We present PTLM (Passive TCP Latency Monitor), a lightweight system for estimating TCP round-trip times (RTT) by passively observing ongoing flows. Unlike active probing approaches such as ICMP ping, PTLM operates entirely in-band by extracting TCP timestamp values (TSval/TSecr) from segments observed at intermediate nodes. This enables accurate latency estimation without injecting probe traffic or modifying endpoints. PTLM computes RTT samples in both directions and aggregates them into an end-to-end view of latency as experienced by applications. Experimental validation in a controlled testbed confirms that PTLM detects RTT variations in real time while maintaining negligible overhead. These results demonstrate PTLM as a practical tool for researchers and operators requiring transparent, TCP-aware latency monitoring.

Latency monitoring is a fundamental requirement for TCP-based services in scenarios where responsiveness must meet strict deadlines, typically below 100 ms. This is particularly relevant for emerging applications such as remote driving in smart mobility testbeds and remote operation of industrial machinery in Industry 5.0 environments. Traditional active monitoring techniques, which rely on probe traffic (e.g., ICMP ping), introduce additional load and may fail to capture the true latency experienced by the target service.In this paper, we present PTLM (Passive TCP Latency Monitor), a system that enables non-intrusive and passive monitoring of TCP latency by directly observing application flows at intermediate nodes along the client–server path. PTLM leverages the principles of passive latency estimation introduced in previous work (e.g., pping) but extends them with a practical and deployable solution specifically designed for enterprise networks and experimental living labs.We report the results of an extensive experimental campaign including feasibility, performance, and reliability tests. In particular, experiments conducted in the MASA (Modena Automotive Smart Area) living lab demonstrate that PTLM can successfully detect latency variations that traditional ping-based methods fail to observe. The results validate PTLM as an effective monitoring solution for latency-critical applications, with direct relevance to enterprise deployments (IGNITE 5.0 project) and experimental testbeds supporting autonomous driving and other time-sensitive services.

In recent years, Intelligent Transport Systems (ITS) have gained significant attention, driven by the European Directive 2010/40/EU, aiming to enhance vehicle safety, traffic efficiency, and overall transport experiences. A cornerstone of ITS is Vehicular Ad-hoc Networks, which enable rapid communication through Dedicated Short-Range Communication for Vehicle-to-Vehicle and Vehicle-to-Infrastructure interactions. Standardized messages such as Cooperative Awareness Messages and Decentralized Environmental Notification Messages have become essential for collision avoidance and other safety-critical applications. However, challenges such as network congestion highlight the importance of Decentralized Congestion Control to ensure reliable communication.This study provides insights to optimize network performance by investigating the mechanisms by which Decentralized Congestion Control tries to mitigate congestion, namely, access categories, transmission power, and modulation coding schemes. The assessment is performed by measuring network performance and QoS under varying conditions through the tool iPerf on a real IEEE 802.11p-based testbed. The results indicate that prioritization classes and transmission power improve performance as expected, while modulation coding schemes can affect data rate in a less intuitive way.

In the context of Intelligent Transportation Systems, a Vehicular Ad-Hoc Network provides communication among nearby vehicles and roadside infrastructure. Commercial devices enabling this type of communication via the IEEE 802.11p standard are expensive and proprietary, limiting research and independent field testing. Furthermore, as the industry is their target client, they integrate features like metal waterproof en-closures or mounting brackets, which often result in relatively cumbersome devices unable to accommodate use by vulnerable road users with limited space and weight-bearing capabilities. Aiming to address these concerns, this paper presents a low-cost and open-source device assembled with off-the-shelf components, which we named Open DSRC Unit or ODU in short. The design minimizes the steps required to implement the unit while maintaining a degree of flexibility for potential extension with future upgrades. As a result, the device can act as both a RoadSide Unit and an On-Board Unit, leaving open the possibility of implementing custom messages and acting with custom roles.

The main objective of connected vehicles is to signif-icantly improve safety for travel participants and nearby people. Cooperative Driving Automation denotes the set of autonomous applications requiring cooperation among connected vehicles, including those with driving automation features engaged. As the operational purpose of the cooperation depends on the application class and the level of driving automation, it follows that the communication strategies and, therefore, the design principles of the human-machine interfaces of participating vehicles must change as well. However, the literature is missing a standardized and coherent strategy for the design and implementation of these interfaces, which may have undesirable impacts on the risk reduction profile of cooperative applications. This paper wants to be a first step in the definition of suitable design principles for such interfaces. We contribute an overview of related solutions providing a selection of relevant works, and we synthesize a set of fundamental human-machine interface design principles for safety - related cooperative applications, hypothesizing an inverse relationship between the level of driving automation and the need for active approaches to maximise the safety of connected vehicles and nearby road users.

The network infrastructure is a key factor for Intelligent Transportation Systems (ITS). Only reliable networking support can enable the low latency required by ITS for safety-related applications, where effectiveness, reliability, and timeliness play crucial roles. These characteristics are required to reach autonomous driving and, tailored for a living lab, are required to test it and design novel applications for connected vehicles and, generally speaking, enhance road safety. This paper introduces a test campaign performed to stress the Modena Automotive Smart Area (MASA) backbone to efficiently evaluate the performance of the network infrastructure before starting the deployment of ITS services, including the installation of smart cameras and Road-Side Units (RSUs). The suite of tests has been designed to accommodate the automotive requirements, with the goal of evaluating the network latency in uncongested and congested situations, including the presence of QoS requirements. The results show that the MASA network is suitable for safety-related applications, introducing less than 100ms of induced latency in severe congested events for all the tested spots. This result is achievable thanks to the backbone characteristics and the partnership with the municipality that shares the physical network with the university, avoiding IST black boxes.

Cooperative awareness messages (CAM) are a family standardized by ETSI to distribute relevant information to all the nodes in vehicle coverage, such as time, position, speed, direction, etc. The parameters needed to trigger these messages, as well as the information contained in the messages themself, can be obtained with a GPS receiver. Our paper aims to increase the efficiency of CAM generation and dissemination in a less-is-more style, considering different GPS devices of different qualities. We achieve this goal by comparing the effectiveness of two GPS, belonging to two different market segments, with an order of magnitude of a price gap between them. We compared them in two different scenarios, similar in layout and traffic behavior but different in terms of GPS coverage conditions.Field trials with DSRC IEEE 802.11p have been carried out in an urban environment where different GPS coverage conditions widely emerge, with the possible presence of large buildings that emphasize all the GPS challenges: reducing the number of visible GPS satellites, introducing multi-path effects, fading, and reducing the signal strength.While having some notable differences in the two scenarios, we found the triggers’ distributions to be similar between both the receiver devices. Instead, the average time between consecutive awareness messages has been found to be significantly lower in the case of bad coverage compared with the open-sky scenario, something that may reflect the presence of unwanted messages triggered by the GPS errors rather than the real vehicle motion. Therefore, we designed a new CAM generation algorithm that also considers the GPS error to reduce the messages transmitted on the radio medium.

Since the definition of the bufferbloat phenomenon, several Linux kernel modules have been introduced in its TCP/IP stack, and there is a lack of experimental studies on their effects when coupled with WLAN technologies, in particular, IEEE 802.11n and IEEE 802.11ac. One essential algorithm introduced is named TCP Small Queues (TSQ) and has the role of limiting the number of packets that a TCP socket can enqueue in the stack, waiting for the physical layer to send the packets before enqueueing extra data. A second significant TCP algorithm is named TCP Pacing (TP) and regulates the pace used by the socket to enqueue packets in the stack, regulating the formation of bursts of data. These mechanisms affect the frame aggregation logic on WLAN networks and compromise the throughput-latency tradeoff of all the TCP variants. This paper presents an experimental evaluation of these techniques investigating the wireless network performance of several TCP congestion control variants under the presence of different TSQ and TP policies, modeling also their interaction.

In this paper, we report and analyze results from a field test campaign aimed at assessing the real-world performance of IEEE 802.11p for safety-related applications. Regulator bodies in the EU and the US recommend the use of domestic-level transmission powers for general usage while granting the opportunity of employing higher powers for safety purposes. While the latter are generally expected to result in higher network performance, we want to quantitatively assess what they translate to in terms of metrics relevant from an application point of view, such as, goodput, round-trip times, jitter, and losses. These are studied and associated with conditions out of the network control at urban and suburban speeds, that is, in absence or in presence of different degrees of congestion, and in complete or partial line-of-sight. In general, we find evidence of an inverse correlation between the degree of congestion and the benefits granted by higher transmission powers. Up to the tested speed, IEEE 802.11p appears able to provide safety guarantees even at relatively long distances, as long as an appropriate transmission power is employed. At the same time, it must be acknowledged that higher power alone cannot overcome the significant dips in performance resulting from highly congested environments and non-line-of-sight scenarios.

This paper shows the inefficiency of TCP BBR in exploiting the Wi-Fi bandwidth. This limitation of BBR has been observed with both IEEE 802.1ln and IEEE 802.11ac, where the mechanism of frame aggregation is used to boost the throughput of data transmission. In the last years, many TCP variants have been introduced to limit the bufferbloat phenomena and bound the latency, through a reduction of the queue backlog injection rate. However, this mechanism impacts on the Wi-Fi frame aggregation logic, impeding TCP congestion controls to reach the full throughput potential of a Wi-Fi interface. While this problem can be solved with TCP CUBIC allowing the sender node to enqueue more packets, for TCP BBR the fix is not the same, as it has a customized pacing algorithm. With this contribution, we propose BBR+, a new BBR version that allows to fine tune the congestion control pace, achieving 6 times more throughput over IEEE 802.1ln channels and 16 times more throughout over IEEE 802.11ac channels, at the cost of an increased latency that is however always less than the latency obtainable with TCP CUBIC.

V2X technologies are based on heterogeneous and safety-related applications that rely on broadcast messages, which are typical in V2V communications, and to unicast messages, which are, instead, typical in V2I/I2V communications. To ease unicast communications from the infrastructure to the vehicle, like rescue operations, addressing the challenging routing aspect for VANETs, we designed an Infrastructured RSSI-based Opportunistic routiNg algorithm for Mobile Adhoc Networks (IRONMAN), also focusing on the energy consumption of the solution developed. IRONMAN takes opportunistic routing decisions based on the RSSI calculated by the Road-Side Units (RSUs), instead of the classical GPS-based solutions. Through a real testbed, we demonstrate that IRONMAN outperforms standard Linux-based routing solutions for ad-hoc networks, like BATMAN and HWMP, providing almost optimal goodput without adding any overhead related to the routing decisions.

In this article, we study the application-level performance of safety-related communications over IEEE 802.11p, presenting the first outcomes resulting from several short- and medium-range field tests. These have been performed with low-cost off-the-shelf equipment operating on the IEEE 801.11p PHY/MAC stack, with devices configured or modified, when necessary, with open-source software to ensure repeatability. In particular, this article discusses the relation that transmission power, antenna quality, distance, and different degrees of congestion, alone and combined, have on round-trip times, TCP Goodput, UDP Goodput, Jitter, and Datagrams or Packets lost. It also details an investigation on the hidden terminal problem, by presenting the results of field tests where destructive interference due to hidden terminals has been artificially introduced at the host premises. Finally, this article presents the metrics obtained by performing initial mobility tests in urban and suburban scenarios. The results are encouraging and attest that, with the appropriate transmission power, it is possible to provide satisfying safety guarantees under the tested conditions.

In this work, we focus on the well-known hidden terminal problem applied to the IEEE 802.11p context, from an infrastructure-centric point of view (I2V). The problem occurs in scenarios where two nodes are out of each other's range while being independently able to communicate with a third that, intuitively speaking, is positioned in the middle of the former two. If the two nodes transmit to the third at the same frequency and at the same time, though not exactly the same signal, the third would be unable to identify the two source signals and, therefore, would suffer from interference. Although the occurrence of this problem may be unlikely in I2V practice, certain applications, such as road safety, require the problem to be addressed anyhow. This article shows the outcomes of field trials conceived to experience first-hand the occurrence of the issue in IEEE 802.11p networks, experiment with different interference management techniques, and design or select the optimal approach to minimize interference. Results unequivocally attest that, in this context, an ad-hoc Time-Division Multiple Access (TDMA) approach provides the best guarantees in any condition. It is also shown how, on the other hand, the common method of using the Request To Send/Clear To Send (RTS/CTS) protocol to prevent the issue in unicast scenarios may result in a disappointing performance, due to the nature of safety-related I2V communications that employ short packets to be delivered with minimum latency.

Future 5G networks will rely on heterogeneous deployment of micro eNodeBs, and is expected that wireless transport interfaces, deployed together with each eNodeB, will provide high-speed backhaul connectivity through the possibility to create an ad-hoc wireless mesh between the deployed micro eNodeBs. Therefore, the backhaul routing strategy will play a fundamental role in exploiting the backhaul resources. In this context, we aim to investigate the advantages and limitations of qualitatively different backhaul routing strategies as UEs download TCP and UDP content. In this paper, we compare a backpressure-based strategy that takes per-packet or per-flow basis (BP-MR) and traditional single path routing (OLSR). Ns-3 simulations show that the per-packet BP-MR variant enhances UDP performance by circumventing congestion but degrades TCP because of reordering. The per-flow BP-MR variant minimizes retransmissions by limiting the excessive reordering, and outperforms OLSR performance for both UDP backhaul traffic and TCP traffic.

202001 bcma

A Seismic Alert System (SAS), also called Earthquake Warning System (EWS) or Earthquake Early Warning System (EEW or EEWS), represents one of the most important measures that can be taken to prevent and minimize earthquake damage. These systems are mainly used to detect P-waves and the faster seismic waves and to subsequently trigger an alarm about the incoming S-waves, the slower and most dangerous seismic waves. In some cases, distributed systems are also able to alert some locations before the impending P-waves strike them. This paper presents Earthcloud, a cloud-based SAS that aims to provide all the former capabilities while retaining financial accessibility. Earthcloud first results, generated from four months of data acquisition, are compared with those coming from other systems. In particular, the paper focuses on processing and communication delays, showing how the Earthcloud new detection strategy may minimize delays. Although a thorough test campaign with more sensor nodes is needed to assess performance reliably, especially for highly dense urban scenarios, initial results are promising, with total latencies for Earthcloud always kept under the 1-second mark, despite being at the expense of solid magnitude estimation.

In this paper, we present and characterize a phenomenon that arises when network congestion is met with packet redundancy at a distributed bottleneck. Congestion typically causes packet drops at the queueing level, especially when Active Queue Management techniques are employed to mitigate the bufferbloat effect. Redundancy may be present in the form of packet replication over different paths, as a solution to increase network resilience and network availability guarantees. When both network congestion and packet redundancy are in place, a counterintuitive throughput-boosting effect may originate from them. We named this phenomenon bufferboost, for the key role played by packet drops introduced to avoid bufferbloat effects. The contributions of this paper are the definition of the buffer-boost phenomenon, its modelling through a mathematical upper bound formulation, its isolation in an emulated environment for reproducibility and its validation through an extensive numerical evaluation that also verifies the introduced model. Results show that bufferboost is a favorable side effect able to boost network throughput.

In recent years, the Linux kernel has adopted an algorithm called TCP Small Queues (TSQ) for reducing queueing latency by controlling buffering in the networking stack. This solution consists of a back-pressure mechanism that limits the number of TCP segments within the sender TCP/IP stack, waiting for packets to actually be transmitted onto the wire before enqueueing further segments. Unfortunately, TSQ prevents the frame aggregation mechanism in the IEEE 802.11n/ac standards from achieving its maximum aggregation, because not enough packets are available in the queue to build aggregates from, which severely limits achievable throughput over wireless links. This paper demonstrates this limitation of TSQ in wireless networks and proposes Controlled TSQ (CoTSQ), a solution that improves TSQ so that it controls the amount of data buffered while allowing the IEEE 802.11n/ac aggregation logic to fully exploit the available channel and achieve high throughput. Results on a real testbed show that CoTSQ leads to a doubling of throughput on 802.11n and up to an order of magnitude improvement in 802.11ac networks, with a negligible latency increase.

A Seismic Alert System (SAS) is one of the most important measures that can be taken to prevent and minimize earthquake damage. The role of a SAS is to detect as soon as possible an earthquake in progress and to then immediately alert about the impending earthquake waves all locations in danger. Clearly, to be of any utility, the communication of the alarm has to be transmitted way faster than the propagation speed of the earthquake waves. This paper presents the rationale, setup and first results of a deployed Earthcloud prototype, an experimental SAS designed to be low-cost, low-power, and cloud-based. Earthcloud is devised to alert locations that are not too close yet not too far from the earthquake epicenter, i.e., those that can benefit from an alarm received from a few seconds to tens of seconds earlier. The system has been designed to leverage existing Internet infrastructures in order to perform both computation and communication in the cloud, minimizing costs and maximizing communication speed and deployment ease. Data harvested by Earthcloud has been cross-referenced with data from the Italian National Institute of Geophysics and Volcanology and, although still in a prototype stage, the first results indicate that the system can correctly detect earthquakes.

Satellite networking is known to suffer from specific network performance issues, such as high latency and low throughput stability; this derives mainly from the high propagation delay, in particular with GEO satellites. This characteristic poses limitations on the benefits that general AQM solutions could introduce; in fact, reducing congestion and mitigating queueing delay is a vital feature that could boost the performance of satellite networks. This paper investigates PINK (Passive INverse feedbacK), a queue management algorithm designed to indirectly impose an individual resource allocation policy in order to mitigate the bufferbloat effect and the network congestion while exploiting the channel throughput and guaranteeing optimal flow fairness without forcing any packet drop. PINK modifies the TCP Acknowledgements (ACKs) segments passing through the Satellite access gateway. The modification consists in replacing the advertised Receive Windows field (RCV.WNDs) with custom values, in order to enforce a particular bandwidth utilization upper bound. To compute new RCV.WND values, PINK needs only the number of active connections, the flows RTT and the transmission channel bandwidth. The design characteristics of this new AQM let it works efficiently in satellite networks, and we validate this statement through several simulations performed with the ns-3 network simulator.

Vehicle-to-Everything (V2X) communications encapsulate significant challenges as a result of the vast number of wireless technologies deployed, nodes mobility and large applications range. One of the greatest challenges is to provide adequate QoS guarantees even in this demanding environment. State-of-the-art solutions for QoS provisioning over wireless links are based on cross-layering packet schedulers that deal both with the QoS guarantees and the wireless link issues. Unfortunately, such an approach is not flexible and requires technology-dependent solutions, with the definition of a different software version for of the each technology deployed. To overcome this issue, we design a modular architecture which permits the use of existing high-performance packet schedulers for wired links over generic wireless technologies, as they are, and at the same time allows the flexibility to adapt to different channel conditions. We also provide a mathematical analysis of the QoS bounds of this novel architecture together with simulations analysis to validate it. This approach paves the way for efficient QoS provisioning and throughput-boosting in V2X systems.

This paper proposes PINK (Passive INverse feedbacK), a queue management algorithm designed to indirectly impose a certain resource allocation policy on defined sets of client hosts. PINK adds intelligence at intermediate nodes that connect client hosts to bottleneck links or to external networks in general, allowing these nodes to dynamically modify the TCP Acknowledgments (ACKs) segments passing through. This is made by setting TCP ACK advertised Receive Windows field (RCV.WNDs) to custom values, in order to enforce a specific bandwidth utilization upper bound. To compute new RCV.WND values, PINK needs only the number of active connections, the flows RTTs and the transmission channel bandwidth. It follows that PINK permits to impose a centralized bandwidth management without the cooperation of clients, which means that no modification or addition whatsoever to end hosts is needed with the goal of enabling a smart environment by simply configuring the access node. Furthermore, as demonstrated in this paper, our proposal does not constraints client hosts performance without purpose; on the contrary, PINK improves efficiency on multiplexed channels by exploiting channel throughput, maintaining a low queuing delay, and guaranteeing optimal flow fairness without forcing any packet drop.

During their day-to-day tasks, public safety operators rely on commercial mobile networks in order to perform data communications and make use of data-based services. In major disasters, however, it is the norm to witness high degrees of network congestion and, often, to experience service outages caused by infrastructural damage. This may strongly hinder the operativity of emergency operators, that expect to rely on resilient communication infrastructures and that demand network availability no matter what. In order for today mobile networks to provide such capabilities, some degree of redundancy should be introduced, either in hardware as infrastructural deployments or in software as data replication mechanisms. This paper focuses on the latter approach. It proposes a network framework to enable collaborative hosts to concurrently forward replicated data over an arbitrary number of channels, in order to compensate for high packet losses or sudden unavailability of routing paths. The data replication and the forwarding behavior are concerted by an SDN controller, that transparently implements a network abstraction to virtually pool hosts network resources. The paper also presents the test bed that has been built and used to extract the emulation results. These demonstrate that the proposal have the potential to strongly improve data communication capabilities in constrained scenarios, in the form of higher data rates and stronger resilience guarantees. Last but not least, it is shown that services availability may be provided in much more cases, even when regular operativity fails.

After a disaster, it is unlikely to expect current terrestrial infrastructures to provide the coverage and the Quality of Service required by emergency operators. Currently, PPDR operators go towards a portable LTE infrastructure that provides an access network to field operators, backhauled by a wireless mesh network that can route traffic from and towards Internet through a satellite gateway. We firstly propose a new per-flow strategy for a backpressure-based routing protocol, BP-MR, and then in the aforementioned environment we analyse the original BP-MR per-packet, the proposed per-flow, and OLSR as routing protocol for the mesh network coupled with three TCP congestion control (Cubic, Hybla, Vegas) performing a typical HTTP transfer. We analysed the throughput, the latency, and the scalability over the number of the flows/operators. An extensive simulation set allows to conclude that employing BP-MR per-flow in the mesh network coupled with TCP Vegas for performing the end-to-end transfer improves the performance for PPDR professionals.

In scenarios where a number of client nodes share a gateway in order to access a high-delay link, the original TCP as well as TCP variants specifically designed to improve performance on those links fail to provide optimal performance; this is due to the indirect inference of both end-to-end congestion and local channel conditions. To mitigate the former, C2ML (Congestion Control Middleware Layer) has been proposed as a tool for centralized and collaborative management of resources that can strongly improve performance for client hosts without affecting network resilience and connection survivability. To complete the picture, this paper proposes DyBRA, an improved version of C2ML with a bandwidth management algorithm designed to dynamically allocate bandwidth among client hosts on the basis of channel conditions, leading to improved performance regardless of the TCP variant employed by clients. Through simulations performed with the ns-3 network simulator, we show how DyBRA strongly improves throughput and TCP friendliness while reducing queue usage and latency.

When multiple nodes share the same gateway or access the same link, the throughput of wireless communications has the unfortunate problem of being limited by traffic congestion; the situation may also further degrade if they establish multiple TCP connections mapped over the same access channel. In end-to- end connections, TCP congestion control plays a critical role in order to not overflow the channel, although most of the network bandwidth bottlenecks are caused by the wireless systems. When TCP is coupled with high-delay channels, the protocol underperform; this is mainly due to long propagation delay, as it is the case with satellite links. To overcome the limitations of TCP on these links, various variants have been proposed, both rate-based or or burst-based. In this paper we present a cooperative transmission control that enhances the effectiveness of TCP asking only for a narrow sender-side modification, operating exclusively at the access stage. It allows to boost throughput over high delay channels, increasing also fairness and friendliness between different TCP algorithms. After the description, we validate its effectiveness and show its properties by means of experimental results, derived from simulations performed on the ns-3 network simulator.

This paper presents preliminary work on PINK (Proactive INjection into acK), an AQM algorithm able to enhance TCP congestion control properties without dropping packets. PINK is a completely transparent solution that does not require any modification to the existing protocol stack of end hosts, and it is particularly suitable for high-delay PPDR systems. The algorithm is based on an explicit feedback scheme, able to enforce a fair bandwidth sharing among clients by modifying the Receive Window in TCP acknowledgements returning to them; such a feedback is computed for each flow, and it only needs the number of active connections, the flows RTTs and the shared link bandwidth. Therefore, PINK is a per-flow stateless AQM that works independently from the TCP algorithm used by clients, and it is fair regardless the flows RTTs, which is a key feature for PPDR systems. The effectiveness of the proposed algorithm is demonstrated using the ns-3 simulator.

Network-enabled services for public protection and disaster relief (PPDR) professionals have been more than necessary in today's emergency situations. Under the extreme circumstances of an emergency, it is essential to have networks which support the required data throughput as well as high availability in spite of high traffic volume, and which minimize the end-to-end delay for applications: however, the choice between the existing technologies is not so easy for PPDR entities, given the high number of parameters associated to satisfying the stringent PPDR requirements, high investments required to permit desirable availability as well as modernization of the existing services (i.e. voice and data) and design constraints posed by network providers as current deployed network reach their boundaries when emergencies occur. This paper analyzes three different emergency scenarios and then presents a subset of the results obtained in terms of financial and economic recommendations, along with technical reports on throughput and end-to-end delay.

The vast majority of efforts aimed to improve network performance are focused on the increase of application throughput. The same holds in the context of Emergency Networks, where operators ask for more bandwidth in order to exploit data-intensive services. Following a steady growth in network capacities and number of users, large buffers have been inserted all over the Internet. Their effects on networks are non-trivial: while they may effectively serve the purpose of exploiting the channel potential, they also create unnecessary delays by damaging the behavior of the most common transport protocol, TCP. Nevertheless, they are being assumed by new congestion control algorithms, especially those tailored for high-latency links such as satellite ones. According to the anywhere-anytime paradigm, these channels represents a key technology for both Emergency and General-Purpose networks. In this paper we first show how buffer lengths impact the perceived delay over satellite links that employ a recently proposed burst-based TCP protocol, and then present a cooperative transmission control that reduce buffers usage and latency while keeping high throughput and flow fairness, thus allowing for a better service provision through satellite links.

This paper describes a novel emergency communication network architecture implemented within the FP7 EU project E-SPONDER [1]. It is characterized by the deployment of heterogeneous wireless systems and also by its holistic approach achieving reliability, high performance, reconfigurability and standalone operation. It is a complete suite of real-time communication technologies built to support the first responders [1] with information services during disaster events. This work investigates the system architecture in an aircraft landing incident and describes a field test carried on at Schiphol airport in Amsterdam. More specifically, a scalable and adaptive telecommunication architecture that ensures voice, video and data between first responders and command centers at all times, even under extreme conditions, is presented. The structure and functionalities of the VoIP subsystem that operates above the proposed heterogeneous E-SPONDER network architecture is described, with a detailed scenario analysis. Finally, the paper presents how the recommended solutions are integrated into an implementable platform.

The packet scheduling problem has been deeply studied for lot of years by researchers in the computer science and telecommunications fields as an important solution that decides the order in which packets are sent over a link in order to provide QoS on a network. Recently, the packet scheduling has become again a challenging topic due to the massive use of wireless technologies (e.g. WiFi, LTE, 4G/5G), both for public and PPDR networks, with which to provide high QoS guarantees is still an open problem. Unfortunately, it is difficult to compare different solutions and actually test them in order to select the most proper packet scheduler for each particular environment.

Wireless communication technologies are critical for public protection and disaster relief (PPDR) professionals during the emergency operations that follow natural or man-made disasters, scenarios in which commercial terrestrial networks often fail to provide the necessary support. The reason is threefold: they simply get disrupted by the disaster, they cannot sustain the sudden surge of network demand or they fail to deliver the necessary bandwidth and/or other QoS guarantees. In every PPDR operation reliable voice communications are critical, especially in the very early stages of the response; nevertheless, there is an increasing demand from the PPDR community for a wider range of data-centric services. While current technologies used for PPDR operations provide a rich set of voice-centric services, they are unable to sustain high-bandwidth data-oriented applications. As the PPDR-TC EU consortium, we propose a hybrid approach to tackle the question of determining the future architecture for Pan-European PPDR networks based on the integration of Terrestrial and Satellite technologies, presenting our first simulation results on the integration of LTE and Satellite Networks.

This paper presents a novel approach to the study of heterogeneous networks. NetBoxIT is a modular, and scalable testbed based on virtualized network emulators. These can be interconnected and run in parallel to reproduce complex networks scenarios within a shared, multi-core hardware platform. We examine the testbed performance with simple emulation trials (related to the reference Emergency Network under study within the EU 7th FP project “A holistic approach towards the development of the first responder of the future”), to verify its realism in assessing multipart networks evaluations.

This work describes the main communication services that have to be provided by any emergency network, together with their bandwidth and network requirements. One of them is voice over IP (VoIP), which is fundamental to coordinate all the necessary rescue operations during a disaster event. However, this service alone is not sufficient for the current needs of any modern (or future) public safety system. Indeed, it has to be accompanied by other important services, like video over IP (VIP) to film the incident area, the transmission of sensor data (e.g., temperature, heartbeat, etc.), and generic file transfer service. In order to design a modern and efficient emergency communication network able to support all of these applications, the first step will be the analysis of the theoretical bandwidth requirements of each network segment. Besides, we will show numerical results, obtained by means of computer simulations, in order to demonstrate that the considered services can be supported, in a reference network architecture, guaranteeing a satisfactory QoS.

In this paper end-to-end performance is evaluated when multiple high-speed access networks, based on the Ethernet Passive Optical Network (EPON) technology, are interconnected through an Optical Burst Switching (OBS) core network. The inter-working unit, or edge node, between these two different networks is properly studied and discussed. A time-based burst assembly algorithm is assumed, which is analyzed and dimensioned to cope with multiple EPONs. The role of a key parameter, such as the EPON cycle time used in the multiple access scheme, is assessed. Transport layer performance of the multi-EPON/OBS network is evaluated through simulations with ns2. Results show that the relationship between the EPON cycle time and the burst assembly time has a fundamental role and requires proper tuning to optimize, e.g., the TCP throughput.

This paper reviews the most important security properties found in computer systems in the context of the network infrastructure for public safety emergency response systems. The goal of this research has been to find what properties most affect the design of an emergency response network, together with the study of the set of the available solutions that can be used to implement these properties in an actual emergency network.

We present a novel approach to heterogeneous networks study based on the implementation of a modular, extensible, and scalable testbed that uses virtualized network emulators. Our goal is to reproduce complex networks scenarios by interconnecting several emulators, concurrently running within a single hardware platform. This paper analyzes the testbed performance with simple emulation trials, to verify if it is suitable for assessing multipart networks evaluations.

In this paper the performance of a hybrid optical network composed of multiple Ethernet Passive Optical Networks (EPONs) at the access connected to an Optical Burst Switching (OBS) core is evaluated. The EPONs make use of the IPACT algorithm to manage the upstream transmissions, whereas the OBS edge node employs a time-based assembly scheme for burst aggregation. This leads to interoperability issues between access and core which are investigated in this paper. The hybrid network scalability is proved by assessing the transport layer performance with different numbers of EPONs served by the same burst assembler and different numbers of Optical Network Units (ONUs) per EPON. The impact of the main design parameters affecting the interaction between EPONs and OBS is estimated. The transport layer performance is evaluated considering heterogeneous traffic flows by simulating both TCP and UDP sources.

This paper presents some of the activities currently performed within the European Union 7thFramework Programme project "A holistic approach towards the development of the first responder of the future". It is a suite of real-time data-centric technologies that will provide enhanced information and communication support to first responders that act during normal and abnormal events. The paper demonstrates the way virtual networks and software routers tools can be appropriate to design and evaluate the telecommunication infrastructure.

In this paper TCP performance over Optical Burst Switching networks is investigated by considering the burst assembly function implemented in the edge nodes. This function has a very important role on end-to-end performance and a edge node architecture is here proposed for improving performance. A multi-queue burst assembly algorithm which is based on an estimation of the current TCP window is proposed and evaluated. Results show that this algorithm can lead to TCP performance improvements of 100%, compared to timer-based assembly operating on a single queue. The OBS network is evaluated through simulations by means of the ns2 simulation tool.

This paper studies TCP performance with different offset times which arise when hybrid multi-granular technology is employed in OBS core nodes. Multi-granular switches are promising devices to match dynamic application needs while optimizing switch costs and feasibility. They foster optical burst sw

This paper proposes a novel bufferless switch architecture based on optical codes, suitable as core node in OBS (optical burst switched) networks. Optical codes are used for coding incoming bursts in order to perform the switching function. To this end the proposed switch architecture employs encoders and decoders and possible output contentions are solved in the wavelength domain by means of variable-input fixed-output wavelength converters.

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 paper determines the performance of TCP when it is employed over a TDD-TD/CDMA architecture which supports different classes of subscribers and adopts an interference-driven admission control policy. The call blocking of the system users is evaluated as well as the TCP throughput that data users experience. This work further shows that when TCP is taken into account, many options come out for the system design in order to optimize radio resources. Furthermore, we show that the TDD system can satisfyingly cope with remarkable traffic loads but a careful tuning of the wireless system is necessary.

The paper determines the performance of TCP when it is employed over a differentiated services network where ATM switches are used as label switching routers, as in Request for Comments 3035. A packet discard scheme is also evaluated with the goal of classifying incoming data flows in different levels of quality of service. In particular, assured forwarding in differentiated services per-hop-behavior, as reported in RFC 2597, is considered as reference scenario. Different degrees of quality of service imply different portions of bandwidth and levels of packet loss for data flows which share the same output link. TCP performance is investigated for low drop precedence flows, or high priority flows, when the considered packet discard scheme is employed during congestion intervals. Numerical results are reported and discussed for different levels of congestion.

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.

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.

In this paper a packet discard scheme is proposed with the goal of providing incoming data flows with output link bandwidth depending on the class of service they belong to. Good performance in terms of throughput and fairness in bandwidth sharing among equal priority flows are also considered as the requirements to meet. The dropping scheme is shown to provide very good results and to represent a possible solution for dealing with the most demanding data flows during congestion intervals. Numerical results are reported and discussed for homogeneous and non-homogeneous data sources.

In a previous paper, one of the authors gave a worst case analysis for the early packet discard (EPD) technique for maintaining packet integrity during overload in ATM switches. This analysis showed that to ensure 100% goodput during overload under worst case conditions requires a buffer with enough storage for one maximum length packet from every active virtual circuit. This paper refines that analysis, using assumptions that are closer to what we expect to see in practice, and examines how EPD performs when the buffer is not large enough to achieve 100% goodput. We show that 100% goodput can be achieved with substantially smaller buffers than predicted by the worst case analysis, although the required buffer space can be significant when the link speed is substantially higher than the rate of the individual virtual circuits. We also show that high goodputs can be achieved with more modest buffer sizes, but that EPD exhibits anomalies with respect to buffer capacity, in that there are situations in which increasing the amount of buffering can cause the goodput to decrease. These results are validated by comparison with simulation.