scholarly journals ANALISIS DAN PERBANDINGAN QUALITY OF SERVICE VIDEO CONFERENCE JITSI DAN BIGBLUEBUTTON PADA VIRTUAL PRIVATE SERVER

2021 ◽  
Vol 4 (2) ◽  
pp. 192-203
Author(s):  
Ida Bagus Ary Indra Iswara ◽  
I Putu Pedro Kastika Yasa

The use of video conferencing technology is increasing due to the COVID-19 pandemic. Bigbluebutton and jitsi are examples of open source video conferencing platforms that can be installed on their own servers. The server is created using a cloud-based virtual machine. Analysis of quality of service which includes delay, packet loss, throughput, and jitter is needed to determine the quality of service and the comparison of the two platforms. Observations were also made on the use of CPU, memory / RAM, and disk usage for each server. There are 3 test scenarios carried out. Each scenario is carried out on each existing VM specification. From this test, it is known that in the delay parameter, the highest bigbluebutton is obtained, which is 35,35 ms. And then the highest jitsi delay is 17,66 ms. In packet loss parameters, jitsi obtained the highest yield, namely 0,29%, while for bigbluebutton only 0,16% of packet loss was the highest. Throughput, bigbluebutton and jitsi all got very bad results. However, bigbluebutton obtained better results, namely, the highest throughput was 5.6%. While Jitsi obtained the highest throughput, namely 2,8%. Whereas for the jitter parameter, jitsi obtained 0,00 ms results on all tests in each VM. Meanwhile, bigbluebutton, get 0,1 ms on test 3 on VM 1

Author(s):  
Ibel Dwi Amiza ◽  
Lindawati Lindawati ◽  
Sopian Soim

Abstract  —  Video Conference is a communication service that can be used to bring together two users (client) or more. Video conferencing can be used for a variety of activities that require communication in real-time without having to come face to face directly. One open-source that can be utilized as a means of communicating is OpenMeetings. OpenMeetings uses IP and in the same network as a means of conducting video conferencing between clients. But if a client is not in the same network, it can utilize the Virtual Private Network (VPN) technology. The VPN can be remote by using a MikroTik router. The Video conferencing service requires fairly high and stable connectivity. Quality of Service (QoS) can be used whether the network is eligible for video conferencing. The QoS parameters used are throughput and packet loss. The QoS test can be done using Wireshark.


2019 ◽  
Vol 5 (2) ◽  
pp. 682
Author(s):  
Sussi . . ◽  
Rendy Munadi . ◽  
Nurwulan Fitriyanti . ◽  
Indra Perdana Putra Sutejo

Perkembangan dunia industri game semakin semarak dengan munculnya teknologi jaringan dibidang Cloud Gaming. Platform Cloud Gaming yang sering digunakan dan sifatnya open source yaitu GamingAnywhere. Dengan menggunakan cloud gaming GamingAnywhere dan platform speech recognition system FreeePIE, client dapat memainkan game berspesifikasi tinggi pada perangkat miliknya yang berspesifikasi lebih rendah dengan sistem input menggunakan perintah suara. Pencinta game yang memiliki kerusakan motoris tangan masih bisa menikmati game dengan inputan suara. Penelitian akan Cloud Gaming yang ada masih terbatas, karena teknologi Cloud Gaming merupakan teknologi yang baru (2013). Penelitian ini ditujukan untuk memberikan informasi mengenai Quality of Service (QoS) dari GamingAnywhere. Dari hasil pengukuran, untuk meraih QoS yang optimal dalam menjalankan game dengan cloud gaming GamingAnywhere, dibutuhkan minimal bandwidth sebesar 3 Mbps. Bila bandwidth yang diberikan kurang dari 3 Mbps, sistem akan mengalami delay yang massif bernilai ± 0.5 detik pada game NEVERBALL dan bernilai ± 1.9 detik pada game 7 Days to Die dan packet loss yang dihasilkan pun akan sangat tinggi.


Author(s):  
Bongga Arifwidodo ◽  
Wasis Rezki Baskoro ◽  
Jafaruddin Gusti Amri Ginting

Era Pandemi menyebabkan peningkatan penggunaan aplikasi konferensi video yang tiba-tiba dan siginifikan. Bagi perusahaan harus segera beradaptasi dalam hal aplikasi komunikasi. Salah satu aplikasi yang bisa digunakan yaitu openmeetings. Openmeetings adalah aplikasi yang digunakan sebagai pengatur konferensi yang terinstal pada server. Umumnya server dibangun menggunakan komponen fisik, namun memiliki keterbatasan sehingga seringkali mengalami penurunan performa dalam segi kecepatan server dalam menjalankan layanan. Salah satu upaya meningkatkan performa server harus menambah atau mengganti perangkat keras sehingga kurang menguntungkan pada biaya operasional. Konsep Cloud selain dapat mengefisienkan biaya operasional server juga handal dalam hal ketersedian layanan. Cloud merupakan sebuah model Client-server, dapat diakses oleh pengguna dimana saja dan kapan saja. Membangun Cloud salah satunya dapat menggunakan openstack, merupakan software open source untuk membangun cloud. Penelitian dilakukan perbandingan harga cloud dengan komponen fisik serta pengujian Video Conference yang di jalankan dalam Cloud untuk mengetahui kinerja Cloud dari sisi Quality of Service (QoS) meliputi delay, packetloss, jitter dan throughput. Hasilnya disimpulkan menggunakan cloud lebih efisien dibandingkan dengan menggunakan server fisik. nilai rata-rata terbesar pada sisi upload adakah throughput sebesar 639.85kbps, delay sebesar 30.22ms, jitter sebesar 10.35ms dan packetloss sebesar 0.94%, untuk nilai rata-rata terbesar pada sisi download adalah throughput sebesar 1,856.55kbps, delay sebesar 10.19ms, jitter sebesar 6.18ms dan packetloss sebesar 0.87%.


2021 ◽  
Vol 1 (2) ◽  
Author(s):  
EL Rama Janistimewa Nuris Sani ◽  
Mohammad Suryawinata

Video Conference is an interactive telecommunication technology or system utilizing the internet network so that it is able to interact with audio video with two or more people without considering the distance or different locations, but video conference requires a stable internet network in accordance with the parameters set by Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON) and to determine the quality of video conference can use Quality of Service (QoS). The purpose of this study is to determine delay, throughput, jitter, and packet loss and if someone is doing video conference services on the network, their use can be further enhanced through QoS than other users who only do ordinary internet services (browsing, chatting, etc.) on the network the same network. Here the author uses the Quality of Service (QoS) method as a regulator of internet bandwidth to maximize the use of video conferencing services so that they can run stably when the flow of internet traffic is quite dense. The results of this study are based on the standard TIPHON calculation using the QoS method in testing more or less affect the quality of the video conference service with the evidence in the packet loss recorded at the time of upload and download, mostly touching the 0% figure compared to without using QoS. That way using the help of the QoS method can increase the level of quality and effectiveness in conducting video conference services.


Author(s):  
Amang Sudarsono ◽  
Anang Siswanto ◽  
Heru Iswanto ◽  
Qoirul Setiawan

Recently, in the distance learning system, video conferencing becomes one of expected course material delivery systems for creating a virtual class such that lecturer and student which are separated at long distance can engage a learning activity as well as face to face learning system. For this reason, the service availability and quality should be able to guaranteed and fulfilled. In this research, we analyze QoS of video conferencing between main campus and sub campus as the implementation of distance learning system in laboratory scale. Our experimental results show that the channel capacity or bandwidth of WAN connection between main campus and sub campus at 128 kbps is able to generate the throughput of video transmission and reception at 281 kbps and 24 kbps, respectively. Meanwhile, throughput of audio transmission and reception is 64 kbps and 26 kbps with the number of total packet loss for video and audio transmission is 84.3% and 29.2%, respectively. In this setting, the total jitter for video and audio transmission is 125 ms and 21 ms, respectively. In this case, there is no packet loss for traffic transmitting and receiving with jitter is not more than 5 ms. We also implemented QoS using Trust CoS model dan Trust DSCP for improving the quality of service in term of jitter up to 12.3% and 22.41%, respectively.Keywords: quality of service, throughput, delay, jitter, packet loss, Trust CoS, Trust DSCP


2016 ◽  
Vol 26 (1) ◽  
pp. 7
Author(s):  
Jose Carlos Tavara Carbajal

RESUMENEste documento tiene como objetivo analizar el comportamiento de la calidad del servicio del protocolo IPv6 sobre el tráfico de video, para esto se realizó sobre un entorno real y se llevó acabo el análisis de resultados a través de un software estadístico de control del tráfico.Palabras Clave.-  Calidad de Servicio, Ancho de Banda, Retardo, Fluctuación de Retardo, Pérdidas de Paquetes.ABSTRACTThis paper has aimed to analyze of the service quality of the IPv6 protocol on video traffic, this was about a real environment and was conducted analysis of results through statistical traffic control software. Key words- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.


2016 ◽  
Vol 26 (1) ◽  
pp. 1
Author(s):  
Jose Carlos Tavara Carbajal

Este documento tiene como objetivo analizar el comportamiento de la calidad del servicio del protocolo IPv6 sobre el tráfico de video, para esto se realizó sobre un entorno real y se llevó acabo el análisis de resultados a través de un software estadístico de control del tráfico.Palabras Clave.-  Calidad de Servicio, Ancho de Banda, Retardo, Fluctuación de Retardo, Pérdidas de Paquetes.ABSTRACT  This paper has aimed to analyze of the service quality of the IPv6 protocol on video traffic, this was about a real environment and was conducted analysis of results through statistical traffic control software.  Key words.- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.


2016 ◽  
Vol 26 (1) ◽  
pp. 7
Author(s):  
Jose Carlos Tavara Carbajal

RESUMENEste documento tiene como objetivo analizar el comportamiento de la calidad del servicio del protocolo IPv6 sobre el tráfico de video, para esto se realizó sobre un entorno real y se llevó acabo el análisis de resultados a través de un software estadístico de control del tráfico.Palabras Clave.-  Calidad de Servicio, Ancho de Banda, Retardo, Fluctuación de Retardo, Pérdidas de Paquetes.ABSTRACTThis paper has aimed to analyze of the service quality of the IPv6 protocol on video traffic, this was about a real environment and was conducted analysis of results through statistical traffic control software. Keywords- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.


Author(s):  
Ida Bagus Ary Indra Iswara ◽  
I Gusti made Ngurah Desnanjaya ◽  
Ida Bagus Gede Sarasvananda ◽  
I Gede Adnyana ◽  
I Dewa Putu Gede Wiyata Putra

2017 ◽  
Vol 3 (2) ◽  
pp. 249-254
Author(s):  
Darmawan Darmawan ◽  
Yayan Syafriyatno

Voice over IP (VoIP) adalah solusi komunikasi suara yang murah karena menggunakan jaringan IP dibanding penggunaan telephone analog yang banyak memakan biaya. Dalam penerapannya, VoIP mengalami permasalahan karena menggunakan teknologi packet switching yang mana penggunaannya bersamaan dengan paket data sehingga timbul delay, jitter, dan packet loss.  Pada penelitian ini, algoritma Low Latency Queuing (LLQ) diterapkan pada router cisco. Algoritma LLQ merupakan gabungan dari algoritma Priority Queuing (PQ) dan Class Based Weight Fair Queuing (CBWFQ) sehingga dapat memprioritaskan paket suara disamping paket data. Algoritma LLQ ini diujikan menggunakan codec GSM FR, G722, dan G711 A-law. Hasil pengujian didapatkan nilai parameter yang tidak jauh berbeda dan memenuhi standar ITU-T.G1010. Nilai delay rata - rata terendah yaitu ketika menggunakan codec G722 sebesar 20,019 ms tetapi G722 memiliki rata - rata jitter yang terbesar yaitu 0,986 ms.  Codec dengan jitter rata – rata terkecil adalah G711 A-law sebesar 0,838 ms. Packet loss untuk semua codec yang diujikan adalah 0%.  Throughput pada paket data terbesar saat menggunakan codec GSM FR yaitu 18,139 kbps. Codec yang direkomendasikan adalah G711 A-law karena lebih stabil dari segi jitter dan codec GSM FR cocok diimplementasikan pada jaringan yang memiliki bandwitdh kecil.


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