VoIP technology is one of the important applications of the network. In addition to using traditional PSTN telephone, people can choose favorable VoIP telephone because of the Internet popularization and High speed Internet. Conference is one of the functions of the VoIP, but not every server supported it. Such as OpenSIPS server we use in NSYSU, didn¡¦t supported it. But FreeSWITCH Server has it. Therefore, for solving this problem, I combile two Servers. Then OpenSIPS can use the Conference function. In this paper, we discuss how to combile OpenSIPS and FreeSWITCH. Besides, I have designed a WEB page interface to simplify the procedure to use conference. By this interface, the users can use it to set the timer to start conference. When the timer is up, all of the participants will receive a conference call. After picking up the phone, the users can enter conference.

The topic of this work is verification of support WebRTC communication through selected Open Source PBX. This work examine demands for WebRTC communications and describes configuration of branch centers for this type of communication. In the theoretical part is reader acquainted with the term WebRTC and with protocols related to this kind of communications. The purpose of this part of the work is to bring the reader closer look to the principles of functioning to ensuring support for this kind of communications. This is also connected with Description of basic interfaces of WebRTC applications. Further the reader finds the configuration of the selected Open Source PBX so that they can make audio-video call between WebRTC clients. This section is divided into three subchapters, each of it deals with the same problems for one of the aforementioned PBX. At the end of each chapter where the PBX PBX is configured step-by-step, test calls are made. These calls are captured by the Wireshark packet analyzer and serve as a demonstration of the WebRTC configuration functionality. At the end of this section, PBXs are compared against each other about WebRTC support. Practical part is dealing with laboratory task for students which are studying subject telecommunication and information systems. In the task students will be configuring WebRTC for PBX Asterisk. The task contains brief description of WebRTC and comments for all steps for configuration. All steps and facts are demonstrated by exemplary configuration files.

This work describes a development of system for network compilation for speech recognition based on Speech Recognition Grammar Specification (SRGS) grammar defined by W3C consortium. Together with the new module, the recognizer was integrated to the FreeSwitch software phone switch using a combination of MRCPv2/SIP/RTP networks protokols and tested.

En el laboratorio de procesamiento y transmisión de voz de la Facultad de Ciencias Físicas y Matemáticas de la Universidad de Chile se desarrollan motores de procesamiento de voz, con los cuales se puede otorgar una amplia gama de servicios relacionados al habla. Para poder brindar estos servicios surge la necesidad de poseer una plataforma de telefonía en la cual se puedan desarrollar aplicaciones que hagan uso de motores de voz y que permita la conectividad con usuarios mediante redes de telefonía pública (PSTN) y redes IP. Además se necesita que las aplicaciones cumplan protocolos estándares para que de esta forma sean compatibles internacionalmente. Como solución se propone la utilización de FreeSWITCH, que corresponde a una plataforma de telefonía de código abierto y en permanente desarrollo. Mediante FreeSWITCH se desarrolla una central telefónica sobre IP brindando conectividad a cada miembro del laboratorio. A su vez, la arquitectura del sistema implementado permite la conexión bidireccional a la PSTN. FreeSWITCH posee un módulo especial para brindar compatibilización con motores de voz, mediante una implementación parcial del protocolo MRCP (Media Resource Control Protocol), que se traduce en que por el momento brinda soporte a motores de reconocimiento automático de voz y motores de texto a voz. Gracias a esto, se tiene un sistema que permite brindar e implementar todo tipo de servicios telefónicos y de procesamiento de voz bajo normas estándares. Las pruebas de calidad de la voz en una llamada entre teléfonos IP arrojan resultados de percepción comparables a los obtenidos en conexiones realizadas en la PSTN. Por otro lado las pruebas realizadas para analizar el comportamiento del motor de reconocimiento bajo las órdenes de un servidor MRCP arrojan un WER de 1,48% superior al que arroja un motor de reconocimiento sin el servidor MRCP pero considerando que el motor de reconocimiento empleado en esta prueba no fue calibrado para su funcionamiento óptimo. Finalmente las pruebas de usabilidad de la aplicación de central telefónica con reconocimiento de voz muestran que la central presta un servicio apto y satisfactorio para los usuarios pero aún así no se demuestra que los usuarios no se incomodan al hablar con una máquina ni que los usuarios prefieren un servicio de reconocimiento de voz ante un servicio de menú con navegación con las teclas del teléfono. Como trabajo futuro se propone extender la funcionalidad del módulo MRCP con el que cuenta FreeSWITCH para poder utilizar las funcionalidades del módulo de verificación de locutor desarrollado en el laboratorio.

This diploma thesis focuses on attacks on PBX Asterisk, FreeSWITCH and Yate in LTS versions. In this work was carried out two types of attacks, including an attack DoS and the attack Teardown. These attacks were carried out using two different protocols, SIP and IAX. During the denial of service attack was monitored CPU usage and detected if its possible to establish call and whether if call can be processed. The Security of PBX was build on two levels. As a first level of security there was used linux based firewall netfilter. The second level of security was ensured with protocols TLS and SRTP.

The aim of this thesis is to install and to configure three Open source PBXes Asterisk, Freeswitch and YATE. Furthermore, the aim is to realize the performance test and stability tests on three different HW configurations with the tester Spirent Abacus 5000. The scripts in bash were created to monitor PBX performance. Another part of the study is to analyze and to compare PBX security and to compare the Open Source PBX with a proprietary PBX Alcatel-Lucent OXE.

This diploma thesis examines and compares several selected libraries of SIP protocol, performance, stability, security and impact of their configuration. The main functions of the signalling protocol are briefly named at the beginning. The following chapters describe the tested PBXs and several stacks for SIP protocol are theoretically compared. The practical part deals with measurements conducted on the load generator Spirent TestCenter C1 which is used for all the performed tests on exchanges. All the mentioned SIP libraries, PBXs and the operating system on which the PBXs were running are open source software.

This work explores, analyzes and rate infuence of VoIP attacks on open source pbx functionality. It describes how voice over IP attacks are achieved according to security standards. There are described concepts and basics of VoIP networks with orientation on facts necessary to understand analyzed actions and measurements in theoretical part. In practical part, there is described realization of attack tests according to instructions with first orientation on initial checking of testing device, pbx’s and security attack possibilities and then complex creation and testing of attack scenarios types such as fuzzing and denial of service attacks.

This master project focuses on the possibilities of protecting the most common free software PBX Asterisk, FreeSWITCH and YATE. In practice, it was verified the behavior of PBX in the attacks and suggested protection against them on one of the most popular distributions of Linux server on CentOS. Tool was created to simulate several types of attacks targeting denial of service. Both protective options PBX themselves and operating system capabilities are used in this work. Comparison was also the possibility of protection of individual PBX with each other. It also includes a brief description of the protocol, topology attacks and recommendation for the operation of softswitches.