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Journal articles on the topic 'VoIP'

Author: Grafiati
Published: 4 June 2021
Last updated: 15 June 2025

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1

Lundblad, Henrik, Gerald Q. Maguire, Charlotte Karlsson-Thur, et al. "Using PET/CT Bone Scan Dynamic Data to Evaluate Tibia Remodeling When a Taylor Spatial Frame Is Used: Short and Longer Term Differences." BioMed Research International 2015 (2015): 1–11. http://dx.doi.org/10.1155/2015/574705.

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Eighteen consecutive patients, treated with a Taylor Spatial Frame for complex tibia conditions, gave their informed consent to undergo Na18F−PET/CT bone scans. We present a Patlak-like analysis utilizing an approximated blood time-activity curve eliminating the need for blood aliquots. Additionally, standardized uptake values (SUV) derived from dynamic acquisitions were compared to this Patlak-like approach. Spherical volumes of interest (VOIs) were drawn to include broken bone, other (normal) bone, and muscle. TheSUVm(t)(m=max, mean) and a series of slopes were computed as(SUVm(ti)-SUVm(tj))/(ti-tj), for pairs of time valuestiandtj. A Patlak-like analysis was performed for the same time values by computing((VOIp(ti)/VOIe(ti))-(VOIp(tj)/VOIe(tj)))/(ti-tj), wherep= broken bone, other bone, and muscle ande= expected activity in a VOI. Paired comparisons between Patlak-like andSUVmslopes showed good agreement by both linear regression and correlation coefficient analysis (r=84%,rs=78%-SUVmax,r=92%, andrs=91%-SUVmean), suggesting static scans could substitute for dynamic studies. Patlak-like slope differences of 0.1 min−1or greater between examinations andSUVmaxdifferences of ~5 usually indicated good remodeling progress, while negative Patlak-like slope differences of −0.06 min−1usually indicated poor remodeling progress in this cohort.
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2

Arif, Rabbai San, Yuli Fitrisia, and Agus Urip Ari Wibowo. "Implementasi Voip Server Berbasis IPV6 Dengan Raspberry PI." Manutech : Jurnal Teknologi Manufaktur 9, no. 01 (2019): 47–54. http://dx.doi.org/10.33504/manutech.v9i01.32.

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Voice over Internet Protocol (VoIP) is a telecommunications technology that is able to pass the communication service in Internet Protocol networks so as to allow communicating between users in an IP network. However VoIP technology still has weakness in the Quality of Service (QoS). VOPI weaknesses is affected by the selection of the physical servers used. In this research, VoIP is configured on Linux operating system with Asterisk as VoIP application server and integrated on a Raspberry Pi by using wired and wireless network as the transmission medium. Because of depletion of IPv4 capacity that can be used on the network, it needs to be applied to VoIP system using the IPv6 network protocol with supports devices. The test results by using a wired transmission medium that has obtained are the average delay is 117.851 ms, jitter is 5.796 ms, packet loss is 0.38%, throughput is 962.861 kbps, 8.33% of CPU usage and 59.33% of memory usage. The analysis shows that the wired transmission media is better than the wireless transmission media and wireless-wired.
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3

ANDRIANTO, HERI, DANIEL SETIADIKARUNIA, and HENDRY RAHARJO. "Evaluasi Kinerja GSM VoIP Gateway pada Sistem IP PBX." ELKOMIKA: Jurnal Teknik Energi Elektrik, Teknik Telekomunikasi, & Teknik Elektronika 9, no. 3 (2021): 731. http://dx.doi.org/10.26760/elkomika.v9i3.731.

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ABSTRAKGSM VoIP Gateway digunakan untuk menghubungkan jaringan VoIP dengan jaringan GSM sehingga memungkinkan VoIP client melakukan komunikasi dengan VoIP client lain melalui jaringan GSM sehingga biaya komunikasi dapat ditekan. Pada penelitian ini, telah dirancang dan direalisasikan sistem IP PBX yang dihubungkan ke jaringan GSM menggunakan GSM VoIP Gateway. Evaluasi kinerja GSM VoIP Gateway pada sistem IP PBX dilakukan dengan mengamati nilai parameter Quality of Service (QoS). Komunikasi antara VoIP client dengan GSM VoIP Gateway dikategorikan pada kualitas layanan VoIP yang baik karena memiliki nilai rata-rata jitter ≤ 5,7 ms, packet loss ≤ 0,18% dan delay ≤ 9,41 ms. Komunikasi antara softphone SIPdroid dengan GSM VoIP Gateway memiliki nilai rata-rata jitter 22,58 ms, paket loss 48,68%, dan delay 14,54 ms, hal ini disebabkan karena komunikasi VoIP menggunakan koneksi WiFi. Selain itu perbedaan spesifikasi perangkat keras dan perangkat lunak juga turut mempengaruhi nilai parameter QoS.Kata kunci: GSM VoIP Gateway, IP PBX, VoIP ABSTRACTGSM VoIP Gateway is used to connect the VoIP network to the GSM network, allowing VoIP clients to communicate with other VoIP clients via the GSM network therefore the communication costs can be reduced. In this research, an IP PBX system connected to a GSM network using a GSM VoIP Gateway has been designed and realized. Performance evaluation of the GSM VoIP Gateway on the IP PBX system is carried out by observing the value of the Quality of Service (QoS) parameter. Communication between the VoIP client and GSM VoIP Gateway is categorized as a good quality VoIP service because it has an average value of jitter ≤ 5.7 ms, packet loss ≤ 0.18% and delay ≤ 9.41 ms. Communication between the SIPdroid softphone and the GSM VoIP Gateway has an average jitter value of 22.58 ms, a packet loss of 48.68%, and a delay of 14.54 ms, due to VoIP communication uses a WiFi connection. In addition, differences on hardware and software specifications also affect the value of QoS parameters.Keywords: GSM VoIP Gateway, IP PBX, VoIP
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4

Child, M. "Briefing: VoIP." ITNOW 47, no. 1 (2004): 28. http://dx.doi.org/10.1093/combul/bwi011.

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5

Johnston, Elizabeth. "Editorial [VoIP]." IEEE Potentials 26, no. 1 (2007): 3. http://dx.doi.org/10.1109/mp.2007.343012.

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6

Gold, Steve. "Securing VoIP." Network Security 2012, no. 3 (2012): 14–17. http://dx.doi.org/10.1016/s1353-4858(12)70046-6.

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7

Soupionis, Yannis, and Dimitris Gritzalis. "Hacking VoIP." Computers & Security 32 (February 2013): 267. http://dx.doi.org/10.1016/j.cose.2012.09.006.

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8

Ebbinghaus, R. "VoIP lessons." Communications Engineer 1, no. 5 (2003): 28–31. http://dx.doi.org/10.1049/ce:20030505.

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Materna, B. "VoIP insecurity." Communications Engineer 4, no. 5 (2006): 39–42. http://dx.doi.org/10.1049/ce:20060507.

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10

Bhebhe, Leo, and Rauli Parkkali. "VoIP Performance over HSPA with Different VoIP Clients." Wireless Personal Communications 58, no. 3 (2010): 613–26. http://dx.doi.org/10.1007/s11277-010-0125-2.

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11

Syafrinal, Syafrinal. "Implementasi VoIP Sebagai Media Komunikasi pada Dinas Perhubungan Komunikasi Informasi dan Telematika Aceh." Jurnal JTIK (Jurnal Teknologi Informasi dan Komunikasi) 3, no. 2 (2019): 64. http://dx.doi.org/10.35870/jtik.v3i2.88.

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The purpose of this study is to design and build voice communication over IP networks using the Briker operating system. The use of VoIP Server to make calls from VoIP clients to fellow VoIP clients, and from people's VoIP numbers to one of the VoIP clients, and also looking for a comparison of public IP client and local IP voice delay at Dishubkomintel Aceh. The research method consisted of several stages namely; 1) System Design, 2) Network Topology, 3) Server-Side Design, 4) Client-Side Design, 5) Operating System and Application Installation, 6) Configuration, 7) VoIP Network Connectivity Testing, and 8) VoIP Work Observation. From the test results, several conclusions can be drawn namely; 1) VoIP Server Briker has a role in handling SIP calls from all registered clients into the Briker Server, 2) Between VoIP clients can communicate two ways with each other when registered into the Briker Server, 3) Calls to public VoIP are made by pressing '9' which used as Outbound routes then continued by pressing the destination number, and 4) Calls from VoIP of the people to VoIP are made by pressing the VoIP phone number of the people connected to the VoIP Server and then received by VRR (Voice Response Response) which will be directed to the extension number headed.Keywords:Voice Communication, IP Network, VoIP, Briker
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Arwa, Fitrah Jihad, Jonny Latuny, and Elvery B. Johannes. "RANCANG BANGUN VOICE OVER INTERNET PROTOCOL (VOIP) DENGAN MENGGUNAKAN RASPBERRY PI 4 PADA FAKULTAS TEKNIK UNIVERSITAS PATTIMURA AMBON." Jurnal ISOMETRI 1, no. 2 (2022): 85–91. http://dx.doi.org/10.30598/isometri.2022.1.2.85-91.

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Penelitian bertujuan untuk membangun sistem sentral dan jaringan telepon berbasis Voice over Internet Protocol (VoIP) dengan memanfaatkan interkoneksi sistem menggunakan jaringan LAN/WLAN di Fakultas Teknik. Pengembangan sistem / jaringan telepon berbasis VoIP ini untuk mengatasi masalah komunikasi internal di lingkungan Fakultas Teknik yang jika menggunakan telepon VoIP dapat mengurangi biaya telepon yang digunakan untuk keperluan administrasi didalam melaksanakan pekerjaan. Metode pengembangan sistem VoIP yang digunakan dalam penelitian ini menggunakan metode system prototyping dimana sistem dibangun dalam dua kategori utama yakni pada bagian hardware dan software. Konstruksi sistem pada bagian hardware berupa deployment jaringan dari server VoIP ke masing-masing unit telepon VoIP yang digunakan. Sedangkan pada bagian software dilakukan scripting process untuk mengkonfigurasi perangkat lunak Asterisk agar dapat berfungsi sebagai suatu sentral telepon VoIP. Hasil dari proses pengembangan / rancang bangun sistem VoIP diperoleh suatu sistem Client-Server VoIP dengan sejumlah client (5 unit telepon) VoIP tipe PIP901. Server menggunakan Raspberry PI4+. Asterisk pada server VoIP menggunakan protokol SIP (Session Initiation Protocol) guna mengatur proses panggil dan terima pada masing-masing unit telepon VoIP yang tersambung. Hasil pengujian menunjukkan sistem dapat berfungsi sesuai konfigurasi pada server serta diperoleh kualitas suara pada telepon VoIP berada pada rentang 0-5000 Hz yang sesuai untuk pendengaran pengguna telepon. Sistem sentral telepon yang dibangun dalam penelitian ini menggunakan alokasi nomor telepon untuk masing-masing unit telepon VoIP dengan standar nomor 4 digit angka dengan format 7001, 7002, 7003, 7004, 7005. Sistem selanjutnya direncanakan untuk digunakan untuk memfasilitasi keperluan komunikasi internal di lingkungan Fakultas Teknik
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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 (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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14

Abualhaj, Mosleh M., Ahmad A. Abu-Shareha, and Sumaya N. Al-Khatib. "Utilizing Voip Packet Header’s Fields to Save the Bandwidth." Transport and Telecommunication Journal 24, no. 1 (2023): 33–42. http://dx.doi.org/10.2478/ttj-2023-0004.

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Abstract Voice over IP (VoIP) is widely utilized by organizations, schools, colleges, and so on. Nevertheless, VoIP numerous challenges that hinder its spread. One of the significant challenges is the poor exploit of the VoIP technology network bandwidth (BW), caused by the huge preamble of the VoIP packet. This paper suggests a novel methodology to manage this huge preamble overhead challenge. The proposed methodology is named runt payload VoIP packet (RPV). The core principle of the RPV methodology is to reemploy and exploit the VoIP packet preamble’s data (fields) that are superfluous by VoIP technology, especially for unicast IP voice calls. Generally, those fields will be used to convey the VoIP packet payload. Consequently, diminish or zero the length of the payload and, therefore, spare the BW. The results of the investigation into the suggested RPV methodology indicated significant enhancement in the BW exploitation of VoIP technology. For instance, the saved BW in the examined environment with the LPC codec came to up to 25.9%.
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Najwaini, Effan, and Ahmad Ashari. "Analisis Kinerja Voip Server pada Wireless Access Point." IJCCS (Indonesian Journal of Computing and Cybernetics Systems) 9, no. 1 (2015): 89. http://dx.doi.org/10.22146/ijccs.6643.

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AbstrakPada komunikasi VoIP (Voice Over IP) kualitas suara dipengaruhi oleh banyak faktor salah satunya yaitu kualitas server. Pemilihan platform PC atau server yang cocok (baik dari segi harga maupun kinerja) merupakan persoalan utama dalam membangun jaringan VoIP. Kinerja server yang jelek akan menurunkan kualitas suara atau bahkan tidak mampu untuk menghubungkan antar pengguna.Dalam penelitian ini dilakukan pengujian terhadap kinerja wireless access point Linksys WRT54GL yang dimanfaatkan sebagai VoIP server. Pengujian dilakukan untuk mengetahui berapa banyak panggilan VoIP yang mampu dilayani oleh wireless access point sebagai VoIP server serta berapa waktu yang diperlukan oleh server tersebut untuk dapat memproses setiap sinyal SIP maupun paket RTP. Berdasarkan hasil pengujian yang dilakukan, VoIP server pada wireless access point mampu melayani komunikasi VoIP dengan baik untuk jumlah panggilan yang sedikit sehingga layak diimplementasikan pada penggunaan skala kecil. Penggunaan metode Native Bridging dalam penanganan media yang dilakukan oleh server dapat meningkatkan jumlah panggilan yang mampu dilayani sebesar 3 hingga 7 kali dibandingkan dengan metode lainnya. Kata kunci—VoIP, Asterisk, Acess Point, WRT54GL, OpenWRT, Kinerja AbstractVoice quality on VoIP communication is caused by many factors, one of which is the quality of the server. Choosing PC platform or server which is suitable is the main issue in developing VoIP network. A bad server performance or not equivalent with the most of users will degrade the sound quality or even not able to connect between users.Tthe test carried out to the performance of the wireless access point Linksys WRT54GL which is used as a VoIP server. The test was carried out to determine how many VoIP calls which are able to be serviced by a wireless access point as a VoIP server and how long the server needs to be able to process every signal of SIP and RTP packet.Based on the test result performed, the VoIP server on the wireless access point is able to serve VoIP communication well for a few calls number, so it is worth to be implemented on the use of small scale. The use of Native Bridging method in handling the media performed by the server can increase the number of calls that were able to be served about 3 to 7 times compared with other methods. Keywords— VoIP, Asterisk, Acess Point, WRT54GL, OpenWRT, Performance
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Ismail, Mohd Nazri. "Best Approach for Video Codec Selection Over VoIP Conversation Using Wireless Local Area Network." International Journal of Interdisciplinary Telecommunications and Networking 3, no. 1 (2011): 36–49. http://dx.doi.org/10.4018/jitn.2011010103.

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This study evaluates video codec performance over VoIP using a campus wireless network. Today, the deployment of VoIP occurs in various platforms, including VoIP over LAN, VoIP over WAN and VoIP over VPN. Therefore, this study defines which video codec provides good video quality over VoIP transmission. The soft phone is used as a medium for communication between two parties. A network management system is used to evaluate and capture the video quality performance over VoIP. The quality of video codec is based on MOS, jitter, delay and packet loss. The experimental scope is limited to G.722 with MP4V-ES, G.726 (16) with H.261 and G.726 (24) with H.264. The results show that audio codec G.722 with MP4V-ES generates good video quality over VoIP using wireless local area network. Whereas audio codec G.726 (16) with H.261 generates low rate video and voice quality performance. Therefore, using the appropriate video and audio, the codec selection increases video quality over VoIP transmission.
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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 (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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18

Hamidi, Eki Ahmad Zaki, Mufid Ridlo Effendi, and Hafizh Wibowo Widodo. "Prototipe Layanan VoIP Pada Jaringan OpenFlow." TELKA - Telekomunikasi, Elektronika, Komputasi dan Kontrol 4, no. 1 (2018): 33–42. http://dx.doi.org/10.15575/telka.v4i1.84.

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Voice Over Internet Protocol (VoIP) adalah sebuah teknologi yang mampu melewatkan trafik suara dan data yang berbentuk paket melalui jaringan IP. Penggunaan IP memungkinkan penghematan biaya dikarenakan tidak perlu membuat sebuah infrastruktur baru untuk komunikasi suara. Dalam mengaplikasikan VoIP saat ini masih menggunakan infrastruktur jaringan konvensional, dengan berkembangnya SDN (Software Defined Network) yang menawarkan paradigma baru dalam dunia jaringan dalam mendesain, mengelola dan mengimplementasikan jaringan, terutama untuk mendukung kebutuhan dan inovasi di bidang ini yg semakin lama semakin kompleks, maka layanan VoIP dapat diaplikasikan pada jaringan OpenFlow yang merupakan implementasi dari konsep SDN (Software Defined Network). Dalam mengimplementasikan VoIP menggunakan pada jaringan OpenFlow, VoIP dapat dibangun dengan menggunakan aplikasi yang bersifat freeware seperti X-Lite sebagai User Agent atau Client, Asterisk sebagai server VoIP dan G.711 sebagai codec dengan memanfaatkan OpenvSwitch yang berfungsi meneruskan paket-paket dalam layanan VoIP. Pada pengujian ini dilakukan terbagi dalam 2 bagian, pengujian pada OpenvSwitch dan VoIP. Pada pengujian OpenvSwitch dapat disimpulkan bahwa flow yang sudah dibuat pada OpenvSwitch dapat berfungsi untuk meneruskan paket-paket dalam layanan VoIP sehingga antara server dan client dapat terhubung. Pengujian VoIP dilakukan sebanyak 10 kali percobaan dengan hasil yang didapatkan dengan rata-rata delay 10,0002421 ms, throughput 171,195 Kbps, jitter 0,424 ms, packet loss 0%.
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Hamidi, Eki Ahmad Zaki, Mufid Ridlo Effendi, and Hafizh Wibowo Widodo. "Prototipe Layanan VoIP Pada Jaringan OpenFlow." TELKA - Telekomunikasi, Elektronika, Komputasi dan Kontrol 4, no. 1 (2018): 33–42. http://dx.doi.org/10.15575/telka.v4n1.33-42.

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Voice Over Internet Protocol (VoIP) adalah sebuah teknologi yang mampu melewatkan trafik suara dan data yang berbentuk paket melalui jaringan IP. Penggunaan IP memungkinkan penghematan biaya dikarenakan tidak perlu membuat sebuah infrastruktur baru untuk komunikasi suara. Dalam mengaplikasikan VoIP saat ini masih menggunakan infrastruktur jaringan konvensional, dengan berkembangnya SDN (Software Defined Network) yang menawarkan paradigma baru dalam dunia jaringan dalam mendesain, mengelola dan mengimplementasikan jaringan, terutama untuk mendukung kebutuhan dan inovasi di bidang ini yg semakin lama semakin kompleks, maka layanan VoIP dapat diaplikasikan pada jaringan OpenFlow yang merupakan implementasi dari konsep SDN (Software Defined Network). Dalam mengimplementasikan VoIP menggunakan pada jaringan OpenFlow, VoIP dapat dibangun dengan menggunakan aplikasi yang bersifat freeware seperti X-Lite sebagai User Agent atau Client, Asterisk sebagai server VoIP dan G.711 sebagai codec dengan memanfaatkan OpenvSwitch yang berfungsi meneruskan paket-paket dalam layanan VoIP. Pada pengujian ini dilakukan terbagi dalam 2 bagian, pengujian pada OpenvSwitch dan VoIP. Pada pengujian OpenvSwitch dapat disimpulkan bahwa flow yang sudah dibuat pada OpenvSwitch dapat berfungsi untuk meneruskan paket-paket dalam layanan VoIP sehingga antara server dan client dapat terhubung. Pengujian VoIP dilakukan sebanyak 10 kali percobaan dengan hasil yang didapatkan dengan rata-rata delay 10,0002421 ms, throughput 171,195 Kbps, jitter 0,424 ms, packet loss 0%.
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20

Basorun, Oluseyi, and Mnse Mnieee Mieee. "Implementing Enterprise VoIP Deployment." Journal of Advance Research in Computer Science & Engineering (ISSN: 2456-3552) 3, no. 4 (2016): 06–16. http://dx.doi.org/10.53555/nncse.v3i4.424.

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Voice over Internet Protocol (VoIP) is perceived as the best example of collaboration technology in today’s telecommunication space. There have being a paradigm shift from regular legacy telephone systems to VoIP over the last decade by most telecommunication and multinational companies globally. One of the huge benefit of VoIP is noticeable reduction in international call rates. In addition to the potential cost savings, VoIP offers the prospect of integrating voice with data and video applications, this goes a long way in increasing workers’ productivity within organizations. Many organizations are already embracing VoIP as replacement for their legacy PBX, others are integrating the existing PBX system with VoIP. This write up will focus on deployment of PBX-VoIP converged solution in enterprise organizations, where cost saving is one of the primary objectives.
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21

Kabachinski, Jeff. "VoIP: Let's Talk." Biomedical Instrumentation & Technology 42, no. 4 (2008): 297–302. http://dx.doi.org/10.2345/0899-8205(2008)42[297:vlt]2.0.co;2.

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22

Stone, A. "Has VoIP arrived?" IEEE Internet Computing 7, no. 6 (2003): 10–11. http://dx.doi.org/10.1109/mic.2003.1250578.

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Bodhani, A. "VOIP - voicing concerns." Engineering & Technology 6, no. 7 (2011): 76–79. http://dx.doi.org/10.1049/et.2011.0713.

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Cherry, S. "The VoIP backlash." IEEE Spectrum 42, no. 10 (2005): 61–63. http://dx.doi.org/10.1109/mspec.2005.1515963.

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POPESCU, Eliza-Elena. "VoIP Security Threats." International Journal of Information Security and Cybercrime 13, no. 1 (2024): 66–70. http://dx.doi.org/10.19107/ijisc.2024.01.06.

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Nowadays, communication is based on a wide variety of solutions, especially using the real-time ones. People use a lot of online tools that offer them audio-video solutions, and the most important thing is that anyone has access to them, and there are easy to use. VoIP (Voice over IP) is one of the most used solutions today, with which everyone can make calls over the internet. Like any other service that is exposed to the internet, it presents risks for the data violation and administrators or any people that use this service, needs to secure the system.
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O'Halloran, Joe. "Zotob and VoIP." Infosecurity Today 2, no. 5 (2005): 10. http://dx.doi.org/10.1016/s1742-6847(05)70316-5.

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O'Halloran, Joe. "Zotob and VoIP." Network Security 2005, no. 9 (2005): 2–20. http://dx.doi.org/10.1016/s1353-4858(05)70276-2.

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Hunter, P. "Thwarting VoIP threats." Information Professional 2, no. 6 (2005): 37–39. http://dx.doi.org/10.1049/inp:20050608.

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Lee, Minkyu, James W. McGowan, and Michael C. Recchione. "Enabling wireless VoIP." Bell Labs Technical Journal 11, no. 4 (2007): 201–15. http://dx.doi.org/10.1002/bltj.20204.

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Deepti. "Voice over Internet Protocol (VOIP): Future Potential." COMPUSOFT: An International Journal of Advanced Computer Technology 03, no. 11 (2014): 1342–49. https://doi.org/10.5281/zenodo.14768362.

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VoIP (voice over IP) delivers standard voice over telephone services over Internet Protocol (IP). VoIP 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. Gateways are the key component required to facilitate IP Telephony. A gateway is used to bridge the traditional circuit switched PSTN with the packet switched Internet. The paper covers software, hardware and protocol requirements followed by weighing the VoIP advantages such as low cost, portability, free and advanced features, bandwidth efficiency, call recording and monitoring against the VoIP disadvantages such as power dependency, quality of voice and service, security, and reliability. With ever increasing internet penetration and better broadband connectivity, VoIP is going to expand further with businesses already using VoIP standalone or in a hybrid format, although our focus and scope here remains VoIP. Mobile VoIP, an infant with less than 4% market share, has so far been focusing on increasing active subscriptions without a sustainable revenue model, but has the potential and is going to see tussle with static VoIP for space in days ahead. 
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Kim, Young-Dong. "Transmission Performance of VoIP Traffic over MANETs." Journal of the Korean Institute of Information and Communication Engineering 14, no. 5 (2010): 1109–16. http://dx.doi.org/10.6109/jkiice.2010.14.5.1109.

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Musbah, Esra Musbah Mohammed, Khalid Hamed Bilal, and Amin Babiker A. Nabi Mustafa. "Comparison of QoS Performance Over WLAN, VoIP4 and VoIP6." International Research Journal of Management, IT & Social Sciences 2, no. 11 (2015): 42. http://dx.doi.org/10.21744/irjmis.v2i11.80.

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VoIP stands for voice over internet protocol. It is one of the most widely used technologies. It enables users to send and transmit media over IP network. The transition from IPv4 to IPv6 provides many benefits for internet IPv6 is more efficient than IPv4. This paper presents a performance analysis of VoIP over WLAN using IPv4 and IPv6 and OPNET software program to simulate the protocols and to investigate the QoS parameters such as jitter, delay variation, packet send, and packet received and throughputs for IP4 and IP6 and compare between them.
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Tandel, Hardik, and Parag H. Rughani. "Forensic Analysis of Asterisk-FreePBX based VoIP Server." International Journal of Emerging Research in Management and Technology 6, no. 8 (2018): 166. http://dx.doi.org/10.23956/ijermt.v6i8.133.

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VoIP – Voice over IP is becoming an important aspect of communication. With invention of high speed Internet together with mobile phones has made it possible to utilize this highly important technology for voice communication. International calls are now possible with VoIP at compatible rates compared to past. The trend made people develop applications based on VoIP not only for private IP PBX but for subscribed users from all over the world. The dependency in whole technology is its VoIP Server, which is not directly exposed to end users, but if can be compromsed then can affect all the connected users. Looking at increasing popularity of VoIP, attackers are now targeting these VoIP servers for various purposes like stealing information or using service to make free calls. It is always difficult to prevent such attacks but investigation may always help in solving such attacks and preventing future attacks with similar modus operandi. These work focuses on forensic analysis of VoIP server asterisk and discusses important artifacts which can be retrieved from affected VoIP server.
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Washima Tuleun. "Design of an asterisk-based VoIP system and the implementation of security solution across the VoIP network." World Journal of Advanced Research and Reviews 23, no. 1 (2024): 875–906. http://dx.doi.org/10.30574/wjarr.2024.23.1.2048.

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Voice over Internet Protocol (VoIP) is a rapidly advancing technology that facilitates the transmission of voice and audio signals over the Internet or an IP-based network in real-time. This technology has seen a significant rise in demand due to its advantages over traditional circuit-switched telephony, including lower call rates, reduced operational costs, easier management, and enhanced call features. However, the growth in VoIP usage has also increased the potential for various security threats and attacks, jeopardizing the privacy, confidentiality, and integrity of transmitted data. This paper presents the design of an Asterisk-based VoIP system and the implementation of a comprehensive security solution across the VoIP network. The study involves an in-depth analysis of VoIP technology, identifying its vulnerabilities and addressing potential threats. A security framework is proposed and implemented to safeguard the VoIP network. The designed system and security solutions are rigorously tested and evaluated to ensure robustness and effectiveness. The findings highlight critical security measures necessary for protecting VoIP infrastructures and provide a framework for future research and development in securing VoIP networks.
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Syaifudin, Imam, Abdul Rahman, Destiarini Destiarini, Achmad Azhari, and Safaruddin Safaruddin. "Penggunaan Voice Over Internet Protocol (VOIP) dengan MPLS di PT. Semen Baturaja." INTECH 3, no. 1 (2022): 22–26. http://dx.doi.org/10.54895/intech.v3i1.1252.

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Teknologi VOIP yang digunakan pada PT. Semen Baturaja (Persero) adalah IP PBX yang terkoneksi lanngsung dengan MPLS indosat. VOIP merupakan singkatan dari Voice Over Internet Protocol atau juga bisa dinamakan sebagai IP Telephony, Digital Phone, atau Internet Telephony. Teknologi ini digunakan untuk berkomunikasi secara jarak jauh dengan memanfaatkan koneksi internet. Dalam penulisan tugas akhir artikel ini mengunakan metode observasi, interview dan literatur. Dengan menggunakan VOIP sebagai komunikasi suara pada PT. Semen Baturaja dapat membberikan keuntungan dan kemudahan dalam melakukan komunikasi dengan cabang perusahaan yang ada di kota lain terutama dari segi biaya. VoIP memiliki keunggulan yang tidak dimiliki oleh PSTN (Public Switched Telephone Network). VOIP sudah banyak diterapkan ke dalam berbagai macam pemanfaatan standar dan juga protocol yang bersifat open source. PT. Semen Baturaja (Persero) telah menggunakan VOIP untuk dapat berhubungan dengan mudah dan baik, untuk itu saya tertarik untuk mendalami apa itu VOIP di PT Semen Baturaja. IP Telephony Internet Telephony Broadband Telephony atau diistilahkan dengan VoIP (Voice Over Internet Protocol) merupakan teknologi yang memanfaatkan Internet Protocol untuk menyediakan komunikasi suara secara elektronis dan real-time.
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Washima, Tuleun. "Design of an asterisk-based VoIP system and the implementation of security solution across the VoIP network." World Journal of Advanced Research and Reviews 23, no. 1 (2024): 875–906. https://doi.org/10.5281/zenodo.14792201.

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Voice over Internet Protocol (VoIP) is a rapidly advancing technology that facilitates the transmission of voice and audio signals over the Internet or an IP-based network in real-time. This technology has seen a significant rise in demand due to its advantages over traditional circuit-switched telephony, including lower call rates, reduced operational costs, easier management, and enhanced call features. However, the growth in VoIP usage has also increased the potential for various security threats and attacks, jeopardizing the privacy, confidentiality, and integrity of transmitted data. This paper presents the design of an Asterisk-based VoIP system and the implementation of a comprehensive security solution across the VoIP network. The study involves an in-depth analysis of VoIP technology, identifying its vulnerabilities and addressing potential threats. A security framework is proposed and implemented to safeguard the VoIP network. The designed system and security solutions are rigorously tested and evaluated to ensure robustness and effectiveness. The findings highlight critical security measures necessary for protecting VoIP infrastructures and provide a framework for future research and development in securing VoIP networks.
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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 (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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Abualhaj, Mosleh M., Mayy M. Al-Tahrawi, and Mahran Al-Zyoud. "Contracting VoIP Packet Payload Down to Zero." Cybernetics and Information Technologies 21, no. 1 (2021): 137–50. http://dx.doi.org/10.2478/cait-2021-0010.

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Abstract The inefficient use of the IP network bandwidth is a fundamental issue that restricts the exponential spreading of Voice over IP (VoIP). The primary reason for this is the big header size of the VoIP packet. In this paper, we propose a method, called Short Voice Frame (SVF), that addresses the big header size of the VoIP packet. The main idea of the SVF method is to make effective use of the VoIP packet header fields that are unneeded to the VoIP technology. In particular, these fields will be used for temporarily buffering the voice frame (VoIP packet payload) data. This will make the VoIP packet payload short or even zero in some cases. The performance evaluation of the proposed SVF method showed that the use of the IP network bandwidth has improved by up to 28.3% when using the G.723.1 codec.
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Try, Ardana Putra, M. Fid Aksara L., and Surimi La. "IMPLEMENTASI SERVER VOIP MENGGUNAKAN ASTERISK PADA JURUSAN TEKNIK INFORMATIKA UNIVERSITAS HALU OLEO." semanTIK Vol 6 No 2 (December 29, 2020): 107–14. https://doi.org/10.5281/zenodo.4399332.

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VoIP (<em>Voice Over Internet Protocol</em>) adalah teknologi yang mampu melewatkan &ldquo;panggilan suara&rdquo;, <em>video</em> dan data melalui jaringan Internet Protocol (IP). VOIP merupakan jaringan komunikasi data yang berbasis <em>packet</em>-<em>switch</em>, sehingga bisa menelepon dengan menggunakan jaringan berbasis IP. Jaringan VoIP dapat dibangun dengan menggunakan jaringan <em>nirkabel</em>. VoIP memungkinkan <em>acces server </em>dan <em>multiservice access concentrator </em>membawa dan mengirim suara dan <em>fax </em>melintasi jaringan IP. <em>S</em><em>erver </em>VOIP<em> yang </em>menggunakan<em> Asterisk, </em>mengharuskan setiap <em>client </em>yang ingin melakukan panggilan untuk menggunakan aplikasi tambahan seperti <em>Zoiper</em> dan <em>MizuDroid</em> untuk saling terkoneksi. Pengujian dilakukan pada autentikasi klien, pengujian panggilan (<em>call</em>) dan pengujian <em>Quality of Services</em> (QoS). Autentikasi klien dan pengujian panggilan, semua dapat dilakukan sesuai dengan yang diharapkan. Untuk pengujian QoS, dari semua pengujian yang telah dilakukan, semua parameter yang digunakan menunjukkan hasil yang sangat baik. Nilai <em>delay</em> setiap pengujian sangat kecil dibawah 1 ms, nilai <em>packet loss </em>di bawah 1% dan nilai <em>throughput</em> di atas 15 MBps
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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 (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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Simarangkir, Manase Sahat H., Adam Puspabhuana, and Bei Harira Irawan. "Pelatihan Implementasi Server VoIP Menggunakan Router Cisco Pada Jaringan Lokal." BEMAS: Jurnal Bermasyarakat 2, no. 1 (2021): 9–18. http://dx.doi.org/10.37373/bemas.v2i1.119.

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Komunikasi data dalam jaringan sangat diperlukan untuk pertukaran data dan informasi. Teknologi komunikasi berbasis IP (Internet Protocol) dapat diimplementasikan menggunakan VoIP (Voice Over Internet Protocol). VoIP dapat digunakan untuk layanan komunikasi yang dapat mengirimkan pesan, suara maupun video menggunakan jaringan internet yang terhubung dengan IP. Kelebihannya adalah efisiensi terhadap bandwidth, efisiensi terhadap biaya pengelolaan. Di SMK Al-Manar Islamic School Cibarusah siswa jurusan TKJ belum sepenuhnya memahami akan peranan VoIP untuk layanan komunikasi dalam jaringan lokal dengan memanfaatkan infrastruktur jaringan yang ada. Pada pelatihan ini dibuat rancangan layanan jaringan komunikasi VoIP menggunakan Router Cisco sebagai server VoIP dan IP Phone sebagai media komunikasinya. Tujuannya adalah supaya siswa/i memiliki kemampuan dalam melakukan konfigurasi Router Cisco untuk layanan VoIP sekaligus dapat mengimplementasikannya pada jaringan lokal. Metode yang digunakan pada pelatihan ini adalah simulasi dan implementasi. Simulasi dilakukan pada rancangan dengan tools cisco packet tracer, kemudian implementasi konfigurasi VoIP diterapkan pada real device Router Cisco. Hasil konfigurasi pada Router cisco langsung dilakukan pengujian komunikasi menggunakan IP phone
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Pratama, Zakaria, Angga Putra Juledi, and Rahma Muti Ah. "Membangun Layanan Telepon Voice Over Internet Protocol Dengan Menggunakan Server Trixbox Di Smk Pemda Rantauprapat." INFORMATIKA 11, no. 3 (2024): 131–37. https://doi.org/10.36987/informatika.v11i3.5010.

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SMK Swasta Pemda Rantauprapat is one of the schools in Labuhanbatu Regency which is one of the oldest vocational schools in Labuhanbatu Regency, where communication has been carried out using paid telephones so that it becomes an additional cost that burdens the school's finances. Along with the rapid development of technology, it has resulted in free telephone services, one of which is VoIP (Voice Over Internet Protocol) technology with VoIP technology which can be used as the right solution to solve this problem. In building a VoIP system, a voip server is needed, namely Trixbox. Trixbox is a Voip Server built on the Linux CentosOs operating system which is open source so it can be developed. Communication using VoIP technology only requires a computer/leptop, microphone, speakers, smartphone and a LAN network using both wired and wireless. With VoIP technology in schools, school principals, vice principals, heads of departments and teachers can communicate without incurring telephone costs.
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Maheswari, K., and A. Balamurugan. "Voice over internet protocol codec performance in interactive streaming environment." i-manager's Journal on Communication Engineering and Systems 13, no. 1 (2024): 16. http://dx.doi.org/10.26634/jcs.13.1.20435.

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Voice over Internet Protocol (VoIP) is an interactive telecommunication technology that suggests shared, cooperative, and quality transactions in the form of communications over the internet. VoIP normally differs from traditional and conventional circuit-based networks in the form of a low rate because all the communication takes place through the internet. The user has to pay only for an internet connection. Real-time voice data transmission is very difficult when compared with ordinary text data transmission. A main challenge in telecom engineering is maintaining quality. The VoIP transmission faces a lot of inconvenience. It suffers from packet loss, low quality, jitter, delay, and on-time delivery. These parameters influence and degrade performance. The codec plays a major role in VoIP transmission. This paper reports on the fundamental functionalities of VoIP, including coder and decoder operations for quality output, broadcast technology, VoIP connection setups, advantages, applications, and emerging trends in VoIP. The performance of various codecs was analyzed to identify the most suitable codec for voice transmission.
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Munadi, Rendy, Iman Hedi Santoso, and Asep Mulyana. "Performance Evaluation for VoIP on Campus." INTERNATIONAL JOURNAL OF COMPUTERS & TECHNOLOGY 10, no. 9 (2013): 2027–35. http://dx.doi.org/10.24297/ijct.v10i9.1382.

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The VoIP Campus implementation is to make the existing VoIP technology become more beneficial for campus stake holder. This VoIP on Campus (VoC) technology make use of a web server, facilitating users to carry out VoIP registration, get and changing account, and also to see others who have register and active in this VoIP network. Basically, this VoC infrastructure uses asterisk as VoIP server and playVoIP as web server interface, those programs included in a server computer. Furthermore, the server interconnected with several servers, such as, PBX, SMS gateway, ENUM server, softphone and smartphone. At this moment, VoC network serve locally, but next time it will be developed so that it could be served in public network, and further VoC network could be connected to VoIP Rakyat, the biggest VoIP network in Indonesia. In this research, VoC network have been tested for several QoS parameters, such as throughput, delay, jitter, packet loss, and MOS. Average value for each parameter, are : 27 kbps throughput, 20.08 ms delay, 3.54 ms jitter, 0.08% packet loss, and 3.3 MOS. Those results indicates that VoC network have a good performance. Â
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Chen, Charlie C., Alanah Mitchell, and Scott Hunsinger. "Understanding Continuance of Using VoIP Applications to Improve Intercultural Communication." International Journal of Social and Organizational Dynamics in IT 2, no. 1 (2012): 1–16. http://dx.doi.org/10.4018/ijsodit.2012010101.

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Voice over Internet Protocol (VoIP) is a cost-effective medium to help learners improve their intercultural communication competency. However, the proliferation of VoIP applications has not accelerated the mass adoption of VoIP technology by users for the purposes of intercultural communication training. This study investigates the possibility of using VoIP technology as a learning tool to educate and train individuals to improve their intercultural communication. The perspectives of information and system qualities are adopted to investigate how to enhance users’ satisfaction and intention to reuse a VoIP technology for intercultural communication competencies. Information quality, system quality, and perceived task technology fit are important predictors of satisfaction. Satisfied users are more likely to continue using a VoIP technology. To test the authors’ expectations, 93 American and 45 Taiwanese subjects participated in a four-week experiment to virtually converse with each other via a VoIP technology (i.e., Skype). After working together one-on-one to communicate, participants completed a survey, resulting in 116 valid data points for analysis. The authors’ findings provide grounds for theoretical and practical implications concerning the adoption of VoIP technology by users as an e-learning tool.
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Putra, Dwi Prastantio. "ANALISIS KEAMANAN VOICE OVER INTERNET PROTOCOL (VOIP) OVER VIRTUAL PRIVATE NETWORK (VPN)." Jurnal Informatika dan Rekayasa Perangkat Lunak 2, no. 3 (2021): 324–33. http://dx.doi.org/10.33365/jatika.v2i3.1232.

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This research was conducted on the basis of the influence of security systems that play a role in encrypting data on VoIP communication systems, with the security technology of PPTP VPN computer networks with the system passing data in a virtual private ip or as a tunnel for secure data transmission media. The results obtained from security analysis for the implementation of security methods on PPTP VPNs, then the data to help developers in terms of building a secure VoIP communication system. Basiclly ZRTP uses the Diffie-Hellman key exchange as a key exchange of communication between clients, which is the key for communication between clients using hashes from Diffie-Hellman and is done peer-to-peer through the VOIP RTP package, while the Point-to-Point Tunneling Protocol (PPTP) is a network protocol that allows the secure transmission of data from the remote client to the server by creating a virtual private network (VPN) through a network of data. TCP/IP or UDP is dedicated to encryption and creates RTP tunnel transport on VoIP communication systems. VoIP communication system research is conducted using 2 security methods, namely VoIP VPN PPTP, VoIP ZRTP, with the aim to find out the results of VoIP communication testing using PPTP and ZRTP VPN security methods if an attack occurs during VoIP communication
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Alvianto, Richard, Samuel Hutagalung, and Franciscus Ati Halim. "RANCANG BANGUN MEKANISME QUALITY OF SERVICE TERHADAP PROTOKOL RTP DAN SIP PADA ARSITEKTUR OPENFLOW." Ultima Computing : Jurnal Sistem Komputer 11, no. 1 (2019): 9–15. http://dx.doi.org/10.31937/sk.v11i1.1093.

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Pada beberapa tahun terakhir, angka dari pengguna Voice Over Internet Protocol (VoIP) terus meningkat, dengan teknologi VoIP yang berkomunikasi melalui satu medium dalam jaringan. Hal ini tentu menimbulkan beberapa dampak terhadap VoIP seperti penggunaan bandwidth tidak terbagi dengan rata sesuai dengan kebutuhan masing-masing paket, dengan tuntutan VoIP yang membutuhkan delay, jitter, packet loss yang seminimal mungkin, untuk menjamin kualitas suara dan memberikan kenyamanan kepada pengguna VoIP. Pada penelitian ini dengan mekanisme Quality of Service (QoS) untuk memberikan prioritas terhadap protokol Real-time Transport Protocol (RTP) dan Session Initiation Protocol (SIP) dalam jaringan dirancang supaya kualitas VoIP tetap terjaga dan menghindari terjadi kemacetan terhadap paket RTP maupun SIP dalam proses antrian dalam jaringan. Analisis dalam penelitian ini dilakukan implementasikan pada emulator mininet dan diuji dengan beberapa parameter QoS, pada skenario mengujian jaringan tersebut dialiri paket dengan kecepatan 100 Mbps untuk menciptakan kondisi trafik yang padat dalam jaringan tersebut dan secara bersamaan dialiri juga trafik RTP, SIP dan data yang merupakan paket yang akan diukur nilai dari delay, jitter, packet loss. Hasil pengukuran dalam jaringan setelah diterapkan QoS menunjukan nilai dari delay, jitter, packet loss dapat berkurang dan juga memenuhi standar ITU-T G.1010 sehingga trafik VoIP dapat terjaga stabilitas dalam jaringan dan pengguna juga merasa nyaman, sedangkan pada kondisi jaringan tidak menerapkan QoS, trafik VoIP memperoleh nilai delay, jitter, packet loss yang cukup tinggi dan juga tidak memenuhi standar dari ITU-T G.1010 menyebabkan pengguna VoIP akan terganggu dengan keterlambatan dan terbuang paket VoIP yang membuat suara yang hilang dalam sebuah percakapan.
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Abhishek, R. Bhat, S. V. Abhishek, Acharya Akash, and P. S. Amruth. "Voice over Internet Protocol (VoIP) - A Review." International Journal of Innovative Science and Research Technology 7, no. 8 (2022): 850–53. https://doi.org/10.5281/zenodo.7045394.

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Voice over Internet protocol (VoIP), is a modern method of communication. The field of &quot;IP Telephony&quot; is growing more and more popular today. Voice over Internet Protocol (VoIP) is the term used to describe the transmission of voice messages over packetswitched IP networks. One of the most recent forms of communication is VoIP. VoIP presents opportunities as well as security challenges, as is the case with the majority of modern technology. In contrast to the traditional circuit-based telephony, it features a distinctive architecture. VoIP is hence vulnerable to a variety of security threats.
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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 (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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Basem, Basma, Atef Z. Ghalwash, and Rowayda A. Sadek. "Multilayer Secured SIP Based VoIP Architecture." International Journal of Computer Theory and Engineering 7, no. 6 (2015): 453–62. http://dx.doi.org/10.7763/ijcte.2015.v7.1002.

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