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Journal articles on the topic 'QoS, VoIP'

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Published: 30 May 2022

Last updated: 31 May 2022

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1

Susiani Pande, Putu Sintia, Pande Ketut Sudiarta, and I. Made Oka Widyantara. "PENGUKURAN KINERJA VOIP DENGAN CODEC G.711?, G.711a DAN G.729 DI MEDIA TRANSMISI NIRKABEL BERBASIS SIP DAN IAX." Jurnal SPEKTRUM 5, no. 1 (June 25, 2018): 21. http://dx.doi.org/10.24843/spektrum.2018.v05.i01.p04.

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Voice over Internet Protocol (VoIP) is a technology that can send real-time data with IP-based networks (Internet Protocol). In VoIP technology with wireless network has several problems that cause the performance of the network to be varied due to the QoS (Quality of Service) include delay, jitter, packet loss and MOS that affect the wireless network. This research uses G.711?, G.711a and G.729 codec based on SIP and IAX server on wireless network which then the QoS result from each codec compared with ITU-T standard which become the reference of whether the network is good or not so that later can realized on campus. In the research results, QoS on wireless IEEE 802.11 b has linear results, whereas QoS wireless in VoIP has fluctuating results because the use of codecs in VoIP on each codec has a large bitrate and different coding techniques and is a feature of wireless networks. The QoS comparison of three codecs produced the best G711 Q7S codecs because the G.711 codec has a bitrate that conforms to the 64 kbps voice communication standard and uses voice coding techniques that match the digital signal encoding technique of PCM (Pulse Code Modulation).

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2

Vakilinia, Shahin, Mohammadhossein Alvandi, Mohammadreza Khalili Shoja, and Iman Vakilinia. "Cross-Layered Secure and QoS Aware Design of VOIP over Wireless Ad-Hoc Networks." International Journal of Business Data Communications and Networking 9, no. 4 (October 2013): 23–45. http://dx.doi.org/10.4018/ijbdcn.2013100102.

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In this paper, Cross-layer design has been used to provide quality of service (QoS) and security at the same time for VOIP over the wireless ad-hoc network. In this paper the authors extend their previous work (i.e. Multi-path Multi-Channel Protocol Design for Secure QoS-Aware VOIP in Wireless Ad-Hoc Networks) by adding transport and application layers considerations. The goal of this paper is to support QoS and security of VOIP simultaneously. Simulation results shows that the proposed cross-layered protocol stack design significantly improve QoS parameters of the VOIP calls under the jamming or Denial-of-service attacks.

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3

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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4

Núñez Cuadrado, Marcelo David, Carlos Andres Jativa Huilcapi, and Román Alcides Lara Cueva. "Performance evaluation of VoIP technology in an extended service set, in concordance with IEEE 802.11g." Sistemas y Telemática 15, no. 42 (October 19, 2017): 85–100. http://dx.doi.org/10.18046/syt.v15i42.2541.

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In this paper, we evaluate the performance in function of the metrics associated to Quality of Service [QoS] and Quality of user Experience [QoE] in an experimental way in the VoIP service for G.711 and G.729 códecs. This was performed over an extended service set based on Wi-Fi technology in concordance with IEEE 802.11g standard using embedded systems. QoS related metrics are obtained by using the intrusive traffic injection technique. In addition, we assessed the QoE using the MOSc [Mean Opinion Score conversational] analysis. The best results were obtained for G.729, reaching up to 25 simultaneous injections with optimal delay, jitter and packet loss values according to the ITU-T recommendation for VoIP. However, the G.711 codec presented a better throughput. On the other hand, QoE evaluation indicates a slight superiority of G.729 in the MOSc appreciation. Finally, we conclude that packet loss and delay are the most influential metrics in VoIP service degradation.

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5

Liang, Jin Hua, and Xuan Zen. "Projects Improving QoS of the Voice Real-Time Data Based on IP Network." Applied Mechanics and Materials 220-223 (November 2012): 2471–74. http://dx.doi.org/10.4028/www.scientific.net/amm.220-223.2471.

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The reasons influencing the VoIP QoS include transmission delay, jitter and data packet drop. The main measures improving the VoIP QoS are the integrated services and the differentiated services. But the integrated service is only suitable for the small-scale network, and the differentiated services can’t guarantee QoS from the source end to the destination end for every IP data stream alone. The paper sets forth a kind of mixed model combining integrated Services with differentiated Services to support the VoIP QoS on the bases analysis of their defects.

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6

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 (August 30, 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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7

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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8

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 (July 9, 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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9

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 (November 1, 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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10

Luca, Robert, Petrica Ciotirnae, and Florin Popescu. "Influence of the QoS Measures for VoIP Traffic in a Congested Network." International Journal of Computers Communications & Control 11, no. 3 (March 24, 2016): 405. http://dx.doi.org/10.15837/ijccc.2016.3.2558.

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The paper revolves around the subject regarding quality of service (QoS) n a telecommunication network. The chosen scenario is based on the transmission of ata and voice packets using a WAN connection, which has a limited bandwidth and mphasize the need of implementing QoS mechanisms in order to fulfill the quality equirements of the traffic, especially for VoIP. This topology will outline the impact nd importance of the QoS implementation, illustrated by the desired quality resulted hrough VoIP traffic simultaneously with maintaining the data conectivity using a ower bandwidth for applications which require a smaller amount of QoS properties, uch as FTP.

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11

EL Khier, Mutasim Mohammed. "QoS of VOIP Over Broadband Network." International Journal of Engineering and Management Research 09, no. 05 (October 31, 2019): 122–25. http://dx.doi.org/10.31033/ijemr.9.5.17.

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12

Jalali, Nasir Ahmad, and Asadullah Tareen. "MPLS-VPN Impact on VOIP-QoS." International Journal of Computer Trends and Technology 67, no. 12 (December 25, 2019): 8–14. http://dx.doi.org/10.14445/22312803/ijctt-v67i12p103.

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13

Doshi, Bharat T., Dominik Eggenschwiler, Aswath Rao, Behrokh Samadi, Y. T. Wang, and James Wolfson. "VoIP network architectures and QoS strategy." Bell Labs Technical Journal 7, no. 4 (April 23, 2003): 41–59. http://dx.doi.org/10.1002/bltj.10033.

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14

Takahashi, A., H. Yoshino, and N. Kitawaki. "Perceptual QoS assessment technologies for VoIP." IEEE Communications Magazine 42, no. 7 (July 2004): 28–34. http://dx.doi.org/10.1109/mcom.2004.1316526.

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15

Rivas, F. Javier, Almudena Díaz, and Pedro Merino. "Obtaining More Realistic Cross-Layer QoS Measurements: A VoIP over LTE Use Case." Journal of Computer Networks and Communications 2013 (2013): 1–10. http://dx.doi.org/10.1155/2013/405858.

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We introduce a real-time experimentation testbed in this paper which enables more realistic analysis of quality of service (QoS) in LTE networks. This testbed is envisioned for the improvement of QoS and quality of experience (QoE) through the experimentation with real devices, services, and radio configurations. Radio configurations suggested in the literature typically arise from simulations; the testbed provides a real and controlled testing environment where such configurations can be validated. The added value of this testbed goes a long way not only in the provision of more realistic results but also in the provision of QoS and QoE cross-layer measurements through the correlation of information collected at different layers: from service and IP levels to radio and protocol parameters. Analyzing the interlayer dependencies will allow us to identify optimal settings for the radio access network and service parameters. This information can be used to suggest new cross-layer optimizations to further improve quality of experience of mobile subscribers. As a use case, we examine VoIP service over LTE, which is currently an open issue.

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Shi, Heng Hua, Xin Xu, Yu Jie Wang, and Yuan Yue Yang. "QoS Control Strategy Simulation and Analysis Based on DiffServ." Advanced Materials Research 538-541 (June 2012): 669–72. http://dx.doi.org/10.4028/www.scientific.net/amr.538-541.669.

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As the development and applying of real-time multimedia such as VoIP, video conferencing and so on, the Internet is required to provide a better QoS support. However, current IP network can only provide ‘Best Effort’ service and can not satisfy different network multimedia traffic quality requirements. Based on the analysis DiffServ architecture and studies its control mechanism, the OPNET simulation design different ToS value in IP header of VoIP traffic, and apply WFQ scheduling algorithm on the bottleneck link. The simulation results compare jitter, delay, delay variation of different ToS value of VoIP traffic, and analyze relationship of QoS and ToS value in the IP network.

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Zach, Petr, Martin Pokorný, and Jiří Balej. "Voice Quality Estimation in Wireless Networks." Acta Universitatis Agriculturae et Silviculturae Mendelianae Brunensis 63, no. 6 (2015): 2179–85. http://dx.doi.org/10.11118/actaun201563062179.

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This article deals with the impact of Wireless (Wi-Fi) networks on the perceived quality of voice services. The Quality of Service (QoS) metrics must be monitored in the computer network during the voice data transmission to ensure proper voice service quality the end-user has paid for, especially in the wireless networks. In addition to the QoS, research area called Quality of Experience (QoE) provides metrics and methods for quality evaluation from the end-user’s perspective. This article focuses on a QoE estimation of Voice over IP (VoIP) calls in the wireless networks using network simulator. Results contribute to voice quality estimation based on characteristics of the wireless network and location of a wireless client.

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Mushtaq, M. Sajid, Scott Fowler, Abdelhamid Mellouk, and Brice Augustin. "QoE/QoS-aware LTE downlink scheduler for VoIP with power saving." Journal of Network and Computer Applications 51 (May 2015): 29–46. http://dx.doi.org/10.1016/j.jnca.2014.02.001.

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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 (May 7, 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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Zotos, Nikolaos, Evangelos Pallis, and Anastasios Kourtis. "Performance Evaluation of Triple Play Services Delivery with E2E QoS Provisioning." International Journal of Digital Multimedia Broadcasting 2010 (2010): 1–14. http://dx.doi.org/10.1155/2010/836501.

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The creation and wide use of new high quality demanding services (VoIP, High Quality Video Streaming) and the delivery of them over already saturated core and access network infrastructures have created the necessity for E2E QoS provisioning. Network Providers use at their infrastructures several kinds of mechanisms and techniques for providing QoS. Most known and widely used technologies are MPLS and DiffServ. The IEEE 802.16-2004 standard (WiMAX) refers to a promising wireless broadband technology with enhanced QoS support algorithms. This document presents an experimental network infrastructure providing E2E QoS, using a combination of MPLS and DiffServ technologies in the core network and WiMAX technology as the wireless access medium for high priority services (VoIP, High Quality Video Streaming) transmission. The main scope is to map the traffic prioritization and classification attributes of the core network to the access network in a way which does not affect the E2E QoS provisioning. The performance evaluation will be done by introducing different kinds of traffic scenarios in a saturated and overloaded network environment. The evaluation will prove that this combination made feasible the E2E QoS provisioning while keeping the initial constrains as well as the services delivered over a wireless network.

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Li, Zhen, Qian Yi Yang, Yi Chen Zhou, and Hui Ren. "Research on QoS Guarantee Technology for Intercom System Based on SIP." Applied Mechanics and Materials 644-650 (September 2014): 2863–67. http://dx.doi.org/10.4028/www.scientific.net/amm.644-650.2863.

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Theatre intercom system based on SIP has solved the defects of traditional theater scheduling, but it doesn't guarantee the quality of service (QoS). This paper introduced the key technology for SIP-based theatre intercom system, including the fundamentals of VOIP and signaling technology. We analysed the main factors influencing QoS, summarized correlative techniques improving QoS for theatre intercom system, including QoS relevant protocols and QoS guarantee technology on terminals. These technologies can be synthetically used, so as to improve the reliability of theatre intercom system.

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Shin, Dong-Yun, and Young-Kil Kim. "Dynamic QoS Mechanism for supporting VoIP Service in Tactical Communication Environment." Journal of the Korean Institute of Information and Communication Engineering 16, no. 9 (September 30, 2012): 2078–83. http://dx.doi.org/10.6109/jkiice.2012.16.9.2078.

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Pertovt, Erik, Kemal Alic, Aleš Švigelj, and Mihael Mohorcic. "Performance Evaluation of VoIP Codecs over Network Coding in Wireless Mesh Networks." WSEAS TRANSACTIONS ON COMMUNICATIONS 20 (December 28, 2021): 185–91. http://dx.doi.org/10.37394/23204.2021.20.24.

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Voice over Internet protocol (VoIP) is used for transmitting voice signals in a packet-switched Internet protocol (IP) networks in real time. For transmitting voice over a wireless mesh networks (WMNs), the analog voice signal has to be digitalized, encoded, and packetized. Codecs based on different Quality of Service (QoS) requirements are used. One of the main QoS requirements is that packets are transmitted through the network in real time; one-way transmission time or End-to-End (ETE) packet delay, and packet delay variation or jitter have to be lower than thresholds. ETE delay depends on various parameters; among them is also network delay. Various mechanisms are used to lower the network delay in WMNs. A promising mechanism, for improving the performance of streaming services such as the case also in VoIP, is network coding. In this paper, we evaluate the benefits of using wireless network coding for VoIP in WMNs. Network coding procedure in combination with various VoIP codecs is used to observe the impact on network delay and jitter of the VoIP application. The simulation results show that network coding can decrease network delay and jitter. Moreover, results show that network coding benefits are codec dependent.

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Lazzez, Amor. "VoIP Technology: Investigation of QoS and Security Issues." International Journal of Information Technology and Computer Science 6, no. 7 (June 8, 2014): 65–76. http://dx.doi.org/10.5815/ijitcs.2014.07.09.

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25

Hu, ZhiGuo, HongRen Yan, Tao Yan, HaiJun Geng, and GuoQing Liu. "Evaluating QoE in VoIP networks with QoS mapping and machine learning algorithms." Neurocomputing 386 (April 2020): 63–83. http://dx.doi.org/10.1016/j.neucom.2019.12.072.

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Pratama, Aditya. "Practicum Module Design and Comparative Analysis of Codecs for VoIP." Jurnal Jartel: Jurnal Jaringan Telekomunikasi 3, no. 2 (November 7, 2016): 7–13. http://dx.doi.org/10.33795/jartel.v3i2.214.

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Several types of codecs in VoIP are G.711, iLBC, GSM and Speex. This study aims to compare the quality of calls through different codecs and design a VoIP practicum module for students. The research method was carried out by conducting literature studies, planning network design, device configuration, determining QoS parameters and testing VoIP calls to determine the quality of VoIP calls through different codecs. From the analysis, it is found that G.711 is an audio codec with very good QoS parameters, namely a delay of 5.28 ms; jitter 5.12 ms; 347.04 kbps throughput, 0% packet loss, and 64 kbps bandwidth. while for audio codec with good enough parameters, namely GSM with a delay of 42.02 ms; jitter 19.46 ms; 39.18 kbps throughput, 0% packet loss and 13kbps bandwidth. In addition to data testing, non-technical testing was carried out on 16 respondents with pre-test, post-test and questionnaire testing. From the pre-test, respondents got an average score of 45, while in the post-test it increased to 70.

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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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Abrar, Muhammad Saleh, and Rudy Rudy. "Implementasi dan Analisa Kinerja VOIP Server Pada Jaringan Wireless LAN Menggunakan Smartphone." Elektrika Borneo 5, no. 1 (April 10, 2019): 1–5. http://dx.doi.org/10.35334/jeb.v5i1.587.

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Penelitian ini bertujuan untuk mengimplementasikan VoIP Server Pada jaringan Wireless LAN di Fakultas Teknik Universitas Borneo Tarakan dengan menggunkan Elastix sebagai server dan aplikasi VoIP Call pada Smartphone menggunakan CsipSimple serta menganalisa kinerja dari server tersbut dengan perangkat lunak Wireshark. Metode yang digunakan dalam penelitian ini yakni dengan metode pengukuran kualitas layanan suara atau QoS (Quality of Service). Pengujian dilakukan indoor dan outdoor. Dengan parameter QoS sepeti delay, throughput, dan packet loss dapat dijadikan sebagai ukuran untuk mengetahui kualitas dari suatu jaringan. Delay yang dihasilkan paling besar di pengujian indoor dengan jarak 11-15 meter yakni sebesar 0.00956464 seconds. Packet loss yang dihasilkan pada range 0,00%, sedangkan standar packet loss yang ditetapkan oleh ITU-T untuk layanan aplikasi VoIP adalah 3%. Jitter yang dihasilkan yakni antara 0,04608 – 0.09485 seconds sedangkan standar yang ditetapkan oleh ITU-T adalah = 0–75 ms.. Throughput yang dihasilkan pada proses pengujian yakni antar 104,551 kbps - 108,905 kbps

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Kim, Beom-Joon. "Study on QoE of the VoIP Service for QoS levels over LTE Mobile Communication System." Journal of the Korea institute of electronic communication sciences 11, no. 3 (March 31, 2016): 309–16. http://dx.doi.org/10.13067/jkiecs.2016.11.3.309.

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El brak, Said, Mohamed El brak, and Driss Benhaddou. "A New QoS Management Scheme for VoIP Application over Wireless Ad Hoc Networks." Journal of Computer Networks and Communications 2014 (2014): 1–10. http://dx.doi.org/10.1155/2014/945695.

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Nowadays, mobile ad hoc networks (MANETs) have to support new applications including VoIP (voice over IP) that impose stringent QoS (quality of service) constraints and requirements. However, VoIP applications make a very inefficient use of the MANET resources. Our work represents a first step toward improving aspects at the network layer by addressing issues from the standpoint of adaptation, claiming that effective adaptation of routing parameters can enhance VoIP quality. The most important contribution is the adaptive OLSR-VA algorithm which provides an integrated environment where VoIP activity is constantly detected and routing parameters are adapted in order to meet the application requirements. To investigate the performance advantage achieved by such algorithm, a number of realistic simulations (MANET scenarios) are performed under different conditions. The most important observation is that performance is satisfactory in terms of the perceived voice quality.

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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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Gohel, Chirag K., and Kamaljit I. Lakhtaria. "Implement VoIP Based IP Telephony with Open Source Asterisk Architecture." International Journal of Interdisciplinary Telecommunications and Networking 2, no. 1 (January 2010): 1–11. http://dx.doi.org/10.4018/jitn.2010010101.

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Asterisk is a leading open source telephony software/system, easily implemented over intranet and internet. Asterisk empowers developers and integrators to create advanced communication solutions. An Asterisk system is a low cost type of a traditional PBX system. Any phone controlled by an Asterisk system can call a VoIP or analog phone controlled or managed by a traditional telephone system or by Asterisk telephone system. In this paper, the authors focus on the deployment and testing of various Open Source Asterisk Services in an enterprise level communication system. Selected services are listed in this paper that can be used to implement a telephone system with good Quality of Services (QoS) and good Quality of Experience (QoE) from the personal user to enterprise level users.

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Meeran, Mohammad, Paul Annus, Muhammad Alam, and Yannick Moullec. "Evaluation of VoIP QoS Performance in Wireless Mesh Networks." Information 8, no. 3 (July 21, 2017): 88. http://dx.doi.org/10.3390/info8030088.

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Saleh, Saad, Zawar Shah, and Adeel Baig. "Improving QoS of IPTV and VoIP over IEEE 802.11n." Computers & Electrical Engineering 43 (April 2015): 92–111. http://dx.doi.org/10.1016/j.compeleceng.2014.10.017.

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Chang, Lin-huang, Tsung-Han Lee, Hung-Chi Chu, Yu-Lung Lo, and Yu-Jen Chen. "QoS-aware path switching for VoIP traffic using SCTP." Computer Standards & Interfaces 35, no. 1 (January 2013): 158–69. http://dx.doi.org/10.1016/j.csi.2012.06.003.

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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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Harahap, Yoga Pradafa, and Aditya Prapanca. "Analisis Algoritma Penjadwalan Priority Queueing (PQ) terhadap Quality of Service (QoS) pada Jaringan Mobile WiMAX menggunakan OPNET Modeler." Journal of Informatics and Computer Science (JINACS) 3, no. 02 (September 8, 2021): 104–12. http://dx.doi.org/10.26740/jinacs.v3n02.p104-112.

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Kualitas sebuah layanan pada jaringan mobile WiMAX bergantung pada service class dan penjadwalan yang digunakan. Priority Queueing merupakan algoritma penjadwalan dimana sebuah antrian diurutkan berdasarkan urutan antrian, paket dengan prioritas tinggi akan dipilih dan mendapatkan urutan paling awal. Adanya algoritma penjadwalan bertujuan untuk adil terhadap QoS pada layanan-layanan yang digunakan seperti FTP, HTTP, video conference dan VoIP. Pada simulator OPNET Modeler yang digunakan layanan menggunakan Type of Service untuk mengidentifikasi sebuah antrian. Lalu layanan tersebut disesuaikan dengan service class yang digunakan pada WiMAX. Hasilnya, pada penggunaan algoritma Priority Queueing nilai throughput terbaik pada masing-masing layanan yaitu, layanan FTP sebesar 2259.30954 bps, layanan HTTP sebesar 3112.70472 bps, video conference sebesar 4000.45667 bps dan, layanan VoIP sebesar 3200.45111 bps. Nilai delay terbaik pada masing-masing layanan yaitu, layanan FTP sebesar 5255.47904 ms, layanan HTTP sebesar 587.16361 ms, layanan video conference sebesar 2.28657 ms dan, layanan VoIP sebesar 260.62925 ms. Nilai jitter terbaik pada masing-masing layanan yaitu, layanan FTP sebesar 0.0000000000142109 ms, layanan HTTP sebesar 0.04461279 ms, layanan video conference sebesar 0.000164906 ms dan, layanan VoIP sebesar 0.00009812018 ms. Nilai packet loss pada masing-masing layanan yaitu, layanan FTP sebesar 0%, layanan HTTP sebesar 0%, layanan video conference sebesar 0.02830% dan, layanan VoIP sebesar 0.00353%. Jadi rata-rata nilai QoS pada 4 layanan yang diuji mendapatkan nilai 3.1875 masuk pada kategori ”Memuaskan”. Kata Kunci— Mobile WiMAX, Priority Queueing, Throughput, Delay, Jitter, Packet Loss.

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Noh, Si-Choon, and Kee-Chun Bang. "A Study on Designing Method of VoIP QoS Management Framework Model under NGN Infrastructure Environment." Journal of Digital Contents Society 12, no. 1 (March 31, 2011): 85–94. http://dx.doi.org/10.9728/dcs.2011.12.1.085.

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Chakraborty, Tamal, Shubhadip Ghosh, Sayanta Barik, Sampriya Kar, and Subhadip Chatterjee. "VoIP-HDK – A novel channel allocation technique for QoS aware VoIP communication over heterogeneous networks." Procedia Computer Science 171 (2020): 62–71. http://dx.doi.org/10.1016/j.procs.2020.04.007.

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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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Abdullah, Hernan Malik. "Perancangan Jaringan Voice Over IP (VoIP) Berbasis Raspberry Pi Untuk Sistem Komunikasi Area Remote." TELKA - Telekomunikasi, Elektronika, Komputasi dan Kontrol 2, no. 1 (May 22, 2016): 36–43. http://dx.doi.org/10.15575/telka.v2i1.12.

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Telekomunikasi saat ini sudah menjadi kebutuhan dasar manusia. Infrastruktur telekomunikasi dibangun di seluruh negeri untuk melayani kebutuhan tersebut. Namun area layanan yang ada masih belum bisa menjangkau seluruh wilayah terutama daerah-daerah terpencil (remote). Penelitian ini bertujuan untuk merancang sistem komunikasi berbasis VoIP yang bisa dimanfaatkan untuk komunikasi antar warga terutama di area yang minim infrastruktur telekomunikasi. Server VoIP dibuat menggunakan Raspberry Pi. Sedangkan fungsi router dan pemancar menggunakan Mikrotik Routerbord 433 dengan antena eksternal. Sedangkan perangkat yang digunakan oleh pengguna (client) bisa berupa handphone android, laptop maupun personal komputer. Hasil pengujian menunjukkan bahwah panggilan dan percakapan bisa dilayani oleh server Voip. Analisa QOS dengan metode MOS menunjukkan bahwa server VoIP hanya bisa melayani 10 percakapan pada saat bersamaan. Adapun jarak jangkau layanan untuk daerah LOS bisa mencapaijarak 500 meter.

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42

Abdullah, Hernan Malik. "Perancangan Jaringan Voice Over IP (VoIP) Berbasis Raspberry Pi Untuk Sistem Komunikasi Area Remote." TELKA - Telekomunikasi, Elektronika, Komputasi dan Kontrol 2, no. 1 (May 22, 2016): 36–43. http://dx.doi.org/10.15575/telka.v2n1.36-43.

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Telekomunikasi saat ini sudah menjadi kebutuhan dasar manusia. Infrastruktur telekomunikasi dibangun di seluruh negeri untuk melayani kebutuhan tersebut. Namun area layanan yang ada masih belum bisa menjangkau seluruh wilayah terutama daerah-daerah terpencil (remote). Penelitian ini bertujuan untuk merancang sistem komunikasi berbasis VoIP yang bisa dimanfaatkan untuk komunikasi antar warga terutama di area yang minim infrastruktur telekomunikasi. Server VoIP dibuat menggunakan Raspberry Pi. Sedangkan fungsi router dan pemancar menggunakan Mikrotik Routerbord 433 dengan antena eksternal. Sedangkan perangkat yang digunakan oleh pengguna (client) bisa berupa handphone android, laptop maupun personal komputer. Hasil pengujian menunjukkan bahwah panggilan dan percakapan bisa dilayani oleh server Voip. Analisa QOS dengan metode MOS menunjukkan bahwa server VoIP hanya bisa melayani 10 percakapan pada saat bersamaan. Adapun jarak jangkau layanan untuk daerah LOS bisa mencapaijarak 500 meter.

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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 (September 15, 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 resultsÂ indicates that VoC network have a good performance.Â Â

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Hoßfeld, Tobias, and Andreas Binzenhöfer. "Analysis of Skype VoIP traffic in UMTS: End-to-end QoS and QoE measurements." Computer Networks 52, no. 3 (February 2008): 650–66. http://dx.doi.org/10.1016/j.comnet.2007.10.008.

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Kim, Bosung, Gyu-Min Lee, Byeong-Hee Roh, Geunkyung Choi, and Ilhyuk Oh. "Resource Allocation and Control System for VoIP QoS Provision in Cognitive Radio Networks." KIISE Transactions on Computing Practices 20, no. 12 (December 15, 2014): 688–93. http://dx.doi.org/10.5626/ktcp.2014.20.12.688.

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Abu Samra, Aiman Ahmed. "Performance Evaluation of the QoS for VoIP using Different CODECS." International Journal for Research in Applied Science and Engineering Technology V, no. VIII (August 30, 2017): 1636–40. http://dx.doi.org/10.22214/ijraset.2017.8232.

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Refaet, Ahmmed, Muhanad A. Ahmed, Qais Aish, and Ali K. Jasim. "VoIP Performance Evaluation and Capacity Estimation Using different QoS Mechanisms." IOP Conference Series: Materials Science and Engineering 881 (August 11, 2020): 012146. http://dx.doi.org/10.1088/1757-899x/881/1/012146.

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Cai, Lin, Yang Xiao, Xuemin (Sherman) Shen, Lin Cai, and Jon W. Mark. "VoIP over WLAN: voice capacity, admission control, QoS, and MAC." International Journal of Communication Systems 19, no. 4 (2006): 491–508. http://dx.doi.org/10.1002/dac.801.

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49

Chua, Teck-kuen, and David Pheanis. "QoS evaluation of sender-based loss-recovery techniques for VoIP." IEEE Network 20, no. 6 (November 2006): 14–22. http://dx.doi.org/10.1109/mnet.2006.273116.

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Skinnemoen, H., A. Vermesan, A. Iuoras, G. Adams, and X. Lobao. "VoIP over DVB-RCS with QoS and bandwidth on demand." IEEE Wireless Communications 12, no. 5 (October 2005): 46–53. http://dx.doi.org/10.1109/mwc.2005.1522104.

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