10 research outputs found

    Understanding Timelines within MPEG Standards

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    (c) 2016 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works.Nowadays, media content can be delivered via diverse broadband and broadcast technologies. Although these different technologies have somehow become rivals, their coordinated usage and convergence, by leveraging of their strengths and complementary characteristics, can bring many benefits to both operators and customers. For example, broadcast TV content can be augmented by on-demand broadband media content to provide enriched and personalized services, such as multi-view TV, audio language selection, and inclusion of real-time web feeds. A piece of evidence is the recent Hybrid Broadcast Broadband TV (HbbTV) standard, which aims at harmonizing the delivery and consumption of (hybrid) broadcast and broadband TV content. A key challenge in these emerging scenarios is the synchronization between the involved media streams, which can be originated by the same or different sources, and delivered via the same or different technologies. To enable synchronized (hybrid) media delivery services, some mechanisms providing timelines at the source side are necessary to accurately time align the involved media streams at the receiver-side. This paper provides a comprehensive review of how clock references (timing) and timestamps (time) are conveyed and interpreted when using the most widespread delivery technologies, such as DVB, RTP/RTCP and MPEG standards (e.g., MPEG-2, MPEG-4, MPEG-DASH, and MMT). It is particularly focused on the format, resolution, frequency, and the position within the bitstream of the fields conveying timing information, as well as on the involved components and packetization aspects. Finally, it provides a survey of proofs of concepts making use of these synchronization related mechanisms. This complete and thorough source of information can be very useful for scholars and practitioners interested in media services with synchronization demands.This work has been funded, partially, by the "Fondo Europeo de Desarrollo Regional" (FEDER) and the Spanish Ministry of Economy and Competitiveness, under its R&D&i Support Program in project with ref TEC2013-45492-R.Yuste, LB.; Boronat Segui, F.; Montagut Climent, MA.; Melvin, H. (2015). Understanding Timelines within MPEG Standards. Communications Surveys and Tutorials, IEEE Communications Society. 18(1):368-400. https://doi.org/10.1109/COMST.2015.2488483S36840018

    MediaSync: Handbook on Multimedia Synchronization

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    This book provides an approachable overview of the most recent advances in the fascinating field of media synchronization (mediasync), gathering contributions from the most representative and influential experts. Understanding the challenges of this field in the current multi-sensory, multi-device, and multi-protocol world is not an easy task. The book revisits the foundations of mediasync, including theoretical frameworks and models, highlights ongoing research efforts, like hybrid broadband broadcast (HBB) delivery and users' perception modeling (i.e., Quality of Experience or QoE), and paves the way for the future (e.g., towards the deployment of multi-sensory and ultra-realistic experiences). Although many advances around mediasync have been devised and deployed, this area of research is getting renewed attention to overcome remaining challenges in the next-generation (heterogeneous and ubiquitous) media ecosystem. Given the significant advances in this research area, its current relevance and the multiple disciplines it involves, the availability of a reference book on mediasync becomes necessary. This book fills the gap in this context. In particular, it addresses key aspects and reviews the most relevant contributions within the mediasync research space, from different perspectives. Mediasync: Handbook on Multimedia Synchronization is the perfect companion for scholars and practitioners that want to acquire strong knowledge about this research area, and also approach the challenges behind ensuring the best mediated experiences, by providing the adequate synchronization between the media elements that constitute these experiences

    Secure VoIP Performance Measurement

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    This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality

    Scalable Multiple Description Coding and Distributed Video Streaming over 3G Mobile Networks

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    In this thesis, a novel Scalable Multiple Description Coding (SMDC) framework is proposed. To address the bandwidth fluctuation, packet loss and heterogeneity problems in the wireless networks and further enhance the error resilience tools in Moving Pictures Experts Group 4 (MPEG-4), the joint design of layered coding (LC) and multiple description coding (MDC) is explored. It leverages a proposed distributed multimedia delivery mobile network (D-MDMN) to provide path diversity to combat streaming video outage due to handoff in Universal Mobile Telecommunications System (UMTS). The corresponding intra-RAN (Radio Access Network) handoff and inter-RAN handoff procedures in D-MDMN are studied in details, which employ the principle of video stream re-establishing to replace the principle of data forwarding in UMTS. Furthermore, a new IP (Internet Protocol) Differentiated Services (DiffServ) video marking algorithm is proposed to support the unequal error protection (UEP) of LC components of SMDC. Performance evaluation is carried through simulation using OPNET Modeler 9. 0. Simulation results show that the proposed handoff procedures in D-MDMN have better performance in terms of handoff latency, end-to-end delay and handoff scalability than that in UMTS. Performance evaluation of our proposed IP DiffServ video marking algorithm is also undertaken, which shows that it is more suitable for video streaming in IP mobile networks compared with the previously proposed DiffServ video marking algorithm (DVMA)

    Treatment-Based Classi?cation in Residential Wireless Access Points

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    IEEE 802.11 wireless access points (APs) act as the central communication hub inside homes, connecting all networked devices to the Internet. Home users run a variety of network applications with diverse Quality-of-Service requirements (QoS) through their APs. However, wireless APs are often the bottleneck in residential networks as broadband connection speeds keep increasing. Because of the lack of QoS support and complicated configuration procedures in most off-the-shelf APs, users can experience QoS degradation with their wireless networks, especially when multiple applications are running concurrently. This dissertation presents CATNAP, Classification And Treatment iN an AP , to provide better QoS support for various applications over residential wireless networks, especially timely delivery for real-time applications and high throughput for download-based applications. CATNAP consists of three major components: supporting functions, classifiers, and treatment modules. The supporting functions collect necessary flow level statistics and feed it into the CATNAP classifiers. Then, the CATNAP classifiers categorize flows along three-dimensions: response-based/non-response-based, interactive/non-interactive, and greedy/non-greedy. Each CATNAP traffic category can be directly mapped to one of the following treatments: push/delay, limited advertised window size/drop, and reserve bandwidth. Based on the classification results, the CATNAP treatment module automatically applies the treatment policy to provide better QoS support. CATNAP is implemented with the NS network simulator, and evaluated against DropTail and Strict Priority Queue (SPQ) under various network and traffic conditions. In most simulation cases, CATNAP provides better QoS supports than DropTail: it lowers queuing delay for multimedia applications such as VoIP, games and video, fairly treats FTP flows with various round trip times, and is even functional when misbehaving UDP traffic is present. Unlike current QoS methods, CATNAP is a plug-and-play solution, automatically classifying and treating flows without any user configuration, or any modification to end hosts or applications

    Supervisión de la Instalación y Montaje de los Sistemas de Comunicaciones Voz, Data, CCTV y CATV con la Tecnología CAT7A en el Nuevo Hospital San Juan de Dios de la Ciudad de Pisco

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    En este trabajo – informe de experiencia profesional haremos llegar una descripción de los sistemas de comunicación (DATA, VOZ, CCTV Y CATV) instalados en el Hospital San Juan de Dios de la ciudad de Pisco. El sistema consta de una red de Cableado Estructurado Categoría 7A, y de un backbone de Fibra Óptica Multimodo 50/125 um. El Cableado Estructurado que fue implementado fue diseñado conforme a los Estándares ANSI/TIA/EIA e ISO 11801. Estos Estándares definen la estructura del Sistema de Cableado de la siguiente manera: Facilidades de Entrada Cuarto de Equipos Cableado Backbone Cuarto de Telecomunicaciones Cableado Horizontal Área de Trabajo Finalmente se realizará una descripción de las pruebas de certificación de cableado estructurado de fibra óptica y la red de cobre (Cable Categoría 7A). PALABRAS CLAVES Cableado estructurado, fibra óptica, data, voz, cctv y catv

    Multimedia-Streaming in Benutzergruppen

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    At the time being, multimedia services using IP technology like IPTV or video on-demand are a hot topic. Technically, they can be classified under the notion of streaming. A server sends media data in a continuous fashion to one or several clients, which consume and display data portions as soon as they arrive. Using a feedback channel customers may influence the play-back, watching programs time-shifted or pausing the program. An enhancement of such streaming services is to watch those movies with a group of people on several devices in parallel. Similar approaches have been developed using IP multicast. However, users cannot control the presentation: pausing or skipping of more unimportant parts is impossible. Moreover, members cannot be added to the session directly within the application. The costream architecture developed in this works offers a collaborative streaming service without these limitations: People may join others watching a movie or invite others to such a collaborative streaming session. Dependent on the desired course of the session the participants' control operations are executed for all users, or the group is split into subgroups to let watchers follow their own time-lines. A group management controls this by means of user roles. Separate from the group management, the so-called association service provides for streaming session control and synchronization among participants. This separation of duties is advantageous in the sense that standard components can be used: For group management, SIP conferencing servers are suitable, whereas session control can best be handled using RTSP proxies as already used for caching of media data. Eventually, the evaluation of this architecture shows that such a service offers both low latency for clients and an acceptable synchronization of media streams to different client devices. Moreover, the communication overhead compared to usual conferencing or streaming systems is very low.Mit Hilfe der IP-Technologie erbrachte Multimedia-Dienste wie IPTV oder Video-on-Demand sind zur Zeit ein gefragtes Thema. Technisch werden solche Dienste unter dem Begriff "Streaming" eingeordnet. Ein Server sendet Mediendaten kontinuierlich an Empfänger, welche die Daten sofort weiterverarbeiten und anzeigen. Über einen Rückkanal hat der Kunde die Möglichkeit der Einflussnahme auf die Wiedergabe. Eine Weiterentwicklung dieser Streaming-Dienste ist die Möglichkeit, gemeinsam mit anderen denselben Film auf mehreren Geräten anzusehen. Ähnliche Ansätze gibt es im Internet bereits durch IP-Multicast. Allerdings können Benutzer hierbei keinen Einfluss auf die Übertragung nehmen - das Überspringen von Teilen ist zum Beispiel nicht möglich. Andere Benutzer können nicht direkt zur Streaming-Sitzung eingeladen werden. Collaborative Streaming ohne solche Einschränkungen bietet die in dieser Arbeit entwickelte costream-Architektur: Sie erlaubt es, andere zum gemeinsamen Betrachten eines Filmes einzuladen oder sich selbst in eine Benutzergruppe einzuklinken. Abhängig vom gewünschten Ablauf der Sitzung wird die Steuerung für alle Teilnehmer durchgeführt oder die Gruppe aufgeteilt. Eine Gruppenverwaltung regelt dies mit Hilfe von Rollenzuweisungen. Davon getrennt sorgt eine weitere Komponente für die Steuerung der Streaming-Sitzungen und die Synchronisation zwischen Teilnehmern. Diese Aufteilung hat den Vorteil, dass von der IETF entwickelte Standardprotokolle eingesetzt werden können. Für die Gruppenverwaltung sind SIP-Konferenzsysteme geeignet, während für die Sitzungssteuerung ein RTSP-Zwischensystem benutzt wurde. Die Evaluierung dieser Architektur zeigt schließlich, dass ein solcher Dienst nicht nur geringe Wartezeiten aufweist, sondern eine akzeptable Synchronisation der Datenströme auf die verschiedenen Ausgabegeräte der Benutzer erreicht wird. Zudem ist der Zusatzaufwand verglichen mit üblichen Konferenz- oder Streaming-Systemen sehr gering

    Online learning on the programmable dataplane

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    This thesis makes the case for managing computer networks with datadriven methods automated statistical inference and control based on measurement data and runtime observations—and argues for their tight integration with programmable dataplane hardware to make management decisions faster and from more precise data. Optimisation, defence, and measurement of networked infrastructure are each challenging tasks in their own right, which are currently dominated by the use of hand-crafted heuristic methods. These become harder to reason about and deploy as networks scale in rates and number of forwarding elements, but their design requires expert knowledge and care around unexpected protocol interactions. This makes tailored, per-deployment or -workload solutions infeasible to develop. Recent advances in machine learning offer capable function approximation and closed-loop control which suit many of these tasks. New, programmable dataplane hardware enables more agility in the network— runtime reprogrammability, precise traffic measurement, and low latency on-path processing. The synthesis of these two developments allows complex decisions to be made on previously unusable state, and made quicker by offloading inference to the network. To justify this argument, I advance the state of the art in data-driven defence of networks, novel dataplane-friendly online reinforcement learning algorithms, and in-network data reduction to allow classification of switchscale data. Each requires co-design aware of the network, and of the failure modes of systems and carried traffic. To make online learning possible in the dataplane, I use fixed-point arithmetic and modify classical (non-neural) approaches to take advantage of the SmartNIC compute model and make use of rich device local state. I show that data-driven solutions still require great care to correctly design, but with the right domain expertise they can improve on pathological cases in DDoS defence, such as protecting legitimate UDP traffic. In-network aggregation to histograms is shown to enable accurate classification from fine temporal effects, and allows hosts to scale such classification to far larger flow counts and traffic volume. Moving reinforcement learning to the dataplane is shown to offer substantial benefits to stateaction latency and online learning throughput versus host machines; allowing policies to react faster to fine-grained network events. The dataplane environment is key in making reactive online learning feasible—to port further algorithms and learnt functions, I collate and analyse the strengths of current and future hardware designs, as well as individual algorithms

    Control of real-time multimedia applications in best-effort networks

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    The increasing demand for real-time multimedia applications and the lack of quality of service (QoS) support in public best-effort or Internet Protocol (IP) networks has prompted many researchers to propose improvements on the QoS of such networks. This research aims to improve the QoS of real-time multimedia applications in public best-effort networks, without modifying the core network infrastructure or the existing codecs of the original media applications. A source buffering control is studied based on a fluid model developed for a single flow transported over a best-effort network while allowing for flow reversal. It is shown that this control is effective for QoS improvement only when there is sufficient flow reversal or packet reordering in the network. An alternate control strategy based on predictive multi-path switching is studied where only two paths are considered as alternate options. Initially, an emulation study is performed, exploring the impact of path loss rate and traffic delay signal frequency content on the proposed control. The study reveals that this control strategy provides the best QoS improvement when the average comprehensive loss rates of the two paths involved are between 5% and 15%, and when the delay signal frequency content is around 0.5 Hz. Linear and nonlinear predictors are developed using actual network data for use in predictive multi-path switching control. The control results show that predictive path switching is better than no path switching, yet no one predictor developed is best for all cases studied. A voting based control strategy is proposed to overcome this problem. The results show that the voting based control strategy results in better performance for all cases studied. An actual voice quality test is performed, proving that predictive path switching is better than no path switching. Despite the improvements obtained, predictive path switching control has some scalability problems and other shortcomings that require further investigation. If there are more paths available to choose from, the increasing overhead in probing traffic might become unacceptable. Further, if most of the VoIP flows on the Internet use this control strategy, then the conclusions of this research might be different, requiring modifications to the proposed approach. Further studies on these problems are needed

    An Integrated Ntp-Rtcp Solution To Audio Skew Detection And Compensation For VOIP Applications

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    The circuit switched POTS (Plain Old Telephone System) preserves the timing relationship between media samples from sender to receiver through use of a common clock. For PC-based Internet multimedia, the existence of separate audio and system clocks on either end-host can introduce significant complications. Much work has taken place in recent years that addresses the issue of system clock skew and its effect on precise delay measurement. In a Voice over IP (VoIP) environment, where adaptive buffering techniques are employed, system and audio clock skew can distort both delay measurement and playout control as well as lead to poor buffer performance. This paper presents a high level mechanism to measure and compensate for the skew relationships between system and audio clocks at each end of a multimedia session. The mechanism utilises both the Network Time Protocol (NTP) and the RTP (Realtime Transport Protocol) Control Protocol or RTCP. Preliminary and positive results are presented from a testbed system and plans for further work are outlined
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