• ETRI Journal
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• 제44권4호
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• pp.543-559
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• 2022

To determine a suitable waveform for Internet of Things (IoT) transmission over low-Earth orbit (LEO) satellites, this paper reports the results of a performance comparison between chirp spread spectrum (CSS)-based LoRa and direct sequence spread spectrum (DSSS)-based Ingenu. The performance of both waveforms was measured in terms of the packet error rate, throughput, and packet loss rate, considering the Doppler effect caused by the high speed of LEO satellites and the interference among multiple terminals. Simulation results indicate that the DSSS scheme is more suitable than the CSS scheme for high-traffic IoT services because of its robustness against interference among multiple terminals. However, the CSS scheme is more suitable than the DSSS scheme for high-mobility IoT services because of its robustness against the Doppler effect. We discuss various solutions, such as the preprocessing of Doppler effect and avoidance of packet collision, to enhance the performance of the DSSS and CSS schemes. The simulation results of the proposed solution show that the enhanced DSSS scheme can be a proper waveform in IoT transmission over LEO satellites for both the high-traffic and high-mobility services.

• 한국멀티미디어학회논문지
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• 제13권5호
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• pp.740-753
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• 2010

WLP 기반 모바일 IP의 무선 네트워크에서, 사용자의 이동에 의한 핸드오프로 발생하는 패킷 손실은 TCP 수율 성능을 심각하게 악화시킬 수 있다. 이러한 문제점을 해결하기 위해, 사용자의 이동에 의하여 손실된 패킷을 복원하는 Bridge Station(BS) 패킷 버퍼링 방식이 제안됐다. BS 패킷 버퍼링 방식을 이용하면 핸드오프 동안 손실되는 패킷들이 이전 BS에서 저장되고, 핸드오프 종료시 저장된 패킷들이 새로운 BS로 포워딩되어, 복구된다. 그러나 WLP 디바이스가 새로운 WLP 서브 네트워크의 혼잡한 BS로 이동한 경우, 이전 BS가 포워딩하는 패킷들은 손실되고, 이전 BS가 포워딩하는 패킷들의 버스트한 도착 특성으로 심화된 혼잡이 BS 내 WLP 디바이스 플로들의 TCP 전송 성능을 저하시킨다. 본 논문에서는 이러한 BS 패킷 버퍼링 방식을 사용하는 WLP 기반 모바일 IP 무선 네트워크에서, AS(Assured Service) WLP 디바이스의 in-profile(IN) 및 전체 패킷 수율 감소를 막기 위해, 핸드오프 시 버퍼링된 out-of-profile(OUT) 패킷을 IN 패킷으로 Re- Marking하는 PBM(Packet Bridge Marker) 방식을 제안한다. 시뮬레이션 결과는 제안하는 PBM 방식을 사용하여 AS WLP 디바이스의 버퍼링된 OUT 패킷의 손실을 막아 핸드오프 시 IN 패킷의 수율뿐만 아니라 전체 패킷 수율도 향상시킬 수 있음을 보인다.

• Journal of information and communication convergence engineering
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• 제8권2호
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• pp.168-172
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• 2010

The Network-based handover still has problems such as the transmission delays and the packet losses in the case of macro mobility, though technological advances have been made in the wireless and mobile communication. For end-to-end handover, the link bandwidth has been reduced in the wireless network due to its burst errors and congestion control. To overcome such problems, we propose a new scheme of the macro handover according to the protocol layer. The proposed macro handover is implemented on the network layer to partially substitute wired signaling for wireless signaling, to flexibly employ buffers, and on the transport layer to postpone its retransmission time. We have performed extensive simulation using ns-2 and the result shows that our proposed scheme outperforms the other existing schemes in terms of transmission delay, packet loss, and data transfer rate during the handovers.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2004년도 ICCAS
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• pp.474-479
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• 2004

This paper describes a feedback-based congestion control algorithm to improve TCP performance over wireless network. In this paper, we adjust the packet marking probability at the router for Max-Min fair sharing of the bandwidth and full utilization of the link. Using the successive ECN (Explicit Congestion Notification), the proposed algorithm regulates the window size to avoid the congestion and sees the packet loss only due to the wireless link error. Based on the asymptotic analysis, it is shown that the proposed algorithm guarantees the QoS of the wireless TCP. The effectiveness of the proposed algorithm is demonstrated by simulations.

• International journal of advanced smart convergence
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• 제3권2호
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• pp.18-21
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• 2014

A control application requirements currently used is very low, such as packet loss rate, minimum delay on sensor networks with quality of service (QoS) requirements some packet delivery guarantee. This paper is the sampling period at the end of the actuator and sensor data transfer related to the Miss ratio for each source sensor node, use the controller and the internal ANFIS. The proposed scheme has the advantages of simplicity, scalability, and General. Simulation results of the proposed scheme can provide QoS support in WSANs.

• 한국정보과학회:학술대회논문집
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• 한국정보과학회 2002년도 봄 학술발표논문집 Vol.29 No.1 (A)
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• pp.217-219
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• 2002

TCP has been designed and tuned as a reliable transfer protocol far wired links. However, it incurs end-to-end performance degradation in wireless environments where packet loss is very high. TCP HACK(Header Checksum Option) is a novel mechanism proposed to improve original TCP in lossy link. It presents an extension to TCP that enables TCP to distinguish packet corruption from congestion in losssy environments. TCP HACK performs well when the sender receives the special ACKs correctly, hut if the ACKs are also lost much, the efficient of TCP HACK will net be prominent. In this paper we present an extension to TCP HACK, which can perform well even when the ACKs are much corrupted.

• 한국멀티미디어학회논문지
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• 제8권6호
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• pp.798-806
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• 2005

In this paper, we suggest a game system that can support network modules for multi-platform based video games, and built a system that can convert from a single-user game to multi-user game. In this system, we bring in an initial delay buffering scheme on clients to handle any periods of latency occurring from the load fluctuation in a network, when a real-time game is played, and shows that stable play for a game is achieved as the result of the scheme. This paper also presents a retransmission algorithm based on the history of game commands to handle drawbacks of UDP mechanism. And, we evaluate the network delay and packet loss using the simulation tool NS2, and shows the case of 0.3 second buffer delay is the most suitable for recovery.

• 한국통신학회논문지
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• 제27권5C호
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• pp.465-473
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• 2002

In this paper the authors propose a theoretic framework to design the Virtual Private Network (VPN) via which the Quality of Services (QoSs) are guaranteed over IP networks. The required QoS is a very strict packet loss probability or a probability that packet delay does not exceed a certain target value in a statistical manner. QoSs are guaranteed by providing a statistical bandwidth similar to equivalent bandwidth, which is computed so that the provided bandwidth is sufficient to guarantee those requirements. Two typical network architectures are considered in constructing VPN, the customer pipe scheme and the Hose scheme, and we propose a method to compute the amount of the required bandwidth for the two schemes. Finally, we investigate the implication of the scheme via numerical experiments.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.131-132
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• 2008

It is necessary to develop a high-speed wireless transmission system, which is able to send medical informations to the emergency medical center during emergency patient transportation. In this research, a system which transmits patient’s vital signs and a real-time audio/video contents of the event has been designed, developed, and the suitability of the system has been verified. Test results indicate that the system is capable of transmitting vital signal data, including 17 numeric data, 12 waveforms and 113 events, reading the affected part by forwarding a $320{\times}240$ pixel image at 2fps. Also, the full-duplex voice transmission of the system at 8bit/64kbps is enough to make stable communication between emergency medical technicians and hospital professionals possible. After numerous hours of driving, the packet loss of patient vital signs is 0.013%.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 I
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• pp.43-48
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• 2003

This paper presents a comparative evaluation of a modified version (A-Protocol) of the Distributed-Queueing Request Update Multiple Access (DQRUMA)Multi-Code Code Division Multiple Access (MC-CDMA) protocol and an enhanced version (P-Protocol) of the DQRUMA/MC-CDMA protocol with a lattice pool for request accesses (LPRA) scheme in a packet-based voice traffic environment. Analytical results agree with the simulation ones and show that the P-Protocol outperforms the A-Protocol in terms of the packet loss rate for voice traffic.