A QoE study of different stream and layout configurations in video conferencing under limited network conditions

One particular problem of QoE research in video conferencing is, that most research in the past concentrated on one-to-one video conferencing or simply video consumption. However, video conferencing with two people (one-to-one) and within a group (multi-party) is different. Particularly, limitations of one participant might have an effect on the QoE of the whole group. This possible effect however is not well studied. Therefore, this paper aims to better understand the impact of individual limitations towards the groups QoE. To do so, we show a study about different video stream configurations and layouts for multi-party conferencing in respect to individual network limitations. For this, we conduct a user study with 20 participants in 5 groups, in a semi-controlled setup. Such a setup, combines supervising participants locally while still using our software infrastructure deployed in the internet. Furthermore, we use an asymmetric experiment design, by putting every participant under a different condition, as this proposes a more realistic scenario. Within our study, we look at three different factors: layout, video quality and network limitations. To foster conversation between participants, the group engaged in a discussion about different survival questions. Our findings show that packet loss and the resulting distortions have a greater impact on the QoE as reducing the video quality by its resolution. Furthermore, our findings indicate that participants are more satisfied in a visually equal layout (showing participants in a similar size) and a more balanced stream configuration

Design and Implement a Hybrid WebRTC SignallingMechanism for Unidirectional & Bi-directional VideoConferencing

WebRTC (Web Real-Time Communication) is a technology that enables browser-to-browser communication. Therefore, a signalling mechanism must be negotiated to create a connection between peers. The main aim of this paper is to create and implement a WebRTC hybrid signalling mechanism named (WebNSM) for video conferencing based on the Socket.io (API) mechanism and Firefox. WebNSM was designed over a combination of different topologies, such as simplex, star and mesh. Therefore it offers several communications at the same time as one-to-one (unidirectional/bidirectional), one-to-many (unidirectional) and many-to-many (bi-directional) without any downloading or installation. In this paper, WebRTC video conferencing was accomplished via LAN and WAN networks, including the evaluation of resources in WebRTC like bandwidth consumption, CPU performance, memory usage, Quality of Experience (QoE) and maximum links and RTPs calculation. This paper presents a novel signalling mechanism among different users, devices and networks to offer multi-party video conferencing using various topologies at the same time, as well as other typical features such as using the same server, determining room initiator, keeping the communication active even if the initiator or another peer leaves, etc. This scenario highlights the limitations of resources and the use of different topologies for WebRTC video conferencing

A service-enabling framework for the session initiation protocol (SIP)

In this dissertation, we propose a framework to provide multimedia communication services. Our proposed framework is based on SIP (Session Initiation Protocol) and has four fundamental properties: it is available, secure, high performing, and oriented to innovations. The framework is not an architecture with a rigid structure. Instead, the framework is a toolkit made up of a set of tools that can be combined in different ways. The combination of these tools provides applications and services with functionality needed to implement a wide variety of multimedia communication services. Applications and services built on top of the framework use different tools within the toolkit in order to provide their desired overall functionality. The functionality provided by the framework includes a number of primitives to be used by applications and services. These primitives mostly relate to multiparty communications and include floor control. The framework also offers support functions that relate to PSTN (Public Switched Telephony Network) interworking, policy control, and consent-based communications. Additionally, the framework contains functions that relate to signalling transport, multihoming, mobility, security, and NAT (Network Address Translation) traversal. The framework also allows building overlay networks when a SIP network infrastructure is not available. In order to test and refine the ideas presented in this dissertation, we have implemented most of them in proof-of-concept prototypes. We have used experiments and simulations to validate our assumptions and obtain new insights

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

TRIM: An architecture for transparent IMS-based mobility

In recent years, the development and deployment of new wired and wireless access net work technologies have made the ubiquitous Internet a reality. Users can access anywhere and anytime to the broad set of value added Internet services, which are delivered by means of the IP protocol. In this context, 3GPP is currently developing the IP Multimedia Subsystem (IMS), as a key element that allows to evolve from the ubiquitous access to the Internet services towards a next generation network model, by providing a set of essen tial facilities such as session control, QoS, charging and service integration. Nevertheless, several open issues still need consideration before the future Internet becomes real, such as supporting user mobility in IP networks. Although mobility support in the Internet is receiving much attention, IMS networks present inherent particularities that require fur ther analysis. The solutions proposed so far for IMS do not support mobility transparently to the end user applications, or address the problem by introducing complex changes to the IMS infrastructure. This paper presents TRIM, an architecture for transparent IMS based mobility. TRIM supports mobility in IMS networks transparently to the end user applications, which are unaware of the handover management procedures executed between the mobile node and the network. We have performed several experiments with a TRIM prototype, using a real IMS testbed with 3G and WLAN access networks, validating the proposal for UDP and TCP based applications.European Community's Seventh Framework ProgramPartially granted by the Madrid Community through the MEDIANET project (S 2009/TIC 1468)Publicad

A Decentralized Session Management Framework for Heterogeneous Ad-Hoc and Fixed Networks

Wireless technologies are continuously evolving. Second generation cellular networks have gained worldwide acceptance. Wireless LANs are commonly deployed in corporations or university campuses, and their diffusion in public hotspots is growing. Third generation cellular systems are yet to affirm everywhere; still, there is an impressive amount of research ongoing for deploying beyond 3G systems. These new wireless technologies combine the characteristics of WLAN based and cellular networks to provide increased bandwidth. The common direction where all the efforts in wireless technologies are headed is towards an IP-based communication. Telephony services have been the killer application for cellular systems; their evolution to packet-switched networks is a natural path. Effective IP telephony signaling protocols, such as the Session Initiation Protocol (SIP) and the H 323 protocol are needed to establish IP-based telephony sessions. However, IP telephony is just one service example of IP-based communication. IP-based multimedia sessions are expected to become popular and offer a wider range of communication capabilities than pure telephony. In order to conjoin the advances of the future wireless technologies with the potential of IP-based multimedia communication, the next step would be to obtain ubiquitous communication capabilities. According to this vision, people must be able to communicate also when no support from an infrastructured network is available, needed or desired. In order to achieve ubiquitous communication, end devices must integrate all the capabilities necessary for IP-based distributed and decentralized communication. Such capabilities are currently missing. For example, it is not possible to utilize native IP telephony signaling protocols in a totally decentralized way. This dissertation presents a solution for deploying the SIP protocol in a decentralized fashion without support of infrastructure servers. The proposed solution is mainly designed to fit the needs of decentralized mobile environments, and can be applied to small scale ad-hoc networks or also bigger networks with hundreds of nodes. A framework allowing discovery of SIP users in ad-hoc networks and the establishment of SIP sessions among them, in a fully distributed and secure way, is described and evaluated. Security support allows ad-hoc users to authenticate the sender of a message, and to verify the integrity of a received message. The distributed session management framework has been extended in order to achieve interoperability with the Internet, and the native Internet applications. With limited extensions to the SIP protocol, we have designed and experimentally validated a SIP gateway allowing SIP signaling between ad-hoc networks with private addressing space and native SIP applications in the Internet. The design is completed by an application level relay that permits instant messaging sessions to be established in heterogeneous environments. The resulting framework constitutes a flexible and effective approach for the pervasive deployment of real time applications.The invention of the phone has radically changed the way people communicate, as it allowed persons to get in contact instantly no matter of their location. However, phone communication has been confined for decades to a fixed location, be it one's own house or a phone boot. The widespread affirmation of cellular technologies has had for fixed telephony a similar impact that the invention of the phone has had on communications years before. With mobile phones, people are enabled to talk with each other anytime and anywhere. Internet has also revolutionized the way people communicate. E-mails have soon become one of the Internet killer applications. Later on, instant messaging, popularly known as chatting, has gained huge consensus among net surfers. Only recently, the use of the Internet for voice communication is becoming mainstream, and the so called Voice over IP (VoIP) applications (Skype is probably the most famous for the masses) are becoming common use. Despite its popularity, Internet still suffers from the inherent limitations that affected early telephony: it is fixed. The usage of Internet on the move still does not constitute the easiest and most satisfactory user experience, due to capabilities and limitations of the access technology, terminals, services and applications. Efforts for mobilizing the Internet are ongoing both in the industrial and in the academic worlds, but several bricks are needed to build the wall of mobile Internet. This dissertation provides one of these bricks, describing a solution that allows the deployment of multimedia applications (chat, VoIP, gaming) in mobile environments. In other words, this dissertation gives solutions for facilitating ubiquitous Internet-based communication, anytime and anywhere. The vision that we want to become true is that Internet must become mobile in the same way as fixed telephony has become mobile thanks to the cellular technology. More than this, we do not want that users are limited by the presence of an infrastructure to communicate with each other. In order to achieve this, we present solutions to deploy Internet-based services and applications in environments where no support from servers is available. In other words, we enable direct device-to-device, user-to-user Internet communication. Our contribution is mainly focused on the steps needed to establish the communication, the so called session establishment or signaling phase. We have validated our signaling framework by building a chat application that utilizes its features and works in server-less environments. The custom server-less solution does not prohibit to connect at the same time with the Internet, so that one can engage in a chess game using direct communication with a person in the proximity while having a chat in progress with a friend using standard Internet services. The challenge that we had to face is that Internet services and applications are usually built implying support from a centralized server. In order to deploy direct user-to-user Internet services, while maintaining interoperability with mainstream services, we had to enhance native Internet services to work without infrastructure support, without sacrificing interoperability with standard Internet applications. To conclude, we have placed our brick on the still yet to be completed wall of mobile Internet. Our hope is that one day, thanks also to this brick, everybody will be able to enjoy Internet-based applications as easily as now it is possible to use mobile telephony services