Best-effort Internet Research Articles

The Effects of Network Outages on User Experience in Augmented Reality Based Remote Collaboration - An Empirical Study

Augmented Reality (AR) applications can enable geographically distant users to collaborate using shared video feeds or interactive 3D holograms, and may be particularly useful in the socially distant context of the Covid-19 pandemic. However, a good user experience is key for their success and could be negatively impacted by network impairments, which are an inevitable occurrence in today's best-effort Internet. In this paper, we present the findings of an empirical user study, aimed at understanding the effects of network outages, on user experience and behavior, in a collaborative AR task. We highlight how network outages affected users in different ways depending on their role in the collaborative task, and how giving users explicit information about poor network conditions helped them deal with some of these negative effects. Furthermore, we report the strategies that users themselves adopted, to deal with outages, such as batching instructions, or shifting to a different spatial referencing style when communicating with their partners. Lastly, based on our findings, we present some design implications for future remote-collaborative AR applications.

QoE Management of Multimedia Streaming Services in Future Networks: A Tutorial and Survey

We provide in this paper a tutorial and a comprehensive survey of QoE management solutions in current and future networks. We start with a high level description of QoE management for multimedia services, which integrates QoE modelling, monitoring, and optimization. This followed by a discussion of HTTP Adaptive Streaming (HAS) solutions as the dominant technique for streaming videos over the best-effort Internet. We then summarize the key elements in SDN/NFV along with an overview of ongoing research projects, standardization activities and use cases related to SDN, NFV, and other emerging applications. We provide a survey of the state-of-the-art of QoE management techniques categorized into three different groups: a) QoE-aware/driven strategies using SDN and/or NFV; b) QoE-aware/driven approaches for adaptive streaming over emerging architectures such as multi-access edge computing, cloud/fog computing, and information-centric networking; and c) extended QoE management approaches in new domains such as immersive augmented and virtual reality, mulsemedia and video gaming applications. Based on the review, we present a list of identified future QoE management challenges regarding emerging multimedia applications, network management and orchestration, network slicing and collaborative service management in softwarized networks. Finally, we provide a discussion on future research directions with a focus on emerging research areas in QoE management, such as QoE-oriented business models, QoE-based big data strategies, and scalability issues in QoE optimization.

A Survey on 360° Video Streaming

Head-mounted displays and 360° videos have become increasingly more popular, delivering a more immersive viewing experience to end users. Streaming 360° videos over the best-effort Internet, however, faces tremendous challenges, because of the high resolution and the short response time requirements. This survey presents the current literature related to 360° video streaming. We start with 360° video streaming systems built for real experiments to investigate the practicality and efficiency of 360° video streaming. We then present the video and viewer datasets, which may be used to drive large-scale simulations and experiments. Different optimization tools in various stages of the 360° video streaming pipeline are discussed in detail. We also present various applications enabled by 360° video streaming. In the appendices, we review the off-the-shelf hardware available at the time of writing and the open research problems.

Multi-Party WebRTC Services Using Delay and Bandwidth Aware SDN-Assisted IP Multicasting of Scalable Video Over 5G Networks

At present, multi-party WebRTC videoconferencing between peers with heterogenous network resources and terminals is enabled over the best-effort Internet using a central selective forwarding unit (SFU), where each peer sends a scalable encoded video stream to the SFU. This connection model avoids the upload bandwidth bottleneck associated with mesh connections; however, it increases peer delay and overall network load (resource consumption) in addition to requiring investment in servers since all video traffic must go through SFU servers. To this effect, we propose a new multi-party WebRTC service model over future 5G networks, where a video service provider (VSP) collaborates with a network service providers (NSP) to offer an NSP-managed service to stream scalable video layers using software-defined networking (SDN)-assisted Internet protocol (IP) multicasting between peers using NSP infrastructure. In the proposed service model, each peer sends a scalable coded video upstream, which is selectively duplicated and forwarded as layer streams at SDN switches in the network, instead of at a central SFU, in a multi-party WebRTC session managed by multicast trees maintained by the SDN controller. Experimental results show that the proposed SDN-assisted IP multicast service architecture is more efficient than the SFU model in terms of end-to-end service delay and overall network resource consumption, while avoiding peer upload bandwidth bottleneck and distributing traffic more evenly across the network. The proposed architecture enables efficient provisioning of premium managed WebRTC services over bandwidth-reserved SDN slices to provide videoconferencing experience with guaranteed video quality over 5G networks.

How Telecommunication Regulations Can Contribute to the Alleviation of Traffic Explosion in the Japanese NGN

Internet traffic is increasing explosively throughout the world, causing congestion. This study aims to find policy solutions for carriers and ISPs in Japan. In 2008, to cope with expected increasing traffic, richer contents, and to meet the various needs of content providers, NTT locals initiated an NGN (Next Generation Network) service, which enables rapid and large-volume data transmission. QoS can also be guaranteed. Four categories of connections, according to service quality, have been put into operation, these being (i) videophones, (ii) broadcasting, including 4K and 8K, (iii) VoIP, and (iv) ordinal best-effort type Internet access. Most of the traffic over NGN is (iv) best-effort type. NTT locals prepared two types of connection to NGN; PPPoE (Point-to-Point protocol over Ethernet) and IPoE (IP over Ethernet). The former is mainly used by small local ISPs, whereas the latter is used by VNEs (Virtual Network Enablers) and large ISPs. The point of connection to NGN is thought to be a cause of congestion. For ISPs to connect to NGN, they must accept the rules set by NTT locals. Accordingly, smooth access for IPSs to the NGN network became matters of policy. This study proposes (i) fair rules for ISPs to connect to NGN; and (ii) a scheme of connection charges based on service quality. NGN should be for all entities to access and utilize for various applications leading to new economies. This study proposes how NGN can avoid congestion and be utilized fully and wisely as the age of IoT and 5G approaches.

Quality of Experience-Centric Management of Adaptive Video Streaming Services

Video streaming applications currently dominate Internet traffic. Particularly, HTTP Adaptive Streaming (HAS) has emerged as the dominant standard for streaming videos over the best-effort Internet, thanks to its capability of matching the video quality to the available network resources. In HAS, the video client is equipped with a heuristic that dynamically decides the most suitable quality to stream the content, based on information such as the perceived network bandwidth or the video player buffer status. The goal of this heuristic is to optimize the quality as perceived by the user, the so-called Quality of Experience (QoE). Despite the many advantages brought by the adaptive streaming principle, optimizing users’ QoE is far from trivial. Current heuristics are still suboptimal when sudden bandwidth drops occur, especially in wireless environments, thus leading to freezes in the video playout, the main factor influencing users’ QoE. This issue is aggravated in case of live events, where the player buffer has to be kept as small as possible in order to reduce the playout delay between the user and the live signal. In light of the above, in recent years, several works have been proposed with the aim of extending the classical purely client-based structure of adaptive video streaming, in order to fully optimize users’ QoE. In this article, a survey is presented of research works on this topic together with a classification based on where the optimization takes place. This classification goes beyond client-based heuristics to investigate the usage of server- and network-assisted architectures and of new application and transport layer protocols. In addition, we outline the major challenges currently arising in the field of multimedia delivery, which are going to be of extreme relevance in future years.

Performance Characterization of Low-Latency Adaptive Streaming From Video Portals

News-based websites and portals provide significant amounts of multimedia content to accompany news stories and articles. In this context, the HTTP adaptive streaming is generally used to deliver video over the best-effort Internet, allowing smooth video playback and an acceptable Quality Of Experience (QoE). To stimulate the user engagement with the provided content, such as browsing between videos, reducing the videos’ startup time has become more and more important: while the current median load time is in the order of seconds, research has shown that the user waiting times must remain below two seconds to achieve an acceptable QoE. In this paper, four complementary components are optimized and integrated into a comprehensive framework for low-latency delivery of news-related video content: 1) server-side encoding with short video segments; 2) HTTP/2 server push at the application layer; 3) server-side user profiling to identify relevant content for a given user; and 4) client-side storage to hold proactively delivered content. Using a large data set of a major Belgian news provider, containing millions of text- and video-based article requests, we show that the proposed framework reduces the videos’ startup time in different mobile network scenarios by over 50%, thereby improving the user interaction and skimming available content.

A Lightweight Fairness-Driven AQM for Regulating Bandwidth Utilization in Best-Effort Routers

The end-to-end congestion control mechanism of transmission control protocol (TCP) is critical to the robustness and fairness of the best-effort Internet. Since it is no longer practical to rely on end-systems to cooperatively deploy congestion control mechanisms, the network itself must now participate in regulating its own resource utilization. To that end, fairness-driven active queue management (AQM) is promising in sharing the scarce bandwidth among competing flows in a fair manner. However, most of the existing fairness-driven AQM schemes cannot provide efficient and fair bandwidth allocation while being scalable. This paper presents a novel fairness-driven AQM scheme, called CHORD (CHOKe with recent drop history) that seeks to maximize fair bandwidth sharing among aggregate flows while retaining the scalability in terms of the minimum possible state space and per-packet processing costs. Fairness is enforced by identifying and restricting high-bandwidth unresponsive flows at the time of congestion with a lightweight control function. The identification mechanism consists of a fixed-size cache to capture the history of recent drops with a state space equal to the size of the cache. The restriction mechanism is stateless with two matching trial phases and an adaptive drawing factor to take a strong punitive measure against the identified high-bandwidth unresponsive flows in proportion to the average buffer occupancy. Comprehensive performance evaluation indicates that among other well-known AQM schemes of comparable complexities, CHORD provides enhanced TCP goodput and intra-protocol fairness and is well-suited for fair bandwidth allocation to aggregate traffic across a wide range of packet and buffer sizes at a bottleneck router.

Securing the Industrial-Tactile Internet of Things With Deterministic Silicon Photonics Switches

Today’s best-effort (BE) Internet of Things (IoT) faces challenges in providing the end-to-end-performance, security, and energy efficiency needed for the Smart Systems of the 21st century. These future smart systems will include smart cities , smart transportation systems , and smart manufacturing . This paper surveys the security weaknesses of the BE IoT. The BE-IoT cannot be partitioned into distinct interference-free virtual networks , which compromises performance, cyber-security, and energy efficiency. The design of a secure deterministic industrial-tactile IoT core network, which can embed millions of distinct secure deterministic virtual networks (SD-VNs) in layer 2, is then presented. Deterministic communications, combined with low-jitter scheduling, offers several benefits: 1) the removal of all congestion, interference, and DOS attacks; 2) a significant reduction in IoT router buffer sizes; 3) a significant reduction in IoT energy use; 4) a reduction of end-to-end IoT delays to the speed of light in fiber; and 5) deterministic packet-switches are relatively easy to synthesize using FPGA technologies. These benefits apply to optical and 5G wireless networks. Future smart systems can reserve their own congestion-free SD-VNs in layer 2 to manage their traffic, with significantly improved performance, security, and energy efficiency. A speed-of-light deterministic IoT core network can transform cloud services in the 21st century by exploiting a new technology: FPGAs combined with silicon photonics transceivers to achieve terabits/second of optical bandwidth. To illustrate the transformational potential, Big Data green cloud computing over a secure deterministic IoT spanning the European Union is explored.

Coordinated computation and setup of multi-layer paths via inter-layer PCE communication: standards, interoperability and deployment

The IETF Path Computation Element (PCE) framework is a prime example of a softwaredefined network virtualisation artifact, originally envisaged about a decade ago for constraintbased routing in the conventional best-effort Internet. Over the years, however, it evolved into the ultimate tool for multi-technology, multidomain, multi-layer, and multi-vendor networking. Carriers have already started to implement PCE-based solutions in their combined Internet IP/MPLS and Wavelength Switched Optical Networks (WSONs). However, significant development and testing is still needed to elevate the PCE concepts from theory to practice. This paper addresses standards, interoperability and deployment of coordinated computation and setup of multi-layer paths from a practical perspective and presents its first implementation in a multi-vendor, interoperable PCE testbed. In our testbed, we try to push the limits of multivendor interoperability, which current research mostly neglects, and thereby identify missing links in the current research and standards necessary to implement and deploy inter-operable multi-layer systems. We point to the key areas that the research and industrial communities will need to address in making the proposed solutions practical, which in turn will drive the upcoming efforts in software-defined networking with inter-operable PCE-based architectures.

End-to-End Resource Management for Federated Delivery of Multimedia Services

Recently, the Internet has become a popular platform for the delivery of multimedia content. Currently, multimedia services are either offered by Over-the-top (OTT) providers or by access ISPs over a managed IP network. As OTT providers offer their content across the best-effort Internet, they cannot offer any Quality of Service (QoS) guarantees to their users. On the other hand, users of managed multimedia services are limited to the relatively small selection of content offered by their own ISP. This article presents a framework that combines the advantages of both existing approaches, by dynamically setting up federations between the stakeholders involved in the content delivery process. Specifically, the framework provides an automated mechanism to set up end-to-end federations for QoS-aware delivery of multimedia content across the Internet. QoS contracts are automatically negotiated between the content provider, its customers, and the intermediary network domains. Additionally, a federated resource reservation algorithm is presented, which allows the framework to identify the optimal set of stakeholders and resources to include within a federation. Its goal is to minimize delivery costs for the content provider, while satisfying customer QoS requirements. Moreover, the presented framework allows intermediary storage sites to be included in these federations, supporting on-the-fly deployment of content caches along the delivery paths. The algorithm was thoroughly evaluated in order to validate our approach and assess the merits of including intermediary storage sites. The results clearly show the benefits of our method, with delivery cost reductions of up to 80 % in the evaluated scenario.

H.264/SVC 스트리밍을 위한 가중치 기반 혼잡 제어 알고리즘

Because best-effort Internet provides no guarantees on packet delay and loss, transient network congestion may cause negative effects on H.264/SVC streaming. Thus, the congestion control is required to adjust bit rate by dropping enhancement layers of H.264/SVC streams. This paper differentiates the video streams according to different levels of importance and proposes weighted-based congestion control algorithms to use the rate-distortion characteristics of streams. To maximize the weighted sum of PSNR values of all streams on a bandwidth-constrained node, this paper proposes WNS(Weighted Near-Sighted) and WFS(Weighted Far-Sighted) algorithms to control the number of enhancement layers of streams. Through simulation, this paper shows that weighted-based congestion control algorithm can efficiently adapt streams to network conditions and analyzes the characteristics of congestion control algorithms.

An efficient relay node selection scheme to improve the performance of P2P-based VoIP applications in Chinese internet

Peer-to-peer (P2P) systems have been widely deployed and used to provide voice-over-IP (VoIP) service in the Internet. However, the current best-effort Internet cannot readily provide the service guarantees that meet the quality standards achieved in the public switched telephone network (PSTN). To address this problem, many studies have demonstrated that exploiting path diversity is a promising approach, such as multi-homing and overlay routing. In this paper, we focus on the overlay scenario and bring forward a previously unexplored approach that exploits the properties of delay space of Internet to select relay node to enhance the performance of P2P-based VoIP applications in Chinese Internet. By conducting intensive Internet measurements, we analyze the properties of delay space of Chinese Internet and show these properties can be readily exploited to select relay node with as small a cost as possible. Exploiting these properties we bring forward an efficient relay node selection scheme to improve the performance of P2P-based VoIP applications in Chinese Internet. Our intensive evaluation by trace-driven simulation shows our scheme is highly efficient and easy to be implemented.

M3FEC: Joint Multiple Description Coding and Forward Error Correction for Interactive Multimedia in Multiple Path Transmission

The best-effort internet has inherent limitations on the end-to-end performance for interactive multimedia communications. This paper presents a multiple description coding (MDC) and forward error correction (FEC) based multiple path transmission schemes for interactive multimedia (M3FEC), which improves the end users' experience by maximizing a rate-distortion (R-D) based optimization problem. The proposed model considers both the network diversity and the application's stringent requirements, and combines the individual merits of the three promising technologies of multiple path overlay routing, MDC and FEC. Extensive numerical analysis and PlanetLab experiments demonstrate that M3FEC successfully combats packet losses, error propagation, and unpredictable network dynamics. This method also significantly increases distortion for interactive multimedia by over 10 dB than traditional IP-layer single path transmission in poor network environments, and outperforms the performance achieved by using MDC or FEC alone.

Adapting a Main-Stream Internet Switch Architecture for Multihop Real-Time Industrial Networks

As real-time industrial control systems scale up, single real-time local area network (LAN) is no longer sufficient; instead, we need real-time switches to merge many real-time LANs into real-time wide area networks (WANs). However, nowadays commercially-off-the-shelf WAN switches are designed for best-effort Internet traffic rather than real-time traffic. To address this problem, we propose a real-time crossbar switch design that minimally modifies, and even simplifies the de facto industrial standard switch design of iSLIP. Specifically, we change the iSLIP request-grant-accept negotiation to deterministic grant. The switch runs periodically with an M cell-time clock-period. Every input port runs per-flow queueing, and every output port deterministically grants input port per-flow queues according to its own M cell-time clock-period schedule. The schedules are created offline. We prove that the global scheduling can be reduced to a preemptive open shop scheduling problem; as long as every input/output needs to send/fetch no more than M cells per M cell-time clock-period, all outputs schedules do not conflict; and the scheduling algorithm takes O(N <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">4</sup> ) time (N is the number of input/output ports). Such design serves real-time periodic/aperiodic traffic in a time-division multiple-access (TDMA) fashion. This simplifies analysis, provides isolation, and results in a close-form end-to-end delay bound. We implemented the proposed real-time switch using Xilinx field programmable gate arrays (FPGAs), and built a distributed control test bed upon the switched networks. Using the test bed, we carried out experiments to compare the implemented real-time switches and iSLIP switches. The results prove the necessity of using real-time switches for real-time industrial control.

A Virtualized Network Architecture for Improving QoS

Recently, researches have shown that today's best-effort Internet with one fits all principle comes to an impasse. Addressed to the different QoS requirements of today's network services, we propose a new network architecture based on network virtualization technology for improving Internet QoS. It divides a thick network into multiple thin virtual networks deploying on the same substrate, each is customized with a special designing goal and runs a customized protocol. Then the traffic with different QoS requirements is classified at the ingress router, and distributed to different virtual networks, which are the most suitable for carrying the special traffic. Also, we can deploy a service on multiple virtual networks, and making them working collaboratively to achieve a better QoS. To verify this idea, a prototype is implemented in LAN-scale network. By some simply designed experiments for comparison, we observe that, by use of our network architecture, service provider with specific QoS requirement can take its choice to choose appropriate virtual networks to achieve better QoS performance.

M2FEC: An effective FEC based multi-path transmission scheme for interactive multimedia communication

The oncoming prosperity of interactive multimedia application triggers significant challenges to current best-effort Internet due to such application’s stringent delay, loss and bandwidth requirements, and Internet’s unpredictable dynamics. Multi-path transmission and error-resilient coding are two promising approaches to alleviate these problems. This paper attempts to introduce error-resilient coding into multi-path transmission to better trade off between multi-path bandwidth resource consumption and reliable media quality. We propose a model for multi-paths interactive multimedia transmission and develop M2FEC—a FEC based transmission scheme which maximizes the overall quality at the client under various constraints based on the proposed model. Numerical simulation and PlanetLab experiments demonstrate the effectiveness and practicability of M2FEC in theory and in empiricism, respectively.

The Cognitive Packet Network: A Survey

Current and future multimedia networks require connections under specific quality of service (QoS) constraints which can no longer be provided by the best-effort Internet. Therefore, ‘smarter’ networks have been proposed in order to cover this need. The cognitive packet network (CPN) is a routing protocol that provides QoS-driven routing and performs self-improvement in a distributed manner, by learning from the experience of special packets, which gather on-line QoS measurements and discover new routes. The CPN was first introduced in 1999 and has been used in several applications since then. Here we provide a comprehensive survey of its variations, applications and experimental performance evaluations.

Cache architecture for on-demand streaming on the Web

On-demand streaming from a remote server through best-effort Internet poses several challenges because of network losses and variable delays. The primary technique used to improve the quality of distributed content service is replication. In the context of the Internet, Web caching is the traditional mechanism that is used. In this article we develop a new staged delivery model for a distributed architecture in which video is streamed from remote servers to edge caches where the video is buffered and then streamed to the client through a last-mile connection. The model uses a novel revolving indexed cache buffer management mechanism at the edge cache and employs selective retransmissions of lost packets between the remote and edge cache for a best-effort recovery of the losses. The new Web cache buffer management scheme includes a dynamic adjustment of cache buffer parameters based on network conditions. In addition, performance of buffer management and retransmission policies at the edge cache is modeled and assessed using a probabilistic analysis of the streaming process as well as system simulations. The influence of different endogenous control parameters on the quality of stream received by the client is studied. Calibration curves on the QoS metrics for different network conditions have been obtained using simulations. Edge cache management can be done using these calibration curves. ISPs can make use of calibration curves to set the values of the endogenous control parameters for specific QoS in real-time streaming operations based on network conditions. A methodology to benchmark transmission characteristics using real-time traffic data is developed to enable effective decision making on edge cache buffer allocation and management strategies.

On the effects of the packet size distribution on the packet loss process

Real-time multimedia applications have to use forward error correction (FEC) anderror concealment techniques to cope with losses in today's best-effort Internet. The efficiency of these solutions is known however to depend on the correlation between losses in the media stream. In this paper we investigate how the packet size distribution affects the packet loss process, that is, the distribution of the number of lost packets in a block, the related FEC performance and the average loss run length. We present mathematical models for the loss process of the MMPP+M/D/1/K and the MMPP+M/M/1/K queues; we validate the models via simulations, and compare the results to simulation results with an MPEG-4 coded video trace. We conclude that the deterministic packet size distribution (PSD) not only results in lower stationary loss probability than the exponential one, but also gives a less correlated loss process, both at a particular average link load and at a particular stationary loss probability as seen by the media stream.Our results show that for applications that can only measure the packet loss probability, the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior. Our results show that the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior and thus can improve the queuing performance by decreasing the variance of its PSD.