Converged multimedia services in emerging Web 2.0 session mobility scenarios

Adeyeye, Michael; Ventura, Neco

doi:10.1007/s11276-011-0394-z

Cited by 10 publications

(5 citation statements)

References 26 publications

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“…Latency in browser-based user-to-user interactions in our work was reported in [2]. A QoS-enabled SIP-based architectural scheme would be required in a bidirectional exchange of time-critical data between Web terminals.…”

Section: Qos-enabled Architectural Schemementioning

confidence: 94%

“…After that, the last part of the paper points out the wide potential area of application of the proposed solution, its easy application to these areas, and the future academic-/industry-oriented research directions that we will follow to extend our work in the next months. The comparison with other technologies in this article does not contain experimental evaluation in terms of performance and accuracy, another article [2] written by the authors reported the performance and accuracy of our work.…”

Section: Introductionmentioning

confidence: 99%

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Emerging research areas in SIP-based converged services for extended Web clients

Adeyeye

Bellavista

2013

World Wide Web

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The convergence of Next Generation Networks and Internet-based rich applications are generating relevant industrial opportunities in the market of mobility-enabled services. Even if this trend is widely recognized, there are still a few industrial-level solutions that effectively support session mobility in a transparent way and with the capability of openly integrating with existing and legacy applications. In this paper we propose a SIP-based hybrid architecture for Web session mobility that offers content sharing and session handoff between Web browsers. In addition, its technical originality includes integrating a SIP stack into a Web browser, thus offering the advantage of extending a Web browser to act as a SIP client. Lastly, a rich set of control services that prevent abuse of content sharing and session handoff are introduced into the proposed system. The implemented solution uses SIP in a standard way to migrate Web sessions between Web browsers; it is made up of a SIP integrated Web client and a converged (SIP and HTTP) Application Server that can be easily used to enable session mobility in any kind of Web-based application. In addition, the implemented system has recently evolved to a framework for developing different kinds of converged services over the Internet, analogously to what is possible with Google Wave and the existing telephony APIs. Finally, the paper reports the evaluation of the proposed framework and of the employed technologies, together with directions of future work, in terms of both extension to other application domains and exploration of research areas/models that can benefit form the adoption of SIP and Web-related solutions.M. Adeyeye ( )

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Section: Qos-enabled Architectural Schemementioning

confidence: 94%

Section: Introductionmentioning

confidence: 99%

Emerging research areas in SIP-based converged services for extended Web clients

Adeyeye

Bellavista

2013

World Wide Web

Self Cite

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“…Other real-time communication solutions or native functionality execution like WebRTC, plugin based solutions [4] or implementation of a Web app with Googles NativeClient are not capable of reacting on external stimula, e.g. if no browser is running it is not possible to take an incoming communication request.…”

Section: Evaluation Of Our Solutionmentioning

confidence: 99%

A Taste of Telco and Web

Zeiss¹,

Gabner²,

Davies³

et al. 2012

PIK - Praxis Der Informationsverarbeitung Und Kommunikation

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This article introduces a novel approach on how to integrate communication services into Web applications running in the browser. The solution is based on two major design decisions: To resolve the need for a business-to-business relationship between Web provider and communication service provider, and to distribute the Model, View and Controller components of an application across different processes. Our approach helps to answer the question on how to efficiently integrate network operator's assets into applications from over the top players. The separation between application control by the Web page and the actual command execution by the native capabilities of the user device opens new opportunities for global reachability of Telco services, easy deployment and redeployment of applications with zero configuration need for users and developers as well as privacy protection by keeping sensitive data within the user domain, e.g. the user's communication device.

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“…There are many implementations of RTC among web browsers [3], [4], [5], [6], [7] and the WebRTC itself [8], [9], [10]. The standard signalling protocol for WebRTC is JavaScript Session Establishment Protocol (JSEP), however, Remote Object Access and Replication (ROAR) has also been used in some existing implementations.…”

Section: Introductionmentioning

confidence: 99%

“…Communication Service Aggregators (CSA) would help customers that run multiple clients on their phones save cost by running their services in cloud and offering them control via a browser. As earlier envisaged in [7], a click-to-call would not require an operator's voice network in order to setup a voice call,. VAS (Value Added Services) would be extended to web developers that can create value and solve problems for customers.…”

Section: Introductionmentioning

confidence: 99%

Determining the signalling overhead of two common WebRTC methods: JSON via XMLHttpRequest and SIP over WebSocket

Adeyeye

Makitla²,

Fogwill³

2013

2013 Africon

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WebReal-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.

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Converged multimedia services in emerging Web 2.0 session mobility scenarios

Cited by 10 publications

References 26 publications

Emerging research areas in SIP-based converged services for extended Web clients

Emerging research areas in SIP-based converged services for extended Web clients

A Taste of Telco and Web

Determining the signalling overhead of two common WebRTC methods: JSON via XMLHttpRequest and SIP over WebSocket

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