Impact of packet loss and delay variation on the quality of real-time video streaming

2015 ◽  
Vol 62 (2) ◽  
pp. 265-275 ◽  
Author(s):  
Jaroslav Frnda ◽  
Miroslav Voznak ◽  
Lukas Sevcik
Keyword(s):  
Real Time ◽  
Video Streaming ◽  
Packet Loss ◽  
Delay Variation

scholarly journals ANALISIS PERFORMASI TRANSMISI VIDEO PADA JARINGAN MULTICAST DAN UNICAST DENGAN MENGGUNAKAN WLAN IEEE 802.11N

2020 ◽  
Vol 5 (2) ◽  
Author(s):  
Mutia Muliana ◽  
Rizal Munadi ◽  
Teuku Yuliar Arif
Keyword(s):  
Quality Of Service ◽  
Real Time ◽  
Video Streaming ◽  
Packet Loss ◽  
Bit Rate ◽  
User Datagram Protocol ◽  
Ieee 802.11N ◽  
Streaming Multicast

Abstrak— Perkembangan pemakaian internet semakin meluas khususnya dalam bidang jaringan. Saat ini orang berkomunikasi tidak hanya dengan suara maupun teks, tetapi juga secara visual maupu   n menggunakan video streaming. Penggunakan akses wireless sangat tinggi namun wilayah coverage wireless yang tersedia dalam satu gedung terbatas sehingga daya tangkap sinyal wireless berbeda  satu dengan yang lain. Penelitian ini bertujuan untuk menganalisis performasi multicast dan unicast  dengan melihat pengaruh bit rate dan terhadap kualitas layanan streaming menggunakan protocol Real Time Protocol (RTP) dan User Datagram Protocol (UDP). Pada peneletian menggunakan dua format video yang berbeda yaitu MPEG 4 dan H.264 dengan WLAN 802.11n menggunakan metode eksperimental untuk evaluasi kinerja dua protokol yang berbeda  dengan  parameter Quality of Service (Qos). Multicast bekerja dengan mengirim data  kepada  banyak  titik sekaligus dan unicast bekerja dengan mengirim data kepada satu client. Standar IEEE 802.11n digunakan untuk menguji performa wifi dalam mentransmisikan video streaming. Hasil dari penelitian menunjukkan dengan multicast menggunakan sebanyak 4 client yaitu mempertimbangkan QoS terbaik yaitu throughput tertinggi, delay terkecil dan packet loss minimum diusulkan penggunaan protokol RTP dengan format video MPEG 4 lebih baik pada sistem transmisi streaming secara Unicast.  Kata Kunci – Video Streaming, Multicast, Unicast,802.11n, QoS.

An Improved Algorithm of Adaptive Multi-Pulse AMR-WB PLC

2013 ◽  
Vol 321-324 ◽  
pp. 2754-2759
Author(s):  
Jun Ying Jia ◽  
Hu Lin ◽  
Jian Wei Sun
Keyword(s):  
Real Time ◽  
Packet Loss ◽  
Mobile Terminal ◽  
Decoding Algorithm ◽  
Basic Principles ◽  
Voice Communication ◽  
Speed Up ◽  
Small Bandwidth ◽  
Adaptive Codebook

Packet loss is common in internet, when use IMS mobile terminal accessed by wireless to do real-time voice communication, the quality of speech is affected a lot. This text summarizes the basic principles of AMR-WB encoding and decoding algorithm and main technology of PLC. Based on AMR-WB encoding and decoding algorithm, it improves an algorithm of multi-plus PLC. The algorithm described in this text can speed up the recovery of adaptive-codebook by small bandwidth cost and effectively prevent the spread of errors, improve the quality of speech.

scholarly journals Scalable Video Streaming for Single-Hop Wireless Networks Using a Contention-Based Access MAC Protocol

2008 ◽  
Vol 2008 ◽  
pp. 1-21
Author(s):  
Monchai Lertsutthiwong ◽  
Thinh Nguyen ◽  
Alan Fern
Keyword(s):  
Wireless Networks ◽  
Video Streaming ◽  
Admission Control ◽  
Packet Loss ◽  
Optimization Problem ◽  
Heuristic Algorithms ◽  
Mac Protocol ◽  
Video Quality ◽  
Streaming Applications

Limited bandwidth and high packet loss rate pose a serious challenge for video streaming applications over wireless networks. Even when packet loss is not present, the bandwidth fluctuation, as a result of an arbitrary number of active flows in an IEEE 802.11 network, can significantly degrade the video quality. This paper aims to enhance the quality of video streaming applications in wireless home networks via a joint optimization of video layer-allocation technique, admission control algorithm, and medium access control (MAC) protocol. Using an Aloha-like MAC protocol, we propose a novel admission control framework, which can be viewed as an optimization problem that maximizes the average quality of admitted videos, given a specified minimum video quality for each flow. We present some hardness results for the optimization problem under various conditions and propose some heuristic algorithms for finding a good solution. In particular, we show that a simple greedy layer-allocation algorithm can perform reasonably well, although it is typically not optimal. Consequently, we present a more expensive heuristic algorithm that guarantees to approximate the optimal solution within a constant factor. Simulation results demonstrate that our proposed framework can improve the video quality up to 26% as compared to those of the existing approaches.

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 REAL-TIME INTERACTIVE MECHANISM FOR LIVE STREAMING ON PEER-TO-PEER NETWORKS

2020 ◽  
pp. 208-215
Author(s):  
Mina N. Abadeer ◽  
Rowayda A. Sadek ◽  
Gamal I. Selim
Keyword(s):  
Real Time ◽  
Video Streaming ◽  
Packet Delay ◽  
Peer To Peer ◽  
Live Streaming ◽  
P2p Streaming ◽  
Neighbor Selection ◽  
Streaming Technology ◽  
Selection Methodology

Quality of live video streaming technology is based on quality of Experiences parameters (QoE). Approaching the peer-to-peer (P2P) or peer-assisted networks as a sympathetic solution is highly required, especially in light of its authentic scalability and its extremely low initial cost requirements. However, the design of robust, efficient, and performing P2P streaming systems remains a high challenge when real-time constraints are part of the quality of service (QoS), as in TV distribution or conferencing applications. One of the P2P main issues that affect the quality of streaming is the neighbor selection methodology. The proposed work presents an effective mesh-based neighbor selection approaches for video streaming – Uniform Peer Distribution Algorithm (UPDA) – based on QoS and QoE Parameters. UPDA shortens the latency to be ranging from 10 ms to 50 ms servicing up to 4000 online peers under failure / recovery tests. Simulation results demonstrate that the proposed UPDA achieves good performance in End-to End delay with a percentage of 10.4 % and packet delay variation about 2% compared to random neighbor selection method.

scholarly journals STUDI ANALISIS EFEK PERBEDAAN LOKASI TERHADAP PERFORMANSI VIDEO STREAMING PADA JARINGAN W-LAN INDOSAT

2013 ◽  
Vol 4 (1) ◽  
Author(s):  
Said Atamimi
Keyword(s):  
Real Time ◽  
Video Streaming ◽  
Packet Loss ◽  
Wireless Lan ◽  
Loss Ratio ◽  
Packet Loss Ratio ◽  
Delay Jitter ◽  
User Coverage

Video streaming adalah aplikasi yang dapat melayani kebutuhan user akan data yang bersifat real time. Dengan adanya teknologi wireless LAN, user akan semakin dimudahkan dalam mengakses informasi seperti video streaming kapan saja dan di lokasi mana saja.Penelitian ini ditujukan agar dapat memperlihatkan hasil video streaming dari beberapa lokasi dalam lingkungan kantor Indosat. Dalam percobaan ini menggunakan beberapa perangkat antara lain satu buah server streaming, satu klien yang menggunakan laptop dan AP yang memang sudah ada dalam jaringan LAN Indosat serta skenario lokasi yang telah ditentukan sebagai tempat pengambilan data. Kemudian dilanjutkan pada tahap pengamatan sistem dengan melakukan peng-capture-an paket untuk mendapatkan data berupa throughput, delay, jitter, dan packet loss ratio dari tiap-tiap lokasi yang telah ditentukan.Hasil Penelitian ini, dengan adanya perbedaan lokasi mengakibatkan perbedaan dari kualitas video streaming berdasarkan parameter- parameter yang telah didapat pada percobaan.Kata Kunci : video streaming, wireless LAN, user, coverage

scholarly journals Analysis Tunneling IPv4 and IPv6 on VoIP Network

2018 ◽  
pp. 337-344 ◽  
Author(s):  
Muhammad Ismu Haji ◽  
Sugeng Purwantoro E.S.G.S ◽  
Satria Perdana Arifin
Keyword(s):  
Real Time ◽  
Packet Loss ◽  
Test Results ◽  
The Real ◽  
Ip Addresses ◽  
Internet Users ◽  
Jitter Delay ◽  
Ipv4 Address

Using of IP addresses is currently still using IPv4. Meanwhile, the availability of the IPv4 address is gradually diminishes. IPv4 has a limited address capacity. IPv6 was developed with a capacity greater than IPv4. Connect between IPv4 and IPv6 without having to interfere with the existing infrastructure. So, methods like tunneling are needed. Tunneling builds a way that IPv4 and IPv6 can communicate. 6to4 tuning makes IPv6 able to communicate with IPv4 over IPv4 infrastructure. Real time communication is needed by internet users to be able to connect to each other. One of the real time communications is VoIP. To find out the quality of tunneling implemented on a VoIP network, it will analyze QoS such as delay, packet loss, and jitter. Delay obtained is 20,01ms for IPv4, 19,99ms for IPv6 and 20,03ms for 6to4. Packet loss obtained 0,01% for IPv4, IPv6 0,01% and 6to4 0,08%. The obtained jitter is 7,96ms for IPv4, IPv6 7.39ms, and 8,48 for 6to4. The test results show that using IPv6 gets a better QoS value than using IPv4 and 6to4 tunneling. The results using 6to4 tunneling obtained the highest QoS value between IPv4 and IPv6. Implementation using 6to4 tunneling results in high results because, IPv6 packets that are sent are wrapped into the IPv4 form to get through the IPv4 infrastructure. 

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