• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1B
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• pp.17-24
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• 2008

The effect of the variable packet size on the LRD characteristic of the MMPP traffic model is investigated. When we generate packet traffic for the performance evaluation of IP packet network, MMPP model can be used to generate packet interarrival time. And a random length of packet size from a certain distribution can be assigned to each packet. However, there is a possibility that the variable packet size might change the LRD characteristic of the original MMPP model. In this study, we investigate this possibility. For this purpose the 'refined traffic' is defined, where packet arrival time is generated according to the MMPP model and a random packet length from a specific distribution is assigned to each generated packet. Hurst parameter of the refined traffic is estimated and compared with the original Hurst parameter, which is the input parameter of the MMPP model. We also investigate the effect of the packet size distribution on the queueing performance of the MMPP traffic model and the relationship between the Hurst parameter and queueing performance.

• Journal of Internet Computing and Services
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• v.18 no.6
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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.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.121-124
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• 1999

In a packet-switched wireless cellular network, a packet destined to a mobile station is queued at a base station and then broadcast over the base station's cell. When an active mobile station leaves a cell, there re-main packets which are destined to the mobile and not yet delivered to it at the cell's base station. For application which are sensitive to packet losses, such back- logged packets must be forwarded to the new base station. Otherwise, an end-to-end retransmission may be required. However, an increase in packet delay is incurred by employing the packet forward scheme, since a packet may be forwarded many times before it is delivered to the destined mobile station. For an enhanced quality-of-service level, it is preferred to reduce tile packet delay time. In this paper, we develop an analytical approximation method for deriving mean packet delay times. Using the approximation and simulation methods, we investigate the effect of network parameters on the packet delay time.

• ETRI Journal
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• v.8 no.3
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• pp.51-66
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• 1986

Packet switching 서비스 특성과 packet mode 터미널 수용을 위한 시스팀의 기능 및 packet switching을 실현한 타 교환기를 살펴보고 이를 토대로 ISDN을 위한 packet switching 실현시 교환기 측면에서의 고려사항과 실현해야 할 기본기능을 기술한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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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].

• Journal of information and communication convergence engineering
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• v.2 no.4
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• pp.214-218
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• 2004

In this paper, we have analyzed TCP performance over wireless correlated fading links with and without Snoop protocol. For a given value of the packet error rate, TCP performance without Snoop protocol is degraded as the fading is getting fast (i.e. the user moves fast). When Snoop protocol is introduced in the base station, TCP performance is enhanced in most wireless environments. Especially the performance enhancement derived from using Snoop protocol is large in fast fading channel. This is because packet errors become random and sporadic in fast fading channel and these random packet errors (mostly single packet errors) can be compensated efficiently by Snoop protocol's local packet retransmissions. But Snoop protocol can't give a large performance improvement in slow fading environments where long bursts of packet errors occur. Concerning to packet error rate, Snoop protocol results in the highest performance enhancement in the channel with mid-high values of packet error rate. This means Snoop protocol cannot fully fulfill its ability under too low or too high packet error rate environments.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1086-1089
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• 2002

In this paper, an adaptive packet acquisition performance of a CDMA/TDD packet radio system is analyzed and simulated in a frequency-selective Rayleigh fading channel. The frequency-selective fading process is assumed to be WSSUS model which is typical for a satellite and a mobile radio communications. The performance is evaluated in terms of packet error probability. In this adaptive packet acquisition scheme, an auxiliary training sequence is placed within data packet to estimate time-varying channel state. From the simulation results, it is confirmed that the proposed acquisition scheme significantly improves packet error performance compared to the conventional fixed acquisition scheme, especially, for the case of low Doppler frequency. The analysis in this paper can be applied to the design of a CDMA/TDD packet radio system.

• Proceedings of the IEEK Conference
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• 2001.06a
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• pp.29-32
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• 2001

This paper has simulated the PER(Packet Error Rate), ratio of received packet and payload BER of Bluetooth system with packet types of ACL(Asynchronous Connection Less) and SCO(Synchronous Connection Oriented) link over wireless Ad-hoc environment. AWGN(Additive White Gaussian Noise) and Rayleigh fading are considered as channel model, and the analysis is based on the baseband model of Bluetooth system. In terms of PER and ratio of received packet, performance of DM1 packet is almost same as those of HV1, HV2 and HV3 packets, the performances of the other packets depend on the packet types. In terms of payload BER performance, there is no difference among HV2 packet of SCO link and DM1, DM3, DM5 packets of ACL link. Moreover, there is no difference among HV3 packet of SCO link and DM1, DM3, DM5 packets of ACL link, too.

• Proceedings of the IEEK Conference
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• 2002.06e
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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.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.347-353
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• 2010

In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.