2,080 research outputs found

    Quality of Service challenges for Voice over Internet Protocol (VoIP) within the wireless environment

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

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    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

    Designing and prototyping WebRTC and IMS integration using open source tools

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    WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

    Voice over Internet Protocol (VOIP): Overview, Direction And Challenges

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    Voice will remain a fundamental communication media that cuts across people of all walks of life. It is therefore important to make it cheap and affordable. To be reliable and affordable over the common Public Switched Telephone Network, change is therefore inevitable to keep abreast with the global technological change. It is on this basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that would not only maximize cost but keep abreast with the global technological change. Keywords: Voice over Internet Protocol (VoIP), Public Switched Telephone Network (PSTN), Session Initiation Protocol (SIP),  multipoint control uni

    Performance evaluation of a technology independent security gateway for Next Generation Networks

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    With the all IP based Next Generation Networks being deployed around the world, the use of real-time multimedia service applications is being extended from normal daily communications to emergency situations. However, currently different emergency providers utilise differing networks and different technologies. As such, conversations could be terminated at the setup phase or data could be transmitted in plaintext should incompatibility issues exit between terminals. To this end, a novel security gateway that can provide the necessary security support for incompatible terminals was proposed, developed and implemented to ensure the successful establishment of secure real-time multimedia conversations. A series of experiments were conducted to evaluate the security gateway through the use 40 Boghe softphone acting as the terminals. The experimental results demonstrate that the best performance of the prototype was achieved by utilising a multithreading and multi-buffering technique, with an average of 582 microseconds processing overhead. Based upon the ITU-Ts 150 milliseconds one way delay recommendation for voice communications, it is envisaged that such a marginal overhead will not be noticed by users in practice

    Mobility Schemes for future networks based on the IMS

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