• Title/Summary/Keyword: 종단 지연

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Design and Analysis of ATM-based Video Stream Switch for Supporting Digital Video Library Service (디지털 비디오 라이브러리 서비스를 지원하는 ATM-기반 비디오 스트림 스위치의 설계 및 분석)

  • Park, Byeong-Seop;Kim, Seong-Su
    • The KIPS Transactions:PartC
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    • v.8C no.2
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    • pp.164-172
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    • 2001
  • 최근 인터넷의 확산과 더불어 디지털 비디오 라이브러리(DVL : Digital Video Library) 서비스에 대한 관심이 고조되고 있다. 그러나 현재의 통신망 대역폭과 스위칭 환경 하에서는 종단간 QoS 보장하는데 많은 제약사항이 존재한다. 따라서 본 논문에서는 비디오 스트림 처리를 효율적으로 수행하여, 지연-처리율 특성을 만족할 수 있는 스트림 스위칭 구조를 제안하고 이에 대한 성능을 분석하였다. 제안된 ATM-기반 스트림 스위치는 각각 다중화되는 CBR(Constant Bit Rate) 및 VBR(Variable Bit Rate) 스트림의 QoS(Quality of Service)를 보장해야만 한다. 성능분석 결과는 제안된 스위치의 처리율이 r=4일 때 약 0.996의 값을 보였으며, 지연시간도 부하가 0.7 이하일 때 2미만으로 특정되었다. 이 결과는 제안된 구조가 적당한 입력 스트림의 그룹핑을 통하여 비디오 응용을 위한 처리율 및 지연 요구사항 QoS를 보장할 수 있음을 보여준다.

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Performance Analysis of Voice over ATM using AAL2 based on Packet Delay Evaluation (ATM망에서 AAL2를 이용한 음성패킷 전송에 관한 성능분석)

  • 김원순;김태준;홍석원;오창석
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.10B
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    • pp.1852-1860
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    • 1999
  • This paper studied performance of the AAL2 for variable rate real time services in ATM network with discrete-time simulation model. In this simulation, input parameters are packet fill delay for AAL2 PDU generation, guard time for ATM cell generation, burstness and number of channels. Though variation of the above mentioned parameters, we obtained end-to end delay variations and throughput, analyzed performance effect of the each parameter for voice packet service.

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A Dynamic TXOP Allocation Scheme for Providing QoS in IEEE 802.11e based Wireless Ad-hoc Networks (IEEE 802.11e 무선 애드혹 네트워크에서 QoS 지원을 위한 동적 TXOP 할당 기법)

  • Ju, Kwang-Sung;Chung, Kwang-Sue
    • Proceedings of the Korean Information Science Society Conference
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    • 2011.06a
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    • pp.373-376
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    • 2011
  • 무선 멀티 홉 네트워크는 채널 상태의 불안정으로 인해 전송 지연, 패킷 손실 등의 발생이 빈번하며 홉 수가 증가할수록 이와 같은 품질 저하는 더욱 심각해진다. 무선 멀티 홉 네트워크에서 QoS를 보장하기 위해 IEEE 802.11e에서는 우선순위가 높은 노드에게 정해진 시간 동안 연속해서 보낼 수 있도록 TXOP(Transmission Opportunity) 파라미터 값을 정해놓고 있다. 하지만 IEEE 802.11e는 단일 홉 네트워크를 가정하여 설계되었기 때문에 멀티 홉 무선 환경에서 홉 수의 증가에 따른 지연 및 손실 증가를 고려하지 않는다. 본 논문에서는 멀티 홉 환경에서의 QoS 지원을 위한 DTA (Dynamic TXOP Allocation) 메커니즘을 제안하였다. DTA 메커니즘은 네트워크 상황을 고려한 동적 TXOP 할당 기법으로, 각 노드의 전송 큐 상태와 종단간 전송 지연 비율을 판별하여 이에 따른 QoS 지원을 위한 동적 TXOP를 할당한다. 이를 통해 불안정한 채널 상태 및 멀티 홉 환경에서 홉 수 증가에 따른 성능 감소를 개선할 수 있었다.

Building Low Delay Application Layer Multicasting Trees for Streaming Services (스트리밍 서비스를 위한 적은 지연의 응용계층 멀티캐스트 트리 구축)

  • Kim, Jong-Gyung
    • The Journal of the Korea Contents Association
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    • v.8 no.10
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    • pp.20-26
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    • 2008
  • The quality of stream remaking is decided the load of a server and Jitter through the traffic of the transmission path between end to end. In order to improve these problems in this paper, I propose tree construction method of low-delay-level-multicast. In this case which the network congestion will be occurred by streaming quality, I also propose the technique which dynamically changes the transmission path. This technique first constructs the overlay structure for relaxing the overload of server. Secondly, in order to decrease Jitter of client, it makes upload bandwidth and low latency balanced. In the evaluation of the performance, this paper showed better enhancement of about $15%{\sim}24%$ than P2CAST[4] in the simulation about node average join count, average bandwidth, service request refusal ratio, RTT measurement of nodes, and node average join count by defect ratio.

Expediting Data through Erasure Coding in Networks with High Coefficient of Variation of Transfer Time (전송시간의 변화가 큰 네트워크에서 이레이저 코딩을 적용한 긴급 데이터 전송 방법 및 성능 분석)

  • Lee, Goo Yeon;Lee, Yong
    • Journal of Digital Contents Society
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    • v.15 no.2
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    • pp.137-145
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    • 2014
  • In this paper, we focus on end-to-end transfer delay improvement by using erasure coding when delivering expediting message consisting of M packets in networks with high coefficient of variation of transfer time. In the scheme, M packets are divided into b groups with each having g packets. Each group is erasure coded with additional r packets and transmitted. Since the first arrived g packets among g+r packets completes the delivery of the group, the delivery time of the expediting message is reduced. For the scheme, we investigate the optimum group size and number of redundancy packets considering delivery delay reduction and additional transmission cost caused by using erasure coding. From the results of the investigation, we see that the proposed scheme is effective in networks having high variability of transfer time and would be very useful and practical especially for the case that expedited deliveries of messages are needed.

Development of Realtime Multimedia Streaming Service using Mobile Smart Devices (모바일 스마트 단말을 활용한 실시간 멀티미디어 스트리밍 서비스 개발)

  • Park, Mi-Ryong;Sim, Han-Eug
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.14 no.4
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    • pp.51-56
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    • 2014
  • Thesedays, there are many smart device applications developed, especially on the using various sensors included in the smart device. Smart devices have several sensors which are camera, GPS, mike, and communication module for collecting ubiquitous environment, and many applications are developed by using such sensors. In this paper, we developed the multimedia stream architecture and examined the smart device applications based on open source with front and back-end server clouds for developing the conceptual architecture. Also, we examined the back-end distributed servers, realtime multimedia stream transferring, multi-media store, and media relay for other server and smart devices. We test the examined architecture on the real target environment to collect the SIP initial setup time, media stream delay, and end-to-end play time. The test results show that there have good network operation environment to provide realtime multimedia services, and we need to improve the end-to-end play time by minimizing the initial setup time.

Dynamic Buffer Allocation for Seamless IPTV Service Considering Handover Time and Jitter (이동망에서 IPTV 서비스 제공 시 핸드오버 시간과 지터를 고려한 동적 버퍼 할당 기법)

  • Oh, Jun-Seok;Lee, Ji-Hyun;Lim, Kyung-Shik
    • The KIPS Transactions:PartC
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    • v.15C no.5
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    • pp.391-398
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    • 2008
  • To provide IPTV service over mobile networks, the mechanism that reduce packet loss and interrupt of multimedia service during the handover should be supported. Especially, buffering based mechanism is preferable for supporting IPTV services in the way of preserving streaming service using stored data and recovering non-received data after handover. But previous research doesn't consider the buffer allocation for applying various environments which can change handover time or end to end delay of relay node. This paper propose DBAHAJ mechanism that optimize buffer size of mobile nodes and relay node for supporting seamless IPTV service over mobile environments. Mobile node determines buffer size by checking handover time and maximum difference of sequence to keep playing video data. And multicast agent recovers packet loss during the handover by sending buffered data. By these two procedure, node supports seamless IPTV service on mobile networks. We confirm performance of this mechanism on NS-2 simulator.

Analytical model for mean web object transfer latency estimation in the narrowband IoT environment (협대역 사물 인터넷 환경에서 웹 객체의 평균 전송시간을 추정하기 위한 해석적 모델)

  • Lee, Yong-Jin
    • Journal of Internet of Things and Convergence
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    • v.1 no.1
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    • pp.1-4
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    • 2015
  • This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

Epidemic & Prediction based Routing Protocol for Delay Tolerant Network (DTN에서 에피데믹과 예측 기반 알고리즘을 이용한 라우팅 프로토콜)

  • Dho, Yoon-Hyung;Lee, Kang-Whan
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2014.05a
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    • pp.404-407
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    • 2014
  • Delay Tolerant Network (DTN) is a networking architecture that is designed to solve network problem in unstable and stressed environments using the Store-Carry-Forward method. Different form general networks, DTN does not guarantee the end-to-end connectivity, it is hard for the exiting TCP/IP based protocols to normally work due to the characteristic such as large latency and unstable link connectivity. And the condition that send massage without the information of the destination occurs frequently. Thus, suitable routing protocols for DTN are required and being researched. In this paper, we propose a novel routing protocol for DTN utilizing epidemic and prediction based algorithm to prevent the defect of previously DTN routing protocols such as the absence of the holistic view of the network. Proposed algorithm predicted destination using the mobility information recorded in neighbor node's history and utilize epidemic based algorithm when occurred condition without destination's information. This algorithm is enhanced in terms of delivery ratio, decreases latency and overhead in sparse network such as DTN.

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QoS Measurement Method of RTP/RTCP for Multimedia Service in BcN (BcN에서 멀티미디어 서비스를 위한 RTP/RTCP의 QoS 측정방법)

  • Lim Jae-Young;Kim Jeong-Ho
    • Proceedings of the Korea Contents Association Conference
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    • 2005.11a
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    • pp.611-615
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    • 2005
  • Having been multimedia services in BcN network, QoS is best important factor. This paper classify existing services and newly to be offered services, analyze quality criterion and measurement method for QoS guarantee in BcN, This paper investigate end-to-end quality criterion, classify measurement method existing and newly to be offered services, search quality criterion, introduce measurement method such as call success rate, packet loss rate, one-way delay, jitter and R-value for end-to-end quality measurement.

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