and ElectroMagnetic Fields (EMFs) constraints. We initially model the problem with a mixed integer linear programming (MILP) formulation. The pursued objective is a weighed function of next-generation Node-B (gNB) installation costs and 5G service coverage level from a massive multiple input multiple output (MIMO) system. In addition, we precisely model restrictive EMF constraints and we integrate scaling parameters to estimate the power radiated by 5G gNBs. Since the considered planning problem is NP-Hard, and therefore very challenging to be solved even for small problem instances, we design an efficient heuristic, calledPLATEA, to practically solve it. Results, obtained over a realistic scenario that includes EMF exposure from pre-5G technologies (e.g., 2G, 3G, 4G), prove that the cellular planning selected byPLATEAensures 5G service and restrictive EMF constraints. However, we demonstrate that the results are strongly affected by: i) the relative weight between gNB installation costs and 5G service coverage level; ii) the scaling parameters to estimate the exposure generated by 5G gNBs; iii) the amount of exposure from pre-5G technologies; and iv) the adopted frequency reuse scheme.

A very popular theory circulating among non-scientific communities claims that the massive deployment of Base Stations (BSs) over the territory, a.k.a. cellular network densification, always triggers an uncontrolled and exponential increase of human exposure to Radio Frequency “Pollution” (RFP). To face such concern in a way that can be understood by the layman, in this work we develop a very simple model to compute the RFP, based on a set of worst-case and conservative assumptions. We then provide closed-form expressions to evaluate the RFP variation in a pair of candidate 5G deployments, subject to different densification levels. Results, obtained over a wide set of representative 5G scenarios, dispel the myth: cellular network densification triggers an RFP decrease (up to three orders of magnitude) when the radiated power from the BS is adjusted to ensure a minimum sensitivity at the cell edge. Eventually, we analyze the conditions under which the RFP may increase when the network is densified (e.g., when the radiated power does not scale with the cell size), proving that the amount of RFP is always controlled. Finally, the results obtained by simulation confirm the outcomes of the RFP model.

Most localization systems rely on measurements gathered from signals emitted by stations whose position is assumed known as ground truth, namely anchors. As demonstrated by a significant bulk of experimental research, location security is threatened when an attacker becomes able to tamper either the signals emitted by the stations, or convince the user that the anchor station is in a different position than the true one. With this paper, we first propose a formal threat model which captures the above-mentioned wide class of attacks, and permits to quantitatively evaluate how tampering of one or more anchor locations undermines the user’s localization accuracy. We specifically derive a Cramér Rao Bound for the localization error, and we assess a number of example scenarios. We believe that our study may provide a useful formal benchmark for the design and analysis of detection and mitigation solutions.

The debate over whether electromagnetic fields (EMFs) exposure poses a danger to human health is recurring and goes back centuries to when our society first began to rely on electricity [1].

Small cells (SCs) mounted on top of Unmanned Aerial Vehicles (UAVs) can be used to boost the radio capacity in hotspot zones. However, UAV-SCs are subject to tight battery constraints, resulting in frequent recharges operated at the ground sites. To meet the UAV-SCs energy demanded to the ground sites, the operator leverages a set of Solar Panels (SPs) and grid connection. In this work, we demonstrate that both i) the level of throughput provided to a set of areas and ii) the amount of energy that is exchanged with the grid by the ground sites play a critical role in such UAV-aided cellular network. We then formulate the J-MATE model to jointly optimize the energy and throughput through revenue and cost components. In addition, we design the BBSR algorithm, which is able to retrieve a solution even for large problem instances. We evaluate J-MATE and BBSR over a realistic scenario composed of dozens of areas and multiple ground sites, showing that: i) both J-MATE and BBSR outperform previous approaches targeting either the throughput maximization or the energy minimization, and ii) the computation time and the memory occupation of BBSR are reduced up to five orders of magnitude compared to J-MATE.

A common concern among the population is that installing new 5G Base Stations (BSs) over a given geographic region may result in an uncontrollable increase of Radio-Frequency “Pollution” (RFP). To face this dispute in a way that can be understood by the layman, we develop a very simple model, which evaluates the RFP at selected distances between the user and the 5G BS locations. We then obtain closed-form expressions to quantify the RFP increase/decrease when comparing a pair of alternative 5G deployments. Results show that a dense 5G deployment is beneficial to the users living in proximity to the 5G BSs, with an abrupt decrease of RFP (up to three orders of magnitude) compared to a sparse deployment. We also analyze scenarios where the user equipment minimum detectable signal threshold is increased, showing that in such cases a (slight) increase of RFP may be experienced.

We focus on the ElectroMagnetic Field (EMF) exposure safety for people living in the vicinity of cellular towers. To this aim, we analyze a large dataset of long-term EMF measurements collected over almost 20 years in more than 2000 measurement points spread over an Italian region. We evaluate the relationship between EMF exposure and the following factors: (i) distance from the closest installation(s), (ii) type of EMF sources in the vicinity, (iii) Base Station (BS) technology, and (iv) EMF regulation updates. Overall, the exposure levels from BSs in the vicinity are below the Italian EMF limits, thus ensuring safety for the population. Moreover, BSs represent the lowest exposure compared to Radio/TV repeaters and other EMF sources. However, the BS EMF exposure in proximity to users exhibits an increasing trend over the last years, which is likely due to the pervasive deployment of multiple technologies and to the EMF regulation updates. As a side consideration, if the EMF levels continue to increase with the current trends, the EMF exposure in proximity to BSs will saturate to the maximum EMF limit by the next 20 years at a distance of 30 meters from the closest BS.

Message Queuing Telemetry Transport (MQTT) is a widespread protocol for topic-based publish-subscribe architectures supporting IoT and social networks applications. MQTT brokers are logical entities that couple publishers and subscribers and play a critical role in such architectures. MQTT brokers can be implemented either as standalone servers or in a cluster configuration. Clusters of brokers increase reliability, availability and overall performance, since operations can be highly parallelized among the brokers that form the cluster. The load-balancing strategy in a cluster usually consists in connecting an incoming client to a randomly selected broker. This random-attach strategy, it is very simple, but generates a significant amount of inter-broker traffic, as we demonstrate through theoretical and experimental evaluations. Inter-broker traffic is an overhead for the system and it increases the CPU load of the brokers, compromising the scaling behavior of the whole cluster. Indeed, we found that a linear increase of the number of brokers forming a cluster does not necessarily provide an equivalent linear gain in performance, and such a scaling penalty can be surprisingly significant, in the order of 40%. To solve this issue and improve performance, we propose a novel load-balancing strategy that envisages the use of multiple MQTT sessions per client to reduce inter-broker traffic and that can be implemented by means of a greedy algorithm. We show feasibility and effectiveness of our strategy for IoT and social-network applications by means of simulations and real measurements. The resulting scaling penalty is reduced to 10%.

On August 14, 2018, a new set of vulnerabilities collectively named “L1 terminal fault” were announced. Systems with microprocessors utilizing out-of-order execution could allow unauthorized disclosure of information residing in the L1data cache, by tweaking the virtual memory abstraction. The vulnerability was therein mentioned for three different scenarios. In this demo-paper, we provide practical evidence about the feasibility of the most complex “VMM” case of an attacker residing in a Virtual Machine (VM), and targeting information leakage from the host OS and other independent VMs.

The expected dramatic growth of connected things raises the issue of how to efficiently organize them, in order to monitor and manage functions and interactions. Information centric networking (ICN) is a communication paradigm that provides content-oriented functionality in the network and at the network level, including content routing, caching, multicast, mobility, data-centric security, and a flexible namespace. Thus, it is a viable solution for supporting Internet of Things (IoT) services without requiring any centralized entity. In this paper, we introduce the lightweight named object solution: a convenient way to represent physical IoT objects in a derived name space, exploiting ICN. We show that this abstraction can: 1) increase the programming simplicity; 2) offer extended functionality, such as augmentation and upgrading, to cope with the “software erosion,” and 3) implement a common interaction logic involving mutual function invocation. We present some proof-of-concept implementations of the proposed abstraction dealing with challenging IoT test cases; we also carry out a performance evaluation in a simulated network scenario.

We target the problem of managing the power states of the servers in a Cloud Data Center (CDC) to jointly minimize the electricity consumption and the maintenance costs derived from the variation of power (and consequently of temperature) on the servers' CPU. More in detail, we consider a set of virtual machines (VMs) and their requirements in terms of CPU and memory across a set of Time Slot (TSs). We then model the consumed electricity by taking into account the VMs processing costs on the servers, the costs for transferring data between the VMs, and the costs for migrating the VMs across the servers. In addition, we employ a material-based fatigue model to compute the maintenance costs needed to repair the CPU, as a consequence of the variation over time of the server power states. After detailing the problem formulation, we design an original algorithm, called Maintenance and Electricity Costs Data Center (MECDC), to solve it. Our results, obtained over several scenarios from a real CDC, show that MECDC largely outperforms two reference algorithms, which instead either target the load balancing or the energy consumption of the servers.

5G is expected to become the dominant technology in the forthcoming years. In this work, we consider a 5G Superfluid network, as an outcome of the H2020 project SUPERFLUIDITY. The project exploits the concept of Reusable Functional Block (RFB), a virtual resource that can be deployed on top of 5G physical nodes. Specifically, we focus on the management of the RFBs in a Superfluid network to deliver a high definition video to the users. We design an efficient algorithm, called P5G, which is based on Particle Swarm Optimization (PSO). Our solution targets different Key Performance Indicators (KPIs), including the maximization of user throughput, or the minimization of the number of used 5G nodes. Results, obtained over a representative scenario, show that P5G is able to wisely manage the RFBs, while always guaranteeing a large throughput to the users.

We target the evaluation of a Superfluid 5G network from an economic point of view. The considered 5G architecture has notably features, such as flexibility, agility, portability and high performance, as shown by the H2020 SUPERFLUIDITY project. The proposed economic model, tailored to the Superfluid network architecture, allows to compute the CAPEX, the OPEX, the Net Present Value (NPV) and the Internal Rate of Return (IRR). Specifically, we apply our model to estimate the impact for the operator of migrating from a legacy 4G to a 5G network. Our preliminary results, obtained over two realistic case studies located in Bologna (Italy) and San Francisco (CA), show that the monthly subscription fee for the subscribers can be kept sufficiently low, i.e., typically around 5 [USD] per user, while allowing a profit for the operator.

This paper explores methodologies, advantages and challenges related to the use of the Information Centric Network technology for developing NoSQL distributed databases, which are expected to play a central role in the forthcoming IoT and BigData era. ICN services make possible to simplify the development of the database software, improve performance, and provide data-level access control. We use our findings to design a NoSQL spatio-temporal database, named OpenGeoBase, and evaluate its performance by using a real data set relevant to Intelligent Transport System applications.

This paper describes a demo of OpenGeoBase, a NoSQL spatio-temporal distributed database that exploits Information Centric Network technology. We show its basic operations and two applications based on it. The first one is an Intelligent Transport System application making possible to discover public transport information (GTFS files) all over the world. The second application concerns a citizen reporting application though which users can post and search for geo-tagged photos.

We consider the problem of evaluating the performance of a 5G network based on reusable components, called Reusable Functional Blocks (RFBs), proposed by the Horizon 2020 SUPERFLUIDITY project. RFBs allow a high level of flexibility, agility, portability and high performance. After formally modelling the RFB entities and the network physical nodes, we optimally formulate the problem of maximizing different Key Performance Indicators (KPIs) on an RFB-based network architecture, in which the RFBs are shared among the nodes, and deployed only where and when they are really needed. Our results, obtained by solving the proposed optimization problem over a simple yet representative scenario, show that the network can be managed in a very efficient way. More in depth, the RFBs are placed into the nodes in accordance with the amount of requested traffic from users and the specific pursued KPI, e.g., maximization of user throughput or minimization of the number of used nodes. Moreover, we evaluate the relationship between the capacity of each node and the number of RFBs deployed on it.

User authentication at Wi-Fi Access Points (APs) is becoming an important issue. Wi-Fi APs are indeed ubiquitous, but existing authentication methods such as WPA/WPA2 static pre-shared secret key (PSK), or 802.1X-based online authentication services (e.g., RADIUS servers/proxies) have their theoretical or practical limitations. In a previous work, we proposed WIFAB, a new authentication mechanism which neither requires online backend access control infrastructure, nor relies on a static pre-shared secret key. In this paper, we extend WI-FAB by removing the need for having a central authority for user authentication and credential issuing. Our main contribution is twofold: (i) adopting decentralized multi-authority CP-ABE, we support the users who have authentication/authorization credentials from multiple authorities. We decouple the user credentials issuing from the management of the WPA2-PSK, so that neither the credential issuing authority can track the users, nor the AP can access the real identity of the users. Considering an extensive attack model, we show that the proposed approach is secure and preserves the privacy of the users. (ii) We provide a real-world implementation of the proposed approach on off-the-shelf embedded hardware to demonstrate its feasibility and efficiency.

We explore the opportunities and design options enabled by novel SDN and NFV technologies, by re-designing a dynamic content delivery network (CDN) service. Our system, named MOSTO, provides performance levels comparable to that of a regular CDN, but does not require the deployment of a large distributed infrastructure. In the process of designing the system, we identify relevant functions that could be integrated in the future Internet infrastructure. Such functions greatly simplify the design and effectiveness of services, such as MOSTO. We demonstrate our system using a mixture of simulation, emulation, testbed experiments, and by realizing a proof-of-concept deployment in a planet-wide commercial cloud system.

MPEG DASH is a widely used standard for adaptive video streaming over HTTP. The conceptual architecture for DASH includes a web server and clients, which download media segments from the server. Clients select the resolution of video segments by using an Adaptive Bit-Rate (ABR) strategy; in particular, a throughput-based ABR is used in the case of live video applications. However, recent papers show that these strategies may suffer from the presence of proxies/caches in the network, which are instrumental in streaming video on a large scale. To face this issue, we propose to extend the MPEG DASH architecture with a Tracker functionality, enabling client-to-client sharing of control information. This extension paves the way to a novel family of Tracker-assisted strategies that allow a greater design flexibility, while solving the specific issue caused by proxies/caches; in addition, its utility goes beyond the problem at hand, as it can be used by other applications as well, e.g. for peer-to-peer streaming.

This paper presents an Android application exploiting a collaborative multi-RAT approach (cellular and Wi-Fi Radio Access Technologies) for adaptively streaming video content. The application has been developed and tested on real Android devices of different vendors. Each device coordinates with the others to download portions of a video stream through its own preferred RAT, and shares the results with the neighboring devices through a proximity, high bit rate network based on Wi-Fi Direct. As a result, the aggregate bandwidth available to the group of collaborating devices allows a faster download and a higher quality of video playback streaming, with respect to what is achievable by a single device. The application is designed to exploit common Android devices along with the key functionalities of an Information Centric Network: routing-by-name, in-network caching and multicasting.

Information Centric Networking (ICN), a novel network paradigm, places the focus on the content instead of the end-hosts. ICN addresses content by names instead of locations and can ease content retrieval and improve network efficiency. Ongoing work attempts to extend the use of ICN to scenarios such as real time communication, group communication, push services, and addresses the issue of migration from the current network and coexistence of different network paradigms. In this work, we argue that by extending the current design of ICN from a “host-to-name” to a “name-to-name” architecture, the utility and efficiency of ICN could be further increased. We propose Internames, an architectural framework in which names are used to identify all entities involved in communication: content, users, devices, logical points, and services. Internames is envisioned to be an overarching name-to-name communication primitive that is fully compatible with ICN principles, accommodates the coexistence (or gradual migration) of different network realms (e.g., IP, ICN, VANET) and is suitable for application scenarios where ICN is somehow limited by its reliance on a “host-to-name” approach. In this paper, we provide early insights into the Internames architecture by leveraging on the work done by the research community and identify components and challenges that require more detailed investigation.

Information Centric Networking (ICN) is paradigm in which the network layer provides users with content addressed “by name”. In-network caching is one of the key functionality to be provided by ICN nodes. To avoid network nodes caching fake contents, it is necessary to verify the validity of data items. A content is deemed to be valid if it verifies three properties: i) integrity: it has not be modified; ii) provenance: it comes from the intended source; iii) relevance: it is indeed the content requested from the user (by using the name of that content). In this paper, we discuss the interplay among three pivotal ICN aspects: caching, validity and naming. Specifically, we will investigate different naming and digital signature schemes, evaluating their speed, overhead and their impact on caching performance. Perhaps counter-intuitively, we find that the relatively slow verification time of today's signatures, which bottlenecks the rate of storing new data items in the network caches, does not come as a critical shortcoming, but may actually even improve the cache hit probability when the LRU caching policy is employed.

Information Centric Networking (ICN) is a new paradigm in which the network layer provides users with content exposed as names, instead of providing communication channels between hosts. We present a P2P application for the live streaming of video contents encoded at multiple bit-rates. The application enables a limited set of neighboring cellular devices to increase the quality of video playback, by cooperatively using their cellular (e.g. 3G) and proximity (e.g. Wi-Fi Direct) wireless connections. The application exploits key functionalities of ICN: routing by name, in network caching and multicast delivery. We developed a prototype of the application and assessed its performance in a test-bed based on Linux devices with 3G connections, the CCNx tool, the VLC player and the MPEG DASH streaming format. Our main contribution is to show how ICN can provide support to emerging video streaming applications, and ease their creation.

In this paper we propose to integrate Software Defined Networking (SDN) principles in Wireless Mesh Networks (WMN) formed by OpenFlow switches. The use of a centralized network controller and the ability to setup arbitrary paths for data flows make SDN a handy tool to deploy fine-grained traffic engineering algorithms in WMNs. However, centralized control may be harmful in multi-hop radio networks formed by commodity devices (e.g. Wireless Community Networks), in which node isolation and network fragmentation are not rare events. To exploit the pros and mitigate the cons, our framework uses the traditional OpenFlow centralized controller to engineer the routing of data traffic, while it uses a distributed controller based on OLSR to route: i) OpenFlow control traffic, ii) data traffic, in case of central controller failure. We implemented and tested our Wireless Mesh Software Defined Network (wmSDN) showing its applicability to a traffic engineering use-case, in which the controller logic balances outgoing traffic among the Internet gateways of the mesh. Albeit simple, this use case allows showing a possible usage of SDN that improves user performance with respect to the case of a traditional mesh with IP forwarding and OLSR routing. The wmSDN software toolkit is formed by Open vSwitch, POX controller, OLSR daemon and our own Bash and Python scripts. The tests have been carried out in an emulation environment based on Linux Containers, NS3 and CORE tools.

Information Centric Networking (ICN) is a new paradigm in which the network layer provides users with content, instead of providing communication channels between hosts, and is aware of the name (or identifiers) of the contents. In this paper, we first describe what, we believe, are the main advantages and components of an ICN infrastructure; then we present an overall architecture for ICN, and then we focus on the main contribution of the paper, which is a route caching technique, designed to improve the scalability of the routing by name functionality.

The Information-Centric Networking (ICN) paradigm is expected to be one of the major innovation of the Future Internet An ICN can be characterized by some key components like: (i) the content-centric request/reply paradigm for data distribution, (ii) route-by-name operations, and (iii) in-network caching. In this paper we focus on a framework for ICN called CONET (COntent NETwork) and in particular on a solution devised under this framework called coCONET. coCONET characteristics make it suitable for deployment in accordance to the Software Defined Networks (SDN) philosophy. In this paper, we will describe how coCONET can be implemented over an OpenFlow (the most popular SDN instantiation, to date) network and how OpenFlow should be modified to better suit the operations of coCONET and, more in general, of ICN solutions.

In this paper we present a cooperative video streaming application running on top of an “Information Centric Network” (ICN). The application could be used on mobile devices to offload the cellular radio interface. We demonstrate our application in a test-bed exploiting the CCNx software implementation of an ICN, the VideoLan tool and the Apple HTTP Live Streaming format.

The contribution of this position paper is twofold: the first one extends the use of social networking in the direction of allowing not only communications among persons but also human-machine interaction and machine to machine collaboration; the second contribution is the presentation of an experimental platform for large-scale testing of innovative mobile social networking applications, including the one described in the first part of the paper.

The aim of this extended abstract is to present Campus++, a location-based publish-subscribe system for intermittently connected delay tolerant networks, exploiting IEEE 802.15.4 devices, and taking into due account the severe constraints deriving from their physical characteristics. We describe our proposed architectural model and how we implemented our solution in a real test-bed. We point out that our system can be easily adapted to operate in a fully distributed, infrastructure-less way, allowing free communications e.g. in disaster areas or in areas in which ”usual” communications means are either non existent or intentionally made unavailable.

The Data Distribution Service (DDS) middleware is enjoying a rapid adoption in high-performance, mission-critical networks. At the same time, the H.264 Scalable Video Coding (SVC) has been recently standardized and it is deemed to be an effective solution for video streaming over a channel with time-varying bandwidth, like the wireless one. In these conditions, it is critical to adapt the video bit-rate to the actual wireless capacity, and bit-rate adaptation is extremely simple for a H.264 SVC video. In this paper we devise, evaluate and demonstrate a technique for streaming H.264 SVC video over a DDS middleware. The contribution is threefold: i) we design a structure of the DDS data-unit able to carry H.264 SVC video-units; ii) we devise a receiver-driven rate-control mechanism based on our DDS data-unit and exploiting specific DDS functionality; iii) we implement and show the effectiveness of our mechanism in an 802.11 wireless scenario, comparing our proposal with other solutions.

In this paper, we propose and evaluate an overlay distribution algorithm for P2P, chunk-based, streaming systems over forest-based topologies. In such systems, the stream is divided in chunks; chunks are delivered by each node in a store-and-forward way. A relaying node starts distributing a chunk only when it has completed its reception from another node. Peers are logically organized in a forest of trees, where each tree includes all peers. The source periodically distributes different chunks to each tree for their delivery. Our key idea consists in employing serial transmission: for each tree, and thus, for each chunk, the source node sends the chunk to its children in series; the same holds for each peer node of the tree, excluding the leaves. Besides this basic idea, the contributions of this paper are: 1) we demonstrate the feasibility of serial transmission over a forest of trees, which is not a trivial problem, unlike the case of parallel transmission; 2) we derive an analytical model to evaluate the system performance; 3) we derive a theoretical bound for the number of nodes reachable in a given time interval or equivalently for the time required to reach a given number of nodes; 4) we prove the optimality of our approach in terms of its capability to reach such bound; and 5) we develop a general simulation package for P2P streaming systems and use it to compare our solution to literature results. Finally, we stress that this paper is focused on the theoretical properties and performance understanding of the proposed distribution algorithm, rather than on its practical implementation in a real system. However, we also briefly describe a practical workable implementation of our algorithm.

In this paper we evaluate analytically the average occupancy of the transmission buffer of a 802.11 station (STA). The station belongs to a Wi-Fi Hot-Spot and exchanges data with a fixed host. The data exchange is regulated by the Transmission Control Protocol (TCP). The research interest is motivated by the fact that several papers assume that in these conditions the STA buffer is nearly empty. On the contrary, we prove that this assumption may be wrong and discuss the consequences of this fact. We test the proposed model by means of ns2 simulation, ascertain its accuracy, and highlight its limits.

Many distribution algorithms have been proposed up to now for P2P real time streaming. However, due to the lack of basic theoretical results and bounds, common sense and intuitions and heuristics have driven their design so far. The consequence is that we can find in the literature a large variety of different choices about the main aspects of a P2P system, such as overlay topology, scheduling process and upload strategy. In this situation, it is difficult to establish unambiguously the absolute goodness of a particular algorithm or even the rationale behind a particular choice or solution. In this paper we propose and evaluate a theory-driven distribution algorithm for P2P real time streaming. We take advantage from a previous theoretical study, where: i) we derived a theoretical performance bound for forest-based overlay topologies regarding the number of nodes reachable in a given time interval or equivalently the time required to reach a given number of nodes; ii) we proved the optimality of streamline, a distribution algorithm based on the serial transmission over forest-based topologies, in terms of its capability to reach such a bound. The streamline algorithm is based on some ideal assumptions that prevent its practical implementation. In this paper we remove these assumptions and present a practical and working algorithm, named operational streamline or simply O-Streamline. We also evaluate the performance of O-Streamline, comparing them with the optimal bounds of streamline.

In this paper we introduce SMILE (simple middleware independent layer), a framework whose main purpose is to facilitate the development of distributed applications. In the SMILE abstraction an application is composed by a set of processes that exchange information. The interfaces of these processes are described using WSDL or by an equivalent UML definition. Using the open source AndroMDA tool and starting from the UML interface specification we are able to generate the skeleton of SMILE applications and most part of their business logic. An application developed using SMILE can run on different middleware platforms just changing its binding, i.e. the code that adapts SMILE to a given middleware. We have implemented bindings to CORBA, JAVA- RMI, JADE, JXTA and to an our own communication mechanism based on SIP suitable for mobile devices. At the end we hint at the usage of SMILE in service composition and present some prototype applications.

Abstract This paper presents a novel Mobile Ad‐hoc NETworks (MANET) multicast protocol, named Overlay Borůvka‐based Ad‐hoc Multicast Protocol (OBAMP), and evaluates its performance. OBAMP is an overlay protocol: it runs only in the end‐systems belonging to the multicast group. OBAMP has three distinctive features, which give to the protocol a good performance in terms of distribution efficiency: (i) its distribution tree closely resembles the minimum spanning tree; (ii) it exploits broadcast communications; (iii) its design limits not only overlay signaling but also network‐layer signaling. In addition, OBAMP can cope with node failures in a very short time. As a consequence, OBAMP has a low latency and a high delivery ratio, even when the group size increases. To prove these statements, we analyze the performance of OBAMP with ns‐2 and compare it with three state‐of‐the‐art protocols, namely ODMRP (a network‐layer protocol), ALMA, and AMRoute (two overlay protocols). The overlay protocols are assumed to use AODV as underlying routing protocol. Also, we stress that we have implemented OBAMP, in Java, and we have tested it on the field, to prove its feasibility; to allow fellow researchers to reproduce and test our work we published all simulation and implementation codes. Copyright © 2007 John Wiley & Sons, Ltd.

This work presents a novel MANET multicast protocol, named overlay Boruvka-based ad-hoc multicast protocol (OBAMP), and evaluates its performance. OBAMP is an overlay protocol: it runs only in the end-systems belonging to the multicast group. User data are distributed over a shared distribution tree formed by a set of non-cyclic UDP tunnels. OBAMP derives the distribution tree by approximating the Boruvka algorithm; the Boruvka algorithm is a classical tool (1926) to find the minimum spanning tree; thus, the distribution tree of OBAMP is an approximation of the minimum spanning tree of the connectivity topology at hand. OBAMP shows three distinctive advantages: i) its distribution tree closely resembles the minimum spanning tree; ii) it exploits broadcast communications (with favourable consequences on its efficiency); iii) its design takes into account not only overlay signalling but also network-layer signalling; thus, the protocol succeeds in limiting the overall signalling load, network+overlay. As a consequence, OBAMP has a low-latency and a high delivery ratio, even when the group size increases. To prove this statement, we analyze the performance of OBAMP with ns-2 and compare it with two state-of-the-art protocols, namely ODMRP (a network-layer protocol) and ALMA (an overlay protocol). Both OBAMP and ALMA are assumed to use AODV as underlying routing protocol. Also, we stress that we have implemented OBAMP, in Java, and we have tested it on the field, to prove its feasibility. To allow fellow researchers to reproduce and test our work we published all simulation and implementation codes, in www.radiolabs.it/obamp.

The actual standards for service authoring, composition and development are not easy to port and to apply for next generation mobile applications. This paper describes some tools that we're developing in the context of the IST-Simple Mobile Service project, whose aim is to ease the authoring and the use of services for mobile devices. We propose a service composition approach using an UML profile very close to the actual standards for Web services definition and authoring, like WSDL and BPEL. We take a glance at SMILE, the run-time support we provide for service execution. Finally we hint at an efficient serialization mechanism based on JSON, a human readable data exchange format less verbose and, in our opinion, more suitable for mobile terminals than XML.

The paper focus is on the minimum mean square error (MSE) estimation methods in RSS-based localization techniques. We in particular discuss three different error criteria: i) minimizing the mean square distance error (distance-based MSE), ii) minimizing the mean square power error (power-based MSE), iii) minimizing the mean square relative distance error (relative distance-based MSE). The goal is to assess the impact of the chosen error criterion on the estimate accuracy. Simulation results show that i) flip ambiguities in the position estimate may occur when the node to be positioned is very close to the signal source, ii) in presence of flip ambiguities, provided that RSS measurements are combined with information about the signal arrival direction, relative distance-based MSE minimization offers better accuracy in a bi-dimensional localization scenario1.

In this paper, we study the problem of maintaining fairness for TCP connections in wireless local area networks (WLANs), based upon the IEEE 802.11 standard and operating in DCF mode. Although this mode of operation ensures fair access to the medium at the MAC level, it does not provide fairness among TCP connections. TCP unfairness may result in significant degradation of performance, leading to users perceiving unsatisfactory quality of service. We propose and analyze a solution capable of enabling TCP fairness with minimal additional complexity. The proposed solution is based on utilizing an IP rate-limiter with an adaptive rate control mechanism and does not require modifying existing standards at the MAC or network layers. The solution is fully compatible with existing devices and can be integrated either within the access point or within and external device (e.g., the network gateway). The performance analysis is performed in a real test-bed and proves the feasibility and effectiveness of our mechanism. To allow fellow researchers to reproduce our work we published on the WEB all the implementation code.

In this paper, we study the problem of maintaining fairness for TCP connections in wireless local area networks (WLANs) based upon the IEEE 802.11 standard. Current implementations of 802.11 use the so-called distributed coordination function (DCF), which provides similar medium access priority to all stations. Although this mode of operation ensures fair access to the medium at the MAC level, it does not provide any provisions for ensuring fairness among the TCP connections. TCP unfairness may result in significant degradation of performance leading to users perceiving unsatisfactory quality of service. We propose and analyze two solutions that are capable of enabling TCP fairness with minimal additional complexity. The proposed solutions are based on utilizing a rate-control mechanism in two modes: static or adaptive. They do not require modifying existing standards at the MAC or network layers. Hence, they are fully compatible with existing devices. Our performance analysis results prove the efficaciousness of our proposed solutions in achieving TCP fairness compared to existing approaches. We have, also, implemented the proposed solutions in an ad-hoc experimental test-bed, and performed measurements to demonstrate the validity of our approach and results

As technology develops, people are using an ever broader and heterogeneous range of ICT devices and network-based services. The result is an enormous burden of complexity on the shoulders of users, service providers and network operators. The goal of the Simplicity project, supported by the European Union, is to reduce this complexity by: (i) providing automatic customization of user access to services and the network; (ii) automatically adapting services to terminal characteristics and user preferences; (iii) orchestrating network capabilities. The Simplicity approach is based on a personalization device, the so-called Simplicity device, which, together with a brokerage framework, offers transparent service configuration and runtime adaptation, according to user preferences and computing/networking context conditions. This paper presents the Simplicity project, briefly discusses its architecture and introduces the Simplicity demonstrator: a proof of concept implementation used to verify our approach, through diverse usage scenarios. In addition, we introduce the main aspects of a follow-up project of Simplicity, named SMS.

Reconfigurable radio systems allow user terminals to access different communication technologies, without duplication of hardware. To this end, terminals need appropriate software (SW) modules, which, in advanced scenarios, can be downloaded on demand from the network. The network element that provides these modules to the terminals may need to know information about the terminal/user, in order to better adapt the SW modules to user needs and terminal capabilities and to grant only requests coming from authorized users. For this reason, such network element needs to access user profile information. In this paper, we propose a distributed approach to retrieve users profile information that does not require the mandatory presence of a public network operator. The goal of this mechanism is to allow a suitably appointed entity to retrieve all the needed information, related to the user/terminal, from a suitable location/server in the network, given some information provided by the terminal itself. Our approach is based on the "Dataweb" paradigm, under standardization in the Oasis consortium

In this paper, we present a solution to control automatically a wireless access network and we focus on load balancing target. The reference scenario is that of a single administrative domain that offers a heterogeneous wireless access. We propose a monitoring procedure able to track the access network context, i.e., the current state of resources, the amount of service demand, and the wireless access points available to mobile terminals. To this end, we exploit a cooperative brokerage framework, distributed between terminals and network. The access network context is the input of a selection process that aims to distribute suitably mobile users among available access points. The objective is to balance the users demand, thus improving the quality of the service. The numerical analysis shows the effectiveness of the proposed system in an 802.11b access network.

An improved IP network service (e.g., for real time services) is expected in the near future in both wired and wireless environments. In this regard, handover capabilities are extremely important and challenging, in particular if their use in operation must be seamless. One of the main steps to achieve seamless handover is the quick discovery of the IP addresses and service capabilities of candidate access routers to hand over to. We present a push-mode-multicast based solution to discover and update timely information about wireless resources. We evaluate the effectiveness of the proposed approach in terms of signaling burden and discovery time with respect to solutions already presented in the literature.

In this paper, we consider a per-call, usage-based tariff model to charge for IP services with guaranteed quality. This model is based on the virtual delay, which is a quality of service (QoS) index that describes an improved IP service provided by a network domain. We show how to compute the virtual delay, and how to make it dependent on the service demand. Then, we demonstrate the effectiveness of our tariff model to tune revenues, blocking probability, and resource utilization in a meaningful application scenario. Our goal is to give some directions for network resource dimensioning and pricing purposes, which depend on the service demand.

Traffic engineering and admission control schemes for QoS support over the Internet share the hypothesis that offered traffic must be policed. Recently, measurement-based techniques have been proposed as effective QoS-guarantee mechanisms. For them, a key element is a thorough understanding of the characteristics of regulated traffic. In this paper, we present an analysis of the traffic output process for a leaky-bucket-regulated voice source with silence suppression, which would be a prime candidate for VoIP service. Specifically, we present a procedure for calculating the mean and the variance of the amount of regulated traffic as a function of the measurement period. We evaluate the quality of our results via simulation.

Advanced mobility management is a key issue in providing nomadic users with QoS-enabled services in the future Internet. To improve the mobile IP performance, a crucial role is played by the discovery and selection of the access router to handover. This procedure is of great importance to enable the other additional mobility-related protocols to speed up handover. This paper presents and evaluates a novel approach for carrying out this task, based on a fully distributed process, which also allows timely inter-technology handovers.

In this paper, we present a per-call usage-based tariff model to charge for IP guaranteed services. The tariff is built on the basis of the virtual delay, which is a value characterizing an improved IP service. We propose a model to compute the virtual delay within an administrative domain from purely technical considerations, taking into account traffic, performance and system parameters. In addition, we analyze the sensitivity of the virtual delay and of the tariff to the involved parameters, taking into account both users' benefit and operators' income. Our goal is to provide directions for dimensioning purposes.

IETF developed two main approaches to provide QoS aware services in the Internet: integrated services (IntServ) and differentiated services (DiffServ). Both have well known pros and cons. The stateful IntServ has a greater level of accuracy and a finer level of granularity. The stateless DiffServ possesses excellent scaling properties, but lacks a standardized admission control scheme and, upon overload in a given service class, degradation of service can occur. To provide QoS in DiffServ, three possible strategies are: (i) plain and heavy over-dimensioning; (ii) admission control at the borders of the DiffServ region, coupled with suitable assumptions on the distribution of the traffic within the region, which can lead to over-dimensioning, even if less severe than the previous one; (iii) per-node (i.e., within the region) admission control. Following RFC2990, we proposed an "admission control function which can determine whether to admit a service differentiated flow along the nominated network path", i.e., the third of the above strategies. This function, named GRIP (Gauge & Gate Reservation with Independent Probing), can provide strict QoS guarantees by means of stateless DiffServ-compliant procedures. This feature is paid for with a potential loss of efficiency, with respect to an ideal, stateful admission control. The goal of this paper is to evaluate analytically such loss of efficiency, in a specific scenario. In other words, we estimate how much resources we can waste if we go stateless and avoid state maintenance functions. The comparison between stateless and stateful approaches is performed under the constraint of strictly offering the same performance levels, in terms of, e.g., loss probability and delay.

We define a business model for a future Internet able to provide customers with a wide variety of application services, which typically need performance guarantees. The reference environment includes a number of entities: customers, network and application service providers, resource and information brokers, and a clearing house. Assuming usage-based tariffs to charge for IP guaranteed services, we propose inter-domain routing algorithms for determining the path along autonomous domains, with the aim of maximizing users' benefit, in terms of QoS perceived and price to be paid.

In the present day Internet, the provision of different levels of performance guarantees, appropriate for the type of supported applications and the willingness of users to pay, is a pressing need. Since flat-rate fees are probably inadequate to charge differentiated services, the need to provide usage-based tariff models to charge IP guaranteed services, and therefore to develop business interactions, is expected. We define the network service commodity, traded in a future Internet scenario, as the transfer of information units between two points of the network, and we present an approach to measure it as a function of so called virtual delay. In our approach, the virtual delay summarizes the parameters characterizing the performance guarantees of the transfer service. Under these assumptions, we propose a pricing law to charge improved IP services depending on the duration of the connection and on the traffic volume exchanged during the communication. The tariff is a function of the actually used and/or reserved network resources, so as to satisfy the requirements of users and network management, respectively.

The assured forwarding per hop behavior (AF PHB) has been devised by the IETF differentiated services (DiffServ) working group to provide drop level differentiation. This paper shows, quite surprisingly, that a standard AF PHB class is semantically capable of supporting per flow admission control. This is obtained by adopting the AF PHB as the core router forwarding mechanism in conjunction with an end point admission control mechanism running at the network edges. Our work appears to prove that the extent of the AF PHB can be larger than was originally expected, since it can support an admission control mechanism, which is deemed necessary in recent RFCs to offer QoS aware services. An interesting related issue is whether the expedited forwarding PHB, devised to support flows with strict delay requirements, and based on a single DSCP tag, should be redesigned as a PHB class to comply with end point admission control. Preliminary performance results show that, under suitable assumptions, our scheme can provide as much as hard QOS guarantees (see Bianchi, G. et al., IEEE Globecom, 2001; http://drake.diei.unipg.it/netweb/GRIP/sub -/tech/sub -/rep.pdf).

Distributed admission control solutions share the idea that no coordination among network routers (ie, explicit signaling) is necessary, when the decision whether to admit or reject a new offered flow is pushed to the edge of the IP network. Proposed solutions differ in the degree of complexity required in internal network routers, and result in a different robustness and effectiveness in controlling the accepted traffic. This paper builds on a recently proposed distributed admission control solution, called GRIP (Gauge and Gate Reservation with Independent Probing), designed to integrate the flexibility and scalability advantages of a fully distributed operation with the performance effectiveness of admission control mechanisms based on traffic measurements. We show that, in the assumption that traffic sources are dual-leaky-bucket shaped, GRIP allows providing deterministic performance guarantees. Tight QoS performances are made possible even in impulsive load conditions (i.e., sudden activation of several flows), thanks to the introduction of a "stack" mechanism in each network node. A thorough performance evaluation of the conservative effects of the stack shows that the throughput reduction brought about by this mechanism is tolerable, and limited to about 15%.

In this paper, we propose a solution to provide deterministic quality of service (QoS) guarantees in Differentiated Services (DiffServ) framework. We define packet classification functions and traffic conditioning functions, with particular emphasis on shaping functions at edge routers. Our proposal is based on aggregated classification states, within core routers. The original aspect of our contribution is the introduction of a connection admission control (CAC) procedure on the selected paths, in order to provide performance guarantees to traffic aggregates. Another key aspect is the low involvement of core routers in traffic management, obtained by pushing complexity and processing load towards the boundary of the DiffServ domain. We analyze which guarantees can be provided to flow aggregates in these conditions. To this end, we make use of network calculus for deriving bounds on delay and losses within each network node.

Current satellite systems operate according to circuit switching transfer modes. To improve flexibility and efficiency, several kinds of packet switching systems have been proposed. However, it appears that full packet switches are still too complex and expensive to be implemented on board the satellites, in the near future. For the time being, dynamic bandwidth allocation capabilities (DBAC) provide a compromise solution when satellite systems are based on classical circuit switches, since the DBAC payload allows changing dynamically the capacity of each connection, without tear-down and set-up. We consider a DBAC satellite system and define algorithms to allocate the bandwidth so as to provide deterministic QoS guarantees. Standard dual leaky buckets (DLBs) regulate the traffic sources. We define bandwidth-handling policies, design connection admission control rules and evaluate the system performance analytically. Results show a significant increase in the bandwidth utilization of our system, when compared with a plain circuit switching solution.

Current satellite systems operate according to circuit switching transfer modes. To improve flexibility and efficiency, several kinds of packet switching systems have been proposed. However, it appears that full packet switches are still considered complex and expensive, when implemented on board the satellites. For the time being, a compromise has been found in satellite networks with dynamic bandwidth allocation capabilities (DBAC). Such systems are based on classical circuit switches, but the DBAC payload allows the changing dynamically of the capacity of each connection, without the need for tearing-down and setting-up again the connection itself. In this paper we consider a DBAC satellite system and define algorithms to allocate the bandwidth so as to provide deterministic and statistical QoS guarantees. The traffic sources are regulated by standard dual leaky buckets (DLB). We define bandwidth-handling policies, design connection admission control rules and evaluate analytically the system performance. As expected, the numerical results show a significant increase of the overall utilisation factor of our system, when compared with a plain circuit-switching solution.

Current satellite systems operate according to circuit switching transfer modes. To improve flexibility and efficiency, several kinds of packet switching systems have been proposed. However, it appears that full packet switches are still too complex and expensive to be implemented on board the satellites in the near future. For the time being, dynamic bandwidth allocation capabilities (DBAC) provide a compromise solution when satellite systems are based on classical circuit switches, since the DBAC payload allows changing dynamically the capacity of each connection, without teardown and setup. We consider a DBAC satellite system, and define algorithms to allocate the bandwidth so as to provide deterministic and statistical QoS guarantees. Standard dual leaky buckets (DLBs) regulate the traffic sources. We define bandwidth-handling policies, design connection admission control rules, and evaluate the system performance analytically. Results show a significant increase in bandwidth utilization of our system, when compared to a plain circuit switching solution.

We describe an ATM system architecture for satellite communications. The proposed architecture includes on-board switching, and supports the ATM traffic categories defined in previous specifications. In this framework, a critical issue is the control of congestion phenomena. In particular, the application of feedback-based control strategies to a satellite network is critical due to the peculiarities of such an environment: very large propagation delay, expensive transfer capacity, and limited on-board processing capability. The available bit rate (ABR) is the ATM service category handled according to a reactive congestion control (RCC). The focus of this paper is the definition of an RCC that is fully compatible with the standard ABR protocols, and that takes into account the constraints of the satellite environment. We also derive an analytical model that allows us to evaluate the performance of the proposed scheme and to dimension the system. The analytical model is validated with simulations.

Among the various video coding techniques, the H.263 algorithm will play a fundamental role for low bit rate services. In this paper we first present the basic concepts of the H.263 video coding procedure and we describe the main characteristics of H.263 Test Model TMN5. In the second part of the paper we propose a simple renewal approach for the performance evaluation of a queueing system loaded with such bit streams. Our model considers the particular structure of H.263 streams and gives accurate results in a well-defined set of the queueing system parameters. The proposed model is validated with simulation results produced using experimental videophone sequences (H.263 test sequences). A performance evaluation study is also presented.

This paper deals with a novel strategy for connection admission control in ATM networks. This strategy, called worst deterministic pattern allocation (WDPA), aims at reaping statistical gain of ATM, through a simple and controllable bandwidth allocation. WDPA is based on the concept of preventively constraining information sources to emit their cells according to a superimposed deterministic mask. Resource allocation is performed taking into account only the parameters of the declared deterministic mask. The same allocation rule is utilized in every network section. Application of WDPA to input and output queueing switches is also discussed and its compatibility with both these architectures is demonstrated. A performance study is presented. A comparison of the efficiency resulting from WDPA application with that relevant to peak allocation and pure statistical allocation is presented. Results show that, by fixing a maximum transit delay, WDPA outperforms statistical allocation if path length exceed few hops.>

An exact analysis of the MMPP/G/1/K queuing model is carried out, yielding the whole buffer occupancy probability distribution and in particular the loss probability. The solution is obtained by an algorithmic procedure and requires only algebraic manipulations. An expression for the Laplace-Stieltjes transform of the virtual waiting time distribution is derived.>

The stochastic fluid flow approach is applied to the analysis of the cell loss performance of an ATM multiplexer. The input traffic stream offered to the multiplexer is the superposition of heterogeneous on-off sources with independent and exponentially distributed on and off times. The focus is on the numerical investigation of the steady-state behavior of models involving very large state spaces. To this end, an efficient algorithm for the evaluation of tight upper and lower bounds of the cell loss probability is developed. The algorithm allows a significant reduction of the computational burden, while yielding a guaranteed overestimate of the error implied by the proposed approximation of the cell loss probability. Numerical results are presented both to assess the tightness of the proposed bounds and to gain insight into the behavior of heterogeneous traffic mixes. The main conclusion, from the multiplexer performance evaluation point of view, is that it is not convenient to mix very different traffic streams in a completely shared FIFO buffer, without some kind of control.>

The authors study the loss performance of an asynchronous transfer mode (ATM) multiplexer, whose input consists of the superposition of a multiplicity of heterogeneous on-off sources. The stochastic fluid flow approach yields the entire buffer occupancy probability distribution in terms of the solution of a linear differential equation system. The focus is on the numerical investigation of the steady-state behavior of models involving very large state spaces. A novel approximate technique is introduced allowing a dramatic reduction of the computational burden, so that the analysis of a large variety of traffic mixes becomes feasible. Many numerical results are presented to support the accuracy of the method. Applications of the analysis method are outlined.>

The loss performance is studied of an ATM (asynchronous transfer mode) multiplexer, whose input consists of the superposition of a multiplicity of homogeneous ON-OFF sources. The approach is based on the approximation of the actual input process by means of a suitably chosen two-state Markov modulated Poisson process (MMPP), as a simple and effective choice for the representation of the superimposed arrival streams. To accurately evaluate the cell loss performance, a matching procedure was developed which allows the derivation of a tight upper bound on the cell loss probability and the characterization of its asymptotic behavior as the buffer size increases. Moreover, this procedure yields insight into the multiplexer performance behavior, as the buffer size and the source parameters are varied, and clarifies the modeling limits implied by the choice of an approximating MMPP with only two states.>

The performance of an asynchronous transfer mode (ATM) multiplexer whose input consists of the superposition of a multiplicity of homogeneous on-off sources modeled by a two-state Markovian process is studied. The approach is based on the approximation of the actual input process by means of a suitably chosen two-state Markov modulated Poisson process (MMPP), as a simple and effective choice for the representation of superposition arrival streams. To evaluate the cell loss performance, a new matching procedure that leads to accurate results compared to simulation is developed. The application limits of the proposed method are also discussed. The outstanding physical meaning of this procedure permits a deep insight into the multiplexer performance behavior as the source parameters and the multiplexer buffer size are varied.>