Options for Securing RTP Sessions

The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism

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

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

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]

Past, present and future of IP telephony

“Copyright © [2008] IEEE. Reprinted from International Conference on Communication Theory, Reliability, and Quality of Service, 2008. CTRQ '08. ISBN:978-0-7695-3190-8. This material is posted here with permission of the IEEE. Internal or personal use of this material is permitted. However, permission to reprint/republish this material for advertising or promotional purposes or for creating new collective works for resale or redistribution must be obtained from the IEEE by writing to [email protected]. By choosing to view this document, you agree to all provisions of the copyright laws protecting it.”Since the late 90's IP telephony, commonly referred to as Voice over IP (VoIP), has been presented as a revolution on communications enabling the possibility to converge historically separated voice and data networks, reducing costs, and integrating voice, data and video on applications. This paper presents a study over the standard VoIP protocols H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and H.248/Megaco. Given the fact that H.323 and SIP are more widespread than the others, we focus our study on them. For each of these protocols we describe and discuss its main capabilities, architecture, stack protocol, and characteristics. We also briefly point their technical limitations. Furthermore, we present the Advanced Multimedia System (AMS) project, a new system that aims to operate on Next Generation Networks (NGN) taking the advantage of its features, and it is viewed as the successor to H.323 and SIP

Programmable Conference Server

Conferencing services for Internet telephony and multimedia can be enhanced by the integration of other Internet services, such as instant messaging, presence notification, directory lookups, location sensing, email and web. These services require a service programming architecture that can easily incorporate new Internet services into the existing conferencing functionalities, such as voice-enabled conference control. W3C has defined the Call Control eXtensible Markup Language (CCXML), along with its VoiceXML, for telephony call control services in a point-to-point call. However, it cannot handle other Internet service events such as presence enabled conferences. In this paper, we propose an architecture combining VoiceXML with our Language for End System Services (LESS) and the Common Gateway Interface (CGI) for multi-party conference service programming that integrates existing Internet services. VoiceXML provides the voice interface to LESS and CGI scripts. Our architecture enables many novel services such as conference setup based on participant location and presence status. We give some examples of the new services and describe our on-going implementation

Multi-user media streaming service for e-learning based web real-time communication technology

Web real-time communication (WebRTC) standards do not define precisely how two browsers establish and control their communication. Therefore, a signalling mechanism/protocol has not specified in WebRTC. The essential goal of this research is to create and apply a WebRTC bi-directional video conferencing based on mesh topology (many-to-many) using Google Chrome, Firefox, Opera, and Explorer. This experiment involved through Ethernet and Wireless of the Internet and 4G networks in e-learning. The signalling mechanism of this experiment has been created and implemented using JavaScript language along with MultiConnection libraries. In addition, an evaluation of quality of experience (QoE), resources, such as bandwidth consumption, and CPU performance was done. In this paper, a novel implementation was accomplished over e-learning using different networks, different browsers, many peers, opening one or many rooms concurrently, defining room initiator, sharing the information of the new user with participants, using user identification (user-id), and so on. Moreover, the paper also highlights the advantages and disadvantages of using WebRTC video conferencing