• Title/Summary/Keyword: Network Adaptive QoS

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HTTP Adaptive Streaming Method for Service-compatible 3D Contents Based on MPEG DASH (MPEG DASH 기반 service-compatible 3D 콘텐츠 대상 HTTP adaptive streaming 적용방안)

  • Park, Gi-Jun;Lee, Gil-Bok;Lee, Jang-Won;Kim, Kyu-Heon
    • Journal of Broadcast Engineering
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    • v.17 no.2
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    • pp.207-222
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    • 2012
  • Recently, many consumer electronics manufacturers have produced 3D devices such as 3DTVs and 3D monitors as interests in a stereoscopic video service are getting increased. However, most of 3D services are focused on local storage or bandwidth guaranteed service since 3D stereoscopic video service require bandwidth more stable and larger. This property causes difficulties in seamless stereoscopic video streaming services under IP based open network environment that cannot guarantee quality of services. In order to achieve a seamless video streaming service the international standard organization MPEG (Moving Pictures Experts Group) has developed the adaptive HTTP streaming technology called as DASH (Dynamic Adaptive Streaming over HTTP). However, the DASH doesn't have obvious scheme which can express the two elementary video streams based service-compatible stereoscopic contents in one segment. Therefore, this paper proposes a scheme of efficient 3D adaptive streaming service based on the DASH, which covers not only frame-packing stereoscopic contents but also service-compatible ones. The 3D adaptive HTTP streaming scheme introduced in this paper is able to provide 3D contents with various qualities to user and also has benefit that single 3D content can be applied to a variety of devices.

An Optimal Power-Throughput Tradeoff Study for MIMO Fading Ad-Hoc Networks

  • Yousefi'zadeh, Homayoun;Jafarkhani, Hamid
    • Journal of Communications and Networks
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    • v.12 no.4
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    • pp.334-345
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    • 2010
  • In this paper, we study optimal tradeoffs of achievable throughput versus consumed power in wireless ad-hoc networks formed by a collection of multiple antenna nodes. Relying on adaptive modulation and/or dynamic channel coding rate allocation techniques for multiple antenna systems, we examine the maximization of throughput under power constraints as well as the minimization of transmission power under throughput constraints. In our examination, we also consider the impacts of enforcing quality of service requirements expressed in the form of channel coding block loss constraints. In order to properly model temporally correlated loss observed in fading wireless channels, we propose the use of finite-state Markov chains. Details of fading statistics of signal-to-interference-noise ratio, an important indicator of transmission quality, are presented. Further, we objectively inspect complexity versus accuracy tradeoff of solving our proposed optimization problems at a global as oppose to a local topology level. Our numerical simulations profile and compare the performance of a variety of scenarios for a number of sample network topologies.

A New Multi-objective Evolutionary Algorithm for Inter-Cloud Service Composition

  • Liu, Li;Gu, Shuxian;Fu, Dongmei;Zhang, Miao;Buyya, Rajkumar
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.12 no.1
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    • pp.1-20
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    • 2018
  • Service composition in the Inter-Cloud raises new challenges that are caused by the different Quality of Service (QoS) requirements of the users, which are served by different geo-distributed Cloud providers. This paper aims to explore how to select and compose such services while considering how to reach high efficiency on cost and response time, low network latency, and high reliability across multiple Cloud providers. A new hybrid multi-objective evolutionary algorithm to perform the above task called LS-NSGA-II-DE is proposed, in which the differential evolution (DE) algorithm uses the adaptive mutation operator and crossover operator to replace the those of the Non-dominated Sorting Genetic Algorithm-II (NSGA-II) to get the better convergence and diversity. At the same time, a Local Search (LS) method is performed for the Non-dominated solution set F{1} in each generation to improve the distribution of the F{1}. The simulation results show that our proposed algorithm performs well in terms of the solution distribution and convergence, and in addition, the optimality ability and scalability are better compared with those of the other algorithms.

A novel Adaptive Re-Marking Strategy for TCP Fairness of DiffServ Assured Services (DiffServ Assured Service에서 TCP 공평성 보장을 위한 적응적인 패킷 Re-Marking 방안)

  • Hur, Kyeong
    • The Journal of Korean Association of Computer Education
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    • v.11 no.2
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    • pp.99-106
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    • 2008
  • In this paper, we have proposed a novel re-marking strategy at tbe ingressive edge router to improve TCP fairness of DiffServ Assured Services. Our re-marking strategy introduces a configuration method of the Temporary Permitted Rate (TPR). By using this new configuration method of TPR, IN packets of greedy TCP flows are re-marked to OUT packets pertinently and constantly whenever the network traffic changes. Simulation Results show that this novel re-marking strategy can regulate the packet transmission rate of each TCP flow to the contract rate without a decrease in the link utilization.

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Developing an Adaptive Multimedia Synchronization Algorithm using Leel of Buffers and Load of Servers (버퍼 레벨과 서버부하를 이용한 적응형 멀티미디어 동기 알고리즘 개발)

  • Song, Joo-Han;Park, Jun-Yul;Koh, In-Seon
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.39 no.6
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    • pp.53-67
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    • 2002
  • The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.

Design and Implementation of Video Streaming Service Quality Control System through Available Bandwidth Management (가용대역폭 관리를 통한 영상 스트리밍 서비스 품질 제어 시스템 설계 및 구현)

  • Lee, In-Sun;Kim, Hyun-Jong;Choi, Seong-Gon
    • The Journal of the Korea Contents Association
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    • v.10 no.9
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    • pp.36-44
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    • 2010
  • In this paper, we propose video quality control system(VQCS) which can control video service quality through the monitoring of end-to-end available bandwidth for video streaming service like IPTV in NGN convergence network. Various multimedia services such as video, voice and gaming service can be provided by IPTV, and these services require large amounts of bandwidth. At this time, video quality degradation like video jerkiness, block distortion and blurring is caused when network available bandwidth is insufficient. Available bandwidth monitoring method is need to stably control video streaming quality. So, we periodically calculate the amount of the packets in link and measure available bandwidth by using total length field in IP header at terminal. Scalability extractor in network selects suitable video streaming data rate based on available bandwidth and transports video streaming with adaptive data rate to prevent video quality deterioration.

A Flexible Handover Scheme for Supporting Seamless Real-Time Services in Wireless Network (무선망에서 끊김 없는 실시간 서비스 제공을 위한 유연성 있는 핸드오버 기법)

  • Cho, Sung-Hyun;Pakr, Sung-Han
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.37 no.1
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    • pp.23-31
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    • 2000
  • In this paper, we propose an adaptive handover scheme in wireless network. The proposed handover scheme is a hybrid type of virtual path extension and rerouting. The proposed handover scheme chooses the virtual path management scheme according to the service QoS instead of the network topology which is used in the previous hybrid handover schemes. The proposed scheme supports the seamless service and small buffering during virtual path rerouting handover through the multicasting service. To evaluate the performance of the proposed scheme, we computed the number of signaling message used for handover and perform computer simulation. The simulation results show that the proposed scheme provides more efficiency in the handover delay and seamless service than the previous methods.

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Decision on Compression Ratios for Real-Time Transfer of Ultrasound Sequences

  • Lee, Jae-Hoon;Sung, Min-Mo;Kim, Hee-Joung;Yoo, Sun-Kwook;Kim, Eun-Kyung;Kim, Dong-Keun;Jung, Suk-Myung;Yoo, Hyung-Sik
    • Proceedings of the Korean Society of Medical Physics Conference
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    • 2002.09a
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    • pp.489-491
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    • 2002
  • The need for video diagnosis in medicine has been increased and real-time transfer of digital video will be an important component in PACS and telemedicine. But, Network environment has certain limitations that the required throughput can not satisfy quality of service (QoS). MPEG-4 ratified as a moving video standard by the ISO/IEC provides very efficient video coding covering the various ranges of low bit-rate in network environment. We implemented MPEG-4 CODEC (coder/decoder) and applied various compression ratios to moving ultrasound images. These images were displayed in random order on a client monitor passed through network. Radiologists determined subjective opinion scores for evaluating clinically acceptable image quality and then these were statistically processed in the t-Test method. Moreover the MPEG-4 decoded images were quantitatively analyzed by computing peak signal-to-noise ratio (PSNR) to objectively evaluate image quality. The bit-rate to maintain clinically acceptable image quality was up to 0.8Mbps. We successfully implemented the adaptive throughput or bit-rate relative to the image quality of ultrasound sequences used MPEG-4 that can be applied for diagnostic performance in real-time.

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Adaptive Power Control Dynamic Range Algorithm in WCDMA Downlink Systems (WCDMA 하향 링크 시스템에서의 적응적 PCDR 알고리즘)

  • 정수성;박형원;임재성
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.8A
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    • pp.918-927
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    • 2004
  • WCDMA system is 3rd generation wireless mobile system specified by 3GPP. In WCDMA downlink, two power control schemes are operated. One is inner loop power control operated in every slot. Another is outer loop power control based on one frame time. Base station (BS) can estimate proper transmission power by these two power control schemes. However, because each MS's transmission power makes a severe effect on BS's performance, BS cannot give excessive transmission power to the specific user. 3GPP defined Power Control Dynamic Range (PCDR) to guarantee proper BS's performance. In this paper, we propose Adaptive PCDR algorithm. By APCDR algorithm, Radio Network Controller (RNC) can estimate each MS's current state using received signal to interference ratio (SIR). APCDR algorithm changes MS's maximum code channel power based on frame. By proposed scheme, each MS can reduce wireless channel effect and endure outages in cell edge. Therefore, each MS can obtain better QoS. Simulation result indicate that APCDR algorithm show more attractive output than fixed PCDR algorithm.

Adaptive Rate Control for Guaranteeing the Delay Bounds of Streaming Service (스트리밍 서비스의 지연한계 보장을 위한 적응적 전송률 제어기법)

  • Koo, Ja-Hon;Chung, Kwang-Sue
    • Journal of KIISE:Information Networking
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    • v.37 no.6
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    • pp.483-488
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    • 2010
  • Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.