The possibilities of SS7 signalling transport over IP network using YATE switch

Diplomová Práce se věnuje problematikou signalizačního systému číslo 7 (SS7), a to zejména přenosem signalizace SS7 přes sítě na bázi IP protokolu. K ověření možností přenosu byla vytvořena konvergovaná síť s využitím Open Source PBX YATE, kde jsou potřebné protokoly implementovány. V úvodu diplomové práce je uveden popis signalizačního systému SS7, který je následován vysvětlením funkce každé z vrstev (MTP2 až aplikační) a přenášených zpráv síti SS7. Dále byla věnována pozornost protokolům, které umožňují přenos SS7 přes IP síť. V diplomové práci byla také popsána architektura PBX YATE, konfigurační soubory a způsoby instalace v operačním systému Linux. Rovněž byly stručně popsány důležité soubory pro realizaci této práce. Experimentální práce byly zahájeny s využitím dvojice virtuálních počítačů, které měly naistalovány dvě různé PBX, a to YATE a Asterisk. Dalším krokem byla realizace konvergované sítě pro ověření teoretických předpokladů. Pro tyto účely již byly využity servery s instalovanými TDM kartami. S pomocí těchto serverů byly ověřeny protokoly SS7, SIGTRAN, implementována MGCP brána a protokol SIP-T. Experimenty byly úspěšné, nicméně lze jistě pokračovat dalšími a ověřit další možnosti.This study examines the use of SS7 signaling system over IP networks by using the open source PBX YATE. At first it starts with describing the SS7 followed by an explanation of the function of each of its levels and the messages that are used within the SS7 network. The study then sheds some light on the ways of using SS7 inside IP network with the use of some protocols. It also discusses the architecture of YATE and its files, and how it is installed in Linux operating system. Finally, it describes the important files for delivering this task. The study was commenced by using two virtual machines that have two different open source PBX's which are YATE and Asterisk, and after acquiring some results by establishing communication between them via the means of SIP trunk, furthermore the study was extended to the laboratory in order to test it over real servers that have TDM cards, in order to apply the study by the means of SS7 protocols, SIGTRAN, MGCP gateway and SIP-T. The experiments have almost delivered successful communications after conducting a configuration for the files on multiple sides.

Investigating call control using MGCP in conjuction with SIP and H.323

Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323

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

Major: Electronics and Communication Engineering

Today, information technology is strategically important to the goals and aspirations of the business enterprises, government and high-level education institutions – university. Universities are facing new challenges with the emerging global economy characterized by the importance of providing faster communication services and improving the productivity and effectiveness of individuals. New challenges such as provides an information network that supports the demands and diversification of university issues. A new network architecture, which is a set of design principles for build a network, is one of the pillar bases. It is the cornerstone that enables the university’s faculty, researchers, students, administrators, and staff to discover, learn, reach out, and serve society. This thesis focuses on the network architecture definitions and fundamental components. Three most important characteristics of high-quality architecture are that: it’s open network architecture; it’s service-oriented characteristics and is an IP network based on packets. There are four important components in the architecture, which are: Services and Network Management, Network Control, Core Switching and Edge Access. The theoretical contribution of this study is a reference model Architecture of University Campus Network that can be followed or adapted to build a robust yet flexible network that respond next generation requirements. The results found are relevant to provide an important complete reference guide to the process of building campus network which nowadays play a very important role. Respectively, the research gives university networks a structured modular model that is reliable, robust and can easily grow

Analysis and testing of voip-subsystems of Ip Brick

Technology in which communication done using IP(Internet Protocol) alternative for the traditional analog systems is voip (voice over internet protocol) .One of the emerging or attractive communication systems in this era is voip. Several technologies within the voip are emerging and more come to near future, services offered by this technology needs internet connections and/or telephone connections. It offers services such as making audio and video calls. VoIP is a medium that converts the analog signal to digital signals[1]. In this dissertation mainly focused on ipbrick voip-subsystems such as webrtc ,asterisk13.8 pbx server and kamilio sip proxy. These are free and open-source technologies available in the marketplace. Integration of these technologies provides real-time communication between the systems such as voice,video and text. Webrtc enables web browsers to have native support for real-time voice, video and data capabilities. As such, end-users do not require additional add-ons or plug-ins to utilize real-time voice and video communication. To allow webrtc to make calls to non-webrtc voip applications, A initiation protocol is needed and one such protocol is session initiation protocol (SIP),which is the standard protocol used for initializing, changing and terminating sessions for multimedia today. It is particularly known for its use in voip applications. Asterisk13.8pbx supports webrtc and it acts as media gateway. In this thesis, we are evaluating and comparing previous and current versions of the ipbrick voip-subsystem which is previously having lab environment of ipbrick OS v6.1 with Asterisk v1.8 as a pbx server and webrtc application such as webrtc2sip and SIP Proxy software called Kamailio which connects two endpoints. In this thesis we are testing ipbrick voip-subsystem with current version of ipbrick OS v6.2 with Asterisk v13.8 as a new pbx server and Webrtc application (SIPML5) on the browser side and a phone on the server side (softphone) establish a phone call between them. The SIP Proxy server Kamailio will act as a intermediary connection, connecting the two endpoints using websockets

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

Decorating Asterisk : experiments in service creation for a multi-protocol telephony environment using open source tools

As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise

Mitigating Denial-of-Service Attacks on VoIP Environment

IP telephony refers to the use of Internet protocols to provide voice, video, and data in one integrated service over LANs, BNs, MANs, not WANs. VoIP provides three key benefits compared to traditional voice telephone services. First, it minimizes the need fro extra wiring in new buildings. Second, it provides easy movement of telephones and the ability of phone numbers to move with the individual. Finally, VoIP is generally cheaper to operate because it requires less network capacity to transmit the same voice telephone call over an increasingly digital telephone network (FitzGerald & Dennis, 2007 p. 519). Unfortunately, benefits of new electronic communications come with proportionate risks. Companies experience losses resulting from attacks on data networks. There are direct losses like economic theft, theft of trade secrets and digital data, as well as indirect losses that include loss of sales, loss of competitive advantage etc. The companies need to develop their security policies to protect their businesses. But the practice of information security has become more complex than ever. The research paper will be about the major DoS threats the company’s VoIP environment can experience as well as best countermeasures that can be used to prevent them and make the VoIP environment and, therefore, company’s networking environment more secure