Networking, VoIP, Security Research Papers (original) (raw)

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- Networking, VoIP, Security, Asterisk Voip
- by Mohd Rahul and +1
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- Wireless Sensor Networks, Cloud Computing, Networking, VoIP, Security
- by Samuel Sotillo
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- Networking, VoIP, Security
- by IT Ambassador Ted Green
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- Machine Learning, Data Mining, Access Control, Cloud Computing

Um dos principais componentes da tecnologia de voz sobre IP são os algoritmos de codificação da fala, conhecidos como codecs. Existe uma grande variedade de codecs e alguns apresentam melhor qualidade que outros, a depender do grau de perda de informação proporcionada pelas técnicas de compressão empregadas. Este trabalho tem por finalidade avaliar a qualidade de alguns codecs de código aberto, tendo como ponto de partida a implementação do procedimento de avaliação definido pela Rec.ITU-T P.834 para derivação do parâmetro de degradação da fala a 0% de perda de pacotes (Ie). Este parâmetro, específico para cada codec, é utilizado pelo Modelo E (Rec. ITU-T G.107) para avaliação da qualidade da fala.

- by Edjair Mota
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- Voip, Networking, VoIP, Security, VoIP over WiFi, VoIP qos

What an alternative. Most people argue that if something works well there is no sense in changing it. This is exactly what is happening with VoIP today. Voice over Internet Protocol usually called VoIP is the transmission of voice, video conferencing, data, faxes over an IP based network. VoIP technology has received much attention due to several emerging application in voice communication. This paper presents a tutorial on a basic way of deploying VoIP using miniSipServer on an existing Metropolitan Area Network (MAN). After which security solution is deployed on the network using Virtual Private Network (VPN)

- by Babatunde Lawall
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- VOIP security, Networking, VoIP, Security, VoIP/SIP/IMS

ASTPP is an Open Source VoIP billing solution for Freeswitch. It supports pre-paid and post-paid billing with call rating, credit control and call reporting. It also provides many other features such as calling cards, least cost routing... more

- by Arpit Modi
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- Information Systems, Structural Engineering, Teacher Education, Voip

A Mobile Ad hoc Network (MANET) is a collection of mobile stations with wireless interfaces which form a temporary network without using any central administration. MANETs are more vulnerable to attacks because they have some specific characteristics as complexity of wireless communication and lack of infrastructure. Hence security is an important requirement in mobile ad hoc networks. One of the attacks against network integrity in MANETs is the Black Hole Attack. In this type of attack all data packets are absorbed by malicious node, hence data loss occurs. In this paper we investigated the impacts of Black Hole attacks on the network performance. We have simulated black hole attacks using Network Simulator 2 (NS-2) and have measured the packet loss in the network without and with a black hole attacks. Also, we measured the packet loss when the number of black hole attacks increases.

VoIP (Voice over Internet) provides delivery of voice information over unsecured IP-based networks like the Internet.
VoIP data, signaling and voice, needs to be secured in such an
environment. Security mechanisms take their toll on VoIP system performance. SIP is dominant signaling protocol for VoIP. This paper measures relative decrease in VoIP performance of system with secured SIP signaling over one without it. It compares SIP with authentication enabled over three transport protocols: UDP, TCP and TLS. Peak throughput of concurrent calls, registration request delay, session request delay, SIP server CPU and RAM usage are measured. Testbed environment consists of Asterisk IP private branch exchange (PBX) as a part of Elastix server, several SIP user agents and SIPp traffic generator. Test results show that performance of SIP over TLS based signaling is four times lower than the SIP signaling over UDP in most metrics.

- by Merima Kulin
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- Networking, VoIP, Security, VoIP/SIP/IMS, Performance Testing

IP telephony is a form telecommunication which involves the transmission of voice, video streams and possibly data on a network. It establishes an interactive communications session between two or more users who are geographically separated.
This project provides a simple, accurate and affordable method of implementing a call log algorithm for communication between IP desk phone and a smart phone technology on a standalone deployment and on an existing Communication and Information Science network.
In this project, Elastix software was installed on a high capacity system that serves the gateway and database server which captures and stores the call logs between the desk phone and the smart phones. This server is then configured to provide to provide a real time voice communication in the faculty of Communication and Information Science. To access this facility, users in the faculty have to install X-lite 4.0 (a digital Softphone) both on their devices which could be a Personal computer, android smart phones, tablets etc.

As a user moves to different locations, how to make the communication service in use follows the user to the current location without being broken off depends on the support of Personal Mobility. Therefore, supporting personal mobility for the " follow-me service " is the goal of this research. We integrate Radio Frequency Identification (RFID), and SIP Express Router (SER) to create a VoIP system which can achieve personal mobility. In our proposed system, as soon as the user moves to different locations, the sensors of the doors/locations can read the RFID Tag of the user and the server can activate an according phone and register to the SER immediately. To avoid multiple registrations, our system will close the phone which the user used before at the same time. The advantage of this research is that it's unnecessary to change the settings of the SER, and neither to use the specific phone. In the implementation, we introduce two modules, Remote Call Server (RCS) and Remote Call Client (RCC), to support personal mobility. Both modules are written in Microsoft© Visual C#.NET and use the same MySQL database with the SIP Proxy Server for reducing the deployment cost. Moreover, the RFID reader directly connects to RCS via a regular USB port. As a result, the reader can immediately transfer the raw data to RCS when it reads some tags. These features make deploying a personal mobility architecture easier and promising.

This paper describes the implementation process of the Free Technologies Open Laboratory (LATL) in the Center for Information Technology Renato Archer (CTI) under the approach of 5W1H framework. The actions of adoption, use and development of ICT by governs remain controversial between daily practice and legislation pro free technologies. The implementation of LATL aims to provide means to promote and empower government units in the use and development of free technologies and open standards for documents and data. The application of 5W1H model proved useful in the definition and evaluation of the actions of cause and effect in the implementation process of the laboratory. Resumo. Este artigo descreve o processo de implantação do Laboratório Aberto de Tecnologias Livres (LATL) no Centro de Tecnologia da Informação Renato Archer (CTI) sob o enfoque do modelo 5W1H. As ações de adoção, uso e desenvolvimento de TIC por governos continuam sendo controversas entre a prática vigente e a legislação pró tecnologias livres. A implantação do LATL visa proporcionar meios para promover e capacitar unidades de governo no uso e desenvolvimento de tecnologias livres e padrões abertos para documentos e dados. A aplicação do modelo 5W1H mostrou-se útil na definição e avaliação das ações de causa e efeito do processo de implantação do laboratório.

In this paper, we present an optimized implementation of secure VoIP protocol stack so that the stack
would fit into the memory and computation budget of constrained embedded systems. The novel approach
that we take to achieve this is to perform cross-layer optimization of buffers and buffer operations.
Buffers and buffer operations are involved in playback, capture, codec transformations, and network I/O.
Following this approach, we have implemented VoIP application functions, RTP, and Secure RTP
protocols in a tightly integrated and highly optimized manner, on the top of the embedded TCP/IP stack,
uIP. We call the protocol stack thus constructed, the uVoIP stack. We have tested the uVoIP stack in
GNU/Linux Operating System using tunnel device for sending and receiving raw packets.

- by Muhammad Zulkifl Hasan
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- Engineering, Wireless Communications, Network Security, Opnet
- by Edjair Mota
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- Voice over IP, Secure and Critical Communicating Systems, Voip, VOIP security

This paper presents a study on IP Security Protocol used to guarantee secure transmissions of Voice over IP on wireless networks. We have investigated and quantified the impact of IPSec security mechanisms on multimedia traffic and... more

- by Edjair Mota
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- Multimedia Communications, Wireless security, Voip, Quality of Service