scholarly journals Performance Evaluation for VoIP on Campus

2013 ◽  
Vol 10 (9) ◽  
pp. 2027-2035
Author(s):  
Rendy Munadi ◽  
Iman Hedi Santoso ◽  
Asep Mulyana
Keyword(s):  
Performance Evaluation ◽  
Packet Loss ◽  
Web Server ◽  
Public Network ◽  
Average Value ◽  
Delay Jitter ◽  
Qos Parameters ◽  
Stake Holder ◽  
Server Computer

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.  

scholarly journals Análisis basado en parámetros QoS para el servicio VoIP sobre wireless en un campus universitario [Analysis based on QoS parameters for VoIP service on a wireless network within a university campus]

2014 ◽  
Author(s):  
Alexander Olave ◽  
Luis Felipe Valencia ◽  
Juan Carlos Cuéllar
Keyword(s):  
Quality Of Service ◽  
Wireless Network ◽  
Packet Loss ◽  
Cost Benefit ◽  
Access Point ◽  
University Campus ◽  
Delay Jitter ◽  
Qos Parameters ◽  
La Red

Resumen Voz sobre IP, VoIP, es uno de los servicios con mayor desarrollo bajo plataformas inalámbricas; actualmente se ha iniciado su implementación como alternativa frente a la PSTN (red pública conmutada). El interés por VoIP radica en su relación costo-beneficio, ya que las organizaciones pueden utilizar la misma plataforma de su red de datos para transmitir voz. Por lo anterior, es importante que la organización tenga claro que, para garantizar el buen funcionamiento del servicio de VoIP, es decir para ofrecer QoS, se debe realizar la medición de parámetros que afectan la calidad del servicio como lo son: el retardo, la variación del retardo, el ancho de banda y la pérdida de paquetes. Este artículo analiza y valida los parámetros de QoS necesarios para garantizar el buen funcionamiento del servicio de VoIP sobre la red inalámbrica del campus de la Universidad Icesi. Se realizan pruebas en diferentes escenarios para mostrar que no solo factores como el retardo, y su variación, influyen en la calidad de servicio, sino que también la intensidad de la señal que recibe el cliente desde los puntos de acceso.Palabras Clave: Voz sobre IP, Calidad de servicio, Pérdida de paquetes, Retardo, Variación del Retardo, Intensidad de Señal. Abstract VoIP is one of the services that has been developing over under this type of wireless platforms and today has begun to implement as an alternative to the PSTN (Public Switched Telephone Network). The interest in VoIP is its cost-benefit ratio, and that organizations can use the same platform for their data network to transmit voice. Therefore it is important that the organization is clear that to ensure the smooth operation of the VoIP service, ie provide QoS, you must perform the measurement of parameters that affect the quality of service such as: delay, jitter, bandwidth, packet loss. In this paper we analyze and validate the QoS parameters needed to ensure the smooth operation of VoIP over wireless network on the Icesi University campus. We performed a series of tests in different scenarios to show that not only factors such as delay and jitter influencing the quality of service, but also the client signal strength received from of the AP (Access Point).Keywords: Voice over IP, Quality of service, Packet Loss, Delay, Delay variation, signal intensity.

scholarly journals Interconnecting Haptic Interfaces with High Update Rates through the Internet

2018 ◽  
Vol 1 (4) ◽  
pp. 51
Author(s):  
George Kokkonis ◽  
Kostas Psannis ◽  
Sotirios Kontogiannis ◽  
Petros Nicopolitidis ◽  
Manos Roumeliotis ◽  
...  
Keyword(s):  
Quality Of Service ◽  
Performance Evaluation ◽  
Real Time ◽  
Packet Loss ◽  
The Internet ◽  
Haptic Interfaces ◽  
Transport Protocols ◽  
Delay Jitter ◽  
Packet Arrival

Real-time transferring of the haptic sense over the Internet is quite a challenging task. This paper outlines the proposed protocols for transferring haptic streams over the Internet. Moreover, it describes the Quality of Service requirements that a network has to fulfill in order to successfully use haptic interfaces with high update rates over the Internet. Extensive simulations and experiments for the performance evaluation of transport protocols for real-time transferring haptic data are carried out. Complements between simulation and real world experiments are discussed. The metrics that are measured for the performance evaluation are delay, jitter, throughput, efficiency, packet loss and one proposed by the authors, packet arrival deviation. The simulation tests reveal which protocols could be used for the transfer of real-time haptic data over the Internet.

Implementation and Analysis of Wireless VoIP Networks with Routerboard Management at State Polytechnic of Malang

2015 ◽  
Vol 1 (1) ◽  
pp. 84-96
Author(s):  
Junaedi Adi Prasetyo
Keyword(s):  
Packet Loss ◽  
Quality Of Services ◽  
Delay Jitter ◽  
A Value ◽  
Data Compilation ◽  
Qos Parameters

In this article, the implementation of wirelessVoIP at State Polytechnic of Malang uses microtic management with the aim of knowing the QoS (Quality of Services) performance between systems without microtic management and systems using microtic management. In the implementation of this system, two tests were carried out, namely QoS testing when without proxy management and when using proxy management. From the two tests, the performance will be compared by doing data compilation using the VQ manager software. The QoS parameters to be taken are delay, jitter, packet loss and throughput. From the measurement it is known that when the VoIP server serves <= 3 calls simultaneously, the MOS value between the managed system (MOS = 3.7) and the system without management (MOS = 3.7) is almost the same because the value of delay and packet loss in the system without management and systems with management did not differ much, namely 107 ms and 83 ms, and the packet loss value was the same, namely 5%. And when serving> 3 simultaneous calls, there is a difference of 0.18 from the MOS value between the managed system (MOS = 3.48) and a system without management (MOS = 3.3) with a value of delay and packet loss for systems without management and systems with management, namely 527 ms and 340 ms, and the packet loss values ??were the same, namely 8% and 7.2%.

scholarly journals Single Board Computer Applications as Multi-Server VoIP

2021 ◽  
Vol 9 (VII) ◽  
pp. 1023-1028
Author(s):  
Martono Dwi Atmadja
Keyword(s):  
Packet Loss ◽  
Internet Protocol ◽  
Local Area ◽  
Computer Applications ◽  
The Public ◽  
Average Value ◽  
Delay Jitter ◽  
Telecommunication Technology ◽  
Internet Networks ◽  
Multi Server

Telecommunication technology is developing along with information technology and several innovations in several audio and data transmission and reception techniques. Innovation and communication technology are hoped to be able to create efficiencies in regards to time, equipment, and cost. The Public Switched Telephone Network (PSTN) telephone technology has experienced integration towards communication using Internet Protocol (IP) networks, better known as Voice over Internet Protocol (VoIP). VoIP Technology transmits conversations digitally through IP-based networks, such as internet networks, Wide Area Networks (WAN), and Local Area Networks (LAN). However, the VoIP cannot fully replace PSTN due to several weaknesses, such as delay, jitter, packet loss, as well as security and echo. Telephones calls using VoIP technology are executed using terminals in the form of computer devices or existing analogue telephones. The benefit of VoIP is that it can be set in all ethernet and IP addresses. Prefixes can be applied for inter-server placements as inter-building telephone networks without the addition of inefficient new cables on single board computers with Elastix installed. Prefix and non-prefix analysis on servers from single board computers can be tested using QoS for bandwidth, jitter, and packet loss codec. The installation of 6 clients, or 3 simultaneous calls resulted in a packet loss value in the prefix Speex codex of 2.34%. The bandwidth in the prefix PCMU codec has an average value of 82.3Kbps, and a non-prefix value of 79.3Kbps, in accordance to the codec standards in the VoIP. The lowest jitter was found in the non-prefix PCMU codec with an average of 51.05ms, with the highest jitter for the prefix Speex codec being 314.65ms.

scholarly journals UNJUK KERJA PENERAPAN TEKNOLOGI VoIP PADA JARINGAN VPN (VIRTUAL PRIVATE NETWORK)

2018 ◽  
Vol 5 (2) ◽  
pp. 1-12
Author(s):  
Hadria Octavia
Keyword(s):  
Operating System ◽  
Packet Loss ◽  
Internet Protocol ◽  
Virtual Private Network ◽  
Voice Over Internet Protocol ◽  
Public Network ◽  
The Public ◽  
Delay Jitter ◽  
Voice Data ◽  
Private Network

VoIP ( Voice over Internet Protocol ) is a technology used for communication in the form of IP based voice media over long distances. The concept of a VPN (Virtual Private Network) in this paper makes a client that is on the public network can be connected to a LAN network. To use the VoIP server in the Linux operating system Trixbox,  whereas for the VPN server using ClearOS and X-lite is used as a softphone to make calls to the client. Of testing at 64kbps bandwidth using the G711 codec produces value performance (delay, jitter, and packet loss ) is not good, so that voice data delivered is less clear. Thus the choice of bandwidth for the G.711 codec 512kbps up is the best solution to get the value of the performance (delay, jitter, and packet loss) better . And a choice of 3 Greed (low, medium, high) on setting bandwidth, high is the best option. Because it can produce the best performance for VoIP VPN technology.

scholarly journals PERFORMANCE EVALUATION OF SDN CONTROLLERS: FLOODLIGHT AND OPENDAYLIGHT

2016 ◽  
Vol 17 (2) ◽  
pp. 47-57 ◽  
Author(s):  
Shiva Rowshanrad ◽  
Vajihe Abdi ◽  
Manijeh Keshtgari
Keyword(s):  
Performance Evaluation ◽  
Confidence Interval ◽  
Packet Loss ◽  
Network Architecture ◽  
Tree Topology ◽  
Use Cases ◽  
Software Defined Network ◽  
Linear Topology ◽  
Significant Difference ◽  
Qos Parameters

Software Defined Network is new network architecture. One of its components is the controller, which is the intelligent part of SDN. Many controllers such as Floodlight, Open Daylight, Maestro, NOX, POX and many others are released. The question is which controller can perform better in which situations. Many works were done to compare controllers regarding architecture, efficiency and controllers’ features. In this paper, two of the most popular controllers, Floodlight and OpenDaylight are compared in terms of Network QoS parameters such as delay and loss in different topologies and network loads. This paper can help researchers to choose the best controller in different use cases such as clouds and multimedia. The results with 95% confidence interval show that OpenDaylight outperforms Floodlight in low loaded networks and also for tree topology in mid loaded networks in terms of latency. Floodlight can outperform OpenDaylight in heavy loaded networks for tree topology in terms of packet loss and in linear topology in terms of latency. There is no significant difference in performance of Floodlight and OpenDaylight controllers in other cases.

scholarly journals Evaluasi Kinerja GSM VoIP Gateway pada Sistem IP PBX

2021 ◽  
Vol 9 (3) ◽  
pp. 731
Author(s):  
HERI ANDRIANTO ◽  
DANIEL SETIADIKARUNIA ◽  
HENDRY RAHARJO
Keyword(s):  
Quality Of Service ◽  
Performance Evaluation ◽  
Packet Loss ◽  
Communication Costs ◽  
Average Value ◽  
Software Specifications ◽  
Gsm Network ◽  
Voip Gateway

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

ANALISIS DE CALIDAD DE SERVICIO EN REDES IPv6

2016 ◽  
Vol 26 (1) ◽  
pp. 7
Author(s):  
Jose Carlos Tavara Carbajal
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Traffic Control ◽  
Video Traffic ◽  
Control Software ◽  
Delay Jitter ◽  
End To End Delay ◽  
Analysis Of Results ◽  
Ipv6 Protocol

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.

ANALISIS DE CALIDAD DE SERVICIO EN REDES IPv6

2016 ◽  
Vol 26 (1) ◽  
pp. 1
Author(s):  
Jose Carlos Tavara Carbajal
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Traffic Control ◽  
Video Traffic ◽  
Control Software ◽  
Delay Jitter ◽  
End To End Delay ◽  
Analysis Of Results ◽  
Ipv6 Protocol

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.

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