Comparison of QoS Performance Over WLAN, VoIP4 and VoIP6

2015 ◽  
Vol 2 (11) ◽  
pp. 42
Author(s):  
Esra Musbah Mohammed Musbah ◽  
Khalid Hamed Bilal ◽  
Amin Babiker A. Nabi Mustafa
Keyword(s):  
Performance Analysis ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Ip Network ◽  
Qos Parameters ◽  
Software Program ◽  
Jitter Delay ◽  
Qos Performance ◽  
Delay Variation ◽  
A Performance

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.

scholarly journals Implementasi Sistem IP PBX menggunakan Briker

2013 ◽  
Vol 1 (2) ◽  
pp. 117
Author(s):  
DWI ARYANTA ◽  
ARSYAD RAMADHAN DARLIS ◽  
ARDHIANSYAH PRATAMA
Keyword(s):  
Operating System ◽  
Quality Of Service ◽  
Packet Loss ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Long Distance ◽  
Qos Parameters ◽  
Jitter Delay ◽  
Ip Phone

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.

PENGARUH PENGGUNAAN CODEC STANDART ITU G.729 TERHADAP SISTEM KOMUNIKASI VOIP

2021 ◽  
Vol 17 (1) ◽  
pp. 11-22
Author(s):  
Wahyu Adi Prijono
Keyword(s):  
Communication System ◽  
Communication Systems ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Data Communications ◽  
Voice Communication ◽  
Internet Telephony ◽  
Voice Signal ◽  
Ip Network ◽  
Packet Switch

Voice over Internet Protocol (VoIP) is a technology that is capable of passing voice traffic, in the form of packets through the network Internet Protocol (IP). IP network itself is a data communications network based packet-switch. The voice signal before experiencing bundled voice coding or format conversion of sound into digital form that can be passed over an IP network. Telephony, Internet telephony, or termed VoIP (Voice Over Internet Protocol.This communication system use VoIP (Voice over Internet Protocol), ie voice calls over data services (internet). This communication was developed using Android-based devices. Based on characteristics, android devices are open source, so users do not need to have a license to be able to have android-based devices. In addition, the android device that must be connected to a SIP (Session Iniation Protocol) is a data service that can be done with a paid subscription of the user of the operator using a conventional pulse. Telecommunications designed will use a hybrid system, the merger between VoIP communications with data communications GSM network. With basic calculations where Coding standards G 729, is a standard that can be used for voice communication system through data networks with rate of 8 Kbps. The implementation of the G729 codec is effect on communication systems VOIP.

QoS Control Strategy Simulation and Analysis Based on DiffServ

2012 ◽  
Vol 538-541 ◽  
pp. 669-672 ◽  
Author(s):  
Heng Hua Shi ◽  
Xin Xu ◽  
Yu Jie Wang ◽  
Yuan Yue Yang
Keyword(s):  
Control Mechanism ◽  
Scheduling Algorithm ◽  
Multimedia Traffic ◽  
Simulation Design ◽  
Ip Network ◽  
Jitter Delay ◽  
Bottleneck Link ◽  
Relationship Of ◽  
Diffserv Architecture ◽  
Delay Variation

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.

Performance analysis of voice over internet protocol via non Markovian loss system with preemptive priority and server break down

OPSEARCH ◽  
2013 ◽  
Vol 51 (1) ◽  
pp. 50-75 ◽  
Author(s):  
A. Senthil Vadivu ◽  
Resham Vinayak ◽  
S. Dharmaraja ◽  
R. Arumuganathan
Keyword(s):  
Performance Analysis ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Preemptive Priority ◽  
Loss System

scholarly journals Pengaruh Kinerja QoS Untuk Layanan VoIP Berdasarkan Jarak Dari Base Station Pada Jaringan WiMAX

2018 ◽  
Vol 4 (2) ◽  
Author(s):  
Murhaban Murhaban ◽  
Muhammad Bilai ◽  
Muhammad Nurtanzia Sutoyo
Keyword(s):  
Quality Of Service ◽  
Internet Protocol ◽  
Mobile Station ◽  
Voice Over Ip ◽  
Base Station ◽  
Voice Over Internet Protocol ◽  
Soft Handover ◽  
Jitter Delay ◽  
Hard Handover

Metode Handover digunakan untuk mempertahankan koneksi tetap terjaga. Hal tersebut berkaitan dengan performansi dikarenakan proses pengalihan kanal trafik secara otomatis pada mobile station untuk berkomunikasi tanpa terjadinya pemutusan hubungan. Faktor utama keberhasilan dalam melakukan handover terletak pada quality of service yang menyediakan tingkat jaminan layanan berbeda-beda dalam mengatur dan memberikan prioritas trafik pada jaringan seperti aplikasi voice over IP (VoIP) atau komunikasi voice memanfaatkan jaringan internet dalam permasalahan berdasarkan jarak base station.. Berdasarkan pengujian yang dilakukan untuk metode hard handover dan metode soft handover berdasarkan jarak base station menggunakan aplikasi voice over internet protocol pada mobile station. Diperoleh hasil dengan  nilai jitter 0.015 ms – 0.21 ms, dan hasil delay 35.5 ms – 45.8 ms hal tersebut membuktikan bahwa pengaruh jitter dan delay terhadap handover dengan aplikasi VoIP masih dalam tahapan toleransi yang diizinka. Dan berdasarkan hasil penelitian ini jarak antara satu base station dengan station lainnya sangat berpengaruh untuk mendapatkan kulaitas layanan yang lebih baik. Kata Kunci : Handover, Jitter, Delay, VoIP, Quality of Service

Implementasi Voip Server Berbasis IPV6 Dengan Raspberry PI

2019 ◽  
Vol 9 (01) ◽  
pp. 47-54
Author(s):  
Rabbai San Arif ◽  
Yuli Fitrisia ◽  
Agus Urip Ari Wibowo
Keyword(s):  
Internet Protocol ◽  
Wireless Transmission ◽  
Raspberry Pi ◽  
Test Results ◽  
Average Delay ◽  
Ipv6 Network ◽  
Ip Network ◽  
Selection Of ◽  
Better Than

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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