We investigate the performance of a Multi-packet reception Adaptive Slotted Access (MASA) scheme in the framework of massive machine-type communication, specifically targeted to update message collection. The need to support timely updates from a large number of uncoordinated devices motivates the analysis of Age of Information (AoI) and energy efficiency. We consider a system model incorporating Successive Interference Cancellation (SIC) and an adaptive access strategy, where users adjust access parameters to attain high spectral efficiency under varying contention levels. Specifically, we propose to adapt the target Signal-to-Noise-plus-Interference Ratio (SNIR), besides the transmission probability. An analytical model accounting for a variable number of active users is introduced and validated through simulations. We further compare the proposed MASA scheme with a baseline (no SIC) stabilized Slotted ALOHA and with the state of the art AIRA-SIC approach. We show that jointly adapting the transmission probability and the target SNIR yields significant performance gains.

The surge of interest in massive Internet of Things (IoT) is renewing the investigation for effective scalable multiple access techniques. Limitation of the cost of coordination and contention among a large number of nodes calls for grant-free approaches, exploiting physical layer techniques to solve collisions. Successive Interference Cancellation (SIC) is a key building block of multi-packet reception for massive IoT. In this paper we explore the large scale performance of SIC as the number of simultaneous overlapping transmission grows. A general model of a SIC receiver is stated for a shared channel withntransmitters. The asymptotic number of correctly decoded packets and spectral efficiency performance are characterized exactly asn→ ∞, for a suitably scaled target Signal to Noise plus Interference Ratio (SNIR). The asymptotic analysis is carried out for any probability distribution of the channel gain, assuming that the average received power level is same for all nodes through power control. The impact of this asymptotic result is discussed as well, highlighting the impact of variability of channel gain on spectral efficiency performance.

Internet of Things (IoT) is stirring a surge of interest in effective methods for sharing communication channels, with nodes transmitting sporadic, short messages. These messages are often related to control systems that collect sensor data to drive process actuation, such as in industries, autonomous vehicles, and environmental control. Traditional approaches that dominate wireless and cellular communications prove most effective when dealing with a limited number of concurrently active nodes, sending relatively large volumes of data. We address a different scenario where numerous nodes generate and transmit short messages according to non-periodic schedules. In such cases, random multiple access becomes the typical approach for sharing the communication channel. We propose a general modeling framework that enables the investigation of the impact of Successive Interference Cancellation (SIC) on two of the main random access paradigms, namely Slotted ALOHA (SA) and Carrier-Sense Multiple Access (CSMA). The key varying parameter is the target Signal to Interference plus Noise Ratio (SINR) at the receiver, directly tied to the spectral efficiency of the adopted coding and modulation scheme. Two different regimes are highlighted that bring the system to work at relative maxima of the sum-rate. We further investigate the impact of different transmission power settings and imperfect interference cancellation. Leveraging on the insight gained in the saturated node scenario, an adaptive algorithm is defined for the dynamic case, where the number of backlogged nodes varies over time. The numerical results provide evidence of a significant potential for grant-free multiple access, calling for practical algorithms to translate this promise into feasible realizations.

A large dataset of workload measurements has been released by Google. The wealth of disclosed data allows a deep dive into real workload patterns. With the aim of providing tools to generate realistic workloads in a simple way, we have extracted from Google’s dataset job arrival times, number of tasks per job, required computation time, and memory of tasks. We define a statistical fitting of the relevant probability distribution, providing a simple tool to build artificial workload traces that mimic real traffic as represented by Google measurements. The workload generation algorithm is assessed by comparison of its mean response time on a test dispatching/scheduling system against the real traffic traces. In spite of being only a first-order generation model, it is shown that the proposed artificial generation can reproduce faithfully the performance of real workload in the case of large server clusters.

Efficient data collection from a multitude of Internet of Things (IoT) devices is crucial for various applications, yet existing solutions often struggle with minimizing access delay and Age of Information (AoI), especially when managing multiple simultaneous transmissions and access strategies. This challenge becomes increasingly critical as IoT deployments continue to expand, demanding robust mechanisms for handling diverse traffic scenarios. In this study, we propose a novel approach leveraging Successive Interference Cancellation (SIC) based on adaptive and fixed parameters schemes to address these limitations. By analyzing both throughput and AoI along with access delay, we demonstrate the effectiveness of our adaptive approach compared to the fixed approach, particularly in scenarios featuring heavy and light traffic. Our findings highlight the pivotal role of adaptive approaches in optimizing data collection processes in IoT ecosystems, with a particular focus on minimizing access delay, AoI, and improving spectral efficiency.

The current Cellular V2X (C-V2X) standard for direct communication between vehicles is based on the so called Semi-Persistent Scheduling (SPS) algorithm. It is based on the multiple access structure of 5G New Radio (NR) and exploits sensing and persistence to realize a randomized multiple access. We consider the application of SPS to support periodic broadcast traffic where no acknowledgments are provided. Persistence is a key feature, which consists of a node using the same resource for its transmissions for multiple frames. The specific resource used is randomly selected from among those that are perceived to be idle, and reselected anew after a randomized number of frames. We define a model to gain insight into the interplay between persistence and key performance metrics for the type of traffic considered, namely probability of successful delivery and Age of Information (AoI). A core version of the model is validated against simulations and used to show that there exists an optimal level that minimizes AoI. The model is then extended to account for a distance-dependent propagation model, allowing further insight into the effects of the interplay between sender-receiver distance and persistence. Finally, an even more detailed model is investigated, using ns-3-based simulations. This further analysis confirms the qualitative behaviors revealed by the analytical model and provides more insight into the complex interactions of system parameters and channel characteristics. The obtained results help to identify the limits of SPS, opening the way to system-principled parameter optimization and design of more powerful variants of the multiple access scheme.

We consider a generalized version of Carrier-Sense Multiple Access (CSMA), where the contention window size is a constant and the back-off probability distribution can be varied. We address the optimization of a weighted throughput metric, identifying the optimal back-off Probability Density Function (PDF). We give a simple fixed-point algorithm to compute the optimal PDF and prove that the solution is unique. The weighted throughput definition caters for aspects other than the mere channel utilization. It reduces to plain utilization (normalized throughput) when all weights are equal to 1. We also reconnect our result to the classic analysis of saturated non-persistent CSMA, as introduced in the seminal paper by Tobagi and Kleinrock, proving formally that the modeling assumptions of that work, that lead to a Geometric PDF of back-off, actually correspond to the throughput-optimal choice, provided that the ratio of the Geometric PDF is suitably chosen.

The stability region of non-persistent CSMA is analyzed in a general heterogeneous network, where stations have different mean packet arrival rates, packet transmission times probability distributions and transmission probabilities. The considered model of CSMA captures the behavior of the well known CSMA/CA, at least as far as stability and throughput evaluation are concerned. The analysis is done both with and without collision detection. Given the characterization of the stability region, throughput-optimal transmission probabilities are identified under airtime fairness, establishing asymptotic upper and lower bounds of the maximum achievable stable throughput. The bounds turn out to be insensitive to the probability distribution of packet transmission times. Numerical results highlight that the obtained bounds are tight not only asymptotically, but also for essentially all values of the number of stations. The insight gained leads to the definition of a distributed adaptive algorithm to adjust the transmission probabilities of stations so as to attain the maximum stable throughput.

Medium Access Control (MAC) main functions encompass contention for channel access, packet scheduling, error control, and data integrity. Channel contention is a collective function involving all stations in the network, while data integrity pertains to data flows of each individual station. We propose a design where contention related functions are separated from other data management functions. The hinge connecting contention and other data management functions is a flow control algorithm, aiming at guaranteeing stability of contention queues and load on the MAC channel. With reference to Carrier-Sense Multiple Access (CSMA), we define an analytical model of contention queues under non saturated traffic. An asymptotic analysis of the model for large number of stations yields a closed form of the optimal flow control rate. The insight gained from the model is used to design an adaptive flow control algorithm that guarantees throughput optimality for all values of the number of stations.

The interplay of physical layer enhancement due to Successive Interference Cancellation (SIC) and classic random access protocols used in most multi-access communication channels is the object of this paper. Considering the classic representatives of random access protocols, Slotted ALOHA and Channel Sensing Multiple Access (CSMA), we show that two operation regimes can be identified as a function of the communication link spectral efficiency. In case of high levels of spectral efficiency, multi packet reception as granted by SIC is of limited usefulness. The obtained overall sum-rate performance are dominated by the MAC protocol algorithm. On the contrary, for low spectral efficiency levels, sum-rate performance are essentially dependent on physical layer SIC capability, while the MAC protocol is of marginal. Limitations due to limited transmission power level dynamic range are shown to induce unfairness among nodes, i.e., nodes closer to the sink achieve better sum-rate and success probability performance. However, the unfairness issues fades away when the system is driven to work around the sum-rate peak achieved for low spectral efficiency communications. This is a major finding of this work: apparently, SIC can boost performance while still maintaining a fair sharing of the achieved sum-rate performance among nodes, irrespective of the quality of their respective communication channels.

Recent advances on WLAN technology have been focused mostly on boosting network capacity by means of a more efficient and flexible physical layer. A new concept is required at MAC level to exploit fully the new capabilities of the PHY layer. In this article, we propose a contention mechanism based on Repeated Contentions (ReCo) in frequency domain. It provides a simple-to-configure, robust and short-term fair algorithm for the random contention component of the MAC protocol. The throughput efficiency of ReCo is not sensitive to the number of contending stations, so that ReCo does not require adaptive tuning of the access parameters for performance optimization. Efficiency and robustness is gained through the power of repeated contention rounds. We also apply the ReCo concept to the emerging IEEE 802.11ax standard, showing how it can boost performance of random access with respect to the current version of IEEE 802.11ax OFDMA Back-Off (OBO). Our proposal is supported by an experimental test-bed that realizes ReCo by means of simultaneous transmission and reception of short tones, which is feasible on top of programmable OFDM PHY layers.

In this work we define a framework for the assessment of the Traffic Matrix (TM) of an Internet Service Provider (ISP) network. The solution, referred to as mrT, i) is completely distributed among network nodes, ii) is based on different Map-Reduce building blocks, iii) and is able to extract the Ingress-Egress nodes traffic relationships starting from raw traces captured on node interfaces. Each network node is able to compute a row of the TM having as input the local trace and low size files sent by other network nodes. mrT can be used for the TM computation of any packet switched network where the source and destination identifiers are unique and global, and computational resources are available in proximity of network nodes. The performance evaluation, carried out with synthetic and real traffic traces, highlight that mrT is a suitable solution for the assessment of the TM in real ISP networks with high traffic volumes. Moreover, the use of the Map-Reduce paradigm allows a reduction of more than the 50% of the execution times, with respect to an SQL-based approach.

Traffic matrix (TM) assessment is a key issue for optimizing network management costs and quality of service. This paper presents a method to measure the intensity of ingress-egress traffic flows on an Internet service providers network that overcomes the limits of the classical measurement-based approaches. The proposed algorithm, called segment routing perturbation traffic (SERPENT), uses a routing perturbation approach enabled by the segment routing paradigm: The paths of a subset of flows are changed so that their intensities can be determined measuring the variation of the load of the network links. The TM is measured in successive steps, called snapshots, in which sets of flows are progressively re-routed and measured, under a maximum link utilization constraint. We state an integer linear programming (ILP) optimization problem to determine the flows to be rerouted in one snapshot. SERPENT is an heuristic offering an efficient solution to the stated ILP. Results show that SERPENT assesses the intensity of more than 80% of flows even when the network is highly stressed, while reducing the configuration cost with respect to classical approaches. Moreover, when used in conjunction with an estimation algorithm, SERPENT allows a reduction of the estimation error by more than 50% with fewer than 5 snapshots.

In this paper we propose ReCo, a robust contention scheme for emerging wireless technologies, whose efficiency is not sensitive to the number of contending stations and to the settings of the contention parameters (such as the contention windows and retry limits). The idea is iterating a basic contention mechanism, devised to select a sub-set of stations among the contending ones, in consecutive elimination rounds, before performing a transmission attempt. Elimination rounds can be performed in the time or frequency domain, with different overheads, according to the physical capabilities of the nodes. Closed analytical formulas are given to dimension the number of contention rounds in order to achieve an arbitrary low collision probability. Simulation results and a real implementation for the time-domain solution demonstrate the effectiveness and robustness of this approach in comparison to IEEE 802.11 DCF.

Traffic matrix estimation in communication network is a long standing problem for its intrinsic difficulty and potential benefit to a vast number of network optimization and management functions. We address the improvement of the traffic matrix estimation by means of selected traffic flow measurements, besides the easily obtained link load measurements. The key contribution of this paper is the definition and assessment of an effective criterion, based on the flow spread parameter, to identify the flows to be measured that reduce the estimation error most. It turns out that a small percentage of flows are enough to drive the estimation error an order of magnitude lower than the one obtained with the classical solution solely based on link load measurements. Our algorithm, referred to as flow spread-based algorithm (FSBA), is also able to distribute measurement tasks fairly among network nodes, taking into account the available forwarding tables space. We also show that FSBA outperforms the state-of-the-art similar approaches. A detailed discussion on how the observation of the desired flows can be performed is done as well, by addressing the SDN paradigm. This is in fact a rapidly growing concept that enables individual flow measurements, though for only a limited number of flows to be practical with current technology.

Implementations of data dissemination protocols in Vehicular Ad-Hoc Networks (VANETs) typically use multi-hopping approaches at the network layer, employing timers whose expiration times determine whether a vehicle should relay a message. These protocols, when they interact with lower access layers, such as the IEEE 802.11p MAC, are often affected by phenomena that impair the performance of the dissemination process. In this paper, we highlight the spurious forwarding phenomenon that arises when using timer-based protocols. We demonstrate the degradation incurred in the packet dissemination performance when the VANET is loaded by a high rate of packet flows. We then propose a probabilistic decimation approach, demonstrate its ability to alleviate the spurious forwarding problem, and discuss its performance as a function of vehicular traffic density and packet flow rate. Comparisons with other probabilistic dissemination protocols highlight the significant performance improvement attained by using our approach, assuring the realization of high throughput rate and packet delivery ratio.

Privacy in communications calls primarily for information flow encryption. Packet traffic flows privacy breaches have been widely demonstrated in point-to-point communications due to information leakage from observable traffic features, like packet length, timestamp, direction. We address a point-to multipoint system, namely a Content Delivery Network, where user clients maintain and use connections with a number of servers. Specifically, we address Google search services: they are conveyed by TLS connections, by using https, either from within user accounts or even without logging as a Google services user. Https is provided to protect communications privacy. Yet, we show that by collecting the encrypted traffic and extracting simple features related to traffic activity and possibly the amount of data sent by servers to clients, effective classifiers of user activity can be realized. Specifically, we are able to distinguish which type of search a user is carrying out, among a given set of alternatives (text, images, maps, video, video on YouTube, news) with average success rates that can exceed 90%.

Traffic flow classification to identify applications and activity of users is widely studied both to understand privacy threats and to support network functions such as usage policies and QoS. For those needs, real time classification is required and classifier's complexity is as important as accuracy, especially given the increasing link speeds also in the access section of the network. We propose the application of a highly efficient classification system, specifically Min-Max neurofuzzy networks trained by PARC algorithm, showing that it achieves very high accuracy, in line with the best performing algorithms onWeka, by considering two traffic data sets collected in different epochs and places. It turns out that required classification model complexity is much lower with Min-Max networks with respect to SVM models, enabling the implementation of effective classification algorithms in real time on inexpensive platforms.

A key paradigm enabled by Vehicular Ad hoc NET- works is the support of location aware push-mode info-tainment services, besides the safety services that motivate the deployment of the technology in the first place. Message and advertisements are spread via multi-hop communications from the originating Road Side Units connected to the Internet. Support of such services relies on robust and efficient dissemination protocols. We define a timer-based vehicular backbone network protocol, where each vehicle can take forwarding decisions only based on the information read in the message header, its current state and local measurements. We analyze its performance taking into account the effect of the IEEE 802.11p MAC layer. A comparison with other literature dissemination protocols is carried out in a highway setting.

Inter-vehicle communications will play an important role in future cars and traffic management in general. Many different services have been proposed in the literature using vehicle-to-vehicle (V2V) and vehicle-to-infrastructure (V2I) communications. These services include safety applications like collision warning, up-to-date traffic information, active navigation, and also infotainment. Efficient data dissemination in vehicular networks (VANET) is of particular interest for both safety and infotainment services. In case of infotainment services, which are the object of our study, data dissemination is used to extend the radio coverage area of a Road Side Unit (RSU) to reach all users that can be interested in a given service. In this work, we propose a family of algorithms to extend the RSU coverage area in VANETs. These algorithms apply simple geometrical rules based on the position of the sending nodes. The algorithms are used to allow the data to cross road junctions and to propagate towards multiple directions, without using beacon or hello messages. By applying a simple geometrical analysis we show that the proposed solutions are able to increase the number of nodes reached by a broadcast message. Moreover, we study the performance of our solution through message flows injected in a dynamic scenario, where a Manhattan street grid is reproduced and populated through SUMO, which is able to provide realistic vehicle flows. The effect of traffic lights on vehicle flows in relation to network performance is also discussed.

Traffic flow features like packet lengths, direction, gap times have been shown to carry significant information on conveyed the traffic flows they belong to, e.g. enabling application classification with high accuracy and even privacy breaking, even if encryption is used. Such a leakage of user related information can be stopped by modifying the traffic flow features, e.g. for packet lengths by padding, fragmenting or inserting dummy packets. We outline a general approach aiming at full masking of an application layer traffic flow; then, we address the trade-off between information leakage and overhead and we define a practical algorithm to achieve partial traffic masking. Experiments are carried out with traffic, captured on real networks. It turns out that overhead can be substantially reduced if requirements on information leakage are not too strict.

Secure SHell is a TCP based protocol designed to enhance with security features telnet and other insecure remote management tools. Due to its versatility, it is often exploited to forward applications (i.e. HTTP, SCP, etc.) into encoded TCP traffic flows. The point which makes challenging the identification of the uses of SSH is that packets are enciphered and instruments based on deep packet inspection (DPI) cannot achieve this task. We approached the problem of early SSH classification with k-means based machine by studying statistical behavior of IP traffic parameters, such as length, arrival time and direction of packets. In this paper we describe tools and networks designed to collect SSH remote administration traffic as well as relevant results obtained for its classification. In particular, our tool identifies remote management traffic out of other SSH encoded appli cations with accuracy up to 90.34.

The interaction between wireless optimized scheduling algorithms and TCP congestion control mechanisms can have adverse effects on the performance of the system. We focus on the queue based max-weight (QBMW) scheduler, a scheduling strategy which is known to be throughput-optimal under unregulated traffic sources. We use fluid modeling to describe the time evolution of the congestion window size and of the wireless buffer, and show by numerical results that under TCP traffic sources the QBMW scheduling policy leads to a very unfair outcome, in which some users may be completely shut off. We also evaluate and discuss the performance achieved by other scheduling policies: the proportional fair (PF) scheduler, and the queue age (QA) scheduler, which takes account of the age of the packets stored in the wireless buffers.

One of the most critical aspects of security problems is the impossibility of accurately checking a system real weaknesses. In a complex and distributed environment this problem is greatly accentuated. During the process of configuration and implementation of the network security policies errors can occur, resulting in holes in security and, consequently, compromising the entire system functionality. These errors are often very hard to detect by performing a manual or visual inspection. For this reason, automatic management of this phase is required. Here we propose an algorithm to automatically tune up the configurations of the network devices in order to avoid unexpected and unwanted network behaviours. This algorithm will be described in all its phases and some results of the software implementation will be shown.

Firewalls and Security Gateways are core elements in network security infrastructure. As networks and services become more complex, managing access-list rules becomes an error-prone task. Conflicts in a policy can cause holes in security, and can often be hard to find while performing only visual or manual inspection. First, we have defined a methodology to systematically classify the severity of rule conflicts; secondly, we have proposed two different solutions to automatically resolve conflicts in a firewall. For one of them we found an algebraic proof of the existence of the solution and the convergence of the algorithm, and then we have made a software implementation to test it.

A single TCP connection with a time varying capacity bottleneck link is considered. The bottleneck capacity varies arbitrarily and independently of the TCP connection traffic; it aims at reproducing typical situations where capacity is modulated by exogenous process, such as wireless channel with link adaptation or bursty channels. The key point is to understand the interplay of the TCP congestion control evolution and the time constants of the bottleneck link capacity time variation. Fluid modeling is used to describe the time evolution of the congestion window size and of the bottleneck buffer content with a completely general capacity time function, ns-2 based simulations are used as mean to assess the fluid model accuracy. Numerical results show the existence of a "resonance" phenomena, that can significantly degrade the TCP long term performance (even more than halved). The degradation depends on the ratio between the fundamental time constant of the link capacity variation and the TCP average round trip time.

Given a set of communication links sharing a channel, delay for delivery of a given amount of data backlogged in link transmitters can be traded off against average power consumption, for a fixed transmission QoS requirement. The main result of this work is the identification of an asymptotically sharp lower bound for the average power needed to carry a given backlog within a delay constraint. The power-delay tradeoff allows to compare a non orthogonal CDMA capacity sharing with an orthogonal TDMA. Numerical examples for a collection of links belonging to an ad-hoc network are discussed.

Packet scheduling and radio resource allocation in an OFDMA system pursue conflicting goals: the latter aims at achieving short-term efficiency, while scheduling aims at guaranteeing fairness among flows in the long term. We propose a scheduler-allocator architecture that successfully manages to integrate both goals in a loose cross-layer strategy. On each time frame the scheduling module selects a list of packets eligible for transmission, with the goal of achieving long-term fairness; the list is then passed to the radio resource allocator, which finds the best allocation given the current channel state. We evaluate the proposed scheme in a single cell scenario. Results point out that our scheme is able to guarantee throughput fairness among flows, while achieving an efficient allocation of radio resources.

In recent years, issues regarding the behavior of TCP in high-speed and long-distance networks have been extensively addressed in the networking research community, both because TCP is the most widespread transport protocol in the current Internet and because bandwidth-delay product continues to grow. The well known problem of TCP in high bandwidthdelay product networks is that the TCP Additive Increase Multiplicative Decrease AIMD probing mechanism is too slow in adapting the sending rate to the end-to-end available bandwidth. To overcome this problem, many modifications have been proposed such as FAST TCP [1], STCP [2], HSTCP [3], HTCP [4], BIC TCP [5] and CUBIC TCP [6]. The goal of this work is to investigate, by using both analytical models and simulation results, optimal sizing of the retransmission and out-of-order TCP buffers in case of modified TCP congestion control settings and to highlight differences between NewReno TCP and SACK TCP packet loss recovery schemes in terms of TCP internal buffers requirements. An important result is that the SACK option turns out to be particularly effective in reducing buffer requirements in the case of very high bandwidthdelay product links.

In this paper, we investigate interactions between TCP and wireless hybrid FEC/ARQ schemes. The aim is to understand what is the best configuration of the wireless link protocol in order to guarantee TCP performance and channel efficiency. Interactions between TCP and different link layer mechanisms are evaluated by means of an analytic model that reproduces: 1) a Rayleigh fading channel with FEC coding, 2) a generic selective repeat ARQ Protocol, and 3) the TCP behavior in a wired-cum-wireless network scenario. The analytic model is validated-by means of ns-based simulations. The analysis represents a contribution to the optimal design of link layer parameters of wireless networks crossed by TCP/IP traffic. The main findings can be summarized as follows: 1) fully reliable ARQ protocols are the best choice for both TCP performance and wireless link efficiency and 2) optimal values of FEC redundancy degree from the point of view of energy efficiency maximizes TCP performance as well.

We analyse a CSMA MAC protocol for ad hoc wireless networks, that uses one control channel and a number of data channels. The data channel employed in each transmission is dynamically selected with an exchange of frames on the control channel. We present simulation results obtained with both the multichannel protocol and the IEEE 802.11 MAC protocol under different scenarios. We show that the multichannel protocol performs better than the single channel MAC under certain conditions, since the presence of hidden nodes has a smaller impact on its performance; we argue that this is mainly due to the separation operated between the control frames and the data frames.

In this paper we investigate the interaction between TCP and wireless ARQ mechanisms. The aim is to understand what is the best reliability degree of the wireless link in order to guarantee TCP performance. For this purpose, we first develop a Markov model for a selective repeat ARQ protocol, widely used in the current wireless environments. Secondly, we design a cross-layer algorithm that, by exploiting the proposed model, can adapt the number of link layer transmission attempts to the end-to-end packet loss rate perceived by TCP. The interaction between TCP and link layer is evaluated in a specific case study (TCP over 3G radio access) by means of simulations carried out by using a very detailed UMTS-TDD simulator based on us. The deployment of the link layer Markov model and of the proposed algorithm allows us to derive some interesting conclusions about the design of retransmission protocols in TCP/IP network environments.

Wireless data is deemed as a major booster of next generation wireless networks. In this context, the support of Internet applications based on the TCP still presents several open issues. The aim of this work is to evaluate the performance of TCP data transfer over the UMTS TD-CDMA air interface, by means of a rather detailed simulation model. The evaluation includes both bulk FTP like data transfer and interactive client-server (WWW) traffic. The distinctive point of view is exploring the interaction between application and transport level protocol functions and lower layers protocols over the radio interface. The interplay of radio access and fixed core network impairments is analyzed as well. Major results deal with balancing the error recovery performance of the RLC against the radio channel correlation and the fixed network impairments.

The analysis and design of TCP/IP networks is a key component in the development of the Internet. We consider how simple analytical modelling of the key network elements can yield sufficiently accurate performance results by means of simulations with actual TCP sources. The defined modelling approach, named the fixed point model, is exploited to state a dimensioning problem of an IP network dominated by TCP traffic. The procedure for the TCP/IP network capacity dimensioning is based on the definition of a user satisfaction index, suitable for a best-effort scenario.

Ultra-wideband (UWB) radio is becoming a promising field for new generation's digital communication systems. This technique, based mainly on the impulse radio paradigm, offers great flexibility and shows enormous potential in view of a future broadband wireless access. We present the main principles to design a multiaccess scheme based on UWB. The potential of UWB is exploited within a distributed ad hoc wireless system, where we describe the principles for the definition of a medium-access control (MAC) for mobile computing applications and we analyze the main performance results derived from simulations. A general framework for radio resource sharing is outlined for classes of traffic requiring both elastic-dynamic and guaranteed-reserved bandwidth. Then, we discuss the issue of supporting the proposed radio resource sharing scheme by means of a distributed MAC protocol.

Wireless data and multimedia are rapidly becoming a dominant issue of radio communications. In this context, the support of Internet applications based on the TCP presents several open issues due to the specific nature of the wireless network segment. The aim of this work is to evaluate by a rather detailed simulation model the performance of TCP data transfer over the TD-CDMA scheme of the UMTS TDD radio interface. The distinctive point of view is exploring the limits of a relatively, simple approach, by adopting the MAC/RLC connection oriented mode of operation for bulk, loss sensitive data. Features and parameters values that optimize TCP data transfer are discussed and their benefit quantified.

The wireless approach to the last mile access (wireless local loop, known as WLL) is becoming increasingly attractive to network operators and service providers since it offers a flexible and cost-effective solution to enable delivery of even broadband services to end customers. In this paper, a full-blown broadband WLL network is presented. The proposal is based on the OFDM-CDMA technique, to which an added dynamic reservation/request MAC protocol is proposed, fully exploiting the OFDM-CDMA platform. Central to our proposal is the support of different QoS profiles, in the context of QoS aware networks. As a case study, the explicit presentation of the IETF integrated services support over our WLL system is addressed. An extensive performance evaluation focused on the MAC layer is then reported. We prove that our scheme achieves high utilization efficiency, as well as a fair share of the available radio capacity, even in the presence of highly heterogeneous traffic mix. Delay performance is provided for both reference traffic models, as well as for measured IP and MPEG traffic traces offered to the system.

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.

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.>

An algorithmic model for the performance evaluation of the medium access control (MAC) protocol for a satellite system in an asynchronous transfer mode (ATM) environment is presented. This model allows analysis of general load configurations and yields the mean value of the performance parameters (e.g. queue lengths and access delays). The model is based on a computer-oriented approach, and the characterization of the key model variables is obtained by means of an iterative procedure. A comparison of the model analysis results to those obtained by a simulative approach shows a satisfactory matching.>

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.>

An access protocol suitable for asynchronous transfer mode (ATM) operation in an integrated service broadband metropolitan area network (MAN) is presented. This protocol is designed for a twin-bus optical network and is characterized by a fully distributed control mechanism. A detailed description of the MAC procedure and the results of its performance evaluation are given. In particular, the high fairness degree and the very high throughput values that are the distinctive features of this protocol are highlighted and compared with those relevant to the current version of the IEEE standard 802.6.>