1,218 research outputs found

    CIAO: A Component Model and its OSGi Framework for Dynamically Adaptable Telephony Applications

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    International audienceIn recent years, thanks to new IP protocols like SIP, telephony applications and services have evolved to other and combine a variety of communication forms including presence status, instant messaging and videoconference. As a result, advanced telephony applications now consist of distributed entities that are involved into multiple heterogeneous, stateful and long-running interactions (sessions). This evolution complicated significantly applications development and calls for more effective solutions. In this paper, we explore the adoption of components for addressing this issue, focusing specifically on the management and coordination of the numerous and various sessions occurring in such applications. The paper presents CIAO, a domain-specific and hierarchical component model for SIP applications. CIAO combines three kinds of component that are Actor, SessionPart and Role and manage them dynamically in accordance with real SIP sessions. By using these features, we are able to break the complexity of SIP entities and provide flexibility for their development. CIAO is implemented above OSGi to experiment the building of concrete SIP applications and enable their dynamic adaptation

    A Tool for VoIP Audio Extraction

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    Cílem práce je vytvořit systém, který dokáže rekonstruovat audio data z VoIP komunikace. Systém rozpozná v záznamu síťového provozu proudy VoIP  paketů a na základě jejich obsahu sestaví přenášený audio signál.  Kromě rozšířeného RTP protokolu je podporován také IAX protokol používaný Asterisk ústřednou, který nabízí zajímavé možnosti a není plně či vůbec podporován dostupnými nástroji. Systém je implementován jako knihovna s minimálním rozhraním.In this thesis, we describe VoIP protocols and design of a system to reconstruct audio data from VoIP communication. The system is able to detect VoIP packet streams in an IP network traffic and assemble an audio signal they carry. RTP and IAX VoIP protocols are supported. Unlike widespread RTP protocol, IAX is not fully supported by available tools although it is used by increasingly popular Asterisk communications project and offers interesting features not found in RTP. The system is implemented as a library with minimal frontend.

    Optimization of Elastic Cloud Brokerage Mechanisms for Future Telecommunication Service Environments

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    Dieser Beitrag ist mit Zustimmung des Rechteinhabers aufgrund einer (DFG geförderten) Allianz- bzw. Nationallizenz frei zugänglich.This publication is with permission of the rights owner freely accessible due to an Alliance licence and a national licence (funded by the DFG, German Research Foundation) respectively.Cloud computing mechanisms and cloud-based services are currently revolutionizing Web as well as telecommunication service platforms and service offerings. Apart from providing infrastructures, platforms and software as a service, mechanism for dynamic allocation of compute and storage resources on-demand, commonly termed as “elastic cloud computing” account for the most important cloud computing functionalities. Resource elasticity allows not only for efficient internal compute and storage resource consumption, but also, through so called hybrid cloud computing mechanisms, for dynamic utilization of external resources on-demand. This capability is especially useful in order to cost-efficiently cope with peakworkloads, allowing service providers to significantly reduce usually required over-provisioned service infrastructures, allowing for “pay-per-use” cost models. With a steadily growing number of cloud providers and with the proliferation of unified cloud computing interfaces, service providers are given free choice of flexibly selecting and utilizing cloud resources from different cloud providers. Cloud brokering systems allow for dynamic selection and utilization of cloud computing resources based on functional (e.g. QoS, SLA, energy consumption) as well as nonfunctional criteria (e.g. costs). The presented work focuses on enhanced cloud brokering mechanisms for telecommunication service platforms, enabling quality telecommunication service assurance, still optimizing cloud resources consumption, i.e. saving costs and energy. Furthermore this work shows that by combining cloud brokering mechanisms with standardized telecommunication service brokering mechanisms an even greater benefit for telecommunication service providers can be achieved as this enables an even better cost-efficiency since different user segments can seamlessly be served by allocating different cloud resources to them in a policy-driven manner

    Potential Applications of IPsec in Next Generation Networks

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    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

    Signaling for Internet Telephony

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    Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

    Peer-to-Peer Communication between Android Mobiles

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    In the current days Voice telephony over mobile is possible considering GSM system which is cost consuming. Wi-Fi technology is a form of telecommunication that allows data and voice transmissions over a wide range of interconnected networks. In this thesis we provides a mechanism for live communication over IP using mobile phones at no cost. The purpose of this research is to design and implement a telephony program that uses WI-FI in p2p (Peer-to-Peer) or WLAN (Wireless Local Area Network) as a means of communication between mobile phones. The system will allow users to establish p2p voice connection through Access Points (AP) and then allow user to make voice conversation, sending SMS (Short Message Service). The current system will only allow for one call per connection, and no call waiting, or conference calls. The group chat application is the number of users connected to the server

    A study on the effects of e-navigation on reducing vessel accidents

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    The dissertation aims to evaluate how and to what extent e-navigation contributes to reducing accidents for SOLAS ships as well as non-SOLAS ships, hoping that the results are referred to IMO Member States when they are implementing enavigation along with the maritime sectors such as shipping companies, crews on board ships and manufactures developing e-navigation related systems. The study focuses on the potential effects of e-navigation based on tool kits of the IMO e-navigation for SOLAS ships and services of SMART-navigation, which is the Korean approach to implementing the e-navigation concept for both SOLAS ships and non-SOLAS ships. The processes and the methodologies that are used by the IMO to assess the effects of e-navigation are investigated. The vessel accidents for all ships in Korean waters and all Korean-flagged ships worldwide during the 5 years from 2009 to 2013 are analyzed. The formula is proposed to calculate the effects of e-navigation on reducing accidents, which can also be used by other Member States of the IMO when they implement e-navigation in their waters. The direct causes of accidents, which are reducible by the risk control options (RCOs), and the RCOs, which are applicable to non-SOLAS ships, are identified. Additionally, an expert questionnaire survey is carried out with a view to supporting the validity of identifying the RCOs and the direct causes. The results are collated and evaluated for the potential effects of e-navigation on reducing accidents, in relation to type of accidents as well as type of ships, for comparison with the results obtained by the IMO and for reference of other Member States. The concluding chapter examines the results of analysis of e-navigation\u27s tool kits and methodologies to assess their effects on reducing accidents, and discusses the potential rate of accident reduction through e-navigation. A number of recommendations are made concerning the need for further investigation in quantifying the coefficient applied to the proposed formula for evaluating the effects of e-navigation
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