This paper addresses the optimization of the perception message periodicity in vehicular networks by minimizing a weighted sum of transmission and freshness costs. The expressions of the two contributions and the closed-form solution to the problem are obtained for the Dynamic Scheme of the New Radio Vehicle-to-Everything Side Link standard. Moreover, a model rooted in queueing theory is proposed to describe a connected automated vehicle as a source of messages about the detected objects. The model provides the statistical description of the number of objects in the vehicle viewing horizon, wherefrom the expression of its autocorrelation function is determined. Through the latter, the degree of similarity between the content of consecutive messages is understood. Lastly, an illustrative suburban scenario is investigated and realistically implemented in a simulator. The results show that increasing the detection range of the connected automated vehicle is beneficial, allowing higher values of the optimal update periodicity, larger message sizes, and lower correlation levels between consecutive messages. This choice leads to the best communication performance, guaranteeing a higher probability of successful packet delivery.

Accurate positioning of vehicles and vulnerable road users is critical for ensuring road safety. Today’s localization relies on the Global Navigation Satellite System (GNSS), but it is often compromised by signal obstructions due to urban infrastructures, buildings, and other vehicles. To overcome these limitations and enhance GNSS localization accuracy, a cooperative positioning method is put forth that leverages direct vehicular communications and the onboard sensors of the vehicle. In the proposed approach, a real-time algorithm selects the vehicle with the lowest positioning error as the anchor node and mandates the anchor to broadcast its coordinates via a suitable message. The receiving vehicles whose sensing region includes the anchor can determine the distance from it and update their position estimate accordingly. In turn, these vehicles may serve as secondary anchors, extending the correction process to vehicles beyond the first tier. By jointly exploiting direct vehicular communications and the local estimate of the distance from the anchor, the proposed approach achieves accurate vehicle localization. In a reference urban intersection, taking into account imperfect direct vehicle-to-vehicle communications and the first tier only, the percentage of vehicles improving the estimate of their position lies between 36% and 54%, depending on the penetration rate of the connected vehicles; these values raise to 62% and 88% when the secondary anchors contribute to propagate the error correction. Furthermore, the statistical distribution of the positioning error exhibits a significant shift toward lower error values, raising the probability that the error is lower than 6 m from 0.05 when no correction is introduced, to 0.87 when all the vehicles are connected and imperfect communications are considered.

The introduction of vehicular communications has long been advocated as compelling to guarantee the safety and stability of platooning, owing to the information that vehicles exchange through the wireless channel. However, connecting the vehicles through radio links introduces delays and persistent packet losses that may compromise the proper platoon functioning. To investigate this issue, the present work replicates through simulation the vehicle dynamics of a Cooperative Adaptive Cruise Control platoon and the transmission impairments that affect the platoon communications, when the 5G New Radio standard for Vehicle-to-Vehicle communications is adopted. The study first characterizes the actual behavior of the platoon when simultaneously struck by a perturbation in the leader’s acceleration and a severe communication outage. In the worst examined setting, it shows that the radio technology is robust enough to guarantee platoon stability. It further demonstrates that a proper selection of the packet scheduling strategy allows to reduce the time gap between consecutive vehicles while still maintaining platoon stability, at the expense of a moderate increase in energy expenditure.

The Internet of Vehicles (IoV) will interconnect vehicles, vulnerable road users, and infrastructure nodes for a safer, more efficient, and digitalized mobility. In the IoV vision, traditional network-based communications will be complemented with direct Sidelink (SL) Vehicle-to-Everything (V2X) communications. To this aim, 3GPP introduced in Release 16 the New Radio (NR) V2X SL technology. The NR V2X SL standard includes important Physical (PHY) layer novelties with respect to LTE V2X SL and NR Uplink/Downlink, and is characterized by a large set of configurable parameters. However, existing NR V2X SL link level studies focus on a confined set of configurations. This limits a thorough understanding of the NR V2X SL link level performance and impacts the accuracy of system level evaluations, which typically leverage link level models to accurately represent the PHY layer performance. These models are based on Look-Up Tables (LUTs) that map the link level performance, e.g., BLock Error Rate (BLER), as a function of the link quality, e.g., Signal-to-Noise Ratio (SNR). This study presents an exhaustive NR V2X SL link level evaluation that analyzes the impact of relevant PHY layer aspects, e.g., modulation and coding scheme, channel model, and transmitter-receiver relative speed, considering the wide set of configurations recommended by 3GPP and ETSI. The obtained standard-compliant LUTs are openly released, representing the largest NR V2X SL link level dataset available. The released dataset represents a valuable asset for the community, as it allows exhaustive NR V2X SL system level investigations under a broad range of settings and configurations.

5G NR V2X complements LTE V2X to support advanced V2X services for connected and automated driving. 5G NR V2X introduces novel features at the MAC layer that are designed to cope with potential packet collisions, and that could help address the LTE V2X MAC inefficiencies observed under aperiodic traffic of variable size. This is the case of the re-evaluation mechanism that is a mandatory MAC feature of 5G NR V2X, and that seeks to avoid possible packet collisions detected before a vehicle transmits in selected resources. Evaluations conducted to date of 5G NR V2X do not consider the re-evaluation mechanism, and have focused on traffic patterns that do not fully account for the traffic variability of advanced V2X services. This paper extends the current state of the art with the first evaluation of a fully standard compliant 5G NR V2X implementation under the traffic patterns recommended by 3GPP for advanced V2X services. Our study shows that 5G NR V2X mode 2 still faces MAC challenges when using semi-persistent scheduling (SPS) to efficiently support aperiodic traffic of variable size.

The Third Generation Partnership Project (3GPP) recently released the 5G New Radio (NR) Vehicle-to-Everything (V2X) standard. NR-V2X is expected to support a large set of enhanced V2X applications, characterized by stringent requirements in terms of latency, reliability and data rate. The use of simulations plays a crucial role in the preliminary assessment of NR-V2X achievable performance, providing valuable insights on the strengths and limits of the new solution. This work introduces MoReV2X, an open-source ns-3 module for the simulation of NR-V2X communications. The development of MoReV2X adheres to 3GPP specifications, includes various traffic models, features an accurate implementation of the NR propagation channel between vehicles and allows to determine a thorough set of Key Performance Indicators (KPIs). A first set of exemplary results is reported to testify the proper functioning of the module. Moreover, the results reveal the impact that different subcarrier spacings have on system performance, when periodic and aperiodic traffic sources are considered.

This article proposes a new architecture for privacy-preserving data mining based on Multi Party Computation (MPC) and secure sums. While traditional MPC approaches rely on a small number of aggregation peers replacing a centralized trusted entity, the current study puts forth a distributed solution that involves all data sources in the aggregation process, with the help of a single server for storing intermediate results. A large-scale scenario is examined and the possibility that data become inaccessible during the aggregation process is considered, a possibility that traditional schemes often neglect. Here, it is explicitly examined, as it might be provoked by intermittent network connectivity or sudden user departures. For increasing system reliability, data sources are organized in multiple sets, called rings, which independently work on the aggregation process. Two different protocol schemes are proposed and their failure probability, i.e., the probability that the data mining output cannot guarantee the desired level of accuracy, is analytically modeled. The privacy degree, the communication cost and the computational complexity that the schemes exhibit are also characterized. Finally, the new protocols are applied to some specific use cases, demonstrating their feasibility and attractiveness.

Network measurement and telemetry techniques are central to the management of today's computer networks. One popular technique with several applications is the estimation of traffic matrices. Existing traffic matrix inference approaches that use statistical methods, often make assumptions on the structure of the matrix that may be invalid. Data-driven methods, instead, often use detailed information about the network topology that may be unavailable or impractical to collect.

This work performs a comparative analysis of the end-to-end quality guaranteed by Voice over LTE (VoLTE), examining several millions of VoLTE calls that employ two popular speech audio codecs, namely, Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wide Band (AMR-WB). To assess call quality, VQmon®, an enhanced version of the standardized E-Model, is utilized. The study reveals to what extent AMR-WB based calls are more robust against network impairments than their narrowband counterparts; it further shows that the dependence of call quality on the packet loss rate is approximately exponential for both types of codec.

This work puts forth an analytical approach to evaluate the recovery failure probability of power-based Non-Orthogonal Multiple Access (NOMA) on the uplink of a 5G cell, the recovery failure being defined as the unfortunate event where the receiver is unable to decode even one out of the n simultaneously received signals. In the examined scenario, Successive Interference Cancellation (SIC) is considered and an arbitrary number of superimposed signals is present. For the Rayleigh fading case, the recovery failure probability is provided in closed-form, clearly outlining its dependency on the signal-to-noise ratio of the users that are simultaneously transmitting, as well as on their distance from the receiver.

This work investigates the performance of a multi- hop scheme for the dissemination of road safety messages on highway segments, employing the recently standardized LTE-A Cellular Vehicle- to- Everything (C-V2X) technology. In order to guarantee a seamless service in areas where cellular coverage is unavailable, vehicles direcly communicate over the unlicenced ITS 5.9 GHz frequency band, operating in accordance to Mode 4 of the C-V2X standard. The proposed scheme and the examined use-case demonstrate that the delivery of safety messages can successfully take place on a dedicated radio channel, as well as on a shared channel where periodic messages are broadcast at the maximum frequency foreseen by ETSI.

This paper proposes to combine peer-to-peer (P2P) and dynamic adaptive streaming over HyperText Transfer Protocol (HTTP) (DASH), leveraging the scalability and self-organization properties of the former and the capability of the latter to adapt the rate of the delivered video to varying operating conditions. The devised P2P-DASH architecture is a multioverlay P2P platform, where every peer implements a decentralized rate control strategy that steers its transitions from one overlay to another, so as to achieve a good viewing quality and also preserve the good functioning of the entire system. At peer's site, the new architecture displays the following salient features: 1) an augmented pool of neighbors, to provide every node with the knowledge of the streaming process within adjacent overlays; 2) a priority mechanism to speed up the delivery of video chunks to peers that newly enter the system or switch from the current to a new overlay, in order to quickly turn them into active peers; and 3) some alternative rate control algorithms to guarantee, with different degrees, more resources to overlays that deliver higher streaming rates, therefore reducing the misalignment among streaming processes which occur in different overlays, and ultimately improve the viewing quality of the single user. System behavior is extensively investigated through simulation, and some clear design guidelines are provided.

Dynamic Adaptive Streaming over HTTP (DASH) is a recently proposed standard that offers different versions of the same media content to adapt the delivery process over the Internet to dynamic bandwidth fluctuations and different user device capabilities. The peer-to-peer (P2P) paradigm for video streaming allows us to leverage the cooperation among peers, guaranteeing the service of video requests with increased scalability and reduced cost. We propose to combine these two approaches in a P2P-DASH architecture, exploiting the potentiality of both. The new platform is made of several swarms and a different DASH representation is streamed within each of them; unlike client-server DASH architectures, where each client autonomously selects which version to download according to current network conditions and to its device resources, we put forth a new rate control strategy implemented at peer site to maintain a good viewing quality to the local user and to simultaneously guarantee the successful operation of the P2P swarms. The effectiveness of the solution is demonstrated through simulation and it indicates that the P2P-DASH platform is able to provide its users with very good performance, much more satisfying than in a conventional P2P environment where DASH is not employed. Through a comparison with a reference DASH system modeled via the Integer Linear Programming (ILP) approach, the new system is shown to outperform such reference architecture. To further validate the proposal, in terms of both robustness and scalability, system behavior is investigated in the critical condition of a flash crowd, showing that the strong upsurge of new users can be successfully revealed and gradually accommodated.

This work proposes a novel P2P-DASH architecture that jointly leverages the intrinsic cooperation property of P2P and the adaptivity of Dynamic Adaptive Streaming over HTTP (DASH) to deliver videos to a large population of users with heterogeneous streaming requests. The system is made of several overlays, each distributing a different DASH representation; every peer within the system autonomously enforces a newly devised rate control strategy that steers its movements from one overlay to another, in order to maintain a good viewing experience and to simultaneously guarantee the successful operation of the P2P overlays. The effectiveness of the solution is demonstrated through simulation and it indicates that the P2P-DASH platform is able to guarantee its users a very good performance, much more satisfying than in a conventional P2P environment where DASH is not employed. Moreover, through the comparison with a reference system modeled via the Integer Linear Programming (ILP) approach,it is shown that our system outperforms such reference architecture in terms of viewing quality; at the same time, the number of users streaming the video at the desired bit rate is extremely close to the optimum of the ILP setting in the most favorable scenarios, and only moderately suboptimal in the worst.

This paper presents an approach to privacy-preserving data mining that relies upon a relatively simple secret sharing scheme. Its main feature is that users, sensitive data owners, are engaged in the secret sharing operations that protect their privacy. They are grouped in independent clouds connected to a central unit, the data miner, that only manages the aggregated data of each cloud, therefore avoiding the disclosure of information belonging to single nodes. We propose two privacy preserving schemes, with different privacy levels and communication costs. When designing them, we assume that some users' data might become inaccessible during the operation of the privacy preserving protocols, due to intermittent network connectivity or sudden user departures, and therefore introduce a new performance metric, the failure probability, defined as the probability that the mining output cannot guarantee the desired level of accuracy. We then discuss the attractive tradeoffs between privacy, accuracy and communication overhead that each scheme exhibits.

New pervasive technologies often reveal many sensitive information about users' habits, seriously compromising the privacy and sometimes even the personal security of people. To cope with this problem, researchers have developed the idea of privacy-preserving data mining which refers to the possibility of releasing aggregate information about the data provided by multiple users, without any information leakage about individual data. These techniques have different privacy levels and communication costs, but all of them can suffer when some users' data becomes inaccessible during the operation of the privacy preserving protocols. It is thus interesting to validate the applicability of such architectures in real-world scenarios. In this paper we experimentally evaluate two promising privac-preserving techniques on PlanetLab, analyzing the execution time and the failure rate that each scheme exhibits.

This work investigates the end-to-end performance of randomized distributed space-time codes with complex Gaussian distribution, when employed in a wireless relay network. The relaying nodes are assumed to adopt a decode-and-forward strategy and transmissions are affected by small and large scale fading phenomena. Extremely tight, analytical approximations of the end-to-end symbol error probability and of the end-to-end outage probability are derived and successfully validated through Monte-Carlo simulation. For the high signal-to-noise ratio regime, a simple, closed-form expression for the symbol error probability is further provided.

In recent years, classical Internet applications have been accompanied by the surging of a great variety of new services and exciting possibilities. Among such broad range, two particular phenomena are highly successful: Online Social Networks (OSNs) and Peer-to-Peer (P2P) services. This paper merges these distinct worlds, via the proposal of a P2P streaming system that takes advantage of the friendship relationships of an underlying OSN, to better distribute videos among the overlay peers that are also friends within the OSN. A category of privileged users is therefore created, that is guaranteed a satisfying viewing experience when the P2P overlay operates in critical conditions, i.e, when bandwidth availability is scarce. We show that the help of direct friends, two-hops away friends and, in the limit, of the entire OSN community brings in considerable advantages to the peers that are OSN members. In particular: the number of those among them that are able to download the entire video significantly increases; the number of video portions they can obtain consistently raises; as desired, when the P2P system is operating in underloaded conditions, a proper functioning is guaranteed to all of its nodes, regardless of their being members of the OSN or plain P2P users.

In recent years P2P technology has gained great popularity not only in file sharing applications but also in the field of video distribution. This work proposes a simple model to assess the performance a P2P system designed for video streaming can achieve, measured in terms of average video delivery rate. The model allows to compute system efficiency in a simple but accurate way through the partition of peers in two distinct populations: bad peers, that are not collaborative, and good peers, that contribute to share their video contents with others. As a meaningful example, it is employed to investigate the behavior of a real P2P prototype subject to high peer dynamics: its effectiveness is proved via experimental results, and sheds a new light on the way a streaming service overlay can be centrally and timely monitored.

Today two keywords more and more frequently recur over the Internet: Peer-to-Peer (P2P) and social networks. P2P, in all of its different declinations, represents a widely-adopted approach for content distribution, particularly for video diffusion. In parallel, the proliferation of social networks is an analogously stunning phenomenon, of unprecedented popularity and scope. In the present work we examine a mesh-based P2P overlay, specifically designed for video streaming, and put forth some modifications to the neighborhood creation and chunk scheduling algorithm the platform adopts, with the goal of favoring peers belonging to a social network and granting them better performance. The improvements that such modifications attain are measured in terms of delivery ratio (throughput) and playback delay. We find that it is possible to guarantee a clear service differentiation, so that social network peers experience an improved viewing experience at the expense of ordinary overlay members, and that the scheduling mechanism modifications warrant the more consistent gains, we also show the role that different percentages of peers belonging to the social network have on the considered metrics. We finally suggest that the attained differentiated service level can be leveraged as an incentive to convince peers of the video overlay to join the social network.

Abstract-Peer-to-peer (P2P) technology for TV broadcasting over the Internet is becoming more and more popular in the very last years. This paper introduces a network-wide metric to assess the efficiency of P2P streaming systems and develops a mathematical model to explain: (i) the scalability of such architectures with the number of peers, as evidenced by recent measurements; (ii) the initial decrement of efficiency (hence, quality) when a sharp increase in the number of peers in system occurs, as reported by experimental data. As for the second point, the proposed model builds upon the fundamental remark that when a peer first joins the system, it has no video content to share with others: its upload contribution is null for an initial time interval and the new peer behaves as a free rider. Three situations concerning the system reaction to the requests of the new entering peers are examined: full compensation; partial compensation; no reaction at all. Depending on the system answer and on its extent, system efficiency is shown to exhibit different time trends.

The paper mathematically frames the relation between window-based and rate-based congestion control algorithms, when the assumption of variable round-trip time conditions is introduced. Under this hypothesis, the local stability analysis of a specific class of source, window-based controls is first developed; then a new, rate-based control suitable for use by multimedia streams is proposed. The novel strategy is demonstrated to guarantee consistent and intra-fair throughput allocation, good responsiveness and satisfying smoothness, even in a first, easy-to-implement version.

This paper mathematically frames the relation between window-based and rate-based source congestion control algorithms when the assumption of variable round-trip time conditions is introduced, and then it proposes a new, rate-based control designed to be suitable for use by multimedia streams. The behavior of the new strategy is assessed by simulation, demonstrating that it exhibits good responsiveness and satisfying smoothness, guaranteeing consistent and intra-fair throughput allocation. Moreover, a simple version of the control can be readily implemented resorting to RTP basic functionalities.

The current paper proposes a novel end-to-end congestion control suitable for multimedia flows, with the primary aim of guaranteeing a good level of smoothness to the connections that employ it. The suggested strategy, termed Adaptive GAIMD, has as its starting point a blind, window-based scheme named General Additive Increase Multiplicative Decrease (GAIMD); it however enhances it through a measurement-based prediction of network conditions, based on the observation of the interarrival times of the most recently received triple duplicate acknowledgments. Such forecast allows the control to selectively switch to the AIMD policy, i.e., to a TCP-like behavior, when the network drifts towards overload conditions. On the other hand, if the network is sensed lightly loaded, the control behaves as GAIMD. In all cases, Adaptive GAIMD guarantees a smoother packet delivery than native GAIMD and TCP would. The validity of the proposed approach is experimentally proved, performing numerous tests and comparisons between Adaptive GAIMD, GAIMD and TCP-Sack, that took place on a geographical link spanning from Modena, Italy, to Los Angeles, United States.

In this paper reference is made to some allocation techniques suitable for the assignment of OVSF (orthogonal variable spreading factor) codes on the downlink of a WCDMA system. Yet, in contrast to what has been previously assumed in literature, this work proposes a criterion for the admission of a new user that explicitly takes into account transmission requirements. In detail, two distinct conditions are to be fulfilled to let a new call enter the system: given a code allocation strategy is selected, the corresponding rules are first applied with the goal of finding a free code to assign the user, fitting the user's data rate needs; additionally, it is required that the signal-to-noise ratio - computed accounting for the presence of the new user - takes on values higher than a proper threshold with a probability sufficiently high (higher than a predefined admission value) for all ongoing connections. Equivalently, system load can be increased, but only up to the point where the assignment of a new code within the tree does not degrade transmission quality below an acceptable level. An accurate description of the underlying physical layer is employed, and it is demonstrated how markedly the requisite of adequate transmission conditions affects the number of admitted users. As it is the interference level to ultimately drive the OVSF code assignment, realistic limits on the performance that some popular code allocation techniques can achieve in a wideband CDMA system are clearly evidenced

Abstract-Alternative code assignment policies for the forward link of a multicarrier DS-CDMA system employing the recently proposed quasi-orthogonal sequences are put forth in this letter. It is demonstrated how different allocation strategies of quasi-orthogonal codes alter the interference level suffered by active units; it is also shown how to shape the rate of interference growth as a function of the number of active users within the reference cell, implementing some specific code allocation patterns. The performance achieved by various schemes is assessed in a frequency selective fading environment, in terms of average bit error probability.

This work proposes an analytical model to investigate the behavior of a radio link control (RLC) protocol on the forward link of a mobile radio system and to relate its performance to transmission design. As a reference case, a multicarrier DS-CDMA architecture employing Walsh and quasi-orthogonal channelization codes, similar to the current cdma2000 system, is considered. Jointly modelling the features of the data link layer and of the underlying physical channels, it is possible to identify the interaction between transmission parameters and the RLC protocol behavior. The latter is assessed by analysis, determining the RLC protocol goodput and the average delivery time suffered by a data unit transferred over the radio channel, when two different coding approaches are adopted: namely, rate compatible punctured convolutional codes and code combining

This paper presents an extensive experimental analysis of two TCP-friendly congestion controls recently proposed in the literature, placing a special emphasis on the wireless setting. This work is motivated in part by the consideration that, to our knowledge, the majority of the results in this area are validated by simulation analysis rather than by experimentation. Testing these algorithms in real environments can help to verify their actual effectiveness for possible employment over the Internet. To reach our goal, we have implemented one representative algorithm among the so-called window-based TCP-friendly congestion controls, namely, the general additive increase multiplicative decrease (GAIMD) strategy, and used the publicly available code for the most popular control of the rate-based family, i.e., the TCP-friendly rate control (TFRC). We have then evaluated their TCP-fairness and smoothness in a test-bed featuring Cisco WAN equipment, as well as in an IEEE 802.11g wireless local area network (WLAN). Based on our results, we first propose and justify the tuning of TFRC to avoid a weird behavior that in the wireless environment this control occasionally exhibited. We next show that the GAIMD strategy reveals non-negligible scalability and smoothness problems that limit its performance, mainly in the radio setting. We empirically demonstrate that its increase/decrease rules, based on a TCP-Reno analytical model, do not guarantee an adequate performance when GAIMD competes with TCP-Sack, a de facto standard for current TCP implementations.

This work investigates the TCP-friendliness of a particular congestion control mechanism, the TCP friendly rate control protocol (TFRC), when it is employed to regulate the transmission rate of nonTCP sources competing for bandwidth with different TCP flows (Reno, Sack and Westwood) in an IEEE 802.11b wireless LAN. Throughput of simultaneously active TFRC and TCP flows have been both experimentally measured and investigated through simulation, revealing that the wireless indoor radio channel plays a significant role on such parameter. Moreover, some useful insights have been derived on the behavior of TFRC and its degree of friendliness in an environment featuring a low degree of statistical multiplexing, such as the wireless context examined.

This letter analytically determines the effects of selective diversity on the performance of a RAKE receiver in dense multipath channels featuring an arbitrary power delay profile. The approach devised permits assessing the symbol-error probability of the RAKE receiver with simple computations, and determining the influence on system behavior of different power delay profiles and number of resolved paths, as well as signal bandwidth.

This paper proposes some novel techniques to accommodate users with different rate requirements in a wideband code-division multiple-access system employing orthogonal variable spreading factor codes. Two simple static code assignment strategies are first considered, and an improvement based on multicode assignment. Then the new idea of tree partitioning is introduced and used to devise a dynamic code reassignment algorithm. The behavior of these different techniques is experimentally investigated, in terms of call blocking probability and number of required reassignments. The tree partitioning method exhibits very good performances.

The work investigates the behavior of a 3rd generation radio link control protocol on the forward link of a multicarrier DS-CDMA architecture that employs Walsh and quasi-orthogonal channelization codes. Protocol performance is assessed analytically, determining the average delay suffered by a data unit transferred over the radio channel. For alternative allocation policies of Walsh and quasi-orthogonal codes, the effects on throughput of different transmission qualities and size of protocol data units are evaluated.

Several code allocation policies, suitable for the forward link of a multicarrier DS-CDMA mobile radio system that employs Walsh and quasi-orthogonal sequences, are proposed. Their effect on system performance is evaluated in the presence of a single class of users, as well as when transmissions of different data rates are supported, owing to the adoption of the variable spreading factor technique.

This paper proposes some novel techniques to accommodate users with different transmission rate requirements in a CDMA system which employs orthogonal variable spreading factor codes. Several static and dynamic code assignment strategies are put forth and their behavior investigated, in terms of call blocking probability and number of required call reassignments. Their efficiency in dealing with various traffic loads is demonstrated, quantitatively showing the superior performance of our dynamic scheme with respect to a so-called "optimal" code assignment strategy recently presented in the literature.

Alternative code assignment policies for the forward link of a multicarrier DS-CDMA system employing the recently proposed quasi-orthogonal sequences are put forth. It is demonstrated how different allocation strategies of quasi-orthogonal codes alter the interference level suffered by active units; it is also shown how to shape the rate of interference growth as a function of the number of active users within the reference cell, by implementing some specific code allocation patterns. The performance, in terms of average bit error probability, achieved by various schemes is assessed in a frequency selective fading environment, and the advantages of different solutions highlighted.

In this study we determine the Erlang capacity of a TDD-TD/CDMA architecture providing multirate services to different classes of subscribers. The quality of service perceived by speech and data users is assessed and some design criteria are evidenced, with reference to both traffic and transmission parameters.

We determine the effects of different diversity schemes on the behavior of a direct-sequence spread spectrum RAKE receiver which operates with random spreading sequences, in a multipath channel with arbitrary power delay profile. Two different receiver architectures, the former employing the L instantaneously strongest paths out of the N/sub r/ resolved ones, and the latter simply the L first paths out of the received N/sub r/, are confronted in terms of symbol error probability, when an M-level PSK system is assumed. The analytical approach devised for assessing such comparison gives the symbol error probability in terms of simple elementary integrals, and immediately evidences the influence on system behavior of the number of resolved paths, bandwidth occupancy of the signal and power delay profile, with a remarkably moderate computational effort.

This paper analyses the behavior of two hybrid time-code division multiple-access (T/CDMA) architectures on the up-link of a macrocellular mobile radio system. For the examined schemes two categories of users-voice and data-share the domain of available resources, made up of time slots and codewords, through two alternative assignment strategies. Both solutions attribute voice users one single resource pair, i.e., one time slot and a single code to employ on that time slot, but differ in the way data users requests are accommodated: they are either simultaneously granted several codes over the same time slot or are assigned a single code over several distinct time slots. Call admission control is performed by a channel assignment algorithm which dynamically attributes resources only if specified levels of transmission quality are met on the radio channels. The blocking and the outage probability of the two classes of users are determined and compared, showing that one of the proposed schemes exhibits better performance and allows to satisfyingly serve a significant percentage of data users.

In this paper we study the performance of a TDD-TD/CDMA architecture with two classes of subscribers, voice and data, when an interference-driven admission control policy is adopted. The blocking and the outage probability of the system users are evaluated under various traffic conditions for several uplink/downlink configurations of the time slots, demonstrating that the time division full duplex approach needs a careful tuning in order to maximize system capacity.

This paper investigates in a microcellular set-up the behavior of a new class of dynamic channel allocation policies, tailored to provide different QoS degrees to distinct user classes of a wireless system. The performance achieved by a network employing these schemes is evaluated in terms of call blocking, taking into account the requirements raised by multi-rate applications. Some design criteria effective to optimize the management of the scarce radio resources are furnished and the goal of providing radio users with both low and high speed connections, suitable for a basic set of audio and video applications, is achieved.

A previous technique for generating two spatially correlated Rayleigh fading envelope sequences is generalized to give an efficient method for generating multiple cross-correlated Rayleigh fading sequences. Any desired cross-correlations can be realized for an arbitrary number of diversity branches. The computational complexity of the new method is compared to the computational complexity of a direct method. It is shown that properly utilizing the underlying block Toeplitz structure of the correlation matrix can give computational savings of several orders of magnitude.