“Análisis comparativo del desempeño de los diferentes CÓDECS utilizados para video conferencia en la empresa ECUAGREENPRODEX S.A. de la ciudad de Guayaquil”

La presente investigación, ha medido la capacidad de los códec de mayor relevancia del mercado como son: H263, H263 HD, H263 P, H264 y MPEG, los que evolucionaron a un nivel de fidelidad que funcionan adecuadamente en ambientes web como en plataformas de escritorio reduciendo las pérdidas de datos y la velocidad de transmisión (ruido) que eventualmente se ponen de manifiesto en las video llamadas. Por otro lado, la plataforma XENSERVER permitió la virtualización del servidor Ubuntu Server,se evaluó también el desempeño de los softphone Ekiga, Zoiper y Linphone los que funcionaron sin perjuicio del ambiente en el que se realizó la video llamada. El VP8 tiene incompatibilidad de plataforma por una forma de licenciamiento y patente que aún no se ha liberado para todos, por lo que no se incluyeron pruebas con este códec que, a la larga también podría incrementar los costos operativos para la empresa. El códec H264 se ha convertido en uno de los más utilizados no solo por su rendimiento sino por su portabilidad permitiéndole a la organización implementarlo en sus video llamadas para los nodos de VPN de 1 MB y 2 MB que tiene la empresa actualmente. Finalmente, se demostró que los códec utilizados para video conferencia tienen mejor rendimiento por los cifrados que utilizan, sin perjuicio de las plataformas en las que se los ejecuta. En cuanto al rendimiento en paquetes transmitidos, el H264 se mantiene sobre el 1.1 Mbps, lo que potencia su rendimiento en la transmisión de datos, sin perjudicar la calidad y velocidad y sin importar la plataforma en la que está ejecutándose. La experiencia de implementar un prototipo de video conferencia para la empresa ECUAGREENPRODEX S.A. me ha permitido evidenciar claramente el funcionamiento y adecuado manejo y mantenimiento de dichos sistemas que le permitan a la empresa mejorar sus líneas de comunicación sin incrementar sus costos operativos debido a que los sistemas utilizados y recomendados para la empresa en su mayoría open source

Core Network Design of Software Defined Radio Testbed

The 4th generation of cellular system (LTE) does not inherit the traditional voice (circuit-switched) capabilities from its predecessors. Instead it relies on its high speed packet-switched core network with IMS (IP Multimedia Subsystem) for voice capabilities. Even though there are temporary solutions available until LTE gets its full deployment and coverage, operators are looking for a long term solution known as VoIMS which uses VoIP with SIP protocol for voice in the LTE network (VoLTE) through the IMS domain. The scope of this thesis work is to design, implement and verify the working of the core network for an LTE type software defined radio (SDR) testbed which is able to initiate, maintain and terminate voice and data connections. First step in this regard is to search and select the tools, programs and technologies that fulfil the network requirement in terms of network performance and user satisfaction. Next is to build, configure and verify the network operations of the designed network. As SDRs are used for testing purposes, the core network is also designed in correspondence to that, i.e., it is a test (lab) core network with configurations that are simple to implement and do not require coding implementation. The core network makes use of the virtualization technology and is realized with the help of open-source solutions, i.e., protocols and technologies that are customizable as required and does not require licensing for their use. These functionalities are implemented with the help of OpenSIPS, an open-source SIP server, DHCP and DNS servers. Demonstration of the core network verifies that successful voice and video call can be made between registered users on two different networks, running VoIP client software on different operating system platforms. The core network provides features such as voice, video, instant messaging, presence, dynamic IP assignment, IP address to name resolution and mobility

Ubiquitous Mesh Networking: application to mobile communication and information dissemination in a rural context

ICT has furthered the social and economic development of societies but, rural African communities have lagged behind due to issues such as sparse population, low household income, a lack of electricity and other basic infrastructure that make it unattractive for telecommunication service providers to extend service provision. Where the service is available, ubiquitous service coverage has not translated into ubiquitous access for individuals because of the associated costs. A community-wide WMN offering VoIP using fixed telephone handsets has been deployed as a viable alternative to the cellular service provider. The effectiveness of this WMN VoIP service springs from the mobile phone usage statistics which showed that the majority of calls made are intra-community. This dissertation has been an effort towards improved communication and access to information for the under-served communities. Key contributions include, mobile VoIP support, translation gateway deployment to make textual information accessible in voice form via the phone, IP-based radio for community information dissemination. The lack of electricity has been mitigated by the use of low-power devices. In order to circumvent the computational challenges posed by the processing and storage limitations of these devices, a decentralised system architecture whereby the processing and storage load are distributed across the mesh nodes has been proposed. High-performance equipment can be stationed at the closest possible place with electricity in the area and connectivity extended to the non-electrified areas using low-power mesh networking devices. Implementation techniques were investigated and performance parameters measured. The quality of service experienced by the user was assessed using objective methods and QoS correlation models. A MOS value of 4.29, i.e. very good, was achieved for the mobile VoIP call quality, with the underlying hardware supporting up to 15 point-to-point simultaneous calls using SIP and the G.711 based codec. Using the PEAQ algorithm to evaluate the IP-based radio, a PEAQ value of 4.15, i.e. good, was achieved. Streaming audio across the network reduces the available bandwidth by 8Kbps per client due to the unicast nature of streaming. Therefore, a multicast approach has been proposed for efficient bandwidth utilization. The quality of the text-to-voice service rendered by the translation gateway had a PESQ score of 1.6 i.e. poor. The poor performance can be attributed to the TTS engine implementation and also to the lack of robustness in the time-alignment module of the PESQ algorithm. The dissertation also proposes the use of the WMN infrastructure as a back-haul to isles of WSNs deployed in areas of interest to provide access to information about environmental variables useful in decision making

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

Detection and analysis of misuse in SIP-based networks

Die Sprachkommunikation über „Voice over IP“-Netzwerke, basierend auf dem Session Initiation Protokoll (SIP), verbreitet sich auf Grund von Funktionalitäts- und Kostenvorteilen zunehmend und wird die klassischen Telefonnetze in den nächsten Jahren vollständig ablösen. Zusätzlich zu den Netzen der Telefonanbieter wird die Sprachkommunikation über das SIP-Protokoll auch im Unternehmens- und Privatanwenderumfeld unverzichtbar. So bietet VoIP die Möglichkeit, sich unabhängig von dem aktuellen Aufenthaltsort über das Internet bei dem jeweiligen Heimatnetzbetreiber oder der eigenen Firma anzumelden und über das dortige Nutzerkonto Gespräche zu führen. Da die Telefonie somit von einer geschlossenen und vergleichsweise sicheren Plattform auf eine viel offenere Plattform in das Internet migriert wird, ergeben sich neue Risiken und Missbrauchsmöglichkeiten im Bereich der Telefonie. In dieser Dissertation werden Angriffe untersucht, die mit der Einführung von SIP-basierten Sprachdiensten im Internet entstehen und nicht aus Bedrohungen der Netzwerkschicht oder aus rechtlichen Vertragsbestimmungen resultieren. Das Ziel dieser Angriffe ist das Erlangen eines finanziellen Vorteils, indem ein Angreifer kompromittierte Zugänge für Auslandstelefonate oder für Anrufe zu Premiumnummern auf Kosten der Anschlussinhaber nutzt („Toll Fraud“). Für die Realisierung der Bedrohungsanalyse und der Angriffserkennung wurden Konzepte, ein Versuchsnetzwerk sowie die notwendigen Softwarekomponenten ergebnisorientiert entwickelt. Im Vergleich zu anderen Forschungsarbeiten wurden Untersuchungen mit Ködersystemen (Honeypots) weiterentwickelt und es wurde ein System für eine verteilte, automatische Angriffserkennung entwickelt. Dafür wurden SIP-Verkehrsdaten über einen Zeitraum von sechs Jahren in zwei Class-C-Netzwerken aufgezeichnet und mit einem neuen Analyseansatz unabhängig von einzelnen SIP-Nachrichten automatisch ausgewertet. Die Ergebnisse des Feldversuches in dieser Dissertation zeigen, dass die Bedrohungen für die SIP-Infrastruktur ansteigen und dass bereits eine Weiterentwicklung und Optimierung der Angriffswerkzeuge nachzuweisen ist. Die zunehmende Anzahl der Toll Fraud-Versuche mit internationalen Anrufzielen (und auch zu Premium-Rufnummern) verdeutlicht, dass bei einem unzureichenden Schutz der SIP-Server für die Nutzer und Betreiber sehr schnell ein erheblicher finanzieller Schaden entstehen kann. Es ist daher unerlässlich, die vorgeschalteten, systematischen Angriffsstufen frühzeitig zu erkennen und Abwehrkomponenten zu benachrichtigen. Für die automatisierte, verteilte Angriffserkennung in Echtzeit und für die Maximierung des Beobachtungsgebietes wurde für diese Dissertation das „Security Sensor System“ entwickelt. Mit Hilfe von leichtgewichtigen Sensoren wurde eine weltweite signaturbasierte Angriffserkennung realisiert. Zusätzlich zu der standortbezogenen Angriffserkennung werden Angriffe durch einen zentralen Dienst korreliert. Dadurch können Angreifer netzwerkübergreifend bzw. länderübergreifend identifiziert und somit Gegenwehrkomponenten in Echtzeit benachrichtigt werden. Der Vergleich der verschiedenen Messstellen im Internet belegt, dass die analysierten Angriffsmuster nicht nur im Netzwerk der Universität Duisburg-Essen, sondern zeitlich zusammenhängend auch an anderen Standorten auftreten. Dadurch wird deutlich, dass die ermittelten Ergebnisse auch für andere Netzwerke gültig sind und dass die Toll Fraud-Problematik bereits für alle Betreiber von SIP-Servern relevant ist.Voice over IP networks based on the Session Initiation Protocol (SIP) are becoming more and more widespread in the Internet due to functionality and cost advantages and will soon replace the classic telephony networks. Therefore, support of open SIP-based interfaces is an increasingly important requirement for IP-based Public Branch eXchanges (PBXs) and provider systems. The VoIP service allows using the personal or company VoIP account from any location worldwide. The migration of the telephony service from a closed and comparatively secure environment to a network with open interfaces creates security issues and opens up new opportunities for misuse and fraud. In this thesis, attacks are analyzed which result from introducing SIP-based voice services and do not belong to the area of contract regulations or attacks on the network layer. The attacker’s goal is to gain immediate financial benefit by making toll calls (international, cellular, premium services) via cracked third party accounts (“Toll Fraud”). To realize the threat analysis and the attack detection concepts, a SIP-based testbed and required software components were developed. In comparison to the related work, analyses with Honeypots were enhanced and a mechanism for automatic, distributed attack detection was realized. Therefore, for gathering the required data, a Honeynet with two class-C networks captured the SIP traffic for a period of six years. The automatic analysis is based on attacks and operates independently of single SIP messages. The field test results of this thesis demonstrate that SIP-based threats increase over time and attack tools are optimized and enhanced. The increasing number of Toll Fraud attempts to international or premium numbers reveals that Toll Fraud attacks can cause the account owner substantial financial damage in a very short amount of time if there is insufficient attack detection and mitigation. Hence, it is necessary to implement an attack detection which is able to identify the different attack stages and sends a notification to mitigation components before a Toll Fraud call is established. In this thesis, the Security Sensor System was developed to maximize the monitoring scope and to realize the distributed, automatic attack detection in real-time. The light-weight sensor component provides worldwide signature-based attack detection. Additional to the location-based attack detection, all attack notifications are sent to a central service which correlates the incoming alarm messages and provides a comprehensive attacker identification to inform mitigation components in real-time. The comparison of different sensor nodes in the Internet shows that the analyzed attack patterns do not only occur in the University testbed, but also temporally coherent in other networks. Thus, the results are valid for different network environments and it is crucial to know that Toll Fraud attacks are already performed in reality

Secure covert communications over streaming media using dynamic steganography

Streaming technologies such as VoIP are widely embedded into commercial and industrial applications, so it is imperative to address data security issues before the problems get really serious. This thesis describes a theoretical and experimental investigation of secure covert communications over streaming media using dynamic steganography. A covert VoIP communications system was developed in C++ to enable the implementation of the work being carried out. A new information theoretical model of secure covert communications over streaming media was constructed to depict the security scenarios in streaming media-based steganographic systems with passive attacks. The model involves a stochastic process that models an information source for covert VoIP communications and the theory of hypothesis testing that analyses the adversary‘s detection performance. The potential of hardware-based true random key generation and chaotic interval selection for innovative applications in covert VoIP communications was explored. Using the read time stamp counter of CPU as an entropy source was designed to generate true random numbers as secret keys for streaming media steganography. A novel interval selection algorithm was devised to choose randomly data embedding locations in VoIP streams using random sequences generated from achaotic process. A dynamic key updating and transmission based steganographic algorithm that includes a one-way cryptographical accumulator integrated into dynamic key exchange for covert VoIP communications, was devised to provide secure key exchange for covert communications over streaming media. The discrete logarithm problem in mathematics and steganalysis using t-test revealed the algorithm has the advantage of being the most solid method of key distribution over a public channel. The effectiveness of the new steganographic algorithm for covert communications over streaming media was examined by means of security analysis, steganalysis using non parameter Mann-Whitney-Wilcoxon statistical testing, and performance and robustness measurements. The algorithm achieved the average data embedding rate of 800 bps, comparable to other related algorithms. The results indicated that the algorithm has no or little impact on real-time VoIP communications in terms of speech quality (< 5% change in PESQ with hidden data), signal distortion (6% change in SNR after steganography) and imperceptibility, and it is more secure and effective in addressing the security problems than other related algorithms

Una aproximación a la Telefonía 2.0: Asterisk

Aunque está visto que cada día cobra mayor relevancia, la comunidad de Asterisk en Español, ha permanecido circunscrita al contenido y desarrollo de las fuentes que provienen del ingles. Con este proyecto pretendo sentar las bases de Asterisk en nuestro idioma, abarcándolas de la siguiente forma: creación de una WIKI exclusiva de Asterisk, desarrollar un Caso Práctico que cubra la teoría, demostrar el caso utilizando una Máquina Virtual que se adjuntara al proyecto y utilizar herramientas de Software Libre como es Asterisk para preservar su filosofía de uso y distribución

Managing law practice technology

Presented by Barron K. Henley, at a seminar by the same name, held November 17, 2020

Verkeersgegevens en artikel 13 Grondwet. Een technische en juridische analyse van het onderscheid tussen verkeersgegevens en inhoud van communicatie

Zie webpagina: http://www.wolfpublishers.com/book.php?id=102