Modelling and Simulation of SIP and IAX Sessions

Import 03/11/2016My thesis is focused on simulating a functioning model of SIP and IAX and compare these two VoIP protocols. This is done by implementing an Asterisk server onto two virtual machines with Ubuntu operating system where I build a trunking system for each protocol, tested it by calling the peers in both directions, captured the traffic passing through and analysed it with Wireshark. The acquired data is then implemented and presented on a chart form for a better view and comparison of the two parallel protocols.Moje práce je zaměřena na simulaci funkčnosti modelu SIP a IAX a porovnání těchto dvou VoIP protokolů. To je provedeno zavedením Asteriskem serveru na dva virtuální počítaček s operačním systémem Ubuntu, kde je vybudován trunking systém pro každý protokol a to tak, že spojuje volající v obou směrech, zachycuje průchod, a analyzuje pomocí Wireshark. Získaná data jsou pak použita a prezentována ve formě grafů pro lepší přehlednost a srovnání obou paralelních protokolů.440 - Katedra telekomunikační technikydobř

A comparative study of in-band and out-of-band VOIP protocols in layer 3 and layer 2.5 environments

For more than a century the classic circuit-switched telephony in the form of PSTN (Public Service Telephone Network) has dominated the world of phone communications (Varshney et al., 2002). The alternative solution of VoIP (Voice over Internet Protocol) or Internet telephony has increased dramatically its share over the years though. Originally started among computer enthusiasts, nowadays it has become a huge research area in both the academic community as well as the industry (Karapantazis and Pavlidou, 2009). Therefore, many VoIP technologies have emerged in order to offer telephony services. However, the performance of these VoIP technologies is a key issue for the sound quality that the end-users receive. When making reference to sound quality PSTN still stands as the benchmark.Against this background, the aim of this project is to evaluate different VoIP signalling protocols in terms of their key performance metrics and the impact of security and packet transport mechanisms on them. In order to reach this aim in-band and out-of-band VoIP signalling protocols are reviewed along with the existing security techniques which protect phone calls and network protocols that relay voice over packet-switched systems. In addition, the various methods and tools that are used in order to carry out performance measurements are examined together with the open source Asterisk VoIP platform. The findings of the literature review are then used in order to design and implement a novel experimental framework which is employed for the evaluation of the in-band and out-of-band VoIP signalling protocols in respect to their key performance networks. The major issue of this framework though is the lack of fine-grained clock synchronisation which is required in order to achieve ultra precise measurements. However, valid results are still extracted. These results show that in-band signalling protocols are highly optimised for VoIP telephony and outperform out-of-band signalling protocols in certain key areas. Furthermore, the use of VoIP specific security mechanisms introduces just a minor overhead whereas the use of Layer 2.5 protocols against the Layer 3 routing protocols does not improve the performance of the VoIP signalling protocols

Evaluation of Voip Technologies As a Replacement for Traditional Pstn Based Pbx Systems

This project deals with a company in the SME sector with offices located in the midlands of Ireland. The company is well established in the field of Agri-Feed Manufacturing Process Control Systems or SCADA systems, and has been established for over 20 years. The communications requirements of the company have changed over these 20 plus years to a mix of various technologies from PSTN lines to Broadband ADSL. The present telephone system has been in use since 1991 and has several questions marks over it in terms of usage costs, usage reporting, support and maintenance and features available. This project is an evaluation of the possible benefits offered by the use of VOIP technologies and Asterisk Open Source PBX as a possible replacement for the existing telephone system in place. It attempts to look at the potential benefits costs, and risks associated with using such a system. A small pilot system is implemented and some key users test this and feedback on its usability is recorded. The current communications infrastructure is analysed in an effort to highlight systems where cost savings or benefits can be made by switching to these other technologies and a report was presented to the management in order to give the required information to make the best possible informed decision about the way forward for the company

Securing media streams in an Asterisk-based environment and evaluating the resulting performance cost

When adding Confidentiality, Integrity and Availability (CIA) to a multi-user VoIP (Voice over IP) system, performance and quality are at risk. The aim of this study is twofold. Firstly, it describes current methods suitable to secure voice streams within a VoIP system and make them available in an Asterisk-based VoIP environment. (Asterisk is a well established, open-source, TDM/VoIP PBX.) Secondly, this study evaluates the performance cost incurred after implementing each security method within the Asterisk-based system, using a special testbed suite, named DRAPA, which was developed expressly for this study. The three security methods implemented and studied were IPSec (Internet Protocol Security), SRTP (Secure Real-time Transport Protocol), and SIAX2 (Secure Inter-Asterisk eXchange 2 protocol). From the experiments, it was found that bandwidth and CPU usage were significantly affected by the addition of CIA. In ranking the three security methods in terms of these two resources, it was found that SRTP incurs the least bandwidth overhead, followed by SIAX2 and then IPSec. Where CPU utilisation is concerned, it was found that SIAX2 incurs the least overhead, followed by IPSec, and then SRTP

Дослідження особливостей впровадження сервісів IP-телефонії в Інтернет ресурси

Актуальність роботи. Телекомунікаційні системи та мережи вдосконалюються кожного дня та набувають неабиякого значення для якісного функціонування тієї чи іншої галузі. Завдяки швидкому темпу зростання швидкості та якості безпроводових мереж з’явилась можливість ведення міжнародних та міжміських дзвінків за низькою вартістю та гарною якістю. Мета і завдання дослідження. Метою роботи є досліження використання IP-телефонії для інтернет ресурсів, аналіз якості та обгрунтування доцільності інтеграції цієї технології в різні сфери бізнесу. Для досягнення поставленої мети необхідно вирішити такі завдання: дослідити можливості та переваги використання IP-телефонії; проаналізувати функції найвідоміших представників IP-телефонії та порівняти їх між собою; дослідити можливості, інтеграції телефонії в уже існуючі веб ресурси та провести базові налаштування для їх роботи; провести підключення IP-телефонії на веб сторінку та запрограмувати її для роботи, дослідити якість проведених підключень, шляхом використання відомих методів для аналізу голосових повідомлень. Об’єкт дослідження – технологія IP-телефонії, додатки FreeSWITCH та Asterisk. Предмет дослідження –Web SIP клієнт на основі FreeSWITCH. Методи дослідження – критичний аналіз технології IP-телефонії в сферах надання телекомунікаційних послуг, порівняння переваг та недоліків лідерів IP-телефонії. Об’єктивний та суб’єктивний аналіз голосових повідомлень, та їх якості з відомими величинами якості. Застосування положень користувацького досвіду встановлення IP-телефонії та її елементів. Наукова новизна одержаних результатів. Удосконалено звязок між клієнтом, які знаходяться на сайті або веб-ресурсі та представниками бізнесу завдяки використанню пари стандарту WebRTC та протокол зв’язку WebSocket. Практичне значення одержаних результатів. Запропоновано використовувати розроблений Web SIP клієнт для Інтернет ресурсів, що дає змогу збільшувати показники зацікавленості клієнтів, продажів продукту та підтримку клієнтів онлайн. Запропоновано сценарії застосування технології IP-телефонії в межах компанії для здійснення внутрішніх та зовнішніх дзвінків, а також дзвінків за межі країни за низькою вартістю та високою якістю.With the development of information technology, the need for high-quality communication is growing, which can be used as a replacement for traditional analog data transmission methods. For online business, the main task is to communicate with potential customers, promote and believe that their product is worthy of attention. This connection must be secure, easy to set up and connect, provide high quality data transmission, cost-effective and scalable. Telecommunication systems and networks are improving every day and are of great importance for the quality functioning of a particular industry. Due to the rapid growth of speed and quality of wireless networks, it is possible to make international and long distance calls at low cost and good quality. IP telephony combines the benefits of using the Internet and traditional telephony. Using this technology, Internet resources can keep customers longer on their pages and offer them the best conditions for using their product or service, as well as using PBX functionality such as call parking, call forwarding, answering machines, voicemails, callbacks and more

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

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

Soluções open-source para os serviços de fax e VPN numa rede empresarial

Tese de Mestrado Integrado. Engenharia Electrotécnica e de Computadores. Universidade do Porto. Faculdade de Engenharia. 201

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods