scholarly journals A Hybrid Rate Adaptation Framework for MPEG-4 FGS Video Streaming Over IP

Author(s):  
Colin Xialin Huang

There are increasing demands for real-time streaming video applications over the Internet. However, the current generation Internet was not originally designed for real-time streaming applications and only provides best-effort services, so there are many challenges in the deployment of video streaming applications over the Internet. This thesis investigates a hybrid end-to-end rate adaptation framework that provides application-level enhancements to achieve Quality of Service (QoS) for MPEG-4 FGS-Encoded video bandwidth on the path and the terminal process capabilities based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the proportion of load status feedbacks from the receivers. The sender and the receivers act together to minimize the possibility of network congestion by adjusting the transmission rate to match the network conditions. This framework achieves inter-receiver fairness in a heterogeneous multicast environment and improves QoS stability for MPEG-4 FGS video streaming over the Internet.

2021 ◽  
Author(s):  
Colin Xialin Huang

There are increasing demands for real-time streaming video applications over the Internet. However, the current generation Internet was not originally designed for real-time streaming applications and only provides best-effort services, so there are many challenges in the deployment of video streaming applications over the Internet. This thesis investigates a hybrid end-to-end rate adaptation framework that provides application-level enhancements to achieve Quality of Service (QoS) for MPEG-4 FGS-Encoded video bandwidth on the path and the terminal process capabilities based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the proportion of load status feedbacks from the receivers. The sender and the receivers act together to minimize the possibility of network congestion by adjusting the transmission rate to match the network conditions. This framework achieves inter-receiver fairness in a heterogeneous multicast environment and improves QoS stability for MPEG-4 FGS video streaming over the Internet.


2013 ◽  
Vol 4 (1) ◽  
Author(s):  
Said Atamimi

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


2014 ◽  
Vol 2014 ◽  
pp. 1-13 ◽  
Author(s):  
Richard Musabe ◽  
Hadi Larijani

3G long term evolution (LTE) introduces stringent needs in order to provide different kinds of traffic with Quality of Service (QoS) characteristics. The major problem with this nature of LTE is that it does not have any paradigm scheduling algorithm that will ideally control the assignment of resources which in turn will improve the user satisfaction. This has become an open subject and different scheduling algorithms have been proposed which are quite challenging and complex. To address this issue, in this paper, we investigate how our proposed algorithm improves the user satisfaction for heterogeneous traffic, that is, best-effort traffic such as file transfer protocol (FTP) and real-time traffic such as voice over internet protocol (VoIP). Our proposed algorithm is formulated using the cross-layer technique. The goal of our proposed algorithm is to maximize the expected total user satisfaction (total-utility) under different constraints. We compared our proposed algorithm with proportional fair (PF), exponential proportional fair (EXP-PF), and U-delay. Using simulations, our proposed algorithm improved the performance of real-time traffic based on throughput, VoIP delay, and VoIP packet loss ratio metrics while PF improved the performance of best-effort traffic based on FTP traffic received, FTP packet loss ratio, and FTP throughput metrics.


2021 ◽  
Author(s):  
Joni Rasanen ◽  
Aaro Altonen ◽  
Alexandre Mercat ◽  
Jarno Vanne

Author(s):  
Maha Z. Mouasher ◽  
Ala' F. Khalifeh

Voice over Internet Protocol (VoIP) systems have been spreading massively during the recent years. However, many challenges are still facing this technology among which is the lossy behavior and the uncontrolled network impairments of the Internet. In this chapter, the authors design and implement a VoIP test-bed utilizing the Adobe Real-Time Media Flow Protocol (RTMFP) that can be used for many voice interactive applications. The test-bed was used to study the effect of changing some voice parameters, mainly the encoding rate and the number of frames per packet as function of the network packet loss. Several experiments were conducted on several voice files over different packet losses, concluding in the best combination of parameters in low, moderate, and high packet loss conditions to improve the performance of voice packets measured by the Perceptual Evaluation of Speech Quality (PESQ) values.


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