Användandet av IP-telefoni har de senaste åren ökat och marknaden förutspås fortsätta att växa globalt. Många vill ta del av de fördelar som den nya tekniken har i form av ny funktionalitet och minskade kostnader. Men att migrera telefoni till datanät medför både nya möjligheter men också nya utmaningar. I den här rapporten undersöks hur Asterisk kan användas som ett bra IP-PBX alternativ. Rapporten behandlar även säkerheten i att ansluta ett system med Asterisk till internet genom intrusionstester som utförs på systemet i grundkonfiguration. Dessa tester resulterar i rekommendationer om hur systemet kan konfigureras för att hålla en hög säkerhetsnivå. / The usage of IP-telephony has increased in recent past and the market is expected to continue to grow globally. Many want to take part in the advantages that the new technology brings in form of functionality and reduced costs. But to migrate telephony to data networks brings both new possibilities but also new challenges. This report examines how Asterisk can be used as a good IP-PBX alternative. The report also addresses the security aspect of connecting a system based on Asterisk to the internet through conducting intrusion tests on the system in standard configuration. These tests result in recommendations on how the system can be configured to keep a high security standard.

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<p>In the thesis the advantages with Text over IP (ToIP) is explained and it is motivated why it is a good idea to integrate this in Asterisk. It also presents an implementation of a ToIP extension in Asterisk.</p><p>ToIP means communicating over a network based on Internet protocols with real-time text. Real-time text means a character is sent to the receiving terminal as soon the sender has typed it or with a small delay.</p><p>In the thesis IM and ToIP is compared in a survey. The result point at IM is not better than ToIP even though it is much more commonly used. VoIP can not replace ToIP either because there are occasions when ToIP is better for instance if the person using it is deaf or if a person want to make a private conversation in a noisy room.</p><p>Asterisk is an IP-PBX. PBX stands for Private Branch Exchange which means a private telephone system which is part of a larger network system that exchange information.</p><p>An IP-PBX is a PBX based on the Internet. Asterisk and many other IP-PBX can also exchange calls between the PSTN ant the Internet. By including ToIP in Asterisk it will be possible to exchange ToIP calls.</p><p>The implementation described is not only including ToIP in Asterisk but also a translation function between the text format called t140 and another text format called t140 with redundancy.</p><p>The idea is to extend the translation function in the future to more text formats.</p>

Workshop "Netz- und Service-Infrastrukturen" Das Inter-Asterisk eXchange (IAX)-Protokoll ermöglicht eine unproblematische Kommunikation zwischen IAX-fähigen VoIP-Systemen. In der Präsentation zu dem Vortrag werden das Protokoll vorgestellt und die Vorteile von IAX skizziert.

The diploma thesis focuses on the VoIP technology optimization and performance management. The diploma thesis presents the theoretical basis of IP telephony and measurement of its quality. The thesis primarily deals with practical measurements of VoIP calls quality. Asterisk softswitch, various types of IP phones and simulated degradation of signal using Linux software router are used for measurements. Procedural diagram of VoIP technology real deployment is designed based on these measurements.

Submitted by Programa de P?s-Gradua??o Engenharia El?trica (ppgee@ifpb.edu.br) on 2018-04-17T17:04:37Z No. of bitstreams: 1 46- Anselmo de Vasconcelos Cavalcante - AN?LISE DOS EFEITOS DE CODECS DE ?UDIO NA AVALIA??O DE DESVIOS VOCAIS.pdf: 2426098 bytes, checksum: 4e9dc4ae38d3b0d367a6d38e535f9a00 (MD5) / Approved for entry into archive by Programa de P?s-Gradua??o Engenharia El?trica (ppgee@ifpb.edu.br) on 2018-04-17T17:05:40Z (GMT) No. of bitstreams: 1 46- Anselmo de Vasconcelos Cavalcante - AN?LISE DOS EFEITOS DE CODECS DE ?UDIO NA AVALIA??O DE DESVIOS VOCAIS.pdf: 2426098 bytes, checksum: 4e9dc4ae38d3b0d367a6d38e535f9a00 (MD5) / Made available in DSpace on 2018-04-17T17:05:40Z (GMT). No. of bitstreams: 1 46- Anselmo de Vasconcelos Cavalcante - AN?LISE DOS EFEITOS DE CODECS DE ?UDIO NA AVALIA??O DE DESVIOS VOCAIS.pdf: 2426098 bytes, checksum: 4e9dc4ae38d3b0d367a6d38e535f9a00 (MD5) Previous issue date: 2018-03-19 / Este trabalho apresenta um estudo sobre as implica??es no uso de diferentes codecs de ?udio na an?lise perceptiva e ac?stica da voz. Um cen?rio de transmiss?o baseado em VoIP foi criado, empregando o Asterisk e o softphone Microsip, para auxiliar no diagn?stico de desvios vocais ? dist?ncia. Para este prop?sito, foram utilizados 36 sinais de vozes sintetizadas e 36 sinais de vozes reais, classificados como normais, com o desvio rugosidade e com o desvio soprosidade. Cada sinal foi submetido a seis transmiss?es, cada uma delas utilizando um codec espec?fico (G.711 Lei A, Speex32, GSM Full Rate, LPCM16, Opus24 e SILK16). Antes e ap?s cada transmiss?o, um especialista em voz realizou a classifica??o dos sinais quanto ao tipo de desvio e, com aux?lio do software VoxMetria, extraiu-se as medidas ac?sticas frequ?ncia fundamental, jitter, shimmer, GNE e desvio padr?o da frequ?ncia fundamental. Observou-se que, dentre os codecs analisados, o Opus24 foi aquele se mostrou o mais promissor para avalia??o da qualidade vocal, por ter sido o ?nico a apresentar taxa de acur?cia acima de 70%, tanto para os sinais sintetizados quanto para os sinais reais analisados. Este codec conseguiu manter os desvios vocais inalterados para avalia??o em mais de 85% dos sinais reais.

This diploma thesis focuses on development of graphical user interface for managing IVR applications. The work is more software oriented, it analyzes current state of the iPBX product belonging to the IPEX a.s. company, describes used technologies and introduces new concept of interactive user interface for generating IVR diagrams together with Asterisk dial plan generator.

This master's thesis is focused on developement of billing system with the options of monitoring individual calls for software exchange Asterisk. Billing of calls is adaptible with the help of group of individual rules, consisting of tariff impulses, numerical prefix, with help of outgoing trunk and cost of the billed unit. The first part of this work is focused on instalation, configuration and preparation of individual components of the billing system. In this work is explained the architecture of the billing system and highlighted the purpose of work of the model database. Next we focused on the purpose and the principal system invidual function of the system including solution. At last there is a simple manual to operate the system with the help of created web interface.

This thesis describes architecture of IMS and shows possibilities of IMS platform testing. Theoretical part describes layer model of the IMS as a whole and then describes it’s individual layers. Next chapters analyse key entities of the IMS, interconnection between reference points and features of protocols used in the IMS. Practical part deals with the introduction of Open IMS Core, which was chosen for the IMS technology testing. Settings necessary to carry out testing and interconnection between PBX Asterisk are shown in next chapters. After introduction of IMS desktop clients is carried out an instant messaging communication within the IMS network. The communication is captured and analysed by Wireshark application. Afterwards there is described how SIP protocol sends messages within the IMS. After a brief introduction to the PBX Asterisk, there are discussed assumptions for the interconnection between Asterisk and IMS. There are also described necessary settings needed for implementation and communication testing itself. The first test is an audio session carried out between the desktop IMS client and IP phone registered to the PBX Asterisk. Communication is captured for the analysis of preparation, conduction and termination of the session. After the successful realization of the audio call, video session has been made. The session was analysed in detail, including statistics of control signals and transmitted packets. There are two laboratory excercises in attachement of this thesis, which will help students to understand the IMS technology and communication options within the IMS network

This thesis project is on helping a Voice over Internet Protocol (VoIP) service provider by improving server side of Opticall AB's Dial over Data solution. Nowadays, VoIP is becoming more and more popular. People use VoIP to call their family and friends every day. It is cheap, especially when users are abroad, because that they do need to pay any roaming fee. Many companies also like their employees to use VoIP, not only because the cost of calling is cheap, but using VoIP means that the company does not need a hardware Private Branch eXchange (PBX) -- while potentially offering all of the same types of services that such a PBX would have offered. As a result the company can replace their hardware PBX with a powerful PC which has Private Branch eXchange PBX software to connect all the employees and their VoIP provider. At the VoIP provider’s side, the provider can provide cheap calls for all users which are connected by Internet. The users can initialize and tear down a session using a VoIP user agent, but how can they place a VoIP call from a mobile device or other devices without a VoIP user agent? Users want to place cheap VoIP call everywhere. VoIP providers want to provide flexible solution to attract and keep users. So they both want to the users to be able to place cheap VoIP call everywhere. Although VoIP user agent are available for many devices as a software running on a computer, a hardware VoIP phone, and even in some mobile devices. However, there are some practical problems with placing a VoIP call from everywhere. The first problem is that not every device can have a VoIP user agent. But if you do not have a VoIP user agent on your device, then it would seem to be difficult to place a VoIP call. The second problem is that you have to connect to a network (probably Internet) to signal that you want to place a call. Thus at a minimum your device has to support connecting to an appropriate network. If your device is connecting to a mobile network, you can send signaling to set up a VoIP call through General Packet Radio Service (GPRS). However, the bandwidth and delay of the GPRS networks of some mobile operators is not suitable for the transfer of encoded voice data, additionally, some mobile operators charge high fees for using GPRS. All of these problems make placing VoIP calls via a mobile device difficult. However, if your mobile device has a VoIP user agent and you have suitable connectivity, then you can easily use VoIP from your mobile device[.] To provide a flexible solution to VoIP everywhere -- even to devices that do not or can not have a VoIP user agent, Opticall AB has designed Dial over Data (DoD) solution. By using this solution, you can place a VoIP call from your mobile device or even fixed phone -- without requiring that the device that you use have a VoIP user agent. This solution also provides a central Internet Protocol-Private Branch eXchange (IP-PBX) which can connect call to and from to Session Initiation Protocol (SIP) phones. Both individuals and companies can use this solution for call cost savings. Max Weltz created the existing DoD solution in an earlier thesis project. This thesis [1] provides a good description of the existing DoD solution. As a result of continued testing and user feedback, Opticall AB has realized that their DoD solution needs to be improved in several area. This thesis project first identified some of the places where improvement was needed, explains why these improvements are necessary, and finally designs, implements, and evaluates these changes to confirm that they are improvements. An important result of this thesis project was a clear demonstration of improvements in configuration of the solution, better presentation of call data records, correct presentation of caller ID, and the ability to use a number of different graphical user interfaces with the improve DoD solution. These improvements should make this solution more attractive to the persons who have to maintain and operate the solution. / Detta examensarbete behandlar förbättringar i serversidan av OptiCall ABs “Dial over Data” (DoD) lösning som tillhandahålls för tjänsteleverantörer av VoIP. VoIP blir mer och mer populärt. Människor använder VoIP för att ringa till sin familj och vänner varje dag. Det är billigt, särskilt när användaren är utomlands, eftersom de inte behöver betala någon roamingavgift. Många företag vill också att deras anställda skall använda IP-telefoni, inte bara därför att kostnaden för att ringa oftast är lägre, utan för att bolaget kan ersätta sin traditionella företagsväxel (PBX) med en kraftfull dator som har PBX programvara för att även ansluta alla anställda till deras VoIP leverantör. VoIP leverantören kan erbjuda billiga samtal till alla användare som också är anslutna via Internet. Användarna kan hantera VoIP samtal med en VoIP user agent, men hur kan de ringa ett VoIP-samtal från en mobil enhet eller andra enheter utan VoIP user agent? Användare vill kunna ringa billiga VoIP-samtal överallt. VoIP-leverantörer vill erbjuda en flexibel lösning för att locka och behålla användare. Även VoIP user agent finns utvecklade till många enheter som en programvara som körs på en dator, en hårdvara VoIP-telefon, och även i vissa mobila enheter. Men det finns vissa praktiska problem med att ringa ett VoIP-samtal från alla platser. Det första problemet är att inte varje enhet kan ha en VoIP user agent. Det andra problemet är att den måste ansluta till ett nätverk (troligen Internet) för att signalera att den vill ringa ett samtal. Om din enhet ansluter till ett mobilnät, kan du skicka signalerar att upprätta ett VoIP-samtal via General Packet Radio Service (GPRS). Dock är bandbredden och fördröjningen i GPRS-nät i vissa operatörers nät inte lämpliga för överföring av tal, dessutom tar vissa mobiloperatörer ut höga avgifter för att använda GPRS. Alla dessa problem gör det svårt att hantera VoIP-samtal via en mobil enhet. Men om din mobila enhet har en VoIP user agent och du har lämplig nätanslutning så kan du enkelt använda VoIP från din mobiltelefon[.] För att erbjuda en flexibel VoIP lösning överallt - även på enheter som inte kan ha en VoIP user agent har OptiCall AB utformad “Dial over Data” (DoD). Genom att använda denna lösning kan du initiera ett VoIP-samtal från din mobiltelefon eller fast telefon - utan att kräva att den enhet som du använder har en VoIP user agent. Denna lösning inkluderar också en central Internet Protocol-Private Branch Exchange (IP-PBX) som kan koppla samtal till och från Session Initiation Protocol (SIP) telefoner. Både privatpersoner och företag kan använda denna lösning för att minska samtalskostnader. Max Weltz vidareutvecklade den befintliga DoD lösning i ett tidigare examensarbete. Denna avhandling [1] ger en god beskrivning av den befintliga DoD lösning. Som ett resultat av fortsatt testning samt synpunkter från användarna har OptiCall AB insett att deras DoD lösning måste förbättras på flera områden. Detta examensarbete har i första hand identifierat några områden där förbättringar behövdes, förklarat varför dessa förbättringar är nödvändiga, och slutligen utvecklat och utvärderat dessa förändringar. Ett viktigt resultat av detta examensarbete visades av en tydlig demonstration av förbättrad utformning av lösningen. Gränssnittet fick bla en bättre presentation av samtalshistorik, mer korrekt nummerpresentation. Dessa förbättringar bör göra denna lösning mer attraktivt för de personer som skall använda och underhålla lösningen.