Xstream-x264: Real-time H.264 streaming with cross-layer integration

We present Xstream-x264: a real-time cross-layer video streaming technique implemented within a well known open-source H.264 video encoder tool x264. Xstream-x264 uses the transport protocol provided indication of the available data rate for corresponding adjustments in the video encoder.We discuss the design, implementation and the quality evaluation methodology utilised with our tool.We demonstrate via experimental results that the streaming video quality greatly improves with the presented cross-layer approach both in terms of lost frame count and the objective video quality metrics Peak Signal to Noise Ratio (PSNR)

Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms

Mechanisms for QoE optimisation of video traffic: a review paper

Transmission of video traffic over the Internet has grown exponentially in the past few years with no sign of waning. This increasing demand for video services has changed user expectation of quality. Various mechanisms have been proposed to optimise the Quality of Experience (QoE) of end users’ video. Studying these approaches are necessary for new methods to be proposed or combination of existing ones to be tailored. We discuss challenges facing the optimisation of QoE for video traffic in this paper. It surveys and classifies these mechanisms based on their functions. The limitation of each of them is identified and future directions are highlighted

Inter-layer turbo coded unequal error protection for multi-layer video transmission

In layered video streaming, the enhancement layers (ELs) must be discarded by the video decoder, when the base layer (BL) is corrupted or lost due to channel impairments. This implies that the transmit power assigned to the ELs is wasted, when the BL is corrupted. To combat this effect, in this treatise we investigate the inter-layer turbo (IL-turbo) code, where the systematic bits of the BL are implanted into the systematic bits of the ELs at the transmitter. At the receiver, when the BL cannot be successfully decoded, the information of the ELs may be utilized by the IL-turbo decoder for the sake of assisting in decoding the BL. Moreover, for providing further insights into the IL technique the benefits of the IL-turbo scheme are analyzed using extrinsic information transfer (EXIT) charts in the scenario of unequal error protection (UEP) coded layered video transmission. Finally, our data partitioning based experiments show that the proposed scheme outperforms the traditional turbo code based UEP scheme by about an Eb/N0 of 1.1 dB at a peak signal-to-noise ratio (PSNR) of 36 dB or 3 dB of PSNR at an Eb/N0 of -5.5 dB at the cost of a complexity increase of 13%

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version

Design and implementation of an on-line demonstrator for a video telephony system over heterogeneus networks

In recent times, Next Generation Mobile Networks (NGMN) enable user's mobility, not needing to be in a fixed place anymore. For this service to be successful, seamless transitions between the different technologies become essential, in order to make possible the always best connected goal. This brings an additional problematic, which is the impairments resulting from a handover between two networks. In order to succesfully plan and continue the development of always on services and mobility management, the approach must be based on user's perception of phenomena such us packet loss, the so-called Quality of Experience (QoE). This is the context in which Mobisense was born, intending a better understanding of NGMN transmission phenomena and resulting quality. Hence, Mobisense project is focused on the evaluation of the quality of service from user's point of view and on the seamless switch provision between video codecs when connections are transferred between two networks. On that purpose, a NGMN test environment was developed for real-time multimedia services. In this environment, specific network conditions can be associated with user's assessments within a realistic model. Mobisense project creates the foundations for the employment of advance prediction methods for real-time mobility management, to make decisions depending on the characteristics measured in the network and the predictions of quality resulting from them. The present degree final project gathers the work carried out to develop an extension for Mobisense testbed, in order to deploy it in a real environment of network technologies, as well as the integration with Quality of Service (QoS) algorithms. Therefore, the aim of this project consists on the development of the software required for the creation of a video thelephony system over NGMN, taking Mobisense and MultiRAT testbeds as starting point. Mobisense brings adaptation in the application layer and user's perception, and MultiRAT provides QoS adaptation and new wireless technologies. Both testbeds combined to explore more QoE aspects in wireless networks of tomorrow.En los últimos tiempos, las NGMN posibilitan la movilidad del usuario, sin ser ya necesaria su permanencia en un lugar fijo. Para el éxito de este servicio se hacen indispensables transiciones continuas entre las diferentes tecnologías, de manera que sea posible el objetivo de "siempre la mejor conexión". Esto lleva consigo una problemática adicional, que son las de ciencias resultantes del handover entre dos redes. Para planificar y continuar satisfactoriamente el desarrollo de servicios always on y la gestión de la movilidad, el enfoque debe ser en base a la percepción del usuario de fenómenos tales como la pérdida de paquetes, la llamada QoE. En este contexto nació el proyecto Mobisense, buscando una mejor comprensión del fenómeno de transmisión NGMN y la calidad resultante. El proyecto Mobisense se centra, por tanto, en la evaluación de la calidad de servicio desde el punto de vista del usuario y en la provisión de cambios continuos entre codecs de vídeo al transmitirse conexiones entre dos redes. Para tal propósito, un entorno de pruebas NGMN fue desarrollado para servicios multimedia en tiempo real. En este entorno, pueden asociarse determinadas condiciones en la red con valoraciones de calidad por parte del usuario en un modelo realista. El proyecto Mobisense sienta las bases para el empleo de métodos avanzados de predicción para la gestión de movilidad en tiempo real, para tomar decisiones dependientes de las características de la red medidas y de las predicciones de calidad derivadas a partir de éstas. El presente proyecto de fin de carrera recoge el trabajo realizado para desarrollar una extensión del testbed Mobisense, de cara a desplegarlo en un entorno real de tecnologías de red, así como la integración de algoritmos de QoS. Por tanto, el objeto de este proyecto consiste en el desarrollo software requerido para la creación de un sistema de videotelefonía sobre NGMN, tomando los testbeds Mobisense y MultiRAT como punto de partida. Mobisense aporta adaptación en la capa de aplicación y la percepción de usuario, y MultiRAT proporciona adaptación QoS y nuevas tecnologías de red. Ambos testbeds se combinan para la exploración más amplia de los aspectos de QoE en las redes inalámbricas del mañana.Ingeniería de Telecomunicació

Survey of Transportation of Adaptive Multimedia Streaming service in Internet

[DE] World Wide Web is the greatest boon towards the technological advancement of modern era. Using the benefits of Internet globally, anywhere and anytime, users can avail the benefits of accessing live and on demand video services. The streaming media systems such as YouTube, Netflix, and Apple Music are reining the multimedia world with frequent popularity among users. A key concern of quality perceived for video streaming applications over Internet is the Quality of Experience (QoE) that users go through. Due to changing network conditions, bit rate and initial delay and the multimedia file freezes or provide poor video quality to the end users, researchers across industry and academia are explored HTTP Adaptive Streaming (HAS), which split the video content into multiple segments and offer the clients at varying qualities. The video player at the client side plays a vital role in buffer management and choosing the appropriate bit rate for each such segment of video to be transmitted. A higher bit rate transmitted video pauses in between whereas, a lower bit rate video lacks in quality, requiring a tradeoff between them. The need of the hour was to adaptively varying the bit rate and video quality to match the transmission media conditions. Further, The main aim of this paper is to give an overview on the state of the art HAS techniques across multimedia and networking domains. A detailed survey was conducted to analyze challenges and solutions in adaptive streaming algorithms, QoE, network protocols, buffering and etc. It also focuses on various challenges on QoE influence factors in a fluctuating network condition, which are often ignored in present HAS methodologies. Furthermore, this survey will enable network and multimedia researchers a fair amount of understanding about the latest happenings of adaptive streaming and the necessary improvements that can be incorporated in future developments.Abdullah, MTA.; Lloret, J.; Canovas Solbes, A.; García-García, L. (2017). Survey of Transportation of Adaptive Multimedia Streaming service in Internet. Network Protocols and Algorithms. 9(1-2):85-125. doi:10.5296/npa.v9i1-2.12412S8512591-

Evaluación de calidad de video en una aplicación P2P :Goalbit

El presente trabajo consiste en la aplicación de la metodología PSQA a una aplicación de streaming de video concreta como GoalBit y en redes de acceso frecuentes como ADSL y UMTS, para determinar la calidad de video percibida por el usuario. El trabajo consiste en tres instancias: 1) Interiorización en video streaming, la plataforma GoalBit, la metodología PSQA y determinación de potenciales parámetros de calidad que afecten al flujo de video. 2) Generar una maqueta de pruebas para emular la red P2P; para ello se contará con un cliente , un servidor y una emulación de red ADSL o 3G. Para la emulación de red se utiliza Netem(Network Emulator)y Netem2. Sobre éstas se analizará el impacto de la red de transporte en los parámetros de calidad definidos, a la vez que se validarán los mismos. 3) Aplicar la metodología PSQA: generando secuencias de prueba, presentándolas a un conjunto de observadores para obtener una medida tipo Mean Opinion Score(MOS), filtrando las mismas y entrenando una Red Neuronal(RN) con ellas para obtener una fórmula de calidad. Como producto se tendrá una fórmula de evaluación dinámica de la calidad basada en un conjunto reducido de parámetros, la cual podrá ser integrada a GoalBit. Esta herramienta es la base de mecanismos de estadística y monitoreo de calidad de la red P2P. Adicionalmente, en base a ella el servidor podría tomar acciones correctivas sobre la codificación, el fraccionamiento y/o la redundancia en esquemas multi-camino o multi-fuente, como se comentará en el Cap.II, sección II.