An N+1 redundant GOP based FEC algorithm for improving Streaming Video Quality due to Packet Loss and Delay Jitter

2007 ◽  
Author(s):  
M.S. Koul ◽  
K.R. Rao
Keyword(s):  
Packet Loss ◽  
Video Quality ◽  
Streaming Video ◽  
Delay Jitter

scholarly journals Analysis Of Quality Of Service (Qos) Youtube Streaming Video Service In Wireless Network In The Environment Faculty Of Science And Technology Uin Sunan Kalijaga

2019 ◽  
Vol 7 (2) ◽  
pp. 30 ◽  
Author(s):  
Asep Wishnu ◽  
Bambang Sugiantoro
Keyword(s):  
Wireless Networks ◽  
Wireless Network ◽  
Packet Loss ◽  
Science And Technology ◽  
Video Quality ◽  
Streaming Video ◽  
Quality Level ◽  
Wireless Internet ◽  
Wired Networks ◽  
Delay Jitter

The growing number of internet users in Indonesia, making the number of users increasing especially video streaming on Youtube service. This increase is based on rapid technological developments, especially PCs, Laptops and Smartphones that use wireless or wireless internet access. The use of streaming video over wireless networks is different from cable networks Because The characteristics of wireless networks are limited Compared to wired networks, and the characteristics of streaming video transmissions that require different handling than traditional text, and the data transmissions. As a first step towards Achieving an optimum Internet network service effort, Applies Action Research This research method by utilizing video quality with 360p, 480p, and 720p. The QOS parameters Analyzed Consist of delay, jitter, throughput, packet loss and bandwidth using wireshark and NetTools for the testing phase. The results of analysis using QoS for streaming video shows the performance of wireless network services at UIN Sunan Kalijaga, Faculty of Science and Technology is still not maximal especially on video with 480p quality, that has a 20 ms delay and jitter quality level ms According -0.0269 to TIPHON is very good. The amount of throughput is 0:55 MBps throughput and the percentage is 3% and the packet loss value is 28%, if it is Categorized by TIPHON standardization bad this value falls into the category. For the average bandwidth used is 329 714 bps value.

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  

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.

Evaluation of packet loss impairment on streaming video

2006 ◽  
Vol 7 (S1) ◽  
pp. 131-136 ◽  
Author(s):  
Hua-xia Rui ◽  
Chong-rong Li ◽  
Sheng-ke Qiu
Keyword(s):  
Packet Loss ◽  
Streaming Video

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.

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