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.

scholarly journals Interconnecting Haptic Interfaces through The Internet

2018 ◽  
Author(s):  
George Kokkonis ◽  
Kostas E. Psannis ◽  
Sotirios Kontogiannis ◽  
Petros Nicopolitidis ◽  
Manos Roumeliotis ◽  
...  
Keyword(s):  
Performance Evaluation ◽  
Real Time ◽  
Video Stream ◽  
The Internet ◽  
Haptic Interfaces ◽  
Transport Protocols ◽  
Sensory Data ◽  
Delay Jitter ◽  
Packet Arrival

Supermedia streams transfer video, audio, haptic and other sensory data. Real -time transfering of supermedia streams over the Internet is quite challenging. This paper outlines the proposed protocols for transferring supermedia streams over the Internet. Moreover, it describes the Quality of Service (QoS) requirements for supermedia applications that a network has to fulfill. Extensive simulations and experiments for the performance evaluation of transport protocols for real time transferring HEVC streams with supermedia data are carried out. Complements, differences and relevancies 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 supermedia data with a HEVC video stream.

RANCANG BANGUN MEKANISME QUALITY OF SERVICE TERHADAP PROTOKOL RTP DAN SIP PADA ARSITEKTUR OPENFLOW

2019 ◽  
Vol 11 (1) ◽  
pp. 9-15
Author(s):  
Richard Alvianto ◽  
Samuel Hutagalung ◽  
Franciscus Ati Halim
Keyword(s):  
Quality Of Service ◽  
Real Time ◽  
Packet Loss ◽  
Internet Protocol ◽  
Session Initiation Protocol ◽  
Voice Over Internet Protocol ◽  
Transport Protocol ◽  
Delay Jitter

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.

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.

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. Keywords- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.

scholarly journals Analisa Perbandingan Quality of Service Voice Over IP dengan Pengujian Codec Menggunakan Algoritma Low Latency Queuing

2017 ◽  
Vol 3 (2) ◽  
pp. 249-254
Author(s):  
Darmawan Darmawan ◽  
Yayan Syafriyatno
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Packet Switching ◽  
Voice Over Ip ◽  
Low Latency ◽  
Delay Jitter

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.

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.

Performance Evaluation of Transport Protocol in Data Distribution Service Middleware

2014 ◽  
Vol 926-930 ◽  
pp. 1984-1987
Author(s):  
Peng Wei Li ◽  
Hong Li Zhao ◽  
Hai Tao Yang ◽  
Shu Sun
Keyword(s):  
Quality Of Service ◽  
Performance Evaluation ◽  
Real Time ◽  
Data Dissemination ◽  
Data Distribution ◽  
Transport Protocol ◽  
Latency Jitter ◽  
The Common ◽  
Data Distribution Service

The DDS middleware provides powerful support for data dissemination in the distributed real-time and embedded (DRE) systems, and supports multiple transport protocol (e.g. TCP, UDP and Multicast) that affect the end-to-end quality of service (QoS) properties (e.g. latency, jitter and reliability).In order to evaluate the performance of the transport protocol and then evaluate the affection on the DDS middleware QoS, this paper first briefly compares the common DDS implementations, and then presents performance evaluation and analysis of the transport protocol in OpenDDS with different environment configurations, at last presents the conclusion.

scholarly journals PENERAPAN METODE HTB DAN DIFFSERV GUNA PENINGKATAN QOS PADA LAYANAN VIDEO STREAMING

2019 ◽  
Vol 9 (3) ◽  
pp. 35-40
Author(s):  
Mitra Unik ◽  
Soni Soni ◽  
Randra Aguslan Pratama
Keyword(s):  
Quality Of Service ◽  
Video Streaming ◽  
Packet Loss ◽  
Digital Video ◽  
Streaming Media ◽  
Fundamental Problem ◽  
Streaming Video ◽  
Live Streaming ◽  
Delay Jitter

Abstract One of the popular internet services in use today is video streaming, either live (live streaming) or pre-recorder. Streaming video is a type of streaming media where data from video files is continuously transmitted over the internet to remote users. This fundamental problem appears to be influenced by the biggest factor which is the limited infrastructure of network resources which causes poor video quality. The process of digital video communication is known to consume quite a large resource, because in general the bandwidth requirements for sending Video and Audio signals. To maintain the quality of the video being played, there are several instruments needed, one of which is a data connection that is required to have Quality of Service (QoS). The parameters used in the measurement of QoS are delay, jitter, packet loss, throughput. This study uses the PPDIO method as a workflow with a Network Lifecycle approach. In this research, there are many factors that influence the quality of video, namely network factors and hardware factors. The test results obtained are not absolute, so it is possible that there will be differences in subsequent testing. Encoding also affects the quality of the video. Bandwidth equalization according to priority when the traffic conditions of all packets are full. Based on a comparative analysis of QoS parameter calculations using HTB and Diffserv methods, a comparison of throughput, jitter and delay does not differ greatly between clients. Keywords: Video Streaming, Diffserv, HTB, QoS Abstrak Salah satu layanan dari internet yang populer digunakan saat ini adalah video streaming, baik secara langsung (live streaming) atau pre-recorder. Streaming video merupakan jenis streaming media dimana data dari file video secara terus menerus dikirimkan melalui jaringan internet ke pengguna jarak jauh. Permasalahan mendasar ini muncul dipengaruhi oleh faktor terbesar yaitu terbatasnya infrastruktur sumber daya jaringan yang menyebabkan kualitas video yang buruk. Proses  komunikasi  digital  video,  diketahui  menghabiskan  resource  yang  cukup  besar, dikarenakan Secara umum kebutuhan bandwidth untuk mengirimkan sinyal Video dan Audio. Guna menjaga kualitas dari video yang dimainkan, terdapat beberapa instrument yang dibutuhkan, salah satunya adalah koneksi data yang wajib memiliki Quality of Service (QoS). Adapun Parameter yang digunakan dalam pengukuran QoS adalah delay, jitter, packet loss, Throughput. Penelitian ini menggunakan metode PPDIO sebagai alur kerja dengan pendekatan Network Lifecycle. Pada penelitian ini didapat Banyak faktor yang mempengaruhi kualitas dari video yaitu faktor jaringan dan faktor dari Hardware. Hasil pengujian didapat tidaklah mutlak sehingga tidak menutup kemungkinan akan ada perbedaan pada pengujian selanjutnya. Encoding juga mempengaruhi kualitas dari video. pemerataan Bandwidth sesuai prioritasnya saat kondisi traffic seluruh paket penuh. Berdasarkan analisa perbandingan perhitungan parameter QoS menggunakan metode HTB dan Diffserv, didapatkan  perbandingan troughput, jitter dan delay yang tidak berbeda jauh antara klien. Kata kunci: Video streaming, Diffserv, HTB, QoS  

scholarly journals Analysis Quality of Service Wireless LAN at University PGRI Yogyakarta

2018 ◽  
Vol 6 (1) ◽  
pp. 9
Author(s):  
Nur Kukuh Wicaksono ◽  
Bambang Sugiantoro
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Wireless Lan ◽  
Educational Institution ◽  
The Internet ◽  
Average Delay ◽  
Observation Methods ◽  
The University ◽  
Internet Network

PGRI University of Yogyakarta is an educational institution that uses the internet as one of the supporting facilities and infrastructures to manage and organize the data and information used by the student to find references about the lecture. PGRI University Yogyakarta has three buildings on the main campus building A building B and C buildings, where each building using wireless LAN as a means for students to use the internet network, the weakness of the wireless LAN network where poor internet network in the wireless LAN network. Thus the researchers wanted to analyze the Quality of Service wireless LAN networks in building A, building B, and C buildings, in each floor.With the existence of quality of the network at PGRI University of Yogyakarta will be done by interviews and observation methods, problems that occur in wireless LAN networks in each building have been prepared in advance, after which it will do an analysis of wireless LAN networks using quality of service parameters, namely delay, packet loss, bandwidth, throughput and factors that influence the wireless network at the University of PGRI Yogyakarta.The results of the measurement and monitoring of Quality of Service wireless LAN at PGRI University of Yogyakarta in building A, building B, C on each floor of the building can be classified in the category of poor with the average delay for each building to around 150 ms and packet loss = 28%, bandwidth = 173523 bits / s and throughput = 22%, and the factors that occurred in the signal range cannot cover every room in every building. From these results it can be concluded that the quality of the wireless LAN at the University PGRI Yogyakarta according to the TIPHON standards categorized as poor.

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