• 인터넷정보학회논문지
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• 제18권6호
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• pp.65-73
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• 2017

Congestion occurring at wireless sensor networks(WSNs) causes packet delay and packet drop, which directly affects overall QoS(Quality of Service) parameters of network. Network congestion is critical when important data is to be transmitted through network. Thus, it is significantly important to effectively control the congestion. In this paper, new mechanism to guarantee reliable transmission for the important data is proposed by considering the importance of packet, configuring packet priority and utilizing the settings in routing process. Using this mechanism, network condition can be maintained without congestion in a way of making packet routed through various routes. Additionally, congestion control using packet service time, packet inter-arrival time and buffer utilization enables to reduce packet delay and prevent packet drop. Performance for the proposed mechanism was evaluated by simulation. The simulation results indicate that the proposed mechanism results to reduction of packet delay and produces positive influence in terms of packet loss rate and network lifetime. It implies that the proposed mechanism contributes to maintaining the network condition to be efficient.

• Journal of applied mathematics & informatics
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• 제22권1_2호
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• pp.501-515
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• 2006

The IEEE 802.11 protocol is the most mature technology for WLANs(Wireless Local Area Networks). However, as the number of stations increases, the delay and throughput performance of IEEE 802.11 MAC(Medium Access Control) degrades severely. In this paper, we present the comprehensive performance analysis of IEEE 802.11 MAC protocol by investigating the MAC layer packet service time when arrival packet sizes have a general probability distribution. We obtain the discrete probability distribution of the MAC layer service time. By using this, we analyze the system throughput and the MAC layer packet service time of IEEE 802.11 MAC protocol in wireless LAN environment. We take some numerical examples for the system throughput and the mean packet service time for several special distributions of arrival packet sizes.

• 산업경영시스템학회지
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• 제30권1호
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• pp.59-65
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• 2007

HPI (High-speed Portable Internet) system which provides high speed internet services is going to be commercialized soon. Since HPI provides simultaneously four different service types such as UGS (Unsolicited Grant Service), rtPS (real time Polling Service), nrtPS(non-real time Polling Service), and BE (Best Effort) under different QoS (Quality of Service) requirements and limited wireless channel resources, efficient packet scheduling mechanisms are necessary to increase the utilization of channels as well as to satisfy the various QoS requirements. This study regards the traffic data to be served as time series and proposes a new packet scheduling algorithm based on the nonparametric statistical test. The performance of the newly proposed algorithm is evaluated through the simulation analysis using a simulator that can evaluate the performance of packet scheduling mechanisms under various values of system parameters and measures such as packet delay time, data transmission rate, number of loss packets, and channel utilization.

• 전기전자학회논문지
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• 제7권2호
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• pp.181-187
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• 2003

초고속 패킷 스위칭 네트워크에서 QoS 보장을 위한 여러 가지 방안 중 우선순위에 의한 패킷처리 방식이 선호되고 있다. 초고속 패킷 스위치에서 우선순위에 따르는 효율적인 패킷 처리는 서비스 성능 향상의 관건이다. 따라서 초고속 네트워크 내의 핵심인 패킷 스위치에서의 효율적인 우선순위 서비스 알고리즘의 개발은 중요하다. 본 논문에서는 효율적인 패킷 스위치 성능 향상 방안으로 큐에 대기하는 패킷의 개수에 따라 서비스시간을 가변적으로 할당할 수 있는 W-iSLIP 스케줄링 알고리즘을 제안하고, 시뮬레이션하여 기존의 알고리즘과 비교 분석하였다. 우선순위에 따른 서비스와 전체 지연시간을 분석한 결과, 전체 지연시간에 있어서는 2.6%, 우선순위 서비스에 있어서는 34.6%의 성능향상이 있는 것으로 분석되었다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2000년도 정기총회 및 학술대회
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• pp.15-19
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• 2000

This paper is about MPEG-4 error resilience tools of video streaming on packet service(ex, Internet). It is need to packetization for MPEG-4 video transport by packet unit on MPEG-4 system, this paper suggest packetization method of minimizing packet error on packet service[1]. FEC(Forward Error Correction) and retransmission is usually used for recovery of packet loss, and this paper suggest applying these method to DMIF(Delivery Multimedia Integration Framework) for minimizing packet loss[2].

• 한국멀티미디어학회논문지
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• 제20권7호
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• pp.1030-1037
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• 2017

Traffic flows of real-time multimedia services such as Internet phone and IPTV are bounded on the end-to-end delay. Packets violating their delay limits will be dropped at a router because of not useful anymore. Service providers promise the quality of their providing services in terms of SLA(Service Level Agreement), and they, especially, have to guarantee the packet loss rates listed in the SLA. This paper is about a method to guarantee the required packet loss rate of each traffic flow keeping the high network resource utilization as well. In details, it assures the required loss rate by adjusting adaptively the timestamps of packets of the flow according to the difference between the required and measured loss rates in the lossy Weighted Fair Queuing(WFQ) scheduler. The proposed method is expected to be highly applicable because of assuring the packet loss rates regardless of the fluctuations of offered traffic load in terms of quality of services and statistical characteristics.

• 정보처리학회논문지C
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• 제19C권2호
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• pp.119-126
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• 2012

본 논문은 현재 상용화되어 국내에서 서비스가 제공되고 있는 이동통신 시스템인 와이브로와 HSDPA (High Speed Downlink Packet Access) 시스템을 통해서 패킷 기반 음성 서비스가 제공되는 경우의 서비스 품질에 관한 것이다. 미래의 체계적인 서비스 품질 관리 방안 수립 과정에서 이미 개발된 품질 측정 소프트웨어[2]를 이용하여 다양한 시나리오 하에서 실제 음성 서비스의 품질을 측정하였다. 측정 결과의 분석을 통해서 현 시점에서 와이브로와 HSDPA 시스템 기반으로 음성 서비스가 제공될 경우 서비스 품질이 상당히 양호할 것임을 확인할 수 있었다. 추가적으로 무선 전송 환경의 변화가 음성 서비스의 품질에 미치는 영향을 알아보기 위한 또 다른 실험을 통해서 MOS(Mean Opinion Score)값의 급격한 저하 현상이 나타나는 무선 품질 지표의 값을 확인할 수 있었다.

• Journal of Communications and Networks
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• 제11권1호
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• pp.36-41
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• 2009

The IEEE 802.16 wireless metropolitan area network (WMAN) standard is designed to correct expensive communication costs in CDMA-based mobile communication systems and limited coverage problems in wireless LAN systems. Thus, the IEEE 802.16e standard can provide mobile high-speed packet access between mobile stations and the Internet service provider through the base station with cheap communication fees. To efficiently accommodate voice over IP (VoIP) services in IEEE 802.16 systems, an uplink quality of service packet-scheduling scheme is proposed, and its performance is evaluated with an NS-2 network simulator in this paper. Numerical results show that this proposed scheme can increase the system capacity by 100% more than in the unsolicited rand service (UGS) scheme and 30% more than the extended real-time polling service (ertPS) scheme, respectively.

• Journal of Communications and Networks
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• 제4권3호
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• pp.209-220
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• 2002

In this paper, we propose a hierarchical packet scheduling technique to closely approximate a hierarchical extension of the generalized processor sharing model, Hierarchical Generalized Processor Sharing (H-GPS). Our approach is to undertake the tasks of service guarantee and hierarchical link sharing in an independent manner so that each task best serves its own objective. The H-GPS model is decomposed into two separate service components: the guaranteed service component to consistently provide performance guarantees over the entire system, and the excess service component to fairly distribute spare bandwidth according to the hierarchical scheduling rule. For tight and harmonized integration of the two service components into a single packet scheduling algorithm, we introduce two novel concepts of distributed virtual time and service credit, and develop a packet version of H-GPS called Hierarchical Fair Queueing (HFQ). We demonstrate the layerindependent performance of the HFQ algorithm through simulation results.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(5)
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• pp.83-86
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• 2002

In this paper while a router is routing all packet to the next hop, it inspects whether there is congestion on this current hop router or not and if the router discovers that it has some congestion, it informs that the packet is experienced to congestion. The packet arrived to next hop including some information about the congestion is processed first and it has wider bandwidth than another packet The amount of congestion is recorded to the DS field of IP header by congestion experience level. In the next hop when the packet including the congestion information is routed, the standard packet dropping ratio of the current router is changed in proportion to congestion experience that is recorded in IP header on of that. When the packet that has experienced congestion before is arrived, the router extends the drop threshold value not to drop the packet. It mean that transferring the audio or video stream, if the packet is already experienced the congestion in another hop, the router can provide the better service quality about 15∼25% than another.