Membangun Server Voip Dengan Asterisk Di Linux Beserta Administrasi User Berbasis Website

The development of computer network allows for rapid pass voice traffic over computer networks or common called VoIP (Voice Over Internet Protocol). Voice over Internet Protocol (VoIP) is a technology capable of passing voice traffic and video or even a form of data packets over a computer network based on IP (Internet protocol). Major emphasis of VoIP (Voice Over Internet Protocol) is the cost. With two locations that are connected to the internet or network then the conversation becomes very low cost even close to free (Rp0, 00 -). This system is designed using an asterisk on fedora 7 -1.6.0.1

PEMBANGUNAN VOIP PADA JARINGAN EXISTING IT TELKOM DENGAN MEMBERDAYAKAN IP-PBX IT TELKOM DAN ASTERISK SERVER BUILT THE VOIP SYSTEM ON IT TELKOM EXISTING NETWORK USING IT TELKOM IP PBX AND ASTERISK SERVER

ABSTRAKSI: Fasilitas komunikasi suara yang biasanya hanya dimiliki oleh jaringan circuit atau PSTN, saat ini mulai bergeser pada jaringan IP sebagai media pengiriman data.Komunikasi suara pada jaringan data (Internet) biasa disebut dengan VoIP (Voice over Internet Protocol ). VoIP dan semua komunikasi multimedia masa depan akan dilewatkan ke dalam satu jaringan data yang biasa dikenal dengan NGN (Next Generation Network). NGN merupakan jaringan data yang dapat mengakomodasi service komunikasi data maupun komunikasi suara. Untuk dapat mengintegrasikan hal tersebut, NGN membutuhkan suatu device yang salah satunya adalah softswitch. Konsep softswitch ini terdapat dalam suatu alat yang biasa disebut dengan IP-PBX.PBX (Private Branch Exchange) adalah suatu sentral telepon yang digunakan untuk melayani komunikasi pada suatu business oficce, sedangkan IP-PBX merupakan suatu PBX yang mampu memberikan layanan circuit maupun layanan yang berbasis IP. IPPBX mampu melakukan proses switching pada komunikasi suara dengan jaringan circuit maupun jaringan IP, serta mampu menginterkoneksikan keduanya.Dalam Tugas Akhir yang berjudul ”Pembangunan VoIP Pada Jaringan Existing IT Telkom Dengan Memberdayakan IP-PBX IT Telkom Dan Asterisk Server” dibuat suatu komunikasi VoIP serta integrasinya dengan jaringan analog PSTN IT Telkom dengan memanfaatkan IP-PBX dan Asterisk server sebagai sentral komunikasi. Pada hasil pembangunan sistem didapatkan hasil bahwa Asterisk hanya memerlukan load processor sebesar 3% dengan kenaikan memory 1 %untuk menangani 9 simultan call analog IP-PBX ke SIP Asterisk. Asterisk menggunakan 1% load processor dan 1% kenaikan memory untuk menangani 9 panggilan SIP Asterisk secara simultan. Nilai MOS untuk kedua analisa diatas menunjukkan hasil sekitar 4.1 yang mempunyai nilai opini baik.Kata Kunci : VoIP, Asterisk, IP-PBX, Integrasi.ABSTRACT: Voice communication facility that usually gived by circuit network or PSTN, in this time start to shift at IP network as media of data delivery.Voice communication on packet data network called by VoIP (Voice over Internet Protocol). VoIP and all future multimedia communication will be transport into an IP network that usually called by NGN (Next Generation Network). NGN is a data network which can accommodate service of data communications and also communications voice. For that, NGN requiring a device which one of them is softswitch. This softswitch concept is including on IP-PBX telephone exchange.PBX (Private branch Exchange) is a telephone exchange that serves analog communication for business office, while IP-PBX is a PBX that serves a voice communication based on circuit and data network. IP-PBX able to handle switching process of voice communication with circuit network and IP network, and also able to interconnect both of that network.In final assignment that entitling “VoIP Development on IT Telkom Existing Network Using IT Telkom IP PBX And Asterisk Server” build a VoIP communication that integrate with analog networks in IT Telkom using an IP PBX and Asterisk Server. On the build the system get the result that Asterisk server use 3 % of load processor and 1% add of memory used when handle 9 simultaneous call from SIP asterisk to analog IP-PBX. Asterisk used 1% of load processor and 1 % add of memory usage when handle 9 simultaneous call From SIP to SIP. The result of Mean Opinion Service (MOS) is a 4.0 until 4.1 for all callsKeyword: VoIP, Asterisk, IP-PBX, Integration

Implementasi Smart Agent IP PBX pada Perusahaan Kelas SOHO

Smart Agent System which is a system that is needed in every company in memajukkan productivity. With the support of the system is then an employee can manage and improve their performance. Advancements in the field of economic development of areas affected by today's technology, comes a lot of online stores that sell a variety of goods on the internet. Most existing online store today is an enterprise-class SOHO (small office Home office) where by using his home as a residence and place of work. And in the implementation the company has few employees with some parts but have the same sales system with large-scale enterprise. In this final project implements a smart agent system by using an IP PBX Asterisk based on SOHO-class companies. IP PBX that the intent is a PC that serves as a mini server or IP-based PBX. While Asterisk is one of the server software VoIP (Voice over Internet Protocol) which can be used to build an IP PBX. In this final project has taken features of asterisk such as call forward, call pickup, conference calling, call parked, mailbox, followme, blacklist, call transfer, musiconhold, time base context and dial plan security. Results obtained in this final project of a smart agent system-based IP PBX Asterisk on SOHO-class company that has the features that support the communication agent. In testing this system obtained MOS values by 4.68 to *. wav file which is the default of asterisk 16bit PCM with sampling frequency of 8KHz. And the result is a qualitative test involves the full-featured IP PBX opinion as much as 97%, opinions need additional features as much as 3% and opinions of the most useful feature as much as 100% for IVR (Interactive Voice Response) of respondents who are perpetrators of SOHO and SOHO users. Keywords: SOHO, Smart Agent, Features, MOS (Mean Opinion Score), Qualitative testing

Video Quality Measurement For 3G Handsets

Internet provides many services. VOIP (Voice over IP) is one such service also known as Internet Telephony or IP Telephony. Using VOIP we can make voice telephony calls, participate in video conferences, etc over data networks (WAN'S and LAN'S) or internet. VOIP operates by first converting voice data into digital form, organizing them into packets, transmitting them through the most convenient route to their destination and finally reassembling them at the destination. Protocols like SIP/RTP, H.323, MGCP are designed which perform all the above steps. This project aims to make a video call from a 3G Mobile to an IP phone via Asterisk Gateway. Asterisk to act as bridge for video call between 3G-IP network must capture the audio/video stream from 3G mobile, convert captured stream into an IP compatible stream and send stream to an IP client and vice-versa. Asterisk needs to support AMR codec for audio and MPEG-4 codec for video and H.324M protocol stack for capturing audio/video streams from 3G Mobile. Asterisk currently supports audio codec's like GSM, G.729, A-law, and U-law. It allows H.261, H.263 video streams as pass-through. It supports VOIP protocols like SIP/RTP, MGCP, and H.323 which allows it to interface with other devices. This project aims to implement AMR codec, H.324M protocol stack, MPEG-4, bridging functions between SIP/RTP-ISDN and 3G Mobile in Asterisk which allows a 3G phone to call a SIP client via Asterisk. This thesis discusses the implementation of AMR in asterisk as well as SIP protocol and SIP soft phones

Desain dan Implementasi Sistem Contac center berbasis Asterisk Server

ABSTRAKSI: Perkembangan teknologi telekomunikasi yang cukup pesat mendorong kebutuhan untuk berkomunikasi mendapatkan suatu informasi melalui telepon semakin tinggi.Di sisi lain, call center yang biasanya digunakan di perkantoran atau universitas untuk memudahkan mendapatkan informasi, masih menggunakan komunikasi circuit based.Padahal dalam dunia komunikasi global saat ini, trend komunikasi mulai bergeser dari jaringan circuit atau PSTN ke model komunikasi melalui IP atau yang biasa disebut VoIP (Voice over IP). Komunikasi yang popular saat ini adalah komunikasi yang berbasis web, dimana web mulai diterapkan untuk menjadi call center yang komunikasinya melalui IP, sehingga fungsi dari call center yang hanya melayani panggilan menjadi contact center yang memiliki fitur panggilan, chat dan FAQ. Untuk bisa mensupport data suara digital yang beroperasi di jalur internet protokol dibutuhkan suatu hardware yang disebut IP PBX yang saat ini fungsi dari IP PBX sudah bisa digantikan oleh software yang disebut Asterisk . Pada tugas Akhir ini dirancang dan direalisasikan suatu contact center system yang memiliki fitur call audio, call audio video, chat dan Frequently Asked Question yang bertujuan untuk memudahkan pengguna mendapatkan suatu informasi dalam berbagai cara. Fitur call baik audio maupun audio video menggunakan konsep VoIP dengan protokol SIP dengan memanfaatkan Asterisk server yang mempunyai fungsi-fungsi PBX. Dari hasil pengujian yang dilakukan, didapatkan hasil dari call audio yaitu one way delay dengan rata-rata 68.977 ms, jitter dengan rata-rata 13.753 ms, dan packet loss dengan rata-rata 4.8 %. Nilai MOS yang didapatkan dari panggilan audio adalah sebesar 4.328 dan untuk panggilan video adalah sebesar 2.04.Kualitas panggilan audio termasuk ke dalam kategori Bagus kerena memiliki nilai MOS 4.328 dari 5, sedangkan kualitas panggilan video bisa dikatakan tidak terlalu bagus karena memiliki nilai MOS 2.04 dari 5.KATA KUNCI: VoIP, Asterisk, Contact Center, SIPABSTRACT: The development of telecommunications technology rapidly making the need to communicate to get information over the phone higher. On the other hand, call center are typically used in the office or university to make it easier to get information, still using circuit based communications.Whereas, in the current world of global communication, the trend began to shift from circuit or PSTN network to the model communication via IP or commonly known as VoIP (Voice over IP). Popular communication in this era is web based communication, where the web began to be applied to be a call center which are the communications using IP, so the function of the call center that serves only call can transform into the contact center which has free calls, chat and FAQ. To support digital voice data lines operating in the internet protocol requires a hardware called IP PBX which is the function of the IP PBX can be replaced by software called Asterisk. The result of this final project is a contact center system that has features call audio, audio-video calls, chat and Frequently Asked Questions te goals to allow users to get the information in various ways. Call features both audio and audio-video using the concept of VoIP with SIP protocol by utilizing Asterisk servers that have a PBX functions. From the results of tests performed, the results obtained from the audio call is one way delay by an average of 68 977 ms, with an average jitter 13 753 ms, and packet loss by an average of 4.8%. MOS values obtained from an audio call is for 4328 and for video calls is at 2:04.The quality of audio calls can be categorized into“Good” because it has a MOS value 4.328 of 5, while the quality of video calls can be said not too good because it has MOS value 2.04 of 5.KEYWORD: VoIP, Asterisk, Contact Center, SI

ANALISIS IMPLEMENTASI INTERKONEKSI SIP SERVER DAN IP PBX PANASONIC DENGAN PENGUJIAN MENGGUNAKAN ENUM SERVER UNTUK LAYANAN VOIP

ABSTRAKSI: Perkembangan teknologi saat ini maju sangat pesat, tidak terlepas pula dalam teknologi jaringan komunikasi. Softswitch merupakan teknologi yang mampu mewujudkan NGN (Next Generation Network), dimana mampu menjembatani jaringan PSTN, PLMN, dan IP (Internet Protocol) dalam 1 infrastruktur yang dapat saling terhubung dengan berbagai layanan yang tersedia. Salah satu layanannya yang saat ini diminati dan menjadi sorotan adalah VoIP (Voice Over Internet Protocol). IP PBX merupakan sentral telepon digital yang berbasis pure IP dan dapat berkomunikasi dengan sentral telepon analog dan sentral IP yang memiliki fitur komunikasi yang cukup banyak. Asterisk (Trixbox) dan Elastix adalah suatu software implementasi SIP Server yang bersifat open source.Pada tugas akhir yang berjudul "Analisis Implementasi Interkoneksi SIP Server dan IP PBX Panasonic Pengujian dengan menggunakan Enum Server Untuk Layanan VoIP" ini akan diberikan satu cara untuk menginterkoneksikan SIP server yang terdiri dari Asterisk( Trixbox) dan Elastix dengan IP PBX Panasonic versi KX-TDE 200 agar client dari ketiga server tersebut dapat saling berkomunikasi dengan manajemen nomor menggunakan ENUM.Hasil pengukuran QoS menunjukkan bahwa 2 skenario interkoneksi tanpa melalui ENUM Server dan dengan melalui ENUM server yang masing-masingnya terdiri dari interkoneksi antara IP PBX Panasonic dengan Asterisk ( Trixbox), IP PBX Panasonic dengan Elastix, dan Asterisk (Trixbox) dengan Elastix yang dilakukan masih memenuhi standar ”baik”, dengan seluruh variasi background trafik yang diberikan, yaitu delay = Kata Kunci : Kata Kunci : NGN, PSTN, VoIP, Softswitch, IP PBX,Qos,PDD , SIPABSTRACT: The development of today\u27s advanced technology very rapidly , not apart in network communication technology . Softswitch is a technology that is capable of realizing the NGN ( Next Generation Network ) , which is able to bridge the PSTN network , PLMN , and IP ( Internet Protocol ) in 1 infrastructure that can connect with a variety of services available . One of the services that are currently in demand and the spotlight is VoIP ( Voice Over Internet Protocol ) . IP PBX is a telephone exchange pure IP -based digital and can communicate with the central telephone exchange analog and IP communications features that quite a lot . Asterisk ( Trixbox ) and Elastix is a software implementation of a SIP server that is open source .At the end of the task entitled " Analysis Implementation of Interconnection SIP Server and IP PBX Panasonic with Testing Using ENUM Server for VoIP Service " will be given a way to interconnect SIP server that consists of Asterisk ( Trixbox ) and the Elastix IP PBX Panasonic KX - TDE 200 version so that the client of the third the server can communicate with the management of a number using ENUM .QoS measurement results showed that the two scenarios without going through the interconnection and ENUM Server with ENUM server via each of which consists of the interconnection between the Panasonic IP PBX with Asterisk ( Trixbox ) , Panasonic IP PBX with Elastix and Asterisk ( Trixbox ) with Elastix done still meet the standard of " good " , with a whole variety of background traffic is given , ie delay = Keyword: Keywords : NGN , PSTN , VoIP , Softswitch , IP PBX , QoS , PD

Pemberdayaan Voip Di ICT Center SMK N 1 Klaten

Abstract – Protocol TCP / IP data network can dikoneksian in various computer world. This protocol is the existence and dibutukan so many people who develop them to lay sound (voice) through this protocol. VoIP Technology (Voice Over Internet Protocol) is an answer to that desire. This technology is able to convert analog sound (human voice) into data packets over a data network and then Internet or private intranet public data packet is passed, so komunikasipun can occur. With the VoIP communication costs can be reduced so as to reduce the cost of investment and conversation.VoIP implementations can be done by designing a wireless VoIP network using Asterisk as a PBX software. Asterisk VoIP server is an open source software in its application only requires a PC server and multiple client PCs connected to each other. Responding to the speed of technological advancement of VoIP technology is now applied in the ICT center both district and provincial levels. Abtraksi – Protokol TCP/IP dapat dikoneksian dalam jaringan data berbagai computer dunia. Protokol ini semakin eksis dan dibutukan sehingga banyak pihak yang mengembangkannya untuk menumpangkan suara (voice) melalui protocol ini. Teknologi VoIP (Voice Over Internet Protocol) merupakan jawaban atas keinginan itu. Teknologi ini mampu mengubah suara analog (suara manusia) menjadi paket data kemudian melalui jaringan data public internet maupun private intranet paket data dilewatkan, sehingga komunikasipun dapat terjadi. Dengan adanya VoIP biaya komunikasi dapat dikurangi sehingga dapat mereduksi biaya investasi dan percakapan.Implementasi VoIP dapat dilakukan dengan merancang suatu jaringan VoIP nirkabel dengan menggunakan software Asterisk sebagai PBX. Asterisk VoIP server ini merupakan suatu software open source yang dalam aplikasinya hanya membutuhkan satu PC server dan beberapa PC client yang terhubung satu sama lain. Menyikapi lajunya kemajuan teknologi ini sekarang teknologi VoIP diterapkan di ICT center baik tingkat Kabupaten maupun Provinsi

Decorating Asterisk : experiments in service creation for a multi-protocol telephony environment using open source tools

As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise