Abstract This paper introduces Biblio-US17, a labeled dataset collected over 6 months from the log files of a popular public website at the University of Seville. It contains 47 million records, each including the method, uniform resource identifier (URI) and associated response code and size of every request received by the web server. Records have been classified as either normal or attack using a comprehensive semi-automated process, which involved signature-based detection, assisted inspection of URIs vocabulary, and substantial expert manual supervision. Unlike comparable datasets, this one offers a genuine real-world perspective on the normal operation of an active website, along with an unbiased proportion of actual attacks (i.e., non-synthetic). This makes it ideal for evaluating and comparing anomaly-based approaches in a realistic environment. Its extensive size and duration also make it valuable for addressing challenges like data shift and insufficient training. This paper describes the collection and labeling processes, dataset structure, and most relevant properties. We also include an example of an application for assessing the performance of a simple anomaly detector. Biblio-US17, now available to the scientific community, can also be used to model the URIs used by current web servers.

The current IEEE 802.11ax standard enhances Wi-Fi networks with a series of new features, such as multi-user (MU) transmission, an Orthogonal Frequency Division Multiple Access (OFDMA) scheme and the ability to multiplex traffic from different access categories (ACs). These features can be utilized to enhance the QoS support for VoIP traffic and optimize the usage of IEEE 802.11ax network resources. This work proposes a new scheme to multiplex VoIP calls in IEEE 802.11ax MU frames and a new scheduler and resource allocation algorithm specifically designed for VoIP data traffic. Our scheduler allocates VoIP packets requiring longer transmission times into the same frame(s), minimizing the channel air-time assigned to VoIP transmission. In addition, unused radio resources in the MU frame are leveraged to transmit best-effort packets along with VoIP packets. For completeness, we also define a call admission control (CAC) algorithm that anticipates channel saturation conditions to ensure VoIP users can maintain a guaranteed level of QoS. Based on simulation results, our scheme is more efficient in reducing channel utilization than other schedulers like multi-user round-robin (RR) (implemented by ns-3) or single-user FIFO. For example, for 30 VoIP stations using the G.711 codec under mixed channel conditions, our scheme reduces by 30% the air-time required to transmit VoIP packets. When coupled with the ability to also send best-effort packets along with VoIP packets, this translates into a higher throughput (i.e. 10 Mbit/s vs 4 Mbit/s) and more simultaneous VoIP users with guaranteed QoS (up to 46 VoIP users vs 26 and 28 users for the multi-user RR and single-user FIFO scheduling algorithms, respectively).

Unmanned Aerial Vehicle (UAV) networks have emerged as a promising means to provide wireless coverage in open geographical areas. Nevertheless, in wireless networks such as WiFi, signal coverage alone is insufficient to guarantee that network performance meets the quality of service (QoS) requirements of real-time communication services, as it also depends on the traffic load produced by ground users sharing the medium access. We formulate a new problem for UAVs optimal deployment in which the QoS level is guaranteed for real-time voice over WiFi (VoWiFi) communications. More specifically, our goal is to dispatch the minimum number of UAVs possible to provide VoWiFi service to a set of ground users subject to coverage, call-blocking probability, and QoS constraints. Optimal solutions are found using well-known heuristics that include K-means clusterization and genetic algorithms. Via numerical results, we show that the WiFi standard revision (e.g. IEEE 802.11a/b/g/n/ac) in use plays an important role in both coverage and QoS performance and hence, in the number of UAVs required to provide the service.

When telecommunication infrastructure is damaged by natural disasters, creating a network that can handle voice channels can be vital for search and rescue missions. Unmanned Aerial Vehicles (UAV) equipped with WiFi access points could be rapidly deployed to provide wireless coverage to ground users. This WiFi access network can in turn be used to provide a reliable communication service to be used in search and rescue missions. We formulate a new problem for UAVs optimal deployment which considers not only WiFi coverage but also the mac sublayer (i.e., quality of service). Our goal is to dispatch the minimum number of UAVs for provisioning a WiFi network that enables reliable VoIP communications in disaster scenarios. Among valid solutions, we choose the one that minimizes energy expenditure at the user’s WiFi interface card in order to extend ground user’s smartphone battery life as much as possible. Solutions are found using well-known heuristics such as K-means clusterization and genetic algorithms. Via numerical results, we show that the IEEE 802.11 standard revision has a decisive impact on the number of UAVs required to cover large areas, and that the user’s average energy expenditure (attributable to communications) can be reduced by limiting the maximum altitude for drones or by increasing the VoIP speech quality.

In this paper, we propose a quality of service (QoS)-sensitive energy efficiency optimization mechanism for 802.11 networks on the basis of the dynamic and simultaneous adjustment of the content window (W ) and retry attempts limit (r) of the media access control (MAC) sublayer.The use of both operational variables let us not only find the optimum operational point regarding energy efficiency but also attain a positive impact on the QoS, which improves the results obtained with current single-variable optimization strategies.The model under consideration includes external noise and does not impose the saturation condition in stations and as such is well suited for real-time industrial applications under noisy channels.Results obtained from simulation confirm the advantages of adjusting simultaneously W and r versus adjusting either one separately, obtaining a slight improvement in energy efficiency and resulting in less loss and delay at the MAC sublayer.

The current best-effort Internet cannot readily provide the service guarantees that VoIP applications often require. Path switching can potentially address this problem without requiring new network mechanisms, simply by leveraging the robustness to performance variations available from connectivity options such as multi-homing and overlays. In this paper, we evaluate the effectiveness and benefits of path switching in improving the quality of VoIP applications, and demonstrate its feasibility through the design and implementation of a prototype gateway. We argue for an application-driven path switching system that accounts for both network path characteristics and application-specific factors (e.g., codec algorithms, playout buffering schemes). We also develop an application path quality estimator based on the ITU-T E-model for voice quality assessment, and an application-driven path switching algorithm that dynamically adapts the time scales over which path switching decisions are made to maximize voice quality. Through network emulation and experiments over a wide-area multi-homed test bed, we show that, with sufficient path diversity, path switching can yield meaningful improvements in voice quality. Hence by exploiting the inherent path diversity of the Internet, application-driven path switching is a viable option in providing quality-of-service to applications.

The increase in IP telephony demands an efficient tool for QoS prediction and measurement in order to allow the planning and design of the underlying networks to meet quality requirements. Among the several tools available for the traditional circuit switched telephony networks, the E-model can be adapted to modern VoIP networks. This paper discusses this adaptation in a corporate WAN scenario in which PC users can communicate with each other. Impairments related to delay and terminals are analysed to derive the proper formulae to be included in the E-model. Impact of other features of the IP networks and terminals in existing factors of the E-model have also been addressed.