• 한국통신학회논문지
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• 제33권1B호
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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.

• 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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 추계종합학술대회 논문집
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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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• 제8권3호
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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.

• 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].

• Journal of information and communication convergence engineering
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• 제2권4호
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -2
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 하계종합학술대회 논문집(1)
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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.

• 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.

• 한국음향학회지
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• 제29권5호
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• pp.347-353
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• 2010

VoIP 시스템의 통화 품질은 패킷의 전송 손실에 의하여 크게 저하되며, 각 패킷의 손실에 의한 음질 저하 정도는 해당 패킷의 특성에 따라 결정된다. 따라서 각 패킷 손실에 의한 음질 저하를 미리 분석하여 전송 과정에서 손실되는 패킷을 선택적으로 조정하면 VoIP 통화 품질을 향상시킬 수 있다. 본 논문에서는 DS 기반의 네트워크를 사용하는 VoIP에서 각 패킷에 전송 우선순위를 할당하여 통화 품질을 향상시키는 구조를 제안하고, 이를 위한 패킷 중요도 결정 방법을 제안한다. Gilbert 모델에 따른 패킷 손실 환경에서 제안한 방법의 성능을 측정하였으며, 객관적 음질 평가와 주관적 음질 평가를 통하여 VoIP 통화 품질이 향상되는 것을 확인하였다.