A QoS monitoring system in a heterogeneous multi-domain DVB-H platform

The MobileTV, IPTV, and DVB standards (DVB-H/T) have been defined to offer mobile users interactive multimedia services with quality of service (QoS) consistency analogous to TV services. However, the market has yet to provide effective and economical solutions for the real-time delivery of such services to the corresponding transmitters over multi-domain IP networks. The monitoring system proposed in this paper enables the QoS in the IP networks involved in the delivery of real-time multimedia content to the transmitters to be ascertained. The system utilizes the QoS parameters defined in MPEG-2 Transport Streams to detect problems occurring in the heterogeneous multi-domain IP networks. The ability to detect problems having an adverse effect on QoS allows appropriate control actions to be determined to recover the QoS across the composite IP network. The design and implementation of the proposed QoS-Monitoring system (QoS-MS) is presented, followed by analysis of experimental results that demonstrate the feasibility of the system

Virtual RTCP: A Case Study of Monitoring and Repair for UDP-based IPTV Systems

IPTV systems have seen widespread deployment, but often lack robust mechanisms for monitoring the quality of experience. This makes it difficult for network operators to ensure that their services match the quality of traditional broadcast TV systems, leading to consumer dissatisfaction. We present a case study of virtual RTCP, a new framework for reception quality monitoring and reporting for UDP-encapsulated MPEG video delivered over IP multicast. We show that this allows incremental deployment of reporting infrastructure, coupled with effective retransmission-based packet loss repair

Understanding Timelines within MPEG Standards

(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

Towards a new generation of transport services adapted to multimedia application

Une connexion d'ordre et de fiabilité partiels (POC, partial order connection) est une connexion de transport autorisée à perdre certains objets mais également à les délivrer dans un ordre éventuellement différent de celui d'émission. L'approche POC établit un lien conceptuel entre les protocoles sans connexion au mieux et les protocoles fiables avec connexion. Le concept de POC est motivé par le fait que dans les réseaux hétérogènes sans connexion tels qu'Internet, les paquets transmis sont susceptibles de se perdre et d'arriver en désordre, entraînant alors une réduction des performances des protocoles usuels. De plus, on montre qu'un protocole associé au transport d'un flux multimédia permet une réduction très sensible de l'utilisation des ressources de communication et de mémorisation ainsi qu'une diminution du temps de transit moyen. Dans cet article, une extension temporelle de POC, nommée TPOC (POC temporisé), est introduite. Elle constitue un cadre conceptuel permettant la prise en compte des exigences de qualité de service des applications multimédias réparties. Une architecture offrant un service TPOC est également introduite et évaluée dans le cadre du transport de vidéo MPEG. Il est ainsi démontré que les connexions POC comblent, non seulement le fossé conceptuel entre les protocoles sans connexion et avec connexion, mais aussi qu'ils surpassent les performances des ces derniers lorsque des données multimédias (telles que la vidéo MPEG) sont transportées

MediaSync: Handbook on Multimedia Synchronization

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

A continuous media transport protocol

A protocol for the transmission of digitized audio and video data must schedule data transmission such that each audio or video sample is available in the receiver\u27s buffer by its play-out deadline without overflowing the allotted buffer space at the receiver. While traditional data communication protocols provide error-free data transmission through the use of time-outs and retransmissions, the unbounded number of retransmissions that a packet may suffer results in an unboundable network transmission delay. Thus, such protocols are not appropriate for the real-time transmission of digitized audio and video data;This dissertation examines the network environments likely to be used for digitized audio and video data transmission and develops an audio/video transport-layer communication protocol which for such network environments. A unique feature of this protocol is the calculation of a transmission schedule which can minimize the buffer requirements at the receiver while still meeting the real-time play-out deadlines of the individual audio and video streams

Streaming Video Performance and Enhancements in Resource-Constrained Wireless Networks

Streaming video is an increasingly popular application in wireless networks. The concept of a live streaming video yields several enticing possibilities: real-time video conferencing, television broadcasting, pay-per-view movie streaming, and more. These ideas have already been explored via the internet and have met with mixed success, largely due to the shortcomings of the underlying network. Taking streaming video to wireless networks, then, poses several significant challenges. Wireless networks are inherently more susceptible to failures and data corruption due to their unstable communications medium. This volatility suggests serious drawbacks for any implementation of streaming video. Video frame errors, jitter, and even complete sync loss are entirely conceivable in a wireless environment. Many of these issues have been undertaken and several approaches to mediation or even solution of these problems are underway. This thesis proposes to use advanced simulation techniques to properly exhaustively permute many vital parameters within a UMTS network and uncover, if they exist, bottlenecks in UMTS performance under considerable network load. This is accomplished via a described testing plan with simulation environment. Additionally this thesis proposes a new UDP-like transport layer specially optimized for streaming media over resource-constrained networks, tested to work with significant improvements under the UMTS cellular networking system. Finally this thesis provides several innovative new methods in the furtherance of the field of streaming media research in resourceconstrained and cellular environments. Overall this thesis makes several important contributes to an exciting and ever-growing field of active research and discussion

HbbTV-compliant Platform for Hybrid Media Delivery and Synchronization on Single- and Multi-Device Scenarios

[EN] The combination of broadcast and broadband (hybrid) technologies for delivering TV related media contents can bring fascinating opportunities. It is motivated by the large amount and diversity of media contents, together with the ubiquity and multiple connectivity capabilities of modern consumption devices. This paper presents an end-to-end platform for the preparation, delivery, and synchronized consumption of related hybrid (broadcast/broadband) media contents on a single device and/or on multiple close-by devices (i.e., a multi-device scenario). It is compatible with the latest version of the Hybrid Broadcast Broadband TV (HbbTV) standard (version 2.0.1). Additionally, it provides adaptive and efficient solutions for key issues not specified in that standard, but that are necessary to successfully deploy hybrid and multidevice media services. Moreover, apart from MPEG-DASH and HTML5, which are the broadband technologies adopted by HbbTV, the platform also provides support for using HTTP Live Streaming and Real-time Transport Protocol and its companion RTP Control Protocol broadband technologies. The presented platform can provide support for many hybrid media services. In this paper, in order to evaluate it, the use case of multi-device and multi-view TV service has been selected. The results of both objective and subjective assessments have been very satisfactory, in terms of performance (stability, smooth playout, delays, and sync accuracy), usability of the platform, usefulness of its functionalities, and the awaken interest in these kinds of platforms.This work was supported in part by the "Fondo Europeo de Desarrollo Regional" and in part by the Spanish Ministry of Economy and Competitiveness through R&D&I Support Program under Grant TEC2013-45492-R.Boronat, F.; Marfil-Reguero, D.; Montagud, M.; Pastor Castillo, FJ. (2017). HbbTV-compliant Platform for Hybrid Media Delivery and Synchronization on Single- and Multi-Device Scenarios. IEEE Transactions on Broadcasting. 1-26. https://doi.org/10.1109/TBC.2017.2781124S12

Experimental Evaluation Platform for Voice Transmission Over Internet of Things (VoIoTs)

Internet of Things (IoTs) is an example of the last advances in Information and Communication Technologies. In particular, with the revolutionary development of Wireless Sensor Network (WSN) technologies, researchers largely focused on take benefits of integration embedded low-cost, low-power WSN technology in a various IoTs applications. Real-time voice transmission over IoTs is one interesting application that began to be explored by many researchers. Thus, this paper presents a performance study for transmission of voice over WSN (VoWSN) with and without presence of Internet. A framework using a Raspberry Pi3 (RPi3) and open source FFmpeg technology for processing, compressing and streaming voice to a remote computer is proposed, implemented and evaluated. The performance of the proposed framework is evaluated by studying its behavior utilizing three audio encoding algorithms: AC3, MP3 and OPUS with different sampling rates and a set of evaluation metrics such as :One-way delay, jitter, Bandwidth (B.W), CPU usage and packet losses