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Prijono, Wahyu Adi. "PENGARUH PENGGUNAAN CODEC STANDART ITU G.729 TERHADAP SISTEM KOMUNIKASI VOIP." SISTEM Jurnal Ilmu Ilmu Teknik 17, no. 1 (March 5, 2021): 11–22. http://dx.doi.org/10.37303/sistem.v17i1.193.
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Voice over Internet Protocol (VoIP) is a technology that is capable of passing voice traffic, in the form of packets through the network Internet Protocol (IP). IP network itself is a data communications network based packet-switch. The voice signal before experiencing bundled voice coding or format conversion of sound into digital form that can be passed over an IP network. Telephony, Internet telephony, or termed VoIP (Voice Over Internet Protocol.This communication system use VoIP (Voice over Internet Protocol), ie voice calls over data services (internet). This communication was developed using Android-based devices. Based on characteristics, android devices are open source, so users do not need to have a license to be able to have android-based devices. In addition, the android device that must be connected to a SIP (Session Iniation Protocol) is a data service that can be done with a paid subscription of the user of the operator using a conventional pulse. Telecommunications designed will use a hybrid system, the merger between VoIP communications with data communications GSM network. With basic calculations where Coding standards G 729, is a standard that can be used for voice communication system through data networks with rate of 8 Kbps. The implementation of the G729 codec is effect on communication systems VOIP.
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Pasarelski, Rosen, and Verginya Todorova. "Analysis of protocols and techniques for transmission of voice over internet protocol." Yearbook Telecommunications 6 (September 29, 2019): 105–13. http://dx.doi.org/10.33919/ytelecomm.19.6.11.
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The purpose of the article is to present the process of the evolution of telecommunication networks across the years and are developing at a very rapid pace, with a tendency towards convergence of services. Voice over IP is the preferred method for more and more telecommunications operators, replacing standard telephony and networking. As a result, it can be noted that VoIP technology allows much more information to be transmitted over the network to serve and improve communication needs than traditional telephony. The authors' contribution is the analysis of voice over IP protocols, which clarifies the concepts and rules in this type of communication and presents the functionalities and components of these communications.
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Shih, Ying-Wei, Ya-Ling Wu, Yi-Shun Wang, and Chiung-Liang Chen. "Investigating the post-adoption stage of Voice over Internet Protocol (VoIP) telephony diffusion." Information Technology & People 30, no. 4 (November 6, 2017): 753–84. http://dx.doi.org/10.1108/itp-02-2016-0032.
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Purpose The purpose of this paper is to investigate the post-adoption stage of Voice over Internet Protocol (VoIP) telephony diffusion, examining usage behavior based on Shih and Venkatesh’s use-diffusion (UD) model. Design/methodology/approach The research model incorporates technology sophistication, complementary technologies, personal innovativeness, self-efficacy, trust propensity, media exposure, subjective norms, and word-of-mouth (WOM) referrals as UD determinants; rate of use and variety of use as usage variables; intense use, specialized use, nonspecialized use, and limited use as UD patterns; and satisfaction and intention to use future-related technologies as UD outcomes. Data used to test the research model were collected using a web-based online questionnaire form; 360 valid responses were obtained. Partial least squares, multinomial logistic regression, and analysis of variance were used to analyze data. Findings The results reveal that variety of use, self-efficacy, propensity to trust, media exposure, subjective norms, and WOM referrals increase rate of use, while complementary technologies, personal innovativeness, self-efficacy, media exposure, and subjective norms widen variety of use; variety of use is essential in predicting UD outcomes; when choosing limited use as the reference category, more than half of the UD determinants are capable of predicting UD patterns; and generally, intense users are more satisfied with VoIP telephony, while limited users have less intention to use future-related technologies. Originality/value The present study focuses on the post-adoption stage, thereby extending the frontiers of research on the diffusion of VoIP telephony. Academics can obtain some evidence of the explanatory power of the UD model in the context of VoIP telephony use, and practitioners can obtain fresh insights into the dynamics of VoIP telephony usage behavior.
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Yulianto, Budi. "Analisis Korelasi Faktor Perilaku Konsumen terhadap Keputusan Penggunaan Teknologi Komunikasi Voip." ComTech: Computer, Mathematics and Engineering Applications 5, no. 1 (June 30, 2014): 236. http://dx.doi.org/10.21512/comtech.v5i1.2619.
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The advancement of communication technology that is combined with computer and the Internet brings Internet Telephony or VoIP (Voice over Internet Protocol). Through VoIP technology, the cost of telecommunications in particular for international direct dialing (IDD) can be reduced. This research analyzes the growth rate of VoIP users, the correlation of the consumer behavior towards using VoIP, and cost comparisons of using telecommunication services between VoIP and other operators. This research is using descriptive analysis method to describe researched object through sampling data collection for hypothesis testing. This research will lead to the conclusion that the use of VoIP for international area will be more advantageous than the use of other operators of GSM (Global System for Mobile), CDMA (Code Division Multiple Access), or the PSTN (Public Switched Telephone Network).
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J. N, Odii, Nwokoma F.O, Onwuama T.U, and Ejem A. "The Technologies of Voice over Internet Protocol (VoIP) Based Telephony System: A Review." International Journal of Computer Trends and Technology 49, no. 4 (July 25, 2017): 217–22. http://dx.doi.org/10.14445/22312803/ijctt-v49p135.
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Honni, Honni. "Rancang Bangun Perangkat Lunak Billing dan Implementasi Voice Over Internet Protocol." ComTech: Computer, Mathematics and Engineering Applications 4, no. 2 (December 1, 2013): 603. http://dx.doi.org/10.21512/comtech.v4i2.2483.
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The rapidly evolving communication system enables applications for telephone communication to be carried over the data network known as VoIP (voice over internet protocol). SIP (session initiation protocol) as the signaling protocol is text-based VoIP which can be implemented easily in comparison with other signalingprotocols. The purpose of this paper is designing and implementing VoIP billing up to the company to provide additional facilities for enterprise customers. The methods start with data collection, analysis, design, development, and implementation. The result achieved is a system of VoIP with SIP and Asterisk software which has functions of PBX to provide additional facilities such as VoIP which is a plus for the company and customers. After implemented, the VoIP system and billing features are found work well.
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Hoanca, Bogdan, and Richard Whitney. "Taking a Byte of Telephony Costs." Journal of Cases on Information Technology 12, no. 4 (October 2010): 18–34. http://dx.doi.org/10.4018/jcit.2010100102.
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In 2006, the University of Alaska Anchorage (UAA) upgraded the telephone system at its main campus in Anchorage from a traditional private branch exchange (PBX) architecture to a Voice over Internet Protocol (VoIP) system. This case describes the organizational decisions that led to the change; the scope and the process of upgrading; and the current status of the new VoIP system. The actual migration to VoIP was completed less than a year after the start of the project. The transition process went smoothly. User satisfaction with the performance of the VoIP system is very high. Based on extensive interviews with decision makers and the technical personnel involved, this case also describes financial considerations (including “creative” ways to stretch a limited budget), outsourcing considerations, training related issues, as well as lessons learned.
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Supendar, Hendra, Yopi Handrianto, and Santoso Setiawan. "Kualitas Pelayanan Dalam Voice Over Internet Protokol Berbasis Shorewall." PIKSEL : Penelitian Ilmu Komputer Sistem Embedded and Logic 7, no. 2 (September 23, 2019): 123–32. http://dx.doi.org/10.33558/piksel.v7i2.1815.
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Abstract
Communication is very important and a success factor of the system at the company. Communication can be done using telephone media and internet media. PT. Interdev Prakarsa has several branches and communication between branches is still done using telephone media, and it is very costly for the company especially since the company is already using internet media in the network. The solution to this problem in this study was designing an internet-based technology as communication between branches, the technology is called Voice over Internet Protocol (VoIP) with an open source operating system and an open source firewall that was Shorewall. Result showed that after installation and testing, this firewall reliable enough to overcome the problem of attack problems from outside. The results of VoIP tightening on service quality found that the application of VoIP did not consume a lot of CPU Benchmarks, i.e. only 0.80 to 1.35 per cent. The bandwidth used is also very small between 86 to 86.8 kbps for incoming calls and 83.4 up to 84.3 kbps for outgoing calls. Communication built between VoIP peripherals has also been tested to run well because the value of delay, jitter and packet loss is included in the good category.
Keywords: VoIP, Shorewall, Comunication
Abstrak
Komunikasi pada sebuah perusahaan sangatlah penting, dimana komunikasi menjadi alat ukur keberhasilan sistem di perusahaan tersebut. Komunikasi yang dilakukan dapat menggunakan media telepon dan media internet. PT. Interdev Prakarsa memiliki beberapa cabang dan komunikasi antar cabang masih dilakukan dengan menggunakan media telepon, dan itu sangatlah menghabiskan cost perusahaan apalagi perusahaan tersebut sudah menggunakan media internet dalam jaringan. Solusi dari permasalahan tersebut adalah dengan mendesain sebuah teknologi yang berbasis jaringan internet sebagai komunikasi antar cabang, teknologi tersebut bernama Voice over Internet Protocol (VoIP) dengan sebuah sistem operasi open source dan sebuah firewall open source yaitu Shorewall dimana setelah dilakukan penginstallan dan pengetesan, firewall ini cukup handal untuk mengatasi masalah masalah serangan dari luar. Hasil dari pengetasan VoIP terhadap kualitas pelayanan didapatkan bahwa penerapan VoIP tidak banyak menghabiskan Benchmark CPU yaitu hanya 0,80 sampai degan 1.35 persen per call. Untuk bandwidth yang di pakai juga sangatlah kecil berada di antara 86 sampai dengan 86.8 kbps untuk panggilan masuk dan 83.4 sampai dengan 84.3 kbps untuk panggilan keluar. Komunikasi yang dibangun antar peripheral VoIP pun teruji berjalan dengan baik karena nilai dari delay, jitter dan packet loss termasuk dalam katagori baik.
Kata kunci: VoIP, Shorewall, Komunikasi
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Nouvatie, Alwalid, Martono Dwi Atmadja, and Waluyo Waluyo. "Implementasi Trunk Interkoneksi Multi Server menggunakan Singleboard Komputer." Jurnal Jartel: Jurnal Jaringan Telekomunikasi 11, no. 2 (June 28, 2021): 96–100. http://dx.doi.org/10.33795/jartel.v11i2.75.
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Voice over Internet Protocol (juga disebut VoIP, IP Telephony, Internet telephony atau Digital Phone) adalah teknologi yang memungkinkan percakapan suara jarak jauh melalui media internet. Data suara diubah menjadi kode digital dan dialirkan melalui jaringan yang mengirimkan paket-paket data, dan bukan lewat sirkuit analog telepon biasa. Dalam komunikasi VoIP, pemakai melakukan hubungan telepon melalui terminal yang berupa PC atau telepon biasa. Dengan bertelepon menggunakan VoIP, banyak keuntungan yang dapat diambil diantaranya adalah dari segi biaya jelas lebih murah dari tarif telepon tradisional, karena jaringan IP bersifat global. IP Phone dapat di tambah, dipindah dan di ubah. Hal ini karena VoIP dapat dipasang di sembarang ethernet dan IP address, tidak seperti telepon konvensional yang harus mempunyai port tersendiri di Sentral atau PBX (Private branch exchange). Dalam penelitian ini mengimplementasikan keterhubungan antar server menggunakan single board computer yang di install sistem operasi Elastix yang bertujuan untuk mengimplementasikan prefix untuk antar server dan menggunakan beberapa codec audio. Hasil penelitian telepon menggunakan prefix dan tanpa prefix sebanyak 6 client atau 3 pasng panggilan secara bersamaan nilai packet loss tertinggi pada codec speex dengan prefix sebesar 2,34%. Nilai bandwidth tertinggi yang digunakan adalah dengan prefix codec PCMU dengan rata-rata 82,3 Kbps dan tanpa prefix 79,3 Kbps. Kata kunci : Server, VoIP, IP Telphony, Internet telephony, Digital Phone, IP Address, PBX, Codec, Prefix.
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Khalifa, Othman O., Raihan Jannati Binti Roslin, and Sharif Shah Newaj Bhuiyan. "Improved voice quality with the combination of transport layer & audio codec for wireless devices." Bulletin of Electrical Engineering and Informatics 8, no. 2 (June 1, 2019): 665–73. http://dx.doi.org/10.11591/eei.v8i2.1490.
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Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec signaling protocol produced the quality speech.
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Nazih, Waleed, Yasser Hifny, Wail Elkilani, Tamer Abdelkader, and Hossam Faheem. "Efficient Detection of Attacks in SIP Based VoIP Networks Using Linear l1-SVM Classifier." International Journal of Computers Communications & Control 14, no. 4 (August 5, 2019): 518–29. http://dx.doi.org/10.15837/ijccc.2019.4.3563.
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The Session Initiation Protocol (SIP) is one of the most common protocols that are used for signaling function in Voice over IP (VoIP) networks. The SIP protocol is very popular because of its flexibility, simplicity, and easy implementation, so it is a target of many attacks. In this paper, we propose a new system to detect the Denial of Service (DoS) attacks (i.e. malformed message and invite flooding) and Spam over Internet Telephony (SPIT) attack in the SIP based VoIP networks using a linear Support Vector Machine with l1 regularization (i.e. l1-SVM) classifier. In our approach, we project the SIP messages into a very high dimensional space using string based n-gram features. Hence, a linear classifier is trained on the top of these features. Our experimental results show that the proposed system detects malformed message, invite flooding, and SPIT attacks with a high accuracy. In addition, the proposed system outperformed other systems significantly in the detection speed.
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Jebrane, Asma, Ahmed Toumanari, Naîma Meddah, and Mohamed Bousseta. "A New Efficient Authenticated and Key Agreement Scheme for SIP Using Digital Signature Algorithm on Elliptic Curves." Journal of Telecommunications and Information Technology, no. 2 (June 30, 2017): 62–68. http://dx.doi.org/10.26636/jtit.2017.109516.
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Voice over Internet Protocol (VoIP) has been recently one of the more popular applications in Internet technology. It benefits lower cost of equipment, operation, and better integration with data applications than voice communications over telephone networks. However, the voice packets delivered over the Internet are not protected. The session initiation protocol (SIP) is widely used signaling protocol that controls communications on the Internet, typically using hypertext transport protocol (HTTP) digest authentication, which is vulnerable to many forms of attacks. This paper proposes a new secure authentication and key agreement scheme based on Digital Signature Algorithm (DSA) and Elliptic Curve Cryptography (ECC) named (ECDSA). Security analysis demonstrates that the proposed scheme can resist various attacks and it can be applied to authenticate the users with different SIP domains.
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Aditya, Yudha, Adian Fatchur Rochim, and Eko Didik Widianto. "Rancang Bangun Sistem Telekomunikasi Konvergen Berbasis Voice over Internet Protocol Menggunakan Virtualbox." Jurnal Teknologi dan Sistem Komputer 3, no. 2 (April 20, 2015): 282. http://dx.doi.org/10.14710/jtsiskom.3.2.2015.282-294.
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Perkembangan teknologi yang sangat pesat, membuat teknologi telekomunikasi semakin berkembang. Voice over Internet Protocol (VoIP), Public Switched Telephone Network (PSTN), Global System for Mobiles (GSM) dan internet adalah teknologi terkini dalam memenuhi kebutuhan seseorang dalam berkomunikasi. Di sebagian besar implementasinya, penyedia layanan PSTN dan GSM memberikan sebuah tarif dalam setiap panggilan yang terjadi. Perancangan dan pembangunan sistem ini bertujuan untuk menciptakan sebuah sistem telekomunikasi berbasis VoIP, yang dapat menghubungkan jaringan lokal, GSM dan internet secara terpusat, demi memenuhi kebutuhan komunikasi seseorang dengan mobilitas tinggi disertai fleksibilitas pengaturan alur panggilan, untuk menghemat anggaran penggunaan layanan telekomunikasi. Metodologi penelitian tugas akhir ini dibagi menjadi 4 tahapan. Tahapan tersebut diantaranya adalah definisi sistem, spesifikasi kebutuhan, konfigurasi sistem dan pengujian sistem. Definisi sistem dibuat dengan mendefinisikan gambaran dan cara kerja sistem secara umum. Spesifikasi kebutuhan dibuat dengan menyesuaikan kebutuhan perangkat keras dan perangkat lunak yang dibutuhkan oleh sistem. Konfigurasi dilakukan untuk mengimplementasikan pengaturan mengenai dialplan, Interactive Voice Response (IVR) dan kotak suara. Pengujian sistem adalah tahap untuk memeriksa keseluruhan fungsi pada sistem. Sistem ini diuji dengan melakukan panggilan dari setiap klien. Hasil dari pengujian menunjukkan bahwa sistem mampu memenuhi kebutuhan komunikasi seseorang dengan mobilitas tinggi. Fleksibilitas pengaturan panggilan membuat sistem dapat berkomunikasi dengan jaringan GSM dan VoIP Rakyat serta dapat menghemat tarif penggunaan layanan telekomunikasi. Sistem juga dapat mencatat aktivitas panggilan dengan memanfaatkan fitur Call Detail Record (CDR). Penelitian ini dapat dijadikan sebagai alternatif bagi perkantoran maupun instansi, untuk menggunakan layanan telekomunikasi secara terpusat, agar penghematan anggaran dalam penggunaan telekomunikasi menjadi lebih efisien.
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Tanoyo, Suryo Aji, Eva Yovita Dwi Utami, and Eva Yovita Dwi Utami. "Unjuk Kerja QoS (Quality of Services) Jaringan Voice over Internet Protocol Berbasis SIP yang Diimplementasikan pada Jaringan Ethernet Gedung FEB-UKSW." Techné : Jurnal Ilmiah Elektroteknika 15, no. 01 (April 1, 2016): 17–26. http://dx.doi.org/10.31358/techne.v15i01.137.
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Jaringan komputer yang diimplementasikan di dalam suatu perkantoran yang lebih banyak dimanfaatkan untuk layanan data dapat dioptimalkan dengan penambahan layanan voice berbasis IP. Voice over Internet Protocol (VoIP) menghemat resource jaringan dibandingkan dengan PSTN (Public Switched telephone Network). Namun demikian implementasi VoIP harus memperhatikan kualitas layanan atau Qualitiy of Service (QoS). Parameter kualitas layanan VoIP antara lain throughput, delay, jitter, dan packet loss. Teknologi VoIP telah dikembangkan dengan menciptakan berbagai macam protocol seperti SIP, H.323, MGCP dan codec seperti G.711, G.723.1, G.726, G.728, G.729 dengantujuan untuk memperbaiki kualitas layanan VoIP. Penelitian ini bertujuan menganalisis kinerja QoS dengan membandingkan variasi codec G.711, G.723.1 dan G.726 pada sebuah rancangan jaringan VoIP berbasis SIP di gedung FEB-UKSW, dengan parameter QoS adalah Throughput, delay, packet loss, jitter. Komunikasi VoIP yang dilakukan terdiri atas komunikasi internal dan komunikasi eksternal. Komunikasi internal mencakup simulasi komunikasi hardphone ke PC. Komunikasi eksternal mencakup simulasi hardphone ke PC eksternal. Dari hasil penelitian, secara umum didapatkan bahwa codec G.711 memiliki kualitas paling baik untuk simulasi komunikasi internal ataupun eksternal dengan menghasilkan rata-rata delay, jitter, packet loss paling rendah.
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Alfaisaly, Noor Nateq, Suhad Qasim Naeem, and Azhar Hussein Neama. "Enhancement of WiMAX networks using OPNET modeler platform." Indonesian Journal of Electrical Engineering and Computer Science 23, no. 3 (September 1, 2021): 1510. http://dx.doi.org/10.11591/ijeecs.v23.i3.pp1510-1519.
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Worldwide interoperability microwave access (WiMAX) is an 802.16 wireless standard that delivers high speed, provides a data rate of 100 Mbps and a coverage area of 50 km. Voice over internet protocol (VoIP) is flexible and offers low-cost telephony for clients over IP. However, there are still many challenges that must be addressed to provide a stable and good quality voice connection over the internet. The performance of various parameters such as multipath channel model and bandwidth over the Star trajectoryWiMAX network were evaluated under a scenario consisting of four cells. Each cell contains one mobile and one base station. Network performance metrics such as throughput and MOS were used to evaluate the best performance of VoIP codecs. Performance was analyzed via OPNET program14.5. The result use of multipath channel model (disable) was better than using the model (ITU pedestrian A). The value of the throughput at 15 dB was approximately 1600 packet/sec, and at -1 dB was its value 1300 packet/se. According to data, the Multipath channel model of the disable type the value of the MOS was better than the ITU Pedestrian A type.
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Luhach, Ravindra, Chandra K. Jha, and Ashish K. Luhach. "Performance Analysis of QMF Filter Bank For Wireless Voip in Pervasive Environment." Recent Patents on Computer Science 12, no. 4 (August 19, 2019): 349–53. http://dx.doi.org/10.2174/2213275911666181018101737.
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Background: Voice over Internet Protocol (VoIP) has emerged as one of the most significant technology in the field of communication and evolved as a substitute to the conventional communication method as the Public Switched Telephone Network (PSTN). Along with the advantages such as scalability and security, VoIP has some threats such as voice quality and interference that must be dealt with. The voice quality in VoIP is degraded when transmitted over a computer network due to delay, jitter and packet loss etc. Packet loss is one of major reasons for the signal quality degradation. Objective: In this research article, Quadrature Mirror Filter Bank (QMF) has been implemented in wireless VoIP system to enhance the quality of the signals transmitted. Results: The performance has been evaluated under varying network conditions of packet loss. Conclusion: Significant improvement has been observed in the quality of VoIP signal.
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Cortés-Mendoza, Jorge M., Andrei Tchernykh, Fermin A. Armenta-Cano, Pascal Bouvry, Alexander Yu Drozdov, and Loic Didelot. "Biobjective VoIP Service Management in Cloud Infrastructure." Scientific Programming 2016 (2016): 1–14. http://dx.doi.org/10.1155/2016/5706790.
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Voice over Internet Protocol (VoIP) allows communication of voice and/or data over the internet in less expensive and reliable manner than traditional ISDN systems. This solution typically allows flexible interconnection between organization and companies on any domains. Cloud VoIP solutions can offer even cheaper and scalable service when virtualized telephone infrastructure is used in the most efficient way. Scheduling and load balancing algorithms are fundamental parts of this approach. Unfortunately, VoIP scheduling techniques do not take into account uncertainty in dynamic and unpredictable cloud environments. In this paper, we formulate the problem of scheduling of VoIP services in distributed cloud environments and propose a new model for biobjective optimization. We consider the special case of the on-line nonclairvoyant dynamic bin-packing problem and discuss solutions for provider cost and quality of service optimization. We propose twenty call allocation strategies and evaluate their performance by comprehensive simulation analysis on real workload considering six months of the MIXvoip company service.
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Atmadja, Martono Dwi. "Single Board Computer Applications as Multi-Server VoIP." International Journal for Research in Applied Science and Engineering Technology 9, no. VII (July 15, 2021): 1023–28. http://dx.doi.org/10.22214/ijraset.2021.36512.
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Telecommunication technology is developing along with information technology and several innovations in several audio and data transmission and reception techniques. Innovation and communication technology are hoped to be able to create efficiencies in regards to time, equipment, and cost. The Public Switched Telephone Network (PSTN) telephone technology has experienced integration towards communication using Internet Protocol (IP) networks, better known as Voice over Internet Protocol (VoIP). VoIP Technology transmits conversations digitally through IP-based networks, such as internet networks, Wide Area Networks (WAN), and Local Area Networks (LAN). However, the VoIP cannot fully replace PSTN due to several weaknesses, such as delay, jitter, packet loss, as well as security and echo. Telephones calls using VoIP technology are executed using terminals in the form of computer devices or existing analogue telephones. The benefit of VoIP is that it can be set in all ethernet and IP addresses. Prefixes can be applied for inter-server placements as inter-building telephone networks without the addition of inefficient new cables on single board computers with Elastix installed. Prefix and non-prefix analysis on servers from single board computers can be tested using QoS for bandwidth, jitter, and packet loss codec. The installation of 6 clients, or 3 simultaneous calls resulted in a packet loss value in the prefix Speex codex of 2.34%. The bandwidth in the prefix PCMU codec has an average value of 82.3Kbps, and a non-prefix value of 79.3Kbps, in accordance to the codec standards in the VoIP. The lowest jitter was found in the non-prefix PCMU codec with an average of 51.05ms, with the highest jitter for the prefix Speex codec being 314.65ms.
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Sudiarta, Pande Ketut, and I. Putu Ardana. "Implementation of Hotspot Network for Internal Campus Communications Utilizing Smartphone and Free Software." Journal of Electrical, Electronics and Informatics 1, no. 1 (February 3, 2017): 33. http://dx.doi.org/10.24843/jeei.2017.v01.i01.p07.
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VoIP (voice over internet protocol) telephone communication using a data network. There is a change in switching from circuit switching technology into packet switching. Phone exchange can now use the Personal Computer equipped VoIP applications. Even the development of mobile phone technology to make VoIP communication can be performed utilizing the Smartphone. VoIP applications commonly use such as WhatsApp, Line etc. However, this application will cut the user data packets and often the quality is not satisfactory due to limited bandwidth and location of the remote server. During this time, Udayana University campus at several locations has been equipped with Hotspot network is mostly used to connect to the Internet. Hotspot network can also be used for voice communications with VoIP technology by adding a VoIP server. This concept not raises communication costs and should produce sound quality will be better because of the close location of the server. Because that researchers need to develop a model of telephone communication network utilizing Smartphone hotspot and students to be able to communicate in a campus environment. Method of this research is to develop a network utilizing hotspots and VoIP telephone exchange using the mini PC installed software FreePBX. On the side of the Smartphone using the free soft phone application and for aircraft used FXS analog phone as a codec. Tests will be performed for the communication between Smartphone and Smartphone to FXS terms of QOS and MOS produced. The results obtained, if the latency and packet loss have a value corresponding to the Real Time Protocol (RTP), the obtained MOS appropriate for the codec used while with the same codec if the value of packet loss and latency results are high then MOS obtained be small or less good quality. So the quality of VoIP is highly dependent on the quality of the signal obtained hotspot. In general, VoIP communication using a Smartphone connected to the server on the network hotspot mini pc can be used as a voice communication on campus.
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ARYANTA, DWI, ARSYAD RAMADHAN DARLIS, and ARDHIANSYAH PRATAMA. "Implementasi Sistem IP PBX menggunakan Briker." ELKOMIKA: Jurnal Teknik Energi Elektrik, Teknik Telekomunikasi, & Teknik Elektronika 1, no. 2 (July 1, 2013): 117. http://dx.doi.org/10.26760/elkomika.v1i2.117.
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ABSTRAKVoIP (Voice over Internet Protocol) adalah komunikasi suara jarak jauh yang digunakan melalui jaringan IP. Pada penelitian ini dirancang sistem IP PBX dengan menggunakan teknologi berbasis VoIP. IP PBX adalah perangkat switching komunikasi telepon dan data berbasis teknologi Internet Protocol (IP) yang mengendalikan ekstension telepon analog maupun ekstension IP Phone. Software VirtualBox digunakan dengan tujuan agar lebih memudahkan dalam sistem pengoperasian Linux yang dimana program untuk membuat IP PBX adalah menggunakan Briker yang bekerja pada Operating System Linux 2.6. Setelah proses penginstalan Briker pada Virtualbox dilakukan implementasi jaringan IP PBX. Setelah mengimplementasikan jaringan IP PBX sesuai dengan topologi, kemudian melakukan pengujian success call rate dan analisis Quality of Service (QoS). Pengukuran QoS menggunakan parameter jitter, delay, dan packet loss yang dihasilkan dalam sistem IP PBX ini. Nilai jitter sesama user Briker (baik pada smartphone maupun komputer) mempunyai rata-rata berada pada nilai 16,77 ms. Sedangkan nilai packetloss yang didapat pada saat terdapat pada saat user 1 sebagai pemanggil telepon adalah 0%. Sedangkan persentase packet loss pada saat user 1 sebagai penerima telepon adalah 0,01%. Nilai delay pada saat berkomunikasi antar user berada pada 11,75 ms. Secara keseluruhan nilai yang didapatkan melalui penelitian ini, dimana hasil pengujian parameter-parameter QOS sesuai dengan standar yang telah direkomendasikan oleh ITU dan didapatkan nilai QoS dengan hasil “baik”.Kata Kunci: Briker, VoIP, QoS, IP PBX, Smartphone.ABSTRACTVoIP (Voice over Internet Protocol) is a long-distance voice communications over IP networks are used. In this study, IP PBX systems designed using VoIP -based technologies. IP PBX is a telephone switching device and data communication technology-based Internet Protocol (IP) which controls the analog phone extensions and IP Phone extensions. VirtualBox software is used in order to make it easier for the Linux operating system to create a program which is using briker IP PBX that works on Linux 2.6 Operating System. After the installation process is done briker on Virtualbox IP PBX network implementation. After implementing the IP PBX network according to the topology, and then do a test call success rate and analysis of Quality of Service (QoS). Measurement of QoS parameters using jitter, delay, and packet loss resulting in the IP PBX system. Jitter value briker fellow users (either on a smartphone or computer) has been on the average value of 16.77 ms. While the values obtained packetloss when there is 1 user when a phone caller is 0%. While the percentage of packet loss at user 1 as a telephone receiver is 0.01%. Delay value when communicating between users located at 11.75 ms. Overall value obtained through this study , where the results of testing the QOS parameters in accordance with the standards recommended by the ITU and the QoS values obtained with the results "good".Keywords: Briker, VoIP, QoS, IP PBX, Smartphone.
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AlShemmary, Ebtesam Najim, and Bahaa Qasim Al-Musawi. "Low Cost VoIP Architecture Using Open Source Software Component in Tertiary Institutions." INTERNATIONAL JOURNAL OF COMPUTERS & TECHNOLOGY 3, no. 1 (August 1, 2012): 11–14. http://dx.doi.org/10.24297/ijct.v3i1a.2721.
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Governments and their agencies are often challenged by high cost and flexible telephonic, Web based data services. Emerging technologies, such as those of Voice over Internet Protocol (VoIP) that allow convergent systems where voice and Web technologies can utilize the same network to provide both services, can be used to improve such services. This paper describe VoIP system for the enterprise network (e.g. company, university) that have been developed based on Asterisk which is a kind of open source software to implement IP-PBX system. Through the development and evaluation, we have confirmed that VoIP system based on Asterisk is very powerful as a whole and most PBX functions to be required for the enterprise network can be realized. Interesting findings include that the University of Kufa has a potential to implement the project. By connecting multiple Asterisk servers located in different sites based on IAX2, large scale enterprise network can be developed. Since the software recommended for installation is open source, the project could be used as a source of valuable information by students who specialize in real-time multi-media systems.
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Muhammad, Shamsuddeen Hassan, and Abdulrasheed Mustapha. "A Form of List Viterbi Algorithm for Decoding Convolutional Codes." U.Porto Journal of Engineering 4, no. 2 (October 31, 2018): 42–48. http://dx.doi.org/10.24840/2183-6493_004.002_0004.
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Viterbi algorithm is a maximum likelihood decoding algorithm. It is used to decode convolutional code in several wireless communication systems, including Wi-Fi. The standard Viterbi algorithm gives just one decoded output, which may be correct or incorrect. Incorrect packets are normally discarded thereby necessitating retransmission and hence resulting in considerable energy loss and delay. Some real-time applications such as Voice over Internet Protocol (VoIP) telephony do not tolerate excessive delay. This makes the conventional Viterbi decoding strategy sub-optimal. In this regard, a modified approach, which involves a form of List Viterbi for decoding the convolutional code is investigated. The technique employed combines the bit-error correction capabilities of both the Viterbi algorithm and the Cyclic Redundancy Check (CRC) procedures. It first uses a form of ‘List Viterbi Algorithm’ (LVA), which generates a list of possible decoded output candidates after the trellis search. The CRC check is then used to determine the presence of correct outcome. Results of experiments conducted using simulation shows considerable improvement in bit-error performance when compared to classical approach.
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Whelan, Lynsay R., and Nathan Wagner. "Technology that Touches Lives: Teleconsultation to Benefit Persons with Upper Limb Loss." International Journal of Telerehabilitation 3, no. 2 (December 20, 2011): 19–22. http://dx.doi.org/10.5195/ijt.2011.6080.
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While over 1.5 million individuals are living with limb loss in the United States (Ziegler-Graham et al., 2008), only 10% of these individuals have a loss that affects an upper limb. Coincident with the relatively low incidence of upper limb loss, is a shortage of the community-based prosthetic rehabilitation experts that can help prosthetic users to more fully integrate their devices into their daily routines. This article describes how expert prosthetists and occupational therapists at Touch Bionics, a manufacturer of advanced upper limb prosthetic devices, employ Voice over the Internet Protocol (VoIP) videoconferencing software telehealth technologies to engage in remote consultation with users of prosthetic devices and/or their local practitioners. The Touch Bionics staff provide follow-up expertise to local prosthetists, occupational therapists, and other health professionals. Contrasted with prior telephone-based consultations, the video-enabled approach provides enhanced capabilities to benefit persons with upper limb loss. Currently, the opportunities for Touch Bionics occupational therapists to fully engage in patient-based services delivered through telehealth technologies are significantly reduced by their need to obtain and maintain professional licenses in multiple states.
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Goode, B. "Voice over Internet protocol (VoIP)." Proceedings of the IEEE 90, no. 9 (September 2002): 1495–517. http://dx.doi.org/10.1109/jproc.2002.802005.
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Svrzić, Slađan, Zoran Miličević, and Zoran Perišić. "Description of the process of tunneling Q signaling in private telecommunication networks." Vojnotehnicki glasnik 69, no. 1 (2021): 31–63. http://dx.doi.org/10.5937/vojtehg69-28117.
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Introduction/purpose: The article should specify the network signaling type Q-SIG, which is standardized especially for implementation in digital telecommunication networks of integrated services (ISDN), emphasizing the possibility of its further application in the Private Telecommunications Network of Integrated Services of the Serbian Armed Forces (PISN of SAF), i.e. in the Private Automatic Telephone Network of the Serbian Armed Forces (PATN of SAF). Methods: An analysis of the existing standards was performed: ECMA355 and ECMA-336 and a synthesis of the possibilities of their application in the PATN of SAF. Results: The procedure for the application of Q-SIG is processed in a situation when the peripheral parts of the PISN of SAF, which operate on the principle of transmission and circuit switching by TDM (Time Division Multiplexing), are connected via a central Core network with the IP (Internet Protocol), which operates on the principle of packet transmission and switching with the SIP (Session Initiation Protocol). A method of the application of the tunneling of encapsulated Q-SIG messages through the IP network, defined by ECMA-355 Standard, has been developed. The necessary functions for mapping the transmission of tunneled signaling messages Q-SIG and mapping voice (and other audio) information to media streams during VoIP (Voice over IP) communication through that network, which are defined by ECMA-336 Standard, are described. Conclusion: The application of ECMA-355 and ECMA-336 Standards is a new solution in the PATN of SAF with the use of the IP network to connect the IP PINX using the Q-SIG tunneling procedures and mapping functions for their transmission and transmission of audio signals. This then opens up a whole range of new possibilities that, with the growth of the Core network and their application, will rapidly contribute to the creation of a broad Telecommunication information system backbone for the implementation of real-time multimedia communications and the transition to Unified Communications (UC).
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Soomro, Tariq Rahim, and Dawar Asfandyar. "Voice over Internet Protocol (VoIP): UAE Perspective." Asian Journal of Information Technology 9, no. 3 (March 1, 2010): 170–78. http://dx.doi.org/10.3923/ajit.2010.170.178.
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Daramola, Oladunni Abosede. "QUALITY OF SERVICE ISSUES IN WIRELESS VOICE OVER INTERNET PROTOCOL." International Journal of Advanced Research in Computer Science and Software Engineering 7, no. 10 (October 30, 2017): 57. http://dx.doi.org/10.23956/ijarcsse.v7i10.386.
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Voice over Internet Protocol (VoIP) is a significant application of the converged network principle where the voice traffic is routed over Internet Protocol shared traffic networks. VoIP traffic was modelled over wireless network and a simulation of the traffic was transmitted over the network. E-model technique was used to analyze the traffic data and also to rate VoIP QoS parameters. The result achieved was mapped to the Mean Opinion Scale to determine the Quality of Service of VoIP over wireless networks. The results shows that QoS in the VoIP communications is significantly impacted by these parameters and the impact varies according to the parameters and also the communication aspects selected for the VoIP traffic analysis.Keywords: VoIP, QoS, E-Model and Mean Opinion Scale
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Windiarto, Ardi, and Kholilatul Wardani. "Rancang Bangun Voice Over Internet Protocol dan GSM Gateway Berbasis Raspberry Pi." TELKA - Telekomunikasi, Elektronika, Komputasi dan Kontrol 5, no. 1 (May 21, 2019): 55–64. http://dx.doi.org/10.15575/telka.v5n1.55-64.
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Makalah ini membahas desain layanan jaringan komunikasi VoIP Server menggunakan Raspberry Pi sebagai alat komunikasi wireless. VoIP server berbasis Raspberry Pi menggunakan sistem operasi RasPBX. Di dalam sistem operasi RasPBX sudah ada software asterisk yang berfungsi sebagai softswicth. Client VoIP menggunakan zoiper sebagai softphone. Alat ini dilengkapi dengan fitur GSM gateway yaitu fitur yang dapat menghubungkan jaringan VoIP ke jaringan GSM. Fitur GSM gateway ini menggunakan modem GSM sebagai jembatan yang menghubungkan jaringan VoIP dengan jaringan GSM. Persentase keberhasilan panggilan VoIP ke VoIP, VoIP ke GSM, dan GSM ke VoIP mencapai 100%. Berdasarkan hasil pengujian Quality of services (QoS) pada panggilan VoIP ke GSM, dihasilkan rata-rata delay sebesar 12,11 ms yang termasuk dalam kategori kualitas baik, Troughput sebesar 0,151, jitter sebesar 0,052 ms yang termasuk dalam kategori kualitas baik, dan packet loss sebesar 0% yang termasuk dalam kategori kualitas sangat baik. Jangkauan maksimal antara client VoIP ke server agar komunikasi berjalan dengan baik adalah 100 meter dalam kondisi Line Of Sight (LOS). Pengujian dengan jarak 25 m dalam kondisi Non Line Of Sight (NLOS), masih menghasilkan komunikasi yang baik. Berdasarkan hasil pengujian kuisioner dari 30 pengguna, dihasilkan nilai MOS 3,88 yang termasuk dalam kategori kualitas cukup baik.
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Handayani, Rini. "Voice over Internet Protocol (VOIP) Pada Jaringan Nirkabel Berbasis Raspberry Pi." KINETIK 2, no. 2 (May 24, 2017): 82. http://dx.doi.org/10.22219/kinetik.v2i2.146.
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Voice Over Internet Protocol (VoIP) merupakan satu teknologi telekomunikasi yang mampu melewatkan layanan komunikasi dalam jaringan Internet Protocol sehingga memungkinkan antar pengguna berkomunikasi suara dalam jaringan IP. Kelebihan dari VoIP ini mampu melakukan efisiensi bandwith dan biaya pengelolaan dengan memanfaatkan Raspberry Pi sebagai server VoIP. Dalam penelitian ini, VoIP dibangun pada Sistem Operasi Linux dengan aplikasi Asterisk dan RasPBX yang diintegrasikan pada Raspberry Pi dengan menggunakan jaringan nirkabel lokal sebagai media transmisi. Sistem ini diujikan dengan menggunakan dua tipe client, yaitu PC dan smartphone dengan mengukur QoS dengan rata-rata delay 0.4463ms, rata-rata throughput 16.36KBps, rata-rata packet loss 0.889% dan jitter 1.102ms.
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Whitlock, Mike. "Voice Over Internet Protocol (VoIP) And One University Application." Review of Business Information Systems (RBIS) 9, no. 4 (October 1, 2005): 1–6. http://dx.doi.org/10.19030/rbis.v9i4.4438.
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Jalendry, Sheetal, and Shradha Verma. "A Detail Review on Voice over Internet Protocol (VoIP)." International Journal of Engineering Trends and Technology 23, no. 4 (May 25, 2015): 161–66. http://dx.doi.org/10.14445/22315381/ijett-v23p232.
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Shaw, Urjashee, and Bobby Sharma. "A Survey Paper on Voice over Internet Protocol (VOIP)." International Journal of Computer Applications 139, no. 2 (April 15, 2016): 16–22. http://dx.doi.org/10.5120/ijca2016909112.
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Huda, Yasdinul, Muhammad Adri, and Yoharmen Arnov. "PERANCANGAN APLIKASI CLIENT UNTUK JARINGAN VOIP (VOICE OVER INTERNET PROTOCOL) BERBASIS ANDROID." Jurnal Teknologi Informasi dan Pendidikan 11, no. 1 (April 30, 2018): 81–93. http://dx.doi.org/10.24036/tip.v11i1.99.
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Komunikasi adalah suatu keharusan dalam operasi suatu agen atau perusahaan. Komunikasi murah menggunakan VOIP akan menyediakan biaya agensi atau perusahaan. Dalam komunikasi diperlukan suatu sistem jaringan yang handal dengan konsep client untuk pengguna suatu sistem dan server sebagai pengelola jaringan yang digunakan oleh klien. VOIP memerlukan server menggunakan PC dengan Ubuntu Linux 12.04 LTE atau Linphone sebagai server VOIP (Voice Over Internet Protocol) dan tanda bintang untuk mengatur lalu lintas suara. perangkat lunak android di sisi klien untuk menjadi antarmuka dalam berkomunikasi dengan klien lain yang dirancang menggunakan Android Studio dengan smartphone android setidaknya 4.4.2 (kitkat). Dan membutuhkan koneksi internet melalui WIFI_UNP sebagai media transmisi (tautan) antara server dan klien. WIFI_UNP adalah media internet yang digunakan oleh mahasiswa, dosen, staf dan karyawan Universitas Negeri Padang untuk mendukung proses pembelajaran dan menambah wawasan tentang teknologi atau sains terbaru. Sedangkan tethering hotspot adalah media internet yang berasal dari koneksi internet dari Smartphone yang dibagikan atau didistribusikan ke pengguna lain untuk internet. Aplikasi VOIP Client ini menyediakan dua fungsi utama: Memanggil dan Mengobrol. On Calling akan diberikan izin untuk melakukan panggilan suara dengan memasukkan nama pengguna klien untuk dihubungi atau nomor VOIP yang telah didaftarkan. Sedangkan Obrolan adalah fitur untuk melakukan obrolan teks secara realtime (sesuai waktu yang sebenarnya)
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Riki, Riki, Aditiya Hermawan, and Yusuf Kurnia. "Voice Over Internet Protocol Based Communication Design (VoIP) With 3CXSystemPhone On Android Smartphone." bit-Tech 1, no. 1 (August 30, 2018): 1–8. http://dx.doi.org/10.32877/bt.v1i1.2.
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TCP\IP protocol can be connected to various computer data networks in the world. This protocol increasingly exists and is needed so that many parties develop it to vote through this protocol. Voice Over Internet Protocol technology is the answer to that desire. This technology is able to convert analog voice (human voice) into data packets then through public internet data networks and private intranet data packets are passed, so that communication can occur. With VoIP communication costs can be reduced so that it can reduce investment costs and conversations (cost saving) or even up to 100% free. VoIP implementation can be done by designing a wireless VoIP network (cable) using 3CXSystemPhone software as a PBX. In this scientific work the software used is 3CXSystemPhone 11.0, where SIP is a VoIP server which is a freeware software, in its application only requires one PC server and several PC clients (2 for example) that are connected to each other
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Meisel, John B., and Michael Needles. "Voice over internet protocol (VoIP) development and public policy implications." info 7, no. 3 (June 2005): 3–15. http://dx.doi.org/10.1108/14636690510596766.
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Roy, Om Prakash, and Vinod Kumar. "A Survey on Voice over Internet Protocol (VoIP) Reliability Research." IOP Conference Series: Materials Science and Engineering 1020 (January 16, 2021): 012015. http://dx.doi.org/10.1088/1757-899x/1020/1/012015.
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Exsan, Maftufin, and Umi Fadlilah. "Pembangunan Infrastruktur Voice Over Internet Protocol di Organisasi Perangkat Daerah Boyolali menggunakan Server Elastix." Emitor: Jurnal Teknik Elektro 17, no. 2 (September 15, 2017): 39–47. http://dx.doi.org/10.23917/emitor.v17i2.6233.
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Perkembangan teknologi informasi dan komunikasi yang semakin maju, didukungdengan jaringan internet yang semakin sering digunakan dalam kehidupan sehari-hari serta pentingnya komunikasi dalam sebuah lingkungan pemerintahan, maka dalam penelitian ini penulis membangun infrastruktur jaringan Voice over Internet Protocol (VoIP) di Organisasi Perangkat Daerah Boyolali (OPD Boyolali) menggunakan Server Elastix. Teknologi ini mampu menyediakan komunikasi yang berjalan pada jaringan internet yang tentunya lebih hemat jika dibandingkan harus menggunakan operator telepon ataupun berlangganan jasa telepon rumah. Hasil penelitian ini berupa jaringan VoIP yang dapat diakses oleh seluruh OPD Boyolali. Hasil pengujian yang lakukan menggunakan aplikasi Wireshark menunjukkan bahwa jaringan VoIP ini memiliki kualitas yang bagus, sehingga komunikasi yang berlangsung dapat berjalan dengan lancar.
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Jeyanthi, N., R. Thandeeswaran, and J. Vinithra. "Rqa based approach to detect and prevent ddos attacks in voip networks." Cybernetics and Information Technologies 14, no. 1 (March 1, 2014): 11–24. http://dx.doi.org/10.2478/cait-2014-0002.
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Abstract Voice over Internet Protocol (VoIP) is a family of technologies for the transmission of voice over Internet. Voice is converted into digital signals and transmitted as data packets. The Session Initiation Protocol (SIP) is an IETF protocol for VoIP and other multimedia. SIP is an application layer protocol for creating, modifying and terminating sessions in VoIP communications. Since SIP is a more flexible and simple protocol, it is quite easy to add features to it. Distributed Denial of Service Attack (DDoS) floods the server with numerous requests from various hosts. Hence, the legitimate clients will not be able to get their intended services. A major concern in VoIP and almost in all network domains is availability rather than data consistency. Most of the surviving techniques could prevent VoIP network only after collision. This paper proposes a Recurrence Quantification based approach to detect and prevent VoIP from a DDoS attack. This model detects the attack at an earlier stage and also helps to prevent from further attacks. In addition, this techniques enables the efficient utilization of resources. QUALNET has been used to simulate the operation of the proposed technology.
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Hanindar, Muthia, and Irwansyah Irwansyah. "PEMANFAATAN TEKNOLOGI VOICE OVER INTERNET PROTOCOL (VoIP) DALAM ONLINE MOBILE GAMES." Jurnal Common 5, no. 1 (July 5, 2021): 25–38. http://dx.doi.org/10.34010/common.v5i1.2863.
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Kemajuan teknologi memungkinkan adanya kemajuan dalam industri video games. Kemunculan perangkat seluler dengan fitur yang semakin canggih memungkinkan khalayak dapat bermain games dimana saja dan kapan saja. Ada waktu luang yang perlu diisi menjadi salah satu alasan utama khalayak dalam bermain game pada ponsel pintar mereka. Kualitas ponsel pintar yang semakin ditingkatkan membuat dunia industri games semakin berkembang dengan adanya fitur-fitur baru dalam game salah satunya fitur komunikasi. Komunikasi sangat diperlukan utamanya dalam permainan yang membutuhkan kerjasama tim. Teknologi Voice over Internet Protocol (VoIP) memungkinkan komunikasi ini terjadi. Dengan menggunakan metode kualitatif, penelitian ini bertujuan untuk melihat kebutuhan khalayak yang dapat dipuaskan melalui media yang dikonsumsi. Melalui teori uses and gratifications 2.0 ditemukan bahwa kebutuhan seseorang dalam bermain game bisa muncul sebelum mengkonsumsi game tersebut ataupun muncul setelah menjelajahi pemakaian media tersebut.
Kata kunci: games, ponsel pintar, komunikasi, VoIP, uses and gratifications 2.0
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Unik, Mitra, and Sunanto Sunanto. "PERANCANGAN TEKNOLOGI VOICE OVER INTERNET PROTOCOL (VoIP) MEMANFAATKAN INFRASTRUKTUR JARINGAN LISTRIK." JURNAL FASILKOM 7, no. 2 (August 11, 2018): 255–59. http://dx.doi.org/10.37859/jf.v7i2.609.
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Kolaborasi teknologi jaringan dan teknologi multimedia memicu munculnya sebuah ide baru dalam menyiapkan teknologi komunikasi audio (voice) yang lebih murah dari segi infrastruktur karena dapat memanfaatkan jaringan listrik sebagai media penghantar jaringan data. Gagasan utama di balik ide komunikasi listrik / Powerline Communication (PLC) yaitu memanfaatkan jaringan distribusi listrik sebagai distribusi data serta dengan konsep memanipulasi jaringan listrik menjadi jaringan komputer. Sehingga memungkinkan komunikasi antar perangkat komputer terjadi khususnya digunakan sebagai alternatif solusi guna mengakomodasi kebutuhan komunikasi data bagi bangunan yang terletak di daerah-daerah yang belum menyediakan jaringan telekomunikasi dasar
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Kumar, Vinod, and Om Prakash Roy. "Improved Reliability of Voice over Internet Protocol(VoIP) using Machine Learning." IOP Conference Series: Materials Science and Engineering 1020 (January 16, 2021): 012025. http://dx.doi.org/10.1088/1757-899x/1020/1/012025.
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Armanto, Armanto Armanto. "IMPLEMENTASI SERVER VoIP (Voice Over Internet Protocol ) PADA KANTOR KECAMATAN SALING KABUPATEN EMPAT LAWANG." Jurnal Sistem Komputer Musirawas (JUSIKOM) 3, no. 2 (December 5, 2018): 114. http://dx.doi.org/10.32767/jusikom.v3i2.344.
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Implementation of VoIP Server At Sub-District Office of Saling Regency Empat Lawang has building structure which has many room. In case communicate sometimes takes a fast time. Like a meeting to be made at that moment. For that to be helpful in time and efficient then it can be applied Conference Call at Saling sub-district office. In this study the network used is Local Area Network (LAN), Using server3CX Phone System, while in developing the system used NDLC (Network Development Life Cycle) method. With VoIP Server Implementation In Sub-District Office of Saling Regency Empat Lawang can make inter-space communication to be efficient and reduce the cost of usage of mobile phone pulse which is too high. Keywords: VoIP server, 3CX Phone System, Conference
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Nisar, Kashif. "Voice Priority Queue Scheduling System Models for VoIP over WLANs." International Journal of Information Communication Technologies and Human Development 5, no. 1 (January 2013): 36–59. http://dx.doi.org/10.4018/jicthd.2013010103.
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The Voice over Internet Protocol (VoIP) is a delay sensitive traffic due to real-time applications on networks. The assessment of voice flow quality in the VoIP is an essential requirement for technical and commercial motivation. The packets of VoIP streaming may experience drops because of the competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delay and requires the voice packets to arrive on time from the sender to the receiver side without any delay over WLAN. The scheduling system model for VoIP traffic is an unresolved problem. In this research paper, the author proposes a new Voice Priority Queue (VPQ) scheduling system models and algorithms for the VoIP over WLANs to solve scheduling issues over IP-based networks. They present new contributions, through the three stages of the VPQ. The VPQ scheduling algorithm is provided as an essential technique in the VoIP communication networks to guarantee the QoS requirements. The design of the VPQ is managed by the limited bandwidth utilization and has been proven to have an efficient performance over WLANs.
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Kolhar, Manjur. "Zeroize: A New Method to Improve the Utilization of 5G Networks When Running VoIP over IPv6." Applied System Innovation 4, no. 4 (September 26, 2021): 72. http://dx.doi.org/10.3390/asi4040072.
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5G technology is spreading extremely quickly. Many services, including Voice Over Internet Protocol (VoIP), have utilized the features of 5G technology to improve their performance. VoIP service is gradually ruling the telecommunication sector due to its various advantages (e.g., free calls). However, VoIP service wastes a substantial share of the VoIP 5G network’s bandwidth due to its lengthy packet header. For instance, the share of the packet header from bandwidth and channel time reaches 85.7% of VoIP 5G networks when using the IPv6 protocol. VoIP designers are exerting considerable efforts to solve this issue. This paper contributes to these efforts by designing a new technique named Zeroize (zero sizes). The core of the Zeroize technique is based on utilizing the unnecessary fields of the IPv6 protocol header to keep the packet payload (voice data), thereby reducing or “zeroizing” the payload of the VoIP packet. The Zeroize technique substantially reduces the expanded bandwidth of VoIP 5G networks, which is reflected in the wasted channel time. The results show that the Zeroize technique reduces the wasted bandwidth by 20% with the G.723.1 codec. Therefore, this technique successfully reduces the bandwidth and channel time of VoIP 5G networks when using the IPv6 protocol.
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Sarkar, Nurul I., and Kashif Nisar. "Performance of VoIP in Wired-Cum-Wireless Ethernet Network." International Journal of Interdisciplinary Telecommunications and Networking 4, no. 4 (October 2012): 1–25. http://dx.doi.org/10.4018/jitn.2012100101.
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The Voice over Internet Protocol (VoIP) is a rapidly growing technology that enables transport of voice over data networks such as Ethernet Local Area Networks (LANs). This growth is due to the integration of voice and data traffic over the existing network infrastructure, low cost, and improved network management offered by the technology. This paper reports on the performance of VoIP traffic characteristics in a wired-cum-wireless Ethernet LAN. The effect of increasing the number of VoIP wireless clients, different voice codec schemes, and packet arrival distributions on system performance is investigated. Through various simulation experiments under realistic network scenarios, such as Small Office Home Office (SOHO) and campus networks, this paper provides an insight into the performance of VoIP over Ethernet LANs. Simulation results show that VoIP clients and voice codec schemes have significant effect on system performance. The authors preformed OPNET-based simulations to validate their experiments.
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Nisar, Kashif. "Fourth Stage of Voice Priority Queue for VoIP over WLANs." International Journal of Interdisciplinary Telecommunications and Networking 4, no. 2 (April 2012): 48–63. http://dx.doi.org/10.4018/jitn.2012040104.
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Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-term transmission for voice and audio traffic and is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling won’t be able to handle the VoIP packets properly and they have drawbacks over real-time applications. The objective of this research is to propose a Fourth Stage of Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic.
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Heriyanto, Agus, Lailis Syafaah, and Amrul Faruq. "Analisis Quality of Services Jaringan VoIP pada VPN menggunakan InterAsteriks Exchange dan Session Initiation Protocol." Techno.Com 19, no. 1 (February 27, 2020): 1–11. http://dx.doi.org/10.33633/tc.v19i1.2753.
Full textAbstract:
Di dalam komunikasi Voice over Internet Protocol (VoIP) mengenal beberapa macam protocol tambahan selain protocol standar internet Transfer Control Protocol/Internet Protocol (TCP/IP), beberapa diantaranya adalah protocol Session Initation Protocol (SIP), Inter-Asterisk eXchange (IAX) dan H.323. Performansi perlu dijaga mengingat VoIP mempunyai kemungkinan melakukan berbagai cara kompresi untuk menciptakan efisiensi saluran dan pemilihan protocol yang tepat. Teknologi VoIP pada dasarnya tidak memiliki jaminan keamanan pada setiap komunikasi. Keamanan ketika melakukan komunikasi suara merupakan sesuatu yang sangat penting karena menyangkut privasi penggunanya. Penggunaan Virtual Private Network (VPN) merupakan salah satu solusi untuk menutup celah keamanan pada kasus di atas. Analisis yang dilakukan pada artikel ini adalah performa yang dihasilkan VoIP yang menggunakan protocol IAX dan SIP. Penelitian ini mengahasilkan kesimpulan bahwa performansi yang paling baik digunakan untuk membangun sistem komunikasi VoIP adalah protocol IAX dengan menggunakan sistem keamanan VPN Point to Point Protocol (PPTP) dikarenakan nilai Quality of Service (QoS) lebih tinggi daripada protocol SIP dan juga terbukti lebih aman saat diterapkan sistem keamanan Virtual Private Network Point to Point Protocol (VPN PPTP).
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Faridah, Faridah. "DESAIN VOIP SERVER MENGGUNAKAN 3CXPHONE SYSTEM DAN SOFTPHONE PADA KANTOR PDAM KABUPATEN KEPULAUAN SELAYAR." ILTEK : Jurnal Teknologi 12, no. 01 (April 11, 2017): 1702–5. http://dx.doi.org/10.47398/iltek.v12i01.398.
Full textAbstract:
TCP/IP (Transmission Control Protocol/Internet Protocol) merupakan Protokol yang dapat dikoneksikan dalam jaringan data diberbagai komputer di dunia. Dalam penerapannya protocol ini dapat dipakai untuk keperluan perangkat lunak (software) pada system operasi, sehingga banyak yang memanfaatkan dan mengembangkan untuk dapat mengirim pesan suara. Teknologi VoIP (Voice over Internet Protocol) adalah salahsatu solusinya. Implementasi VoIP dapat dilakukan dengan merancang suatu jaringan nirkabel, menggunakan software 3CX Phone System dan IP PBX yang dibuat khusus untuk system operasi Windows dengan menggunakan protocol standar SIP. Dimana dalam pengeturan dan penggunaanya jauh lebih mudah. Teknologi ini mamp umengubah suara analog menjadi paket data kemudian melalui jaringan data public internet maupun private internet paket data tersebut dilewatkan, sehingga komunikasi dapat berjalan. Dengan adanya VoIP biaya komunikasi dapat dikurangi sehingga dapat mereduksi biaya percakapan bahkansecara gratis.
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Hirzan, Alauddin Maulana, Nazrulazhar Bahaman, and Whisnumurti Adhiwibowo. "Voice Over Internet Protocol Performance Evaluation in 6to4 Tunneling Network." Jurnal Transformatika 18, no. 1 (August 6, 2020): 108. http://dx.doi.org/10.26623/transformatika.v18i1.2356.
Full textAbstract:
<span lang="EN-US">Registry reported that their regional already in exhausted state. The IPv6 was proposed to substitute IPv4 network, but the implementation of this version cased many problems such as hardware compatibility. As temporary solution to this problem, 6to4 tunneling transition mechanism is introduced as one of many solutions. This mechanism used IPv4 network as communication media between two IPv6 networks. Thus, this kind of mechanism will affect the performance of Voice over Internet Protocol. VoIP demanded real-time communication by using UDP protocol between nodes. Unlike normal communication mode, real-time mode required data to be sent immediately ignoring the quality of data. This research evaluated the performance of 6to4 tunneling mechanism for Voice over Internet Protocol’s communication between two nodes in native IPv6 networks. </span>
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Aminah, Nina Siti, Muhamamad Reza Ramadhani Raharjo, and Maman Budiman. "Low-cost wireless mesh communications based on openWRT and voice over internet protocol." International Journal of Electrical and Computer Engineering (IJECE) 11, no. 6 (December 1, 2021): 5119. http://dx.doi.org/10.11591/ijece.v11i6.pp5119-5126.
Full textAbstract:
Technology makes it easier for us to communicate over a distance. However, there are still many remote areas that find it difficult to communicate. This is due to the fact that communication infrastructure in some areas is expensive to build while the profit will be low. This paper proposes to combine voice over internet protocol (VoIP) over mesh network implemented on openWRT router. The routers are performing mesh functions. We set up a VoIP server on a router and enabled session initiation protocol (SIP) clients on other routers. Therefore, we only need routers as a means of communication. The experiment showed very good results, in the line-of-sight (LOS) condition, they are limited to reception distances up to 145 meters while in the non-line-of-sight (NLOS) condition, they are limited to reception distances up to 55 meters.