• The Journal of the Korea Contents Association
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• v.18 no.3
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• pp.34-40
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• 2018

In this study, the optimal buffer size is calculated for seamless video playback on a mobile device. Buffer means the memory space for multimedia packet which arrives in mobile device for video play such as VOD service. If the buffer size is too large, latency time before video playback can be longer. However, if it is too short, playback service can be paused because of shortage of packets arrived. Hence, the optimal buffer size insures QoS of video playback on mobile devices. We model the process of buffering into a discret-time queueing model. Mean busy period length and mean waiting time of Geo/G/1 queue with N-policy is analyzed. After then, we uses the main performance measures to present numerical examples to decide the optimal buffer size on mobile devices. Our results enhance the user satisfaction by insuring the seamless playback and minimizing the initial delay time in VOD streaming process.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.81-85
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• 2009

본 논문은 VoIP, IPTV, VoD 등의 실시간 서비스 품질을 네트워크 노드에서 모니터링함으로써네트워크에서 발생한 품질이상을 분석할 수 있는 스위치 칩을 설계하였다. 인터넷 서비스의 특성상 단대단 서비스에 기반한 실시간서비스는 품질이상이 발생한 위치를 정확하게 분석할 수 없기 때문에 유지보수에 어려움이 있다. 이러한 문제를 해결하기 위해 본 논문에서는 실시간서비스에 해당하는 플로우를 등록하고 해당 플로우가 장치내에서 발생한 패킷손실, RTP 시퀀스 넘버를 참조하여 이전 장치에서의 패킷손실, 패킷의 IAT(Inter Arrival Time), 대역폭, 그리고 장치내 지연을 실시간으로 측정할 수 있는 기능을 가진 AVB(Audio/Video Bridge)칩을 구현하기 위해 IEEE802.1AS를 기만한 시간동기 프로토콜의 정확성을 시뮬레이션하고, FPGA를 이용하여 구현한 AVB 스위치칩에서 타임스템프의 정확성을 확인함으로써 실시간서비스의 품질을 네트워크에서 실시간으로 모니터링 할 수 있는 가능성을 확인하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.529-531
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• 2002

VoIP기술의 확산과 그에 따른 IP기반의 망을 포함한 packet, Multimedia서비스를 제공하는 서비스가 늘고있다. 그러나 실시한 서비스를 필요로 하는 이 서비스의 경우에는 그에 따르는 경로에 확실한 대역폭의 조건이 요구되고 있다. 하지만 best-effort방식을 사용하는 기존의 망에서는 대역폭의 확실한 보장이 이루지기 힘들다. 하지만 MPLS와 트래픽 엔지니어링 기술의 발전으로 다중경로 패킷 전달 및 동적인 부하 제어가 가능하게 되었다. 곧 차별화된 서비스에 따른 대역폭의 할당이 이루어질 수 있다. 동적인 부하 제어는 네트웍의 상태를 고려하여 경로간의 부하를 조절함으로써 네트웍의 효율성을 높일 수 있으나, 상태 정보의 시간차이에 의해 불안정한 상태에 이르기 쉽다. 본 논문에서는 차별화된 서비스를 하기 위해 부하의 할당을 안정적으로 변화시키면서도 효율적인 부하로 용량에 맞게 흐름을 받아들이고 패킷을 전송하는 기법을 제안한다.

• 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.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.1A
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• pp.13-23
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• 2007

In this paper, a packet scheduling scheme that supports real-time traffic having multi-level delay constraints in OFDMA systems is proposed. The proposed scheme pursues to satisfy the delay constraint first, and then manage the residual radio resource in order to enhance the overall throughput. A parameters named tolerable delay time (TDT) is newly defined to deal with the differentiated behaviors of packet scheduling according to the delay constraint level. Assuming that the packets violating the delay constraint are discarded, the proposed scheme is evaluated in terms of the packet loss probability, throughput, channel utilization. It is then compared with existing schemes for real-time traffic support such as the Exponential Scheduling (EXP) scheme, the Modified Largest Weighted Delay First (M-LWDF) scheme, and the Round robin scheme. The numerical results show that the proposed scheduling scheme performs much better than the aforementioned scheduling schemes in terms of the packet loss probability, while slightly better in terms of throughput and channel utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.12A
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• pp.1843-1851
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• 2000

ITU-R 권고안을 토대로 미래 IMT-2000 무선망에서의 소요 서비스 채널수 및 지상계 주파수 요구량을 분석한다. IMT-2000 가입자가 요구하는 서비스를 크게 회선과 패킷교환 서비스로 분류하고 옥내, 보행자 및 차량용의 가입자 이용 환경을 고려한다. 본 논문에서는 2010년경 각종 IMT-2000 서비스에 대한 시장성과 무선 전송 기술의 파라미터 값을 이용하여 가입자의 기준 트래픽과 셀내 발생 트래픽을 계산하고 필요한 무선 채널수 및 소요 주파수를 산출한다. 또한, ITU-R에서 권고한 기준 트래픽을 바탕으로 서로 다른 가입자 수와 트래픽 분포 및 이용 환경에서 필요한 무선 채널수와 소요 주파수를 산출하며, 채널수와 주파수 요구량에 대한 패킷 서비스 지연시간의 민감도를 분석한다.

• The Transactions of the Korea Information Processing Society
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• v.4 no.5
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• pp.1308-1318
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• 1997

This paper proposes a new touthind algorithm, the Bandwidth-Delay-Loss based Routing(BDLR) algotithm, which supports the selection of an effcienet routing path by cinsidering the characteristics and QoS requirements of intergarted servies over the Untegrated Serives Packet Network(ISPN), and also compareas it with other touting algorithms by simulating their perfomances on the various combinations of the realtime and non-realtime traffic over the ISPN. The simulation shows that the BDLR algorithm takes great advantages on transmisson dealy, the satisfiability of QoS requirements, and the adapation of traffic envirment over the other routing algorithms priposed for ISPN until now.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2191-2204
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• 1997

An efficient packet transmit scheduling algorithm should be able to allocate the resource to each connection fairly based on the bandwidth and quality of service requirements negotiated during the call admission procedure and it should be able to isolate the effects of users that are behaving badly. In this paper, we propose an effective round-robin packet transmit scheduling mechanism, which we call the delay tolerant packet reserving scheme (DTPRS) based on delay QoS requirments. The proposed scheme can not only provide fairness and but also reduce delay, delay variation, and packet loss rate by reserving output link time slots of delay tolerant packets and assigning the reserved slotsto delay urgent packets. Our scheme is applicable to high speed networks including ATM network because it only requires O(1) work to process a packet, and is simple enough to implement.