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The rapid development of the digital healthcare and the electronic health records (EHR) requires smooth networking infrastructure to access data using Hypertext Transfer Protocol (HTTP)-based applications. The new HTTP/3 standard should provide performance and security improvements over HTTP/2. The goal of our work was to test the performance of HTTP/2 and HTTP/3 in the context of the EHRs. We used 45,000 test FHIR Patient resources downloaded and uploaded using 20, 50, 100 and 200 resources per Bundle, which resulted in 2251, 901, 451 and 226 HTTP GET and POST requests respectively. The first test downloading 20 resources per Bundle showed that HTTP/3 outperformed HTTP/2 in the local (mean request time 16.57 ms ± 7.2 standard deviation [SD]) and in the remote network (71.45 ms ± 43.5 SD) which is almost 3 times faster. In the 50 and 100 resources per Bundle test the HTTP/3 protocol demonstrated again more than two times gain in downloading performance for remote requests with mean request time 91.13 ms ± 34.54 SD and 88.09 ms ± 21.66 SD respectively. Furthermore, HTTP/3 outperformed HTTP/2 in the constructed clinical dataset remote transfer. In the upload tests HTTP/3 showed only a slight gain in performance merely in the remote network. The HTTP/3 protocol is a relatively new development and a major improvement for the worldwide web. This new technology is still missing in the digital health and EHRs. Its use could offer a major performance gain in situations where data is gathered from multiple remote locations. [ABSTRACT FROM AUTHOR]

High-density point clouds expressing attractive 3D images are attracting attention. These gigantic media require large bandwidth allocations, making them problematic to stream to resource-constrained hand-held devices. This paper proposes a method for point cloud compression and streaming of large point clouds using a web server. Storing large point cloud videos on a web server allows users to publish data sets without using additional applications or sending large amounts of data ahead of time. HTTP/2 improves transfer efficiency by compressing headers into binary format and reduces latency by sending many multiplexed requests using a single TCP session. The large size of the point cloud data and the current limitations of wireless channels demand point cloud compression methods. MATLAB is used for the purpose of analysis, at results show an average compression ratio of 30:1. Our proposed system achieves lower average latency between successive frames resulting in an increased frame rate to 72.46 fps when streaming 4M points, and 82.51 fps in the case of streaming 800K points compared to much lower rates in conventional work.

This article offers a comprehensive overview of recent literature on the HTTP/2 protocol and conducts an analysis of the security threats and DDoS attack typologies associated with HTTP/2. The investigation revealed that the introduction of new features in HTTP/2 has significantly improved the network transmission speed and utilization. However, these advancements have also brought forth a series of emerging network security risks. This study examines the current state of the art in DDoS attacks tailored for HTTP/2 and their detection methods, proposing future research directions in the field of attack detection. By analyzing the distinctive features of HTTP/2 protocol, the study suggests extending DDoS attack detection techniques established for HTTP/1 to the realm of HTTP/2. Furthermore, the research underscores the ease with which adversaries can exploit the intrinsic multiplexing in HTTP/2 to launch a large number of malicious requests, leading to severe depletion of network bandwidth and exhaustion of valuable server resources. Additionally, it highlights the potential applicability of deep learning algorithms in the context of the HTTP/2 protocol. Additionally, the article proposes strategies to address challenges associated with DDoS attacks and the scarcity of adequate datasets for HTTP/2. This research contributes to a comprehensive understanding of the security implications surrounding the HTTP/2 protocol and provides valuable insights for advancing DDoS attack detection technologies.

New versions of HTTP protocols have been developed to overcome many of the limitations of the original HTTP/1.1 protocol and its underlying transport mechanism over TCP. In this paper, we investigated the performance of modern Internet protocols such as HTTP/2 over TCP and HTTP/3 over QUIC in high-latency satellite links. The goal was to uncover the interaction of the new features of HTTP such as parallel streams and optimized security handshake with modern congestion control algorithms such as CUBIC and BBR over high-latency links. An experimental satellite network emulation testbed was developed for the evaluation. The study analyzed several user-level web performance metrics such as average page load time, First Contentful Paint and Largest Contentful Paint. The results indicate an overhead problem with HTTP/3 that becomes more significant when using a loss-based congestion control algorithm such as CUBIC which is widely used on the Internet. Also, the results highlight the significance of the web page structure and how objects are distributed in it. Among the various Internet protocols evaluated, the results show that HTTP/3 over QUIC will perform better by an average of 35% than HTTP/2 over TCP in satellites links specifically with a more aggressive congestion algorithm such as BBR. This can be attributed to the non-blocking stream multiplexing feature of QUIC and the reduced TLS handshake of HTTP/3.

HTTP Adaptive Streaming (HAS) plays a key role in over-the-top video streaming with the ability to reduce the video stall duration by adapting the quality of transmitted video segments to the network conditions. However, HAS still suffers from two problems. First, it incurs variations in video quality because of throughput fluctuation. Adaptive bitrate (ABR) algorithms at the HAS client usually select a low-quality segment when the throughput drops to avoid stall events, which impairs the Quality of Experience (QoE) of the end-users. Second, many ABR algorithms choose the lowest-quality segments at the beginning of a video streaming session to ramp up the playout buffer early on. Although this strategy decreases the startup time, clients can be annoyed as they have to watch a low-quality video initially. To address these issues, we introduced the H2BR technique (H TTP/2-B ased R etransmission) (Nguyen et al. 33) that utilizes certain features of HTTP/2 (including server push, multiplexing, stream priority, and stream termination) for late transmissions of higher-quality versions of video segments already in the client buffer, in order to improve video quality. Although H2BR was shown to enhance the QoE, limited streaming scenarios were considered resulting in a lack of general conclusions on H2BR's performance. Thus, this article provides a profound evaluation to answer three open questions: (i) how H2BR's performance is impacted by parameters at the server side (i.e., various encoding specifications), at the network side (i.e., packet loss rate), and at the client side (i.e., buffer size) on the performance of H2BR; (ii) how H2BR outperforms other state-of-the-art approaches in different configurations of the parameters above; (iii) how to effectively utilize H2BR on top of ABR algorithms in various streaming scenarios. The experimental results show that H2BR's performance increases with the buffer size and decreases with increasing packet loss rates and/or video segment duration. The number of quality levels can negatively or positively impact on H2BR's performance, depending on the ABR algorithm deployed. In general, H2BR is able to enhance the video quality by up to 17% and 14% in scalable video streaming and in non-scalable video streaming, respectively. Compared with an existing retransmission technique (i.e., SQUAD Wang et al., ACM Trans Multimed Comput Commun Applic (TOMM) 13(3s): 45, 49), H2BR shows better results with more than 10% in QoE and 9% in the average video quality. [ABSTRACT FROM AUTHOR]

Magnetic Resonance Imaging (MRI) is a critical tool in the detection and characterization of brain tumors, offering precise information on tumor location, size, and properties. This information is vital for healthcare professionals to devise personalized treatment plans and select suitable therapeutic approaches while preserving neurological functions. Presented here is an innovative AI-based MRI tumor diagnosis system, accessible to a wide range of medical professionals including radiologists, specialists, and registrars. The system integrates advanced web service applications, providing an improved user experience. The diagnosis system is built utilizindg a Y-Net architecture with transfer learning, demonstrating a classification loss of 0.2152 and an accuracy rate of 0.9075. Remarkably, for segmentation tasks, the system achieves an accuracy of 0.9983, a dice coefficient of 0.9036, and an Intersection over Union (IoU) of 0.8225. Hosted on Amazon's cloud servers, the system is readily available to medical center personnel, ensuring widespread accessibility. For deployment on the Amazon cloud, a virtual machine running the Ubuntu Linux distribution was established, and the training model was successfully deployed online. To bolster the reliability of interactions among various users, the deployment of HTTP/2 for internet connections is recommended. This protocol significantly enhances the handling of substantial MRI data, fostering improved transfer speeds and effective collaboration in the diagnosis and treatment of brain tumors. This implementation culminates in a sophisticated Picture Archiving and Communication System (PACS) with cloud-based functionalities for the storage, retrieval, management, and distribution of medical images and associated patient data. The MRI PACS system guarantees precise and error-minimized interactions among users, excelling in data interchange speed and accuracy. Simultaneously, the system's core AI functionalities, including segmentation, classification, localization, and tumor area calculation, are executed flawlessly, ensuring integrity and no data loss or corruption. [ABSTRACT FROM AUTHOR]

New versions of HTTP protocols have been developed to overcome many of the limitations of the original HTTP/1.1 protocol and its underlying transport mechanism over TCP. In this paper, we investigated the performance of modern Internet protocols such as HTTP/2 over TCP and HTTP/3 over QUIC in high-latency satellite links. The goal was to uncover the interaction of the new features of HTTP such as parallel streams and optimized security handshake with modern congestion control algorithms such as CUBIC and BBR over high-latency links. An experimental satellite network emulation testbed was developed for the evaluation. The study analyzed several user-level web performance metrics such as average page load time, First Contentful Paint and Largest Contentful Paint. The results indicate an overhead problem with HTTP/3 that becomes more significant when using a loss-based congestion control algorithm such as CUBIC which is widely used on the Internet. Also, the results highlight the significance of the web page structure and how objects are distributed in it. Among the various Internet protocols evaluated, the results show that HTTP/3 over QUIC will perform better by an average of 35% than HTTP/2 over TCP in satellites links specifically with a more aggressive congestion algorithm such as BBR. This can be attributed to the non-blocking stream multiplexing feature of QUIC and the reduced TLS handshake of HTTP/3. [ABSTRACT FROM AUTHOR]

The extensive use of the internet has resulted in novel technologies and protocol improvisation. Hypertext transfer protocol/1.1 (HTTP/1.1) is widely adapted on the internet. However, HTTP/2 is found to be more efficient over transport control protocol (TCP). The HTTP/2 protocol can withstand the payload overhead when compared to HTTP/1.1 by multiplexing multiple requests. However, both the protocols are highly susceptible to applicationlevel denial of service (DoS) attacks. In this research, a slow-rate DoS attack called Slowloris is detected over Apache2 servers enabled with both versions of HTTP in traditional networks and software defined networks (SDN). Server metrics such as server connection time to the webpage, latency in receiving a response from the server, page load time, response-response gap, and inter-packet arrival time at the server are monitored to analyze attack activity. A Monte Carlo simulation is used to estimate threshold values for server connection time and latency for attack detection. This work is implemented in a lab environment using virtual machines, Ryu controller, zodiac FX OpenFlow switch and Apache2 servers. This study also highlights SDN's security benefits over traditional networks. [ABSTRACT FROM AUTHOR]

With the inevitable dominance of video traffic on the Internet, Internet service providers (ISP) are striving to deliver video streaming with high quality. Video resolution, as a direct reflection of video quality, is a key factor of the video quality of experience (QoE). Since the displayed information of video cannot be observed by ISPs, ISPs can only measure the video resolution from traffic. However, with HTTP/2 being gradually adopted in video services, the multiplexing feature of HTTP/2 allows audio and video chunks to be mixed during transmission, making existing monitoring approaches unusable. In this article, we propose a method called H2CI to monitor resolution for adaptive encrypted video traffic under HTTP/2. We consider the size of the mixed data for identification. Specifically, H2CI consists of a length restoration method to extract restored fingerprints and a fingerprint-matching method for fine-grained resolution identification. The experimental results show that H2CI can achieve more than 98% accuracy for fine-grained resolution identification. Our method can be effectively applied to infer the adaptation behavior of encrypted video streaming and monitor the QoE of video services under HTTP/2. [ABSTRACT FROM AUTHOR]

Applying 4K, (Ultra HD) Real-time video streaming via the internet network, with low bitrate and low latency, is the challenge this paper addresses. Compression technology and transfer links are the important elements that influence video quality. So, to deliver video over the internet or another fixed capacity medium, it is essential to compress the video to more controllable bitrates (customarily in the 1-20 Mbps range). In this study, the video quality is examined using the H.265/HEVC compression standard, and the relationship between quality of video and bitrate flow is investigated using various constant rate factors, GOP patterns, quantization parameters, RC-lookahead, and other types of video motion sequences. The ultra)high-definition video source is used, down sampled and encoded at multiple resolutions of (3480x2160), (1920x1080), (1280x720), (704x576), (352x288), and (176x144). To determine the best H265 feature configuration for each resolution experiments were conducted that resulted in a PSNR of 36 dB at the specified bitrate. The resolution is selected by delivery (encoder resource) based on the end)user application. While video streaming adapted to the available bandwidth is achieved via embedding a controller with MPEG DASH protocol at the client-side. Video streaming Adaptation methods allow the delivery of content that is encoded at different representations of video quality and bitrate and then dividing each representation into chunks of time. Through this paper, we propose to utilize HTTP/2 as a protocol to achieve low latency video streaming focusing on live streaming video avoiding the problem of HTTP/1. [ABSTRACT FROM AUTHOR]

The use of robust security solutions is a must for the Internet of Things (IoT) devices and their applications: regulators in different countries are creating frameworks for certifying those devices with an acceptable security level. However, even for already certified devices, security protocols have to be updated when a breach is found or a certain version becomes obsolete. Many approaches for securing IoT applications are nowadays based on the integration of a security layer [e.g., using transport layer security, (TLS)], but this may result in difficulties when upgrading the security algorithms, as the whole application has to be updated. This fact may shorten the life of IoT devices. As a way to overcome these difficulties, this paper presents IoTsafe, a novel approach relying on secure socket shell (SSH), a feasible alternative to secure communications in IoT applications based on hypertext transfer protocol (HTTP and HTTP/2). In order to illustrate its advantages, a comparison between the traditional approach (HTTP with TLS) and our scheme (HTTP with SSH) is performed over low-power wireless personal area networks (6loWPAN) through 802.15.4 interfaces. The results show that the proposed approach not only provides a more robust and easy-to-update solution, but it also brings an improvement to the overall performance in terms of goodput and energy consumption. Core server stress tests are also presented, and the server performance is also analyzed in terms of RAM consumption and escalation strategies.

Distributed Denial of Service (DDoS) attacks using the HTTP protocol have started gaining popularity in recent years. A recent trend in this direction has been the use of computationally expensive requests to launch attacks. These attacks, called Asymmetric Workload attacks can bring down servers using limited resources, and are extremely difficult to detect. The introduction of HTTP/2 has been welcomed by developers because it improves user experience and efficiency. This was made possible by the ability to transport HTTP requests and their associated inline resources simultaneously by using Multiplexing and Server Push. However multiplexing has made request traffic bursty and rendered DDoS detection mechanisms based on connection limiting obsolete. Contrary to its intention, multiplexing can also be misused to launch sophisticated DDoS attacks using multiple high workload requests in a single TCP connection. However, sufficient research has not been done in this area. Existing research demonstrates that the HTTP/2 protocol allows users to launch DDoS attacks easily, but does not focus on whether an HTTP/2 server can handle DDoS attacks more efficiently or not. Also, sufficient research has not been done on the possibility of Multiplexing and Server Push being misused. In this work, we analyse the performance of an HTTP/2 server compared to an HTTP/1.1 server under an Asymmetric DDoS attack for the same load. We propose a new DDoS attack vector called a Multiplexed Asymmetric DDoS attack, which uses multiplexing in a different way than intended. We show that such an attack can bring down a server with just a few attacking clients. We also show that a Multiplexed Asymmetric Attack on a server with Server Push enabled can trigger an egress network layer flood in addition to an application layer attack. [ABSTRACT FROM AUTHOR]

HTTP/2 video streaming has gotten a lot of attention in the development of multimedia technologies over the last few years. In HTTP/2, the server push mechanism allows the server to deliver more video segments to the client within a single request in order to deal with the requests explosion problem. As a result, recent research efforts have been focusing on utilizing such a feature to enhance the streaming experience while reducing the request-related overhead. However, current works only optimize the performance of a single client, without necessary concerns of possible influences on other clients in the same network. When multiple streaming clients compete for a shared bandwidth in HTTP/1.1, they are likely to suffer from unfairness, which is defined as the inequality in their bitrate selections. For HTTP/1.1, existing works have proven that the network-assisted solutions are effective in solving the unfairness problem. However, the feasibility of utilizing such an approach for the HTTP/2 server push has not been investigated. Therefore, in this paper, a novel proxy-based framework is proposed to overcome the unfairness problem in adaptive streaming over HTTP/2 with the server push. Experimental results confirm the outperformance of the proposed framework in ensuring the fairness, assisting the clients with avoiding rebuffering events and lowering bitrate degradation amplitude, while maintaining the mechanism of the server push feature. [ABSTRACT FROM AUTHOR]

Dynamic adaptive streaming (DAS) over HTTP has been widely deployed over the Internet. However, due to the pull-based nature of HTTP/1.1, there exists intolerable streaming latency and high request overhead in the current DAS systems. With dynamic k-push, HTTP/2 live streaming promises to achieve low live latency with less overhead and small segment duration. In this paper, we propose a quality of experience (QoE) driven adaptive k-push mechanism (QK-Push) for HTTP/2 live streaming. The client just sends one request to set push length ($K$) and bitrate (v) parameters and the server would push back $K$ segments in a batch. To determine k-push parameters, a probabilistic buffer model is first designed to avoid buffer underflow/overflow. Also, three QoE objective functions are designed to ensure the high streaming quality (bitrate), playback continuity, and smoothness. QK-Push casts this multi-objective optimization problem as a Pareto optimal problem. To solve it, a Nash bargaining solution is designed to balance the needs for video quality, bitrate smoothness, and request overhead. Finally, the segments in each push cycle are selected by solving the Nash problem with a discrete space Lagrangian method. We implement an HTTP/2 live streaming prototype system, with the QK-Push algorithm over modified dash.js and media presentation description. To evaluate the performances, the extensive live streaming experiments are carried out over a controllable network test bed and real Internet trace. The results demonstrate that the proposed QK-Push algorithm is able to improve the average bitrate up to 13%, reduce the bitrate oscillations up to 81%, decrease the startup delay up to 58%, and increase the estimate the mean opinion score up to 12% compared to the current HTTP/1.1 system. [ABSTRACT FROM AUTHOR]

HTTP/2 video streaming has gotten a lot of attention in the development of multimedia technologies over the last few years. In HTTP/2, the server push mechanism allows the server to deliver more video segments to the client within a single request in order to deal with the requests explosion problem. As a result, recent research efforts have been focusing on utilizing such a feature to enhance the streaming experience while reducing the request-related overhead. However, current works only optimize the performance of a single client, without necessary concerns of possible influences on other clients in the same network. When multiple streaming clients compete for a shared bandwidth in HTTP/1.1, they are likely to suffer from unfairness, which is defined as the inequality in their bitrate selections. For HTTP/1.1, existing works have proven that the network-assisted solutions are effective in solving the unfairness problem. However, the feasibility of utilizing such an approach for the HTTP/2 server push has not been investigated. Therefore, in this paper, a novel proxy-based framework is proposed to overcome the unfairness problem in adaptive streaming over HTTP/2 with the server push. Experimental results confirm the outperformance of the proposed framework in ensuring the fairness, assisting the clients with avoiding rebuffering events and lowering bitrate degradation amplitude, while maintaining the mechanism of the server push feature.

HTTP adaptive streaming (HAS) has become popular for delivering videos over the Internet. However, most HAS services are currently based on the traditional HTTP/1.1, which has limitations for live streaming. Meanwhile, the new HTTP/2 was standardized in 2015 with many improvements. In this letter, we present a novel HTTP/2-based adaptation scheme for low-delay live streaming over mobile networks. The proposed scheme leverages two HTTP/2 features called server push and stream termination to enable a shorter camera-to-display delay. Experimental results with small buffer sizes show that our scheme can provide users with high video bitrate and stable buffer level. Furthermore, the buffer size in live streaming can be reduced to two sec without sacrificing video quality. [ABSTRACT FROM AUTHOR]

Over the last years, streaming of multimedia content has become more prominent than ever. To meet increasing user requirements, the concept of HTTP Adaptive Streaming (HAS) has recently been introduced. In HAS, video content is temporally divided into multiple segments, each encoded at several quality levels. A rate adaptation heuristic selects the quality level for every segment, allowing the client to take into account the observed available bandwidth and the buffer filling level when deciding the most appropriate quality level for every new video segment. Despite the ability of HAS to deal with changing network conditions, a low average quality and a large camera-to-display delay are often observed in live streaming scenarios. In the meantime, the HTTP/2 protocol was standardized in February 2015, providing new features which target a reduction of the page loading time in web browsing. In this paper, we propose a novel push-based approach for HAS, in which HTTP/2’s push feature is used to actively push segments from server to client. Using this approach with video segments with a sub-second duration, referred to as super-short segments, it is possible to reduce the startup time and end-to-end delay in HAS live streaming. Evaluation of the proposed approach, through emulation of a multi-client scenario with highly variable bandwidth and latency, shows that the startup time can be reduced with 31.2% compared to traditional solutions over HTTP/1.1 in mobile, high-latency networks. Furthermore, the end-to-end delay in live streaming scenarios can be reduced with 4 s, while providing the content at similar video quality. [ABSTRACT FROM AUTHOR]

Responsiveness is a critical metric for web performance. Recent work in the Internet Engineering Task Force (IETF) and World Wide Web Consortium (W3C) has resulted in a new set of web protocols, including definition of the hypertext transfer protocol version 2 (HTTP/2) and a corresponding set of TCP updates. Together, these have been designed to reduce the web page download latency compared with HTTP/1.1. This paper describes the main features of the new protocols and discusses the impact of path delay on their expected performance. It then presents a set of tests to evaluate whether current implementations of the new protocols can offer benefit with an operational satellite access network and suggests how the specifications can evolve and be tuned to further enhance performance for long network paths. Copyright © 2016 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

Mobile Cloud Networks group together mobile users and clouds containing content servers. Hence, they are an ideal framework for media content delivery. Streamswitching adaptive video players cope well with some limitations of Mobile Cloud Networks as low bandwidth and bandwidth variability in access network. Nonetheless, other limitations, as cloud congestion, are difficult to be managed by the video players. This paper presents a system for discovering fault situations at the cloud (e.g., cloud congestion) and notifying to the video player, which will take appropriate actions for saving the quality of media transmission. In proposed implementation the video application is DASH-capable and adaptation action may be both stream rate adaptation and content server adaptation. The communication between client and server uses “bidirectional” communication feature of HTTP/2 thanks to the new deployed modules running DASH over HTTP/2 in both client’s and server’s applications.

In HTTP Adaptive Streaming, video content is temporally divided into multiple segments, each encoded at several quality levels. The client can adapt the requested video quality to network changes, generally resulting in a smoother playback. Unfortunately, live streaming solutions still often suffer from playout freezes and a large end-to-end delay. By reducing the segment duration, the client can use a smaller temporal buffer and respond even faster to network changes. However, since segments are requested subsequently, this approach is susceptible to high round-trip times. In this letter, we discuss the merits of an HTTP/2 push-based approach. We present the details of a measurement study on the available bandwidth in real 4G/LTE networks, and analyze the induced bit-rate overhead for HEVC-encoded video segments with a sub-second duration. Through an extensive evaluation with the generated video content, we show that the proposed approach results in a higher video quality (+7.5%) and a lower freeze time (−50.4%), and allows to reduce the live delay compared with traditional solutions over HTTP/1.1. [ABSTRACT FROM PUBLISHER]

HTTP/2 is the second major version of the HTTP protocol published by the internet engineering steering group. The protocol is designed to improve reliability and performance Such enhancements have thus delineated the protocol as being more vulnerable to distributed denial-of-service (DDoS) attacks when compared to its predecessor. Recent phenomenon showed that legitimate traffic or flash crowds could have high-traffic flow characteristics as seen in DDoS attacks. In this paper, we demonstrate that legitimate HTTP/2 flash crowd traffic can be launched to cause denial of service. To the best of our knowledge, no previous study has been conducted to analyse the effect of both DDoS as well as flash crowd traffic against HTTP/2 services. Results obtained prove the effect of such attacks when tested under four varying protocol-dependant attack scenarios. [ABSTRACT FROM AUTHOR]

Mobile Cloud Networks group together mobile users and clouds containing content servers. Hence, they are an ideal framework for media content delivery. Streamswitching adaptive video players cope well with some limitations of Mobile Cloud Networks as low bandwidth and bandwidth variability in access network. Nonetheless, other limitations, as cloud congestion, are difficult to be managed by the video players. This paper presents a system for discovering fault situations at the cloud (e.g., cloud congestion) and notifying to the video player, which will take appropriate actions for saving the quality of media transmission. In proposed implementation the video application is DASH-capable and adaptation action may be both stream rate adaptation and content server adaptation. The communication between client and server uses \bidirectional" communication feature of HTTP/2 thanks to the new deployed modules running DASH over HTTP/2 in both client's and server's applications. [ABSTRACT FROM AUTHOR]

The continuing advances in computer graphics and Internet bandwidths are supporting a gradual convergence between multi-user virtual worlds (MUVW), such as Second Life and OpenSim (SL/OS), and the nascent 3D Web. However, significant networking barriers remain to exploiting these capabilities for developing the 3D Web. These barriers include latency of content update and firewall blocking. In MUVWs the firewall and latency problems are related as the SL/OS network protocols designed over twelve years ago sought to minimise latency through the use of multiple concurrent UDP-based virtual circuits. Most firewall administrators are loathe to open up over fifty unknown UDP ports to accommodate such applications. New protocols now being deployed on the web such as SPDY, HTTP/2 and QUIC seek to reduce latency and routinely traverse firewalls. One of the key goals of the convergence between MUVWs and the 3D Web is for MUVW functionality to be provided in a standard web browser, with optional links to other autonomous virtual worlds. It follows that as an incremental step towards the 3D Web the use of these new web protocols in MUVWs should be researched. This paper details traffic management approaches in SL/OS MUVWs, clarifies 3D Web concepts and terminology, explains the functionality provided by the new web protocols, and provides a mapping which postulates how their features can be exploited for the benefit of MUVWs as part of the convergence with the 3D Web. [ABSTRACT FROM AUTHOR]

The introduction of HTTP/2 has led to a dramatic change in web traffic. The steady flow of requests in HTTP/1.1 has been replaced by bursts of multiple requests, largely due to the introduction of multiplexing in HTTP/2 which allows users to send multiple requests through a single connection. This feature was introduced in order to reduce the page loading time by multiplexing a web page and its associated resources in a single connection. While this feature has significantly improved user experience, it can be misused to launch sophisticated application layer DDoS attacks against HTTP/2 servers. Instead of the intended use of multiplexing, attackers can force the web server to process multiple random requests simultaneously, leading to increased server usage. The use of computationally intensive requests can further exacerbate the situation. These attacks, called Multiplexed Asymmetric Attacks, pose a dangerous threat to HTTP/2 servers and stem from the lack of verification of the multiplexed requests. In this work, an approach to model an HTTP/2 request set as a fuzzy multiset is presented. The proposed approach uses a combination of relative cardinality and request workload to detect multiplexed AL-DDoS attacks. Experiments on open source datasets demonstrate that the proposed approach is able to detect multiplexed AL-DDoS attacks with an accuracy of around 95%, while maintaining a low False Positive Rate (FPR) of around 3%. • Proposes a detection mechanism for multiplexed asymmetric attacks on HTTP/2 servers. • Describes the concept of request sets and valid request sets. • Models an HTTP/2 request set as a Fuzzy Multiset. • Proposes a quantitative approach for describing the validity of a request set. • Demonstrates the efficiency of the proposed detection mechanism. [ABSTRACT FROM AUTHOR]