A Cloud Platform-as-a-Service for Multimedia Conferencing Service Provisioning

Multimedia conferencing is the real-time exchange of multimedia content between multiple parties. It is the basis of a wide range of applications (e.g., multimedia multiplayer game). Cloud-based provisioning of the conferencing services on which these applications rely will bring benefits, such as easy service provisioning and elastic scalability. However, it remains a big challenge. This paper proposes a PaaS for conferencing service provisioning. The proposed PaaS is based on a business model from the state of the art. It relies on conferencing IaaSs that, instead of VMs, offer conferencing substrates (e.g., dial-in signaling, video mixer and audio mixer). The PaaS enables composition of new conferences from substrates on the fly. This has been prototyped in this paper and, in order to evaluate it, a conferencing IaaS is also implemented. Performance measurements are also made.Comment: 6 pages, 6 figures, IEEE ISCC 201

Reflections on security options for the real-time transport protocol framework

The Real-time Transport Protocol (RTP) supports a range of video conferencing, telephony, and streaming video ap- plications, but offers few native security features. We discuss the problem of securing RTP, considering the range of applications. We outline why this makes RTP a difficult protocol to secure, and describe the approach we have recently proposed in the IETF to provide security for RTP applications. This approach treats RTP as a framework with a set of extensible security building blocks, and prescribes mandatory-to-implement security at the level of different application classes, rather than at the level of the media transport protocol

TINA as a virtual market place for telecommunication and information services: the VITAL experiment

The VITAL (Validation of Integrated Telecommunication Architectures for the Long-Term) project has defined, implemented and demonstrated an open distributed telecommunication architecture (ODTA) for deploying, managing and using a set of heterogeneous multimedia, multi-party, and mobility services. The architecture was based on the latest specifications released by TINA-C. The architecture was challenged in a set of trials by means of a heterogeneous set of applications. Some of the applications were developed within the project from scratch, while some others focused on integrating commercially available applications. The applications were selected in such a way as to assure full coverage of the architecture implementation and reflect a realistic use of it. The VITAL experience of refining and implementing TINA specifications and challenging the resulting platform by a heterogeneous set of services has proven the openness, flexibility and reusability of TINA. This paper describes the VITAL approach when choosing the different services and how they challenge and interact with the architecture, focusing especially on the service architecture and the Ret reference point definitions. The VITAL adjustments and enhancements to the TINA architecture are described. This paper contributes to proving that the TINA-based VITAL ODTA allows for easy and cost-effective development and deployment of advanced end-user and operator services, and can indeed act as the basis for a virtual market place for telecommunications service

Web Conferencing Traffic - An Analysis using DimDim as Example

In this paper, we present an evaluation of the Ethernet traffic for host and attendees of the popular opensource web conferencing system DimDim. While traditional Internet-centric approaches such as the MBONE have been used over the past decades, current trends for web-based conference systems make exclusive use of application-layer multicast. To allow for network dimensioning and QoS provisioning, an understanding of the underlying traffic characteristics is required. We find in our exemplary evaluations that the host of a web conference session produces a large amount of Ethernet traffic, largely due to the required control of the conference session, that is heavily-tailed distributed and exhibits additionally long-range dependence. For different groups of activities within a web conference session, we find distinctive characteristics of the generated traffic

Towards a Framework for Modelling Multimedia Conferencing Calls in the Next Generation Network

This paper is concerned with the creation of a multiparty multimedia conferencing application which can be used in Next Generation Networks. It begins by suggesting ways in which conferencing can be modeled with a focus on separating signaling and media transfer functionality. Enabling technologies which could support the modeling framework derived and which are compatible with Next Generation Network (NGN) principles are reviewed. Finally, a design and implementation for a simple multimedia conferencing application are described

Peer-to-Peer Conferencing using Blockchain, WebRTC and SIP

  The owner of the centralized video platform has more control over uploaded content than the content producer does. But the other Blockchain-based decentralized video services are attempting to reduce ad pressure and get rid of middlemen. The article suggests a combination of a safe encryption technique and an access control mechanism created "with technology" to create a successful decentralized video streaming platform built on the Blockchain. Peer-to-peer (P2P) overlays are one of the complicated network applications and services that have been migrated to the Web as a result of the increasing support for Web Real-Time Communication (WebRTC) standard in modern browsers for real-time communications. The expansion of access networks’ bandwidth also makes it possible for end users to start their own content businesses. This paper presents a preliminary proposal of metrics and technologies to move toward a decentralized cooperative architecture for large-scale, real-time live stream content de- livery based on WebRTC, without the requirement of a Content Delivery Network (CDN) infrastructure. The paper takes into account the light of the aforementioned aspects [6]

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