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

ACUTA Journal of Telecommunications in Higher Education

In This Issue President\u27s Message From the ACUTA CEO RIP for TDM IPTV: The Future of Gable TV Not All SIP Trunking ls Problem Free lnterview: Four Campuses Look at lPv6, SIB and More lPv6: What You Don\u27t Know CAN Hurt You Moving from the Old to the New 2013 Award Winners lnstitutional Excellence Award Honorable Mention: Abilene Christian University Virtual La

A Tool for VoIP Audio Extraction

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.

Novel Architecture for Routing Packetized Voice Over Existing Internet Infrastructure Without Using the Internet Protocol

Summary This article addresses a novel method to route Voice over the internet without using the Internet Protocol (IP). It talks about the problems of the current methodology to transfer Voice or Video traffic over IP (VoIP), and proposes a solution to these problems. This proposed solution consists of a new protocol that routes Voice and Video traffic over the Internet infrastructure. This novel protocol runs in parallel and independently from the Internet data routing protocols. Then it gives more detailed design description of this new protocol, and discusses different implementation methods. This is the first time a document on this technology has been written. Key words: Voice over IP, Internet Protocol, Routing, Bandwidth. Current Technology Voice over the Internet Protocol (VoIP) is a standards based technology and a widely used method to transfer and route voice, video and multimedia traffic over the Internet. There are many standard bodies, like the Internet Engineering task Force (IETF) and the International Telecommunication Union (ITU), with standards that address VoIP. A similar technology defined by the IETF standard RFC326

Sinalização de media gateways em redes de próxima geração

Mestrado em Engenharia Electrónica e TelecomunicaçõesCom o grande crescimento das comunicações móveis e fixas, o acesso à Internet tornou-se cada vez mais numa exigência, colocando à industria das Telecomunicações, especialmente aos operadores, grandes desafios. Serviços comuns como chamadas de voz, podem agora ser oferecidos pelos Internet Service Providers (ISPs) aos seus clientes sobre a forma de serviço Voice over IP (VoIP). Este serviço deixou de ser exclusivo das redes Public Switched Telephone Network/Integrated Services Digital Network (PSTN/ISDN) e passou a ser fornecido também na Internet. Mas devido à necessidade de manter as tradicionais redes PSTN/ISDN, houve a necessidade de criar um ambiente de convergência, não só para estas redes mas também para outros tipos de redes de acesso, independentemente da tecnologia. É neste campo que os organismos de normalização e os operadores têm dado os seus contributos, criando uma rede de controlo e de transporte comum baseada em IP para a convergência de serviços. Inicialmente o 3rd Generation Partnership Project (3GPP) definiu uma arquitectura de convergência móvel com a rede IP, constituída por elementos de controlo, transporte e serviço, de nome IP Multimedia Subsystem (IMS). Mais tarde, esta arquitectura serviu de base (core) para o grupo TISPAN do European Telecommunications Standard Institute (ETSI) na normalização das Redes de Próxima Geração. Esta Dissertação pretende dar uma resposta à convergência fixo-móvel no âmbito da arquitectura PSTN/ISDN Emulation Subsystem (PES) do TISPAN. Este sistema permite que todos os clientes de uma Rede de Próxima Geração de um operador acedam a serviços das redes PSTN/ISDN e Digital Subscriber Line (DSL) de uma forma simples e imperceptível. Com este intuito foram desenvolvidos cenários de testes para os sistemas Trunking e de Acesso da arquitectura PES, tendo como objectivo final a sua integração na plataforma de próxima geração Service Handling on ip NETworks (SHipNET). Esta Dissertação experimenta várias situações reais de chamadas de voz sobre os cenários de testes, e inicia a implementação de um novo elemento definido para a arquitectura PES, Access Gateway Control Function (AGCF), para o controlo de Media Gateways nas redes de Acesso. ABSTRACT: With the big growth of mobile and fixed communications, Internet access has become a requirement, putting the telecommunication industry, and especially the operators, in front of a major challenge. Services such as voice calls can now be offered by Internet Service Providers (ISPs) to their customers. This service is no longer exclusive of Public Switched Telephone Network/Integrated Services Digital Network (PSTN/ISDN) and is now provided also through the Internet. But, because of the need to maintain the traditional PSTN/ISDN networks, there was a need to create a convergence, not only for these networks but also for other types of access networks, regardless of technology. The standards bodies and operators have made their contributions to create a network of control and transport policy, based on IP, for the services convergence. In the beginning the 3rd Generation Partnership Project (3GPP) defined an architecture for mobile convergence with IP network, made up of control, transport and service elements, called IP Multimedia Subsystem (IMS). Later, the core IMS served the ETSI TISPAN group in standardization of Next Generation Networks. This thesis aims to give an answer for fixed-mobile convergence within the architecture defined by TISPAN PSTN/ISDN Emulation Subsystem (PES). This system, formed by a Trunking, originally defined by the 3GPP IMS, and Access part, allows all customers of a Next Generation Network operator, access to PSTN/ISDN and Digital Subscriber Line (DSL) network services in a simple way. With this purpose, scenarios were developed for Trunking and Access systems of PES arquitecture, with the goal to integrate into the next generation platform Service Handling on ip NETworks (SHipNET). This thesis tests several real situations of voice calls on testing scenarios, and begins the implementation of a new element defined for PES arquitecture, Access Gateway Control Function (AGCF), for Media Gateways control purpose in access networks

Broadband services virtual operator for bitstream open-access networks: business case and infrastructure

The aim of this thesis is to study the technical and business model for the creation of a virtual operator for bitstream access networks in Catalonia, in the modality of carrier’s carrier. The substrate for this project is Xarxa Oberta network. Xarxa Oberta is a project which is under way at present and that will ultimately provide a backbone network connecting all municipalities in Catalonia. Open access networks are being deployed across Spain to provide access to broadband services where this was previously either not possible or not competitive. Additionally, multiple access networks are already deployed which currently subcontract interconnection and advanced services from third parties. Xarxa Oberta offers the opportunity to create a bitstream virtual operator that offers its services to local service providers in several access networks across Catalonia. The technical model uses both Fiber to the Home (FTTH) and Hybrid Fibre-Coax (HFC) technologies to develop a reference model for access network deployments, including both existing and open access network operators as potential customers. A complete open source stack is proposed to deploy Network Management System (NMS) and Operation and Business Support Systems (OSS/BSS). The infrastructure is defined, including equipment selection and deployment. A business plan is detailed to analyse the viability of the project. The key activities, resources, channels, costs, revenues, etc are presented in detail. This business plan serves as a basis for the business model, which studies the economic viability of the operator. Various scenarios are studied, each with different values for the main parameters (such as network size, number of networks, etc.). The results of these models give a structured view of the viability of the business for this virtual operator