AN ANALYSIS OF VOICE OVER INTERNET PROTOCOL (VOIP) AND ITS SECURITY IMPLEMENTATION

Voice over Internet Protocol (VoIP) has been in existence for a number of years but only quite recently has it developed into mass adoption. As VoIP technology penetrates worldwide telecommunications markets, the advancements achieved in performance, cost reduction, and feature supportmake VoIP a convincingproposition for service providers, equipment manufacturers, and end users. Since the introduction of mass-market VoIP services over broadband Internet in 2004, security and safeguarding are becoming a more important obligation in VoIP solutions. The purpose of this final year project is to study and analyze VoIP and implement the security aspect using Secure Real-time Transport Protocol (SRTP) end-to-end media encryption in the Universiti Teknologi PETRONAS (UTP) laboratory. Extensive research, evaluation of case studies, literature reviews, network analysis, as well as testing and experimentation are the methods employed in achieving a secure and reliable VoIP network. With the given time frame and adequate resources, the study and analysis of VoIP and implementation of SRTP should prove to be very successful

Enterprise network convergence: path to cost optimization

During the past two decades, telecommunications has evolved a great deal. In the eighties, people were using television, radio and telephone as their communication systems. Eventually, the introduction of the Internet and the WWW immensely transformed the telecommunications industry. This internet revolution brought about a huge change in the way businesses communicated and operated. Enterprise networks now had an increasing demand for more bandwidth as they started to embrace newer technologies. The requirements of the enterprise networks grew as the applications and services that were used in the network expanded. This stipulation for fast and high performance communication systems has now led to the emergence of converged network solutions. Enterprises across the globe are investigating new ways to implement voice, video, and data over a single network for various reasons – to optimize network costs, to restructure their communication system, to extend next generation networking abilities, or to bridge the gap between their corporate network and the existing technological progress. To date, organizations had multiple network services to support a range of communication needs. Investing in this type of multiple communication infrastructures limits the networks ability to provide resourceful bandwidth optimization services throughout the system. Thus, as the requirements for the corporate networks to handle dynamic traffic grow day by day, the need for a more effective and efficient network arises. A converged network is the solution for enterprises aspiring to employ advanced applications and innovative services. This thesis will emphasize the importance of converging network infrastructure and prove that it leads to cost savings. It discusses the characteristics, architecture, and relevant protocols of the voice, data and video traffic over both traditional infrastructure and converged architecture. While IP-based networks present excellent quality for non real-time data networking, the network by itself is not capable of providing reliable, quality and secure services for real-time traffic. In order for IP networks to perform reliable and timely transmission of real-time data, additional mechanisms to reduce delay, jitter and packet loss are required. Therefore, this thesis will also discuss the important mechanisms for running real-time traffic like voice and video over an IP network. Lastly, it will also provide an example of an enterprise network specifications (voice, video and data), and present an in depth cost analysis of a typical network vs. a converged network to prove that converged infrastructures provide significant savings

Multimedia

The nowadays ubiquitous and effortless digital data capture and processing capabilities offered by the majority of devices, lead to an unprecedented penetration of multimedia content in our everyday life. To make the most of this phenomenon, the rapidly increasing volume and usage of digitised content requires constant re-evaluation and adaptation of multimedia methodologies, in order to meet the relentless change of requirements from both the user and system perspectives. Advances in Multimedia provides readers with an overview of the ever-growing field of multimedia by bringing together various research studies and surveys from different subfields that point out such important aspects. Some of the main topics that this book deals with include: multimedia management in peer-to-peer structures & wireless networks, security characteristics in multimedia, semantic gap bridging for multimedia content and novel multimedia applications

Designing and optimization of VOIP PBX infrastructure

In the recent decade, communication has stirred from the old wired medium such as public switched telephone network (PSTN) to the Internet. Present, Voice over Internet Protocol (VoIP) Technology used for communication on internet by means of packet switching technique. Several years ago, an internet protocol (IP) based organism was launched, which is known as Private Branch Exchange "PBX", as a substitute of common PSTN systems. For free communication, probably you must have to be pleased with starting of domestic calls. Although, fairly in few cases, VoIP services can considerably condense our periodical phone bills. For instance, if someone makes frequent global phone calls, VoIP talk service is the actual savings treat which cannot achieve by using regular switched phone. VoIP talk services strength help to trim down your phone bills if you deal with a lot of long-distance (international) and as well as domestic phone calls. However, with the VoIP success, threats and challenges also stay behind. In this dissertation, by penetration testing one will know that how to find network vulnerabilities how to attack them to exploit the network for unhealthy activities and also will know about some security techniques to secure a network. And the results will be achieved by penetration testing will indicate of proven of artefact and would be helpful to enhance the level of network security to build a more secure network in future

An improved medium access control protocol for real-time applications in WLANs and its firmware development

The IEEE 802.11 Wireless Local Area Network (WLAN), commonly known as Wi-Fi, has emerged as a popular internet access technology and researchers are continuously working on improvement of the quality of service (QoS) in WLAN by proposing new and efficient schemes. Voice and video over Internet Protocol (VVoIP) applications are becoming very popular in Wi-Fi enabled portable/handheld devices because of recent technological advancements and lower service costs. Different from normal voice and video streaming, these applications demand symmetric throughput for the upstream and downstream. Existing Wi-Fi standards are optimised for generic internet applications and fail to provide symmetric throughput due to traffic bottleneck at access points. Performance analysis and benchmarking is an integral part of WLAN research, and in the majority of the cases, this is done through computer simulation using popular network simulators such as Network Simulator ff 2 (NS-2) or OPNET. While computer simulation is an excellent approach for saving time and money, results generated from computer simulations do not always match practical observations. This is why, for proper assessment of the merits of a proposed system in WLAN, a trial on a practical hardware platform is highly recommended and is often a requirement. In this thesis work, with a view to address the abovementioned challenges for facilitating VoIP and VVoIP services over Wi-Fi, two key contributions are made: i) formulating a suitable medium access control (MAC) protocol to address symmetric traffic scenario and ii) firmware development of this newly devised MAC protocol for real WLAN hardware. The proposed solution shows signifocant improvements over existing standards by supporting higher number of stations with strict QoS criteria. The proposed hardware platform is available off-the-shelf in the market and is a cost effective way of generating and evaluating performance results on a hardware system

Designing and prototyping WebRTC and IMS integration using open source tools

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