AI-based colon polyp detection systems have received much attention, and several products and prototypes report good results. In silico verification is a crucial step when developing such systems, but very few compare human versus AI performance. This paper, therefore, describes methods and results for an in silico test of an AI model with two different versions for polyp detection in colonoscopy and compares them to the performance of endoscopist doctors who reviewed the same colonoscopy video clips. The two versions have different thresholds for false positive rate reduction. Our models perform polyp detection within the range of the endoscopists' performance, although faster, showing a potential for use in a clinical setting. For the AI and the endoscopists alike, the results show a trade-off between high sensitivity and high specificity; to achieve perfect detection, one will also get abundance of false positives. This can cause alarm fatigue in a clinical setting.

The BioMedia 2019 ACM Multimedia Grand Challenge is the first in a series of competitions focusing on the use of multimedia for different medical use-cases. In this year's challenge, the participants are asked to develop efficient algorithms which automatically detect a variety of findings commonly identified in the gastrointestinal (GI) tract (a part of the human digestive system). The purpose of this task is to develop methods to aid medical doctors performing routine endoscopy inspections of the GI tract. In this paper, we give a detailed description of the four different tasks of this year's challenge, present the datasets used for training and testing, and discuss how each submission is evaluated both qualitatively and quantitatively.

With 360° panorama video technology becoming commonplace, the need for efficient streaming methods for such videos arises. We go beyond the existing on-demand solutions and present a live streaming system which strikes a trade-off between bandwidth usage and the video quality in the user’s field-of-view. We have created an architecture that combines RTP and DASH to deliver 360° VR content to a Huawei set-top-box and a Samsung Galaxy S7. Our system multiplexes a single HEVC hardware decoder to provide faster quality switching than at the traditional GOP boundaries. We demonstrate the performance and illustrate the trade-offs through real-world experiments where we can report comparable bandwidth savings to existing on-demand approaches, but with faster quality switches when the field-of-view changes.

This demo dramatically illustrates how replacing 'Classic' TCP congestion control (Reno, Cubic, etc.) with a 'Scalable' alternative like Data Centre TCP (DCTCP) keeps queuing delay ultra-low; not just for a select few light applications like voice or gaming, but even when a variety of interactive applications all heavily load the same (emulated) Internet access. DCTCP has so far been confined to data centres because it is too aggressive---it starves Classic TCP flows. To allow DCTCP to be exploited on the public Internet, we developed DualQ Coupled Active Queue Management (AQM), which allows the two TCP types to safely co-exist. Visitors can test all these claims. As well as running Web-based apps, they can pan and zoom a panoramic video of a football stadium on a touch-screen, and experience how their personalized HD scene seems to stick to their finger, even though it is encoded on the fly on servers accessed via an emulated delay, representing 'the cloud'. A pair of VR goggles can be used at the same time, making a similar point. The demo provides a dashboard so that visitors can not only experience the interactivity of each application live, but they can also quantify it via a wide range of performance stats, updated live. It also includes controls so visitors can configure different TCP variants, AQMs, network parameters and background loads and immediately test the effect.

Time-dependent applications using TCP often send thin-stream traffic, characterised by small packets and high inter-transmission-times. Retransmissions after packet loss can result in very high delays for such flows as they often cannot trigger fast retransmit. Redundant Data Bundling is a mechanism that preempts the experience of loss for a flow by piggybacking unacknowledged segments with new data as long as the total packet size is lower than the flow maximum segment size. Although successful at reducing retransmission latency, this mechanism had design issues leaving it open for abuse, effectively making it unsuitable for general Internet deployment. In this paper, we have redesigned the RDB mechanism to make it safe for deployment. We improve the trigger for when to apply it and evaluate its fairness towards competing traffic. Extensive experimental results confirm that our proposed modifications allows for inter-flow fairness while maintaining the significant latency reductions from the original RDB mechanism.

All computer games present some delay between human input and results being displayed on the screen, even when no networking is involved. A well-balanced discussion of delay-tolerance levels in computer games requires an understanding of how much delay is added locally, as well as in the network. This demonstration uses a typical gaming setup wired to an oscilloscope to show how long the total, local delay is. Participants may also bring their own computers and games so they can measure delays in the games or other software.

Data-bundling is a useful technique that decreases the delivery delay of packet streams when they are transmitted over noisy channels and are subject to retransmission-based error control. In this paper, we investigate the packet delay statistics for a fully reliable selective repeat automatic repeat request (SR ARQ) where a data-bundling mechanism is employed. In more detail, we discuss a model for data-bundling to analyze the SR ARQ mechanism over wireless channels based on Markov chains. We evaluate various channel error distributions and analyze the buffer occupancy to check if the data-bundling mechanism provides efficient results. We further analyze the queueing, delivery and overall delay statistics at link layer. We found that using data-bundling can improve the delay performance of the SR ARQ mechanism, especially when bursty channels with heavily correlated errors are considered. Thus, this technique can bring useful improvements for real-time services, multimedia, and other delay-sensitive applications over wireless networks.

A recent proposed update to TCP congestion control, TCP-newCWV, has targeted congestion control for rate-limited applications. These methods need to be explored in the context of rate-adaptive applications, such as DASH. The new method enables a client to exploit the persistence of a DASH connection and enables the DASH server to rapidly resume transmission of a series of video segments using a single TCP connection. Another technique, called 'Pacing' smoothes DASH bursts when there is no TCP ACK clock, and is shown to significantly reduce burst loss. These two methods in combination can increase the application performance. This paper investigates the effect of implementing these techniques on a DASH flow in different congestion scenarios and whether the method can promote better capacity sharing while minimizing the latency experienced by other flows sharing a common network bottleneck. The results confirm that newCWV with Pacing provides a benefit as a platform for DASH transport.

We present a dataset -- a real-world, server-side packet trace [8] -- from Anarchy Online [9]. Anarchy Online is a science fiction-themed massively multiplayer online roleplaying game (MMORPG), published and developed by Funcom [7]. We present statistics from the network traffic and show that it is a representative dataset for similar games. From the dataset, one can extract several key characteristics from such scenarios like payload sizes, packet rates, data delivery latencies, retransmission statistics, loss rates and stream correlation. The dataset can be used several ways: by replaying the game traffic, components like congestion control mechanisms, middlewares, packet schedulers, router queue behaviour, etc. can be analysed. Based on the observed statistics from the trace, such interactive game traffic shows completely different behaviour compared to the greedy, high-rate streams most network mechanisms are designed for, e.g., file download, video streaming and web-surfing. We hope that the dataset can be used to push research forward in the field of system support for games.

Consumer broadband services are increasingly a mix of TCP-based and UDP-based applications, often with quite distinct requirements for interactivity and network performance. Consumers can experience degraded service when application traffic collides at a congestion point between home LANs, service provider edge networks and fractional-Mbit/sec `broadband' links. We illustrate two key issues that arise from the impact of TCP-based data transfers on real-time traffic (such as VoIP or online games) sharing a broadband link. First, well-intentioned modifications to traditional TCP congestion control can noticeably increase the latencies experienced by VoIP or online games. Second, superficially-similar packet dropping rules in broadband gateways can induce distinctly different packet loss rates in VoIP and online game traffic. Our observations provide cautionary guidance to researchers who model such traffic mixes, and to vendors implementing equipment at either end of consumer links.

No abstract available.

A well known challenge with mobile video streaming is fluctuating bandwidth. As the client devices move in and out of network coverage areas, the users may experience varying signal strengths, competition for the available resources and periods of network outage. These conditions have a significant effect on video quality.

A large number of network services rely on IP and reliable transport protocols. For applications that provide abundant data for transmission, loss is usually handled satisfactorily, even if the application is latency-sensitive (Wang et al. 2004 ). For data streams where small packets are sent intermittently, however, applications can occasionally experience extreme latencies (Griwodz and Halvorsen 2006 ). As it is not uncommon that such thin-stream applications are time-dependent, any unnecessarily induced delay can have severe consequences for the service provided. Massively Multiplayer Online Games (MMOGs) are a defining example of thin streams . Many MMOGs (like World of Warcraft and Age of Conan) use TCP for the benefits of reliability, in-order delivery and NAT/firewall traversal. It has been shown that TCP has several shortcomings with respect to the latency requirements of thin streams because of the way it handles retransmissions (Griwodz and Halvorsen 2006 ). As such, an alternative to TCP may be SCTP (Stewart et al. 2000 ), which was originally developed to meet the requirements of signaling transport. In this paper, we evaluate the Linux-kernel SCTP implementation in the context of thin streams. To address the identified latency challenges, we propose sender-side only enhancements that reduce the application-layer latency in a manner that is compatible with unmodified receivers. These enhancements can be switched on by applications and are used only when the system identifies the stream as thin. To evaluate the latency performance, we have performed several tests over various real networks and over an emulated network, varying parameters like RTT, packet loss and amount of competing cross traffic. When comparing our modifications with SCTP on Linux and FreeBSD and TCP New Reno, our results show great latency improvements and indicate the need for a separate handling of thin and thick streams.

A wide range of Internet-based services that use reliable transport protocols display what we call thin-stream properties. This means that the application sends data with such a low rate that the retransmission mechanisms of the transport protocol are not fully effective. In time-dependent scenarios where the user experience depends on the data delivery latency, packet loss can be devastating for the service quality. In order to reduce application-layer latency when packets are lost, we have implemented modifications to the TCP retransmission mechanisms in the Linux kernel. The changes are only active when thin-stream properties are detected, thus not affecting TCP behaviour when the stream is not thin. In this paper, we show the latency improvements from these thin-stream modifications. We have tested several thin-stream applications like remote terminals (SSH) and audio conferencing (Skype), and we evaluate the user experience with and without the TCP modifications. Our experimental results show that our modifications allow TCP to recover earlier from packet loss. Furthermore, user surveys indicate that the majority of users easily detect improvements in the perceived quality of the tested applications.

TCP is frequently used for interactive multimedia applications like online games and voice-over-IP (VoIP) because it avoids firewall issues. However, traffic analysis shows that these streams usually have small packets and a low packet rate, and that in case of loss, severe latency penalties occur for all existing TCP variations in Linux [5]. In this demonstration, we show how small TCP enhancements greatly improve the perceived quality of such low latency, interactive applications.