Asterisk Voip

This study entails the simulation and implementation of voice over a data network, a telephony system using an IP PBX solution. A technology called voice over Internet Protocol (VoIP), or Internet telephony means that voice is carried over an IP network. Voice, which is an analogue signal, is converted to digital data, which is then disassembled and transmitted through a network only to be reconverted back to an analogue signal on the other end using a Linux based IP-PBX solution called Asterisk. This service can be properly managed and deployed over a network with less stress and expenses. The IP PBX main server also has integrated in it other communication services such as Voice mails, IVR’s, all embedded in the IP PBX SYSTEM. This technology promises an evolutionary leap beyond the standard telephone service we have been accustomed to, as well as a host of benefits. The new technology transmits voice signals the same way email is sent, using the Internet’s data-transfer protocols to break conversations into digital packets that can be sent on lower-cost, more efficient “packet-switched” networks. This project was able to address the persistent communication problem which existed in the departments by allowing users to communicate with the services the solution provided with less stress and comfort.
Target market includes: Corporate organizations, Universities, Health care, Airports, Hotels, Banks etc. This project is economic, cost effective, gives full control to the administrator and provides mobility, feasible, Peer-to-Peer phone calls. The contents of IP PBX System, supplemented by a good number of necessary and descriptive drawings which makes this project report very easy to understand

Session Initiation Protocol (SIP) is a standard protocol for multimedia as a product of Internet Engineering Task Force (IETF) and has been used as a standard VoIP. This protocol combines mobile technology and the world internet. A session in SIP networks can be voice calls, e-mails, text messages or video streaming. The advantages of SIP is a network operator can use to control all of communication in the network, not just the voice. SIP protocol comes from the Simple Mail Transport Protocol (SMTP) and Hypertext Transport Protocol (HTTP). In this research Voice over IP service (VoIP) SIP on the local network using a Graphical Network Simulator (GNS3) analyzed by observing the Quality of Service (QoS) that is the value of the delay, jitter, packet loss and throughput with Wireshark uses that standard G. 114. There one model of the network using OSPF routing and simulation C7200 Cisco routers and type of communication network connecting cable.
Keywords : VOIP, QoS, OSPF

Nowadays information technology, especially the Internet developed very rapidly, which is actually a Internet computers connected to each other. Telephony technology is also developed very fast and there is some alternative to use VoIP beside analog telephone because the cost is cheaper. VoIP also use codec that can compress voice data but the quality is still good. This research design an open source system of Asterisk server because company need of VoIP that can support traditional analog telephony system. Beside design an open source system, some codec technology is also tested, which are G.711 as commonly codec and also G.729 and G.723.1 as propiteary codecs, offering less bandwidth and more clearly sound than G.711. G.729 and G.723.1 is limited for one user only so it can be tested only for one user. After codec testing is arranged then an interconnection system of PSTN or analog telephony system is also tested. Using Linksys SPA-3102 interconnection to analog telephony is also tested and worked for one client. Abstrak Saat ini teknologi informasi, terutama Internet berkembang sangat pesat, sehingga ada teknologi jaringan internet yang saling menghubungkan komputer tersebut. Teknologi telephony juga berkembang sangat cepat dan ada beberapa alternatif untuk menggunakan VoIP disamping telepon analog karena biayanya lebih murah. VoIP menggunakan codec yang bisa mengkompresi data suara namun kualitasnya tetap bagus. Penelitian ini merancang sistem open source server Asterisk karena perusahaan membutuhkan VoIP yang dapat mendukung sistem telepon analog. Selain merancang sistem open source, beberapa teknologi codec juga diuji, yaitu G.711 sebagai codec yang berlaku umum dan juga G.729 dan G.723.1 sebagai codec propiteary, yang menawarkan bandwidth lebih sedikit dengan suara yang lebih jelas daripada G.711. G.729 dan G.723.1 terbatas hanya untuk satu pengguna sehingga hanya bisa diuji untuk satu pengguna saja. Setelah pengujian codec dilakukan maka sistem interkoneksi PSTN atau sistem telepon analog juga diuji. Interkoneksi dilakukan dengan voice gateway Linksys SPA-3102 dihubungkan ke telepon analog juga diuji dan dilakukan untuk satu klien.

Abstract - CV. SatuAtap Camp adalah perusahaan yang bergerak di bidang teknologi informasi. Jasa yang ditawarkan berupa pembuatan website, web kilat, jaga server dan layanan domain serta web hosting. Sekarang ini CV. SatuAtap Camp telah memiliki banyak pelanggan yang tersebar di seluruh wilayah Indonesia. Saat ini CV. SatuAtap Camp sudah memiliki satu kantor untuk operasional usaha.

Voice over Internet Protocol adalah salah satu solusi untuk membuat jaringan komunikasi alternatif yang tentu lebih efektif. Hal ini karena teknologi Voice over Internet Protocol menggunakan internet sebagai media komunikasi. Untuk terhubung ke jaringan telepon publik menggunakan modem GSM yang dijadikan gateway. Dengan begitu pelanggan dari jaringan telekomunikasi seluler / telepon publik bisa menghubungi karyawan pada teknologi Voice over Internet Protocol. Penggunaan komputer server Voice over Internet Protocol cukup menggunakan raspberry pi dan asterisk.
Situasi ini juga didukung oleh pengembang aplikasi ponsel, yang saat ini sudah banyak aplikasi gratis untuk komunikasi Voice over Internet Protocol  melalui telepon seluler. Untuk terhubung ke internet publik / WAN digunakan VPN. Dengan begitu komunikasi bisa lebih efektif dan lebih optimal dalam pelayanan konsumen dan pelanggan bagi CV. SatuAtap Camp.

Keywords - CV. SatuAtap Camp, Voice over Internet Protocol, Raspberry Pi, Asterisk, GSM, Gateway.

VoIPTech Solutions Being one of the major VOIP service providers in world, has been providing Expert asterisk services and support of customization at a very affordable price.To know more visit our website or call us at +91-7008220621 today.

In this paper, we studied the Voice over IP Quality of Service measurement techniques and Quality of Service implementations. We find out techniques that gives better Quality as well as their weaknesses. We have studied E-Model, Mean Opinion Score (MOS) optimization and other traditional algorithms as well as soft computing methods to find Quality of Service of Voice over IP. A thorough study of getting maximum Quality of service is presented here.

Resumen. El presente trabajo estudió GoogleTTS, Espeak y Festival como alternativas de TTS (Text To Speech) para la implementación de servicios VoIP en una red LAN universitaria, con el objetivo principal de implementar el servicio de consulta de notas de las asignaturas de los distintos programas de postgrado, teniendo como base tecnológica la infraestructura de VoIP institucional conformada por CUCM Cisco, Asterisk, AGI, PHP, Webservice y Troncales Lógicas de VoIP; que interopera con el Sistema de Posgrado implementado con Postgres y Java. Se definió la escala MOS y el Modelo E como estándar para la ponderación y medición de la Calidad de Voz y Tiempo de Respuesta de las llamadas con cada uno de TTS implementados. Se utilizó teléfonos IP CISCO y Softphones para aplicar una encuesta a 50 estudiantes de postgrado, teniendo como resultado que; Festival alcanza un valor nominal de 4 respecto a la escala MOS correspondiente a BUENO, superando el valor nominal de 3 obtenido por Espeak y GoogleeTTS correspondiente a ACEPTABLE. Se corroboró las encuestas realizadas mediante pruebas técnicas que utilizaron herramientas software como Myspeed (monitoreo) y Sipp (Esfuerzo). Se puede decir de manera general que el uso de software libre y específicamente el uso de Festival como TTS permitió gestionar de manera satisfactoria el plan de marcado y el listado de estados de notas en la implantación del servicio de VoIP meta a muy bajo costo.

The demands of people for Information Technology related services are very high and this needs to be catered for. In view of this, Our solution which is in the form of a service is developed to contribute to the ever dynamism of the social networking sector of Information Technology. The solution was developed by adopting the Agile methodology. This methodology is customer centered and considers the continual inputs of the customer as a vital key in streamlining the process to produce a better product/service.

Voice over IP (VoIP) is a blanket description for any service that delivers standard voice telephone services over Internet Protocol (IP). Computers to transfer data and files between computers normally use Internet protocol.
"Voice over IP is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP (internet protocol) network where it is reassembled, decompressed, and converted back into an analog wave form.." The transmission of sound over a packet switched network in this manner is an order of magnitude more efficient than the transmission of sound over a circuit switched network.

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 (LCR), did management, reseller management etc.

Cloud computing as a powerful economic stimulus widely being adopted by many companies. However, the management of cloud infrastructure is a challenging task. Reliability, security, quality of service, and cost-efficiency are important issues in these systems. They require resource optimization at multiple layers of the infrastructure and applications. The complexity of cloud computing systems makes infeasible the optimal resource allocation, especially in presence of uncertainty of very dynamic and unpredictable environment. Hence, load balancing algorithms are a fundamental part of the research in cloud computing. We formulate the problem of load balancing in distributed computer environments and review several algorithms. The goal is to understand the main characteristics of dynamic load balancing algorithms and how they can be adapted for the domain of VoIP computations on hybrid clouds. We conclude by showing how none of these works directly addresses the problem space of the c...

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.

Perkembangan teknologi khususnya teknologi informasi membawa perubahan yang sangat mendasar bagi dunia telekomunikasi. Dalam teknologi komunikasi, komunikasi suara merupakan satu hal yang akan menjadi bagian yang sangat penting, karena saat ini komunikasi suara dianggap komunikasi yang paling praktis. VoIP  sebagai  alternatif  sarana komunikasi suara di  Amik Sigma. Namun, pengembangan lebih lanjut harus dilakukan apabila sistem VoIP hasil implementasi hendak dijadikan pengganti sistem Public Switched Telephone Network (PSTN) dan PBX. Layanan - layanan seperti yang diberikan oleh sistem penggunaan telepon berbasis VoIP memberi banyak keuntungan terutama dari segi biaya jelas lebih murah dari biaya telepon tradisiona. Jaringan IP bersifat global harus dapat direplikasi oleh sistem VoIP secara keseluruhan, agar transisi dapat berjalan dengan mulus. Dalam penelitian ini pembahasan perancangan server VoIP menggunakan Tri Box, X Lite untuk Komputer dan 3CX untuk Smart Phone. X Lite dan 3XC menyediakan layananan instant messaging, video call, video conference.

Cloud computing as a powerful economic stimulus widely being adopted by many companies. However, the management of cloud infrastructure is a challenging task. Reliability, security, quality of service, and cost-efficiency are important issues in these systems. They require resource optimization at multiple layers of the infrastructure and applications. The complexity of cloud computing systems makes infeasible the optimal resource allocation, especially in presence of uncertainty of very dynamic and unpredictable environment. Hence, load balancing algorithms are a fundamental part of the research in cloud computing. We formulate the problem of load balancing in distributed computer environments and review several algorithms. The goal is to understand the main characteristics of dynamic load balancing algorithms and how they can be adapted for the domain of VoIP computations on hybrid clouds. We conclude by showing how none of these works directly addresses the problem space of the c...

In particolare il lavoro riguarda la realizzazione di un servizio di conferenza telefonica/VoIP multiutente di cui ho curato l'analisi delle funzionalità da fornire all'utente, l'analisi delle problematiche di scalabilità e affidabilità e l'implementazione di un prototipo del prodotto finale.

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.

... Mosiuoa Tsietsi, Zelalem Shibeshi, Alfredo Terzoli, George Wells Rhodes University Department of Computer Science Grahamstown, South Africa ... Additionally, web-based systems are considered to be the best type of system to address the needs of a live virtual classroom [2]. ...

SUMMARY The widespread use of Session Initiation Protocol as a signalling protocol has created various challenges. An important one is that its throughput can be severely degraded when an overload happens in the proxy server because of several retransmissions from the user agent. One common approach to overcome this problem is 'load balancing'. A balancer needs to know the status of proxy servers, which are continuously gathered implicitly or explicitly. Implicit methods have averagely less overhead than explicit ones. This paper attempts to prevent throughput reduction by balancing the loads among available proxy servers properly using an implicit mechanism called History Weighted Average Response time. The proposed algorithm is robust because it incurs no extra processing to proxy servers. The novelty of the mechanism is making use of 'response time history' to estimate the load being currently processed on servers. By implementing in a real testbed, throughput and scalability are improved compared with an important state-of-the-art similar algorithm. This improvement stems from no need for modification in SIP protocol, easy implementation and application, simple computations for making decision and no need for extra feedback between servers and load balancer.

— The Session Initiation Protocol (SIP) is an application-layer control protocol for creating, modifying and terminating multimedia sessions. An open issue is the control of overload that occurs when a SIP server lacks sufficient CPU and memory resources to process all messages. We prove that the problem of overload control in SIP network with a set of n servers and limited resources is in the form of NP-hard. This paper proposes a Load-Balanced Call Admission Controller (LB-CAC), based on a heuristic mathematical model to determine an optimal resource allocation in such a way that maximizes call admission rates regarding the limited resources of the SIP servers. LB-CAC determines the optimal " call admission rates " and " signaling paths " for admitted calls along optimal allocation of CPU and memory resources of the SIP servers through a new linear programming model. This happens by acquiring some critical information of SIP servers. An assessment of the numerical and experimental results demonstrates the efficiency of the proposed method.

An important problem one needs to deal with in
a Voice over IP system is server overload. One way for pre-
venting such problems is to rely on prediction techniques for
the incoming traffic, namely as to proactively scale the avail-
able resources. Anticipating the computational load induced
on processors by incoming requests can be used to optimize
load distribution and resource allocation. In this study, the
authors look at how the user profiles, peak hours or call pat-
terns are shaped for a real system and, in a second step, at
constructing a model that is capable of predicting trends.

Dynamic optimization based on incoming load analysis and prediction is considered to be an innovative
approach in order to prevent the overload of the servers in a Voice over IP system. The ongoing
project is in an early stage of study and the followings are the current vision and concept regarding
it. The information gathered by inspecting the real system of an IT company, MixVoIP, (probe server
and sensors spread inside the cloud) and by analyzing the data provided by the predictive algorithm,
will be used to optimize load distribution and resource allocation. The implementation in the real-life
environment should lead to an improvement of the service offered but also to a sensible reduction of
the associated carbon emissions, e.g. as a result of an improved load management, reduced idle CPU
times or optimally exploited resources.