Sip Based Mobile Voice Over Ip Client For Wireess Networks

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2008Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2008Bu tez SIP tabanlı mobile bir VoIP istemcisinin tasarımını ve gerçeklenmesini tanımlar. Bu tez temelde çoktürel ağlar üzerinde çalışabilen bir VoIP istemcisi tasarımının çözülmesi gereken iki sorununun üzerinde yogunlaşır. Birinci ve en zorlu sorun farklı erişim teknolojileri arasında kullanıcıya fark ettirmeden yer değişim desteği sağlanmasıdır. Bu tezde, kullanıcıya fark ettirmeden el değiştirme yönetimi, uygulama katmanında, multimedya oturumunu başlatmak, sonlandırmak ve değiştirmek için kullanılan Oturum Başlatma Protokolü (SIP) kullanılarak ele alınmıştır. SIP yaygın bir şekilde kabul edilmekte olan bir VoIP standartıdır. Kullanıcıya fark ettirmeden el değiştirmeyi destekleyebilmek için, VoIP istemcisi üzerinde çalışan SIP tabanlı bir bağlantı yöneticisi önerilmiştir. Bağlantı yöneticisi yeni ağlar keşfettiğinde, adaylar listesinden bir ağ seçer ve hali hazırda yürütülmekte olan iletişimi kullanıcıya fark ettirmeden yeni ağ arayüzüne aktarır. Dolayısı ile, bu birim Wi-Fi, 3G gibi çoktürel ağlar arasında dolaşmayı sağlar. İkinci sorun ise, en kaliteli çağrı (arama) desteğini sağlamaktır. En kaliteli çağrı desteği, iletişim kurmak isteyen tarafların farklı türden ağlara bağlı olmaları durumunda, VoIP uygulamasının iletişim tipine (yarı-çift yönlü yada tam-çift yönlü) karar vermebilmesi demektir. Örneğin, eğer iletişim kurmak isteyen taraflardan biri bir GSM ağındaysa, en iyi çağrı kalitesini yakalayabilmek için, iletişim yarı-çift yönlü olarak kurulmalıdır. Bu tez, bahsedilen özelliği desteklemek için, istemci tabanlı bir karar mekanizması önerir. Bu karar mekanizması, iletişim kurulmak istenen tarafa, istemcinin içinde bulunduğu ağa göre belirlenmiş iletişim tipini içeren bir davet iletisi gönderir. Diğer istemci bu davet iletisini aldıktan sonra, aynı karar mekanizması, iletişimi “bas-konuş VoIP” yada “tam-çift yönlü VoIP” olarak ayarlar.This thesis describes the design and the implementation of a SIP-based mobile VoIP client. It mainly focuses on two challenges of designing a VoIP client which works on heterogeneous network environments. One and the most challenging problem is the provision of seamless mobility support among different access technologies. In this thesis, seamless handover management is handled at the application layer by using Session initiation protocol (SIP), which is used to initiate, terminate, and modify multimedia session. SIP is becoming a widely accepted standard for VoIP. To support seamless handover, a SIP based connection manager is proposed on VoIP client application. As new networks are discovered by the connection manager, it selects a new network from the candidate list and transfers the current communication to the new network interface seamlessly. Therefore, this module provides roaming across heterogeneous networks such as Wi-Fi, 3G. Second problem is providing the best effort call quality support. It means that if the communication parties are in dissimilar networks, the VoIP application should decide the communication type (half-duplex or full-duplex). For instance, if one of the communication parties is in a GSM network, then the communication should be established as a half-duplex manner to achieve best call quality. This thesis proposes a client-based decision mechanism to support this property. This decision mechanism sends an invite message including the communication type (half-duplex or full-duplex) of the client according to the network in which it operates to the other communication party. After the other client receives this invite message, same decision mechanism adjusts the communication as either a “push to talk VoIP” or a “full-duplex VoIP”.Yüksek LisansM.Sc

Voice over IP

The area that this thesis covers is Voice over IP (or IP Telephony as it is sometimes called) over Private networks and not over the Internet. There is a distinction to be made between the two even though the term is loosely applied to both. IP Telephony over Private Networks involve calls made over private WANs using IP telephony protocols while IP Telephony over the Internet involve calls made over the public Internet using IP telephony protocols. Since the network is private, service is reliable because the network owner can control how resources are allocated to various applications, such as telephony services. The public Internet on the other hand is a public, largely unmanaged network that offers no reliable service guarantee. Calls placed over the Internet can be low in quality, but given the low price, some find this solution attractive. What started off as an Internet Revolution with free phone calls being offered to the general public using their multimedia computers has turned into a telecommunication revolution where enterprises are beginning to converge their data and voice networks into one network. In retrospect, an enterprise\u27s data networks are being leveraged for telephony. The communication industry has come full circle. Earlier in the decade data was being transmitted over the public voice networks and now voice is just another application which is/will be run over the enterprises existing data networks. We shall see in this thesis the problems that are encountered while sending Voice over Data networks using the underlying IP Protocol and the corrective steps taken by the Industry to resolve these multitudes of issues. Paul M. Zam who is collaborating in this Joint Thesis/project on VoIP will substantiate this theoretical research with his practical findings. On reading this paper the reader will gain an insight in the issues revolving the implementation of VoIP in an enterprises private network as well the technical data, which sheds more light on the same. Thus the premise of this joint thesis/project is to analyze the current status of the technology and present a business case scenario where an organization will be able to use this information

Junos OS Security Configuration Guide

This preface provides the following guidelines for using the Junos OS Security Configuration Guide: • J Series and SRX Series Documentation and Release Notes on page xli • Objectives on page xlii • Audience on page xlii • Supported Routing Platforms on page xlii • Document Conventions on page xlii • Documentation Feedback on page xliv • Requesting Technical Support on page xliv Juniper Networks supports a technical book program to publish books by Juniper Networks engineers and subject matter experts with book publishers around the world. These books go beyond the technical documentation to explore the nuances of network architecture, deployment, and administration using the Junos operating system (Junos OS) and Juniper Networks devices. In addition, the Juniper Networks Technical Library, published in conjunction with O'Reilly Media, explores improving network security, reliability, and availability using Junos OS configuration techniques. All the books are for sale at technical bookstores and book outlets around the world. The current list can be viewed at http://www.juniper.net/books .Junos OS for SRX Series Services Gateways integrates the world-class network security and routing capabilities of Juniper Networks. Junos OS includes a wide range of packet-based filtering, class-of-service (CoS) classifiers, and traffic-shaping features as well as a rich, extensive set of flow-based security features including policies, screens, network address translation (NAT), and other flow-based services. Traffic that enters and exits services gateway is processed according to features you configure, such as packet filters, security policies, and screens. For example, the software can determine: • Whether the packet is allowed into the device • Which firewall screens to apply to the packet • The route the packet takes to reach its destination • Which CoS to apply to the packet, if any • Whether to apply NAT to translate the packet’s IP address • Whether the packet requires an Application Layer Gateway (ALG

A novel distributed architecture for IoT image processing using low-cost devices and open internet standards

Industry 4.0 can be defined as the integration of computers and automation to current industrial processes, with addition of smart and autonomous systems leveraged by machine learning techniques. In this scenario, a compact, dependable and fast controller is desired, featuring low energy consumption, easily programming and maintenance, with no mobile parts. Nowadays, computing power in single board computers, e.g. the Raspberry Pi among others, has been increased at a very important rate. In just three generations, Pi computers offer almost a two-fold speed gain, when compared to first models. Its design, an underlying video driver with general capabilities of regular OSes, makes them quite suitable to build image processing systems at very low cost, with no mobile parts and low energy consumption. However, designing such a system for industrial image processing is a tough challenge, since it implies to integrate cameras, image processing libraries, database servers and application software with graphical user interface, in an already resource constrained device. This work presents a new architecture for this kind of systems, by means of open internet standards, using a self-contained, high performance web server to publish a RESTful API and a set of web pages that use latest HTML5 capabilities to manage USB webcams and system data. This proposal also integrates OpenCV as a compiled script on client-side using the new WASM paradigm, with an optimized storage for images using -industry-standard RDBMS and a modular design that can target Windows and Linux as well.Sociedad Argentina de Informática e Investigación Operativ

Creation of value with open source software in the telecommunications field

Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200

An Ontology for Network Services

Most of the network service specifications are implemented using relational databases or XML schemas. However, those specifications are not flexible and expressive enough to be extended with new service classes, different corporate policies, network configurations and deployment strategies; thus, most of the QoS management operations are implemented as hard-coded software components. This paper presents a novel approach in the specification of IP network services, using F-logic knowledge representation framework, aiming to include, in the same specification, the high-level service requirements, the network model and the necessary operations for the deployment of multiple network services

Secure VoIP Performance Measurement

This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality

Design and Performance Evaluation of Passive Optical Networks

Currently, new housing developments in many places around the world are built with fiber-based connections to the home, and network providers are conducting field-testing and experiments with fiber access. In order to provide a worthy alternative to the existing infrastructures, the new technology should be, among other things, cost-efficient, broad-banded, and easy to maintain and deploy. It must also support all existing services as well as offer new required services. These services include voice, data, and video/television-broadcast traffic. In this project we will carry out exploration of some of the aspects of the QoS bandwidth allocation in the 802.3ah EPON architecture and the GPON architecture in a multimedia environment. Several general traffic types will be defined, that would represent real traffic in the network, each with its own QoS requirements (bandwidth, delay, etc.). Sometimes we will use real traffic (voice, video, data). Moreover, another goal of this project is to provide an operating and configuration reference tool-like manual facilitating the functional and performance analysis of this kind of networks. This tool-like manual will include step-by-step the way to discover the behaviour of these networks. Due to the required extension of this document, the manual has been included in Annex C

Design, implementation and evaluation of unified communications on-premises and over the Cloud

Unified Communication (UC) is the integration of two or more real time communication systems into one platform. Integrating core communication systems into one overall enterprise level system delivers more than just cost saving. These real-time interactive communication services and applications over Internet Protocol (IP) have become critical in boosting employee accessibility and efficiency, improving customer support and fostering business agility. However, some small and medium-sized businesses (SMBs) are far from implementing this solution due to the high cost of initial deployment and ongoing support. Cloud based UC solution, UC as a Service (UCaaS), is now itself a maturing technology in the marketplace and it has revolutionized the IT industry, being the powerful platform that many businesses are choosing to migrate their on-premises UC solution onto. UCaaS solution has the potential to reduce the capital and operational expenses associated with deploying UC on their own. In this paper, we will discuss and demonstrate an open source on-premises UC solution, viz. “Asterisk” for use by businesses, and report on some performance tests using SIPp. This paper also discusses and demonstrates an open source UCaaS solution. The contribution from this research is the provision of technical advice to businesses in deploying UC and UCaaS, which is manageable in terms of cost, ease of deployment and support