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

Optimization of protection techniques based on FEC codes for the transmission of multimedia packetized streams

Esta tesis presenta dos modelos novedosos de arquitecturas basadas en esquemas FEC con el fin de proteger flujos de paquetes con contenido multimedial, para comunicaciones en tiempo real y en canales donde las pérdidas se producen en ráfagas. El objetivo de estos diseños ha sido maximizar la eficiencia de los códigos FEC considerados. Por un lado, el primer modelo busca alcanzar un menor coste computacional para los códigos de Reed- Solomon, ya que su conocida capacidad de recuperación para todo tipo de canales necesita un coste computacional elevado. Por otro lado, en el caso de los códigos LDPC, se ha perseguido aumentar la capacidad de recuperación de estos códigos operando en canales con errores en ráfagas, teniendo en cuenta que los códigos LDPC no están directamente diseñados para este tipo de entorno. El modelo aplicado a los códigos de Reed-Solomon se denomina inter-packet symbol approach. Este esquema consiste en una estructura alternativa que asocia los bits de los símbolos del código en distintos paquetes. Esta característica permite aprovechar de forma mejor la capacidad de recuperación de los códigos de Reed-Solomon frente a pérdidas de paquetes en ráfagas. Las prestaciones de este esquema han sido estudiadas en términos de tiempo de codificación/decodificación versus capacidad de recuperación y han sido comparados con otros esquemas propuestos en literatura. El análisis teórico ha demostrado que el enfoque propuesto permite la utilización de Campos de Galois de menor dimensión con respecto a otras soluciones. Esto se traduce en una disminución del tiempo de codificación/decodificación requerido, mientras que mantiene una capacidad de recuperación comparable. Aunque la utilización de los códigos LDPC está típicamente orientada hacía canales con errores uniformemente distribuidos (canales sin memoria) y para bloques de información largos, esta tesis surgiere el uso de este tipo de códigos FEC a nivel de aplicación, para canales con pérdidas en ráfagas y para entornos de comunicación de tiempo real, es decir, con una latencia de transmisión muy baja. Para satisfacer estas limitaciones, la configuración apropiada de los parámetros de un código LDPC ha sido determinada usando bloques de información pequeños y adaptando el código FEC de modo que sea capaz de recuperar paquetes perdidos en canales con errores en ráfagas. Para ello, primeramente se ha diseñado un algoritmo que realiza una estimación de las capacidades de recuperación del código LDPC para un canal con pérdidas en ráfagas. Una vez caracterizado el código, se ha diseñado un segundo algoritmo que optimiza la estructura del código en términos de capacidad de recuperación para las características especificas del canal con memoria, generado una versión modificada del código LDPC, adaptada al canal con perdidas en ráfagas. Finalmente, los dos esquemas FEC propuestos, han sido evaluado experimentalmente en entornos de simulación usando canales con errores en ráfagas y se han comparado con otras soluciones y esquemas ya existentes. ABSTRACT This thesis presents two enhanced FEC-based schemes to protect real-time packetized multimedia streams in bursty channels. The objective of these novel architectures has been the optimization of existing FEC codes, that is, Reed-Solomon codes and LDPC codes. On the one hand, the optimization is focused on the achievement of a lower computational cost for Reed-Solomon codes, since their well known robust recovery capability against any type of losses needs a high complexity. On the other hand, in the case of LDPC codes, the optimization is addressed to increase the recovery capabilities for a bursty channel, since they are not specifically designed for the scenario considered in this thesis. The scheme based on Reed-Solomon codes is called inter-packet symbol approach, and it consists in an alternative bit structure that allocates each symbol of a Reed- Solomon code in several media packets. This characteristic permits to exploit better the recovery capability of Reed-Solomon codes against bursty packet losses. The performance of this scheme has been studied in terms of encoding/decoding time versus recovery capability, and compared with other proposed schemes in the literature. The theoretical analysis has shown that the proposed approach allows the use of a lower size of the Galois Fields compared to other solutions. This lower size results in a decrease of the required encoding/decoding time while keeping a comparable recovery capability. Although the use of LDPC codes is typically addressed for channels where losses are uniformly distributed (memoryless channels) and for large information blocks, this thesis suggests the use of this type of FEC codes at the application layer, in bursty channels and for real-time scenario, where low transmission latency is requested. To fulfill these constraints, the appropriate configuration parameters of an LDPC scheme have been determined using small blocks of information and adapting the FEC code to be capable of recovering packet losses in bursty environments. This purpose is achieved in two steps. The first step is performed by an algorithm that estimates the recovery capability if a given LDPC code in a burst packet loss network. The second step is the optimization of the code: an algorithm optimizes the code structure in terms of recovery capability against the specific behavior of the channel with memory, generating a burst oriented version of the considered LDPC code. Finally, for both proposed FEC schemes, experimental results have been carried out in a simulated transmission channel to assess the performances of the schemes and compared to several other schemes

An analysis of IP Telephony Signaling using the Session Initiation Protocol (SIP)

This paper examines both the emergence of IP telephony in the telecommunications industry and the Session Initiation Protocol (SIP) as a method for providing signaling services for IP telephony networks. The technical as well as cost advantages of IP telephony are addressed and the various SIP components, addressing mechanisms, protocol messages, and protocol functionality are discussed. The technical benefits of SIP are examined and a brief comparison is made between SIP and H.323, an ITU umbrella specification and a leading alternative to SIP. SIP and the latest version of H.323 are found to be relatively comparable, although due to the inherent simplicity in developing SIP implementations and applications, SIP has the potential to challenge H.323's dominance in the IP telephony signaling market space

An SNMP-based audio distribution service architecture

Dissertação de mestrado em Engenharia de Redes e Serviços TelemáticosThe constant growth of integration and popularity of “Internet of Things” devices is affecting home automation systems, where new technologies were introduced, in the recent years for this particular sector. These automation systems integrate devices that can be anywhere in the house, connected to a home network, either through a wire or wireless connection. A home automation system can be used to control air conditioning, lighting, pool control systems, home-entertainment systems and much more. Within the field of home-entertainment systems, the best known technologies are the Digital Living Network Alliance and the Digital Audio Access Protocol, which provide interoperability to allow sharing of digital media content between devices across a home network. However, these technologies have the disadvantage of being proprietary, maintaining restrict documentation access, complex architectures and concepts and not optimal to specific purposes, like audio distribution. The main goal of this project was to prove that is possible to use standardized protocols, such as the Simple Network Manager Protocol and open source tools in order to develop a music distribution service that allows the implementation of similar features than the ones already existing proprietary technologies. As such, the implementation prototype system allows a user to manage and play audio from a music collection that is stored in a single home audio server. The system architecture enables audio streaming between the server and the various devices in the same local network. Further more, the music collection, can integrate virtual audio files that are available from external music sources, like iTunes, etc.O constante crescimento de integração e popularidade da “Internet das coisas” tem atualmente afetado sistemas de domótica, onde cada vez mais tecnologias têm vindo a ser desenvolvidas nos últimos anos para este sector em particular. Estes sistemas de domótica integram dispositivos que podem estar em qualquer parte de uma casa, ligados à rede seja através de um cabo ou por wireless. Um sistema de domótica pode ser usado para controlar: ar condicionado, iluminação, sistemas de controlo de piscinas, sistemas de entretenimento, entre outros. Na área de sistemas de entretenimento, as tecnologias mais conhecidas são Digital Living Network Alliance e Digital Audio Access Protocol, que fornecem interoperabilidade de modo a permitir a partilha de conteúdos digitais multimédia entre dispositivos que se encontram na mesma rede local. Contudo, possuem a desvantagem de serem tecnologias proprietárias, com documentação e manuais restritos, arquiteturas e conceitos complexos, e não otimizados para fins específicos, tal distribuição de áudio. O principal objetivo deste projeto foi provar que é possível usar protocolos normalizados, como o Simple Network Manager Protocol e ferramentas open source de forma a desenvolver um serviço de distribuição de música que permite a implementação de funcionalidades semelhantes às tecnologias proprietárias já existentes. Assim, o protótipo implementado permite a um utilizador gerir e reproduzir áudio de uma coleção de música que se esteja armazenada num servidor de áudio domestico. A arquitetura permite streaming de áudio entre o servidor e os diferentes dispositivos que se encontram na mesma rede local. Consequentemente, a coleção de música pode integrar ficheiros de áudio visuais que estejam acessíveis através de fontes externas de música, como por exemplo: iTunes, etc

Produção de Televisão UHD para Eventos em Direto

Embora os formatos de ultra alta definição apresentem um enorme avanço em relação aos formatos de alta definição, o seu aparecimento por ser recente e obrigar a uma remodelação que vai desde a fonte do conteúdo, à cadeia de produção e distribuição, ate à televisão do consumidor final, está a ser explorado num carácter ainda embrionário pelos canais de televisão. Para além disso, a indústria ainda procura as melhores soluções para adaptar estes conteúdos às novas realidades de produção, nomeadamente no que diz respeito ao transporte do vídeo nos formatos de UHD. Com os custos associados às redes de IP de alta velocidade a descerem continuamente, aliado ao facto de permitirem alta escalabilidade e flexibilidade, esta torna-se uma solução atrativa para esse transporte. No entanto, as ferramentas de transporte de vídeo de ultra alta definição sobre IP ainda estão a ser uniformizadas e estudadas para se adaptarem as exigências da atual realidade televisiva, o que torna a sua implementação um desafio