In this paper, we consider networking technologies like Software-Defined Networks (SDN) and Network Functions Virtualization (NFV) to analyze, as a first step, the opportunities that directly address ...the limitations of today's network architectures and lay a foundation for concrete interactions between network providers and Over-The-Top (OTT) service providers. To study the strength of adopting these two emerging technologies in operator networks, we apply this analysis on the case of WebRTC communication services. Indeed, WebRTC presents significant changes in the communication model which makes it an interesting case of study. Thus, our work initiates and discusses the importance of a "Network-as-a-Service" model between network operators and OTT Communication Service Providers (CSPs) in an SDN-NFV network environment.
The web application paradigm has become a dominant choice due to the ubiquity of web browsers across PCs and mobile devices, and the layer of abstraction that they provide. However, browsers have ...traditionally communicated only with servers, never from one browser to another. The WebRTC API overcomes this limitation, providing further avenues for the web application paradigm to evolve. In this paper we present Cyclon.p2p an implementation of the Cyclon peer sampling protocol using the WebRTC API. We provide a detailed overview of our system, and conduct a thorough evaluation of the implementation in real-world and simulated experiments. Our results strongly suggest that our implementation is feasible for large scale deployment in real world Web applications.
Les services de communication, du courrier postal à la téléphonie, en passant par la voix et la vidéo sur IP (Internet Protocol), la messagerie électronique, les salons de discussion sur Internet, ...les visioconférences ou les télécommunications immersives ont évolué au fil du temps. Un système de communication voix-vidéo sur IP est réalisé grâce à deux couches architecturales fondamentales : la couche de signalisation et la couche média. Le protocole de signalisation est utilisé pour créer, modifier et terminer des sessions multimédias entre des participants. La couche de signalisation est divisée en deux sous-couches - la couche de service et celle de contrôle - selon la spécification de l’IP Multimedia Subsystem (IMS). Deux systèmes de communication largement utilisés sont l’IMS et SIP Pair-à- Pair (P2P SIP). Les fournisseurs de services, qui se comportent en tant qu’intermédiaires entre appelants et appelés, implémentent les systèmes de communication, contrôlant strictement la couche signalisation. Or ces fournisseurs de services ne prennent pas en compte la diversité des utilisateurs. Cette thèse identifie trois barrières technologiques dans les systèmes de communication actuels et plus précisément concernant la couche de signalisation. I. Un manque d’ouverture et de flexibilité dans la couche de signalisation pour les utilisateurs. II. Un développement difficile des services basés sur le réseau et les sessions. III. Une complexification du la couche de signalisation lors d’un très grand nombre d’appels. Ces barrières technologiques gênent l’innovation des utilisateurs avec ces services de communication. Basé sur les barrières technologiques listées cidessus, le but initial de cette thèse est de définir un concept et une architecture de système de communication dans lequel chaque individu devient un fournisseur de service. Le concept, "My Own Communication Service Provider" (MOCSP) et le système MOCSP sont proposés, accompagné d’un diagramme de séquence. Ensuite, la thèse fournit une analyse qui compare le système MOCSP avec les systèmes de communication existants en termes d’ouverture et de flexibilité. La seconde partie de la thèse présente des solutions pour les services basés sur le réseau ou les sessions, mettant en avant le système MOCSP proposé. Deux services innovants, user mobility et partial session transfer/retrieval (PSTR) sont pris comme exemples de services basés sur le réseau ou les sessions. Les services basés sur un réseau ou des sessions interagissent avec une session ou sont exécutés dans une session. Dans les deux cas, une seule entité fonctionnelle entre l’appelant et l’appelé déclenche le flux multimédia pendant l’initialisation de l’appel et/ou en cours de communication. De plus, la coopération entre le contrôle d’appel réseau et les différents pairs est facilement réalisé. La dernière partie de la thèse est dédiée à l’extension de MOCSP en cas de forte densité d’appels, elle inclut une analyse comparative. Cette analyse dépend de quatre facteurs - limite de passage à l’échelle, niveau de complexité, ressources de calcul requises et délais d’établissement de session - qui sont considérés pour évaluer le passage à l’échelle de la couche de signalisation. L’analyse comparative montre clairement que la solution basée sur MOCSP est simple et améliore l’usage effectif des ressources de calcul par rapport aux systèmes de communication traditionnels
Different communication services from delivery of written letters to telephones, voice/video over Internet Protocol(IP), email, Internet chat rooms, and video/audio conferences, immersive communications have evolved over time. A communication system of voice/video over IP is the realization of a two fundamental layered architecture, signaling layer and media layer. The signaling protocol is used to create, modify, and terminate media sessions between participants. The signaling layer is further divided into two layers, service layer and service control layer, in the IP Multimedia Subsystem (IMS) specification. Two widely used communication systems are IMS, and Peer-to-Peer Session Initiation Protocol (P2P SIP). Service providers, who behave as brokers between callers and callees, implement communication systems, heavily controlling the signaling layer. These providers do not take the diversity aspect of end users into account. This dissertation identifies three technical barriers in the current communication systems especially in the signaling layer. Those are: I. lack of openness and flexibility in the signaling layer for end users. II. difficulty of development of network-based, session-based services. III. the signaling layer becomes complex during the high call rate. These technical barriers hinder the end-user innovation with communication services. Based on the above listed technical barriers, the first part of this thesis defines a concept and architecture for a communication system in which an individual user becomes the service provider. The concept, My Own Communication Service Provider (MOCSP) and MOCSP system is proposed and followed by a call flow. Later, this thesis provides an analysis that compares the MOCSP system with existing communication systems in terms of openness and flexibility. The second part of this thesis presents solutions for network-based, session based services, leveraging the proposed MOCSP system. Two innovative services, user mobility and partial session transfer/retrieval are considered as examples for network-based, session-based services. The network-based, sessionbased services interwork with a session or are executed within a session. In both cases, a single functional entity between caller and callee consistently enables the media flow during the call initiation and/or mid-call. In addition, the cooperation of network call control and end-points is easily achieved. The last part of the thesis is devoted to extending the MOCSP for a high call rate and includes a preliminary comparative analysis. This analysis depends on four factors - scalability limit, complexity level, needed computing resources and session setup latency - that are considered to specify the scalability of the signaling layer. The preliminary analysis clearly shows that the MOCSP based solution is simple and has potential for improving the effective usage of computing resources over the traditional communication systems
Web Real-Time Communication (WebRTC) introduces real-time multimedia communication as native capabilities of Web browsers. With the adoption of WebRTC the Web browsers will be able to use WebRTC to ...communicate with one another (peer-to-peer), and with WebSocket servers such as Mobicents SIP Servlets and other server technologies that support WebSocket communication to enable SIP-to-WebRTC communication. This paper outlines the potential of WebRTC and discusses the two common methods of doing real-time communication in Web browsers through WebRTC. The methods are JavaScript Object Notation (JSON) via XMLHttpRequest (XHR) and Session Initiation Protocol (SIP) via WebSocket. A three-user WebRTC video chat prototype application was developed and used to evaluate both methods. Additional signalling overhead introduced into a browser by each method was determined. The results showed WebRTC-SIP/WS has more overhead than WebRTC-JSON/XHR. These signalling overhead findings are useful in that they could help application developers make decision on their choice of technologies and protocols when developing WebRTC-supported applications.
Telemedicine for emergency care management using WebRTC Vidul, A. P.; Hari, Shibin; Pranave, K. P. ...
2015 International Conference on Advances in Computing, Communications and Informatics (ICACCI),
08/2015
Conference Proceeding
With the rapid advancement and development in the field of real time communications, telemedicine has reached great heights and had helped save millions of lives during emergency situations. The use ...of telemedicine is longstanding but its application in emergency care management is still in its developmental stage. Currently there are several emergency telemedicine systems available in the market that uses up-to-date vehicle electronics, latest telecommunications technology and specialized software. However these systems are highly sophisticated, bulky, and expensive and are employed by very few health centers. Thus these life saving services are not available to large portion of the population especially those living in the rural areas. The aim of this paper is to come up with a new idea where these Tele Emergency services can be implemented in a much efficient, economical and less sophisticated manner so that these services can be provided widely. Here we propose a new Emergency Telemedicine Application for emergency care management which uses WebRTC for real time communication. This system just requires a mobile device with internet connection with either chrome or Mozilla browser installed in it. The device is carried within the ambulance to conduct an initial assessment of the patient and later brought to the nearest health center where further treatment is carried under the assistance of specialists whose telepresence is provided by WebRTC enabled devices.
Mediation-based communications Gregoire, Jean-Charles
2017 20th Conference on Innovations in Clouds, Internet and Networks (ICIN),
2017-March
Conference Proceeding
IP telephony has evolved following a structure inherited from traditional telephony, that is, a relay-based, station to station model. This philosophy has deeply influenced the evolution of the ...traditional support protocols, SIP/SDP. However, with the emergence of an embedded form of voice and video communications, supported by new platforms such as WebRTC, other models for user communications become possible. In this paper, we consider that communications are no longer established from station to station, but mediated by an independent entity at the request of a party. This approach leads to simplifications in call management and the underlying protocols, as well as opening the door to new forms of call models and features.
RTMFP and WebRTC standards for Web-based real-time media data sharing applications are provided significant gains. These types of applications and are among the topics researched intensely. A model ...for the web based real time transmission of media data is suggested within this study. WebRTC and RTMFP standards are compared with regard to the factors affecting the service quality such as packets for signaling and connectability as to be between different NAT types, addressing specifically the techniques that are used for NAT transmission. Advantages and disadvantages are demonstrated.
P2P live video streaming in WebRTC Rhinow, Florian; Veloso, Pablo Porto; Puyelo, Carlos ...
2014 World Congress on Computer Applications and Information Systems (WCCAIS),
01/2014
Conference Proceeding, Journal Article
In this paper, we analyse the feasibility of implementing live video streaming protocols into web applications with the use of WebRTC. As a result of demand the distribution of video content requires ...ever increasing bandwidth. Although, specialised programs exist to distribute video content efficiently, web pages have up until recently not been able to leverage these technologies. WebRTC could serve as a solution by enabling peer-to-peer communication directly between browsers without any need for a server as an intermediary. The feasibility analysisis accompanied by a practical implementation of a peer-to-peer streaming protocol in WebRTC that runs natively in all browsers and an identification of optimal settings for such a protocol. Our work highlights current limitations and future challenges when implementing sophisticated peer-to-peer solutions using a technology that is still in its infancy. Finally, we provide preliminary experimental data on WebRTC which measures the performance of such a system in a laboratory environment.
In this paper we present the implementation of a WebRTC gateway service that can forward ad-hoc RTP data plane traffic from a browser inside a local hospital network to a browser on a local home ...network. The gateway leverages the same infrastructure used by the hospital to tunnel sensor and control data for medical devices in home-care deployments. In our use case, doctors at hospitals can only access port 80 through the hospital firewall on external machines, and they need to communicate with patients who are typically behind a NAT in a local WiFi network. VPN solutions only work for staff but not between patients and staff. Our solution solves this problem by redirecting all WebRTC traffic through a gateway service on the local network that has a secure tunnel established with a public gateway. The public gateway redirects traffic from multiple concurrent streams securely between local gateway services that connect to it. The local gateways also communicate with browsers on their local network to mimic a direct browser-to-browser connection without having to change the browser runtime. We have demonstrated that this technique works well within the hospital network and arbitrary patient networks, without the need for any individual host configuration. In our evaluation we show that the latency overhead is 18-20 ms for each concurrent stream added to the same gateway service, which is not discernible with a naked eye until you have more than 10 concurrent streams.
This paper deals with a numerical solution enabling universities to extend the functionality of their distance learning platform to improve not only their model of delivery of educational content but ...also their evaluation of learners' knowledge. Indeed, this paper proposes a solution for e-learning platform Moodle, to create virtual classrooms, integrating video, audio, chat, audio recording of different actors. This solution integrates an audio knowledge assessment tool for language courses and allows learners to learn to write or do online assignments requiring extensive use of symbols and mathematical tools. Our solution which is built around the WebRTC technology, is used directly with popular browsers without installing plugins as well as from computers as from smartphones and tablets. Our solution has been tested by the Virtual University of Senegal actors and allowed teachers of language and mathematics to assess students online as it should be without necessarily making use of multiple choice or having to modify the tests to take in charge the difficulties associated with the use of platforms for online assessments.
