• ETRI Journal
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• v.44 no.4
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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.

• Journal of Korea Multimedia Society
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• v.13 no.5
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• pp.740-753
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• 2010

Performance of TCP can be severely degraded in WLP-based Mobile IP wireless networks where packet loss not related to network congestion occurs frequently during WLP-based inter-subnetwork handoff by user mobility. To resolve such a problem in the networks using WLP-based Mobile IP, the packet buffering method recovering seamlessly the packets dropped due to user mobility has been proposed. The packet buffering method at a bridge station recovers those packets dropped during handoff by forwarding buffered packets at the old bridge station to the WLP device. But, when the WLP device moves to a congested bridge station in a new WLP foreign subnetwork, those buffered packets forwarded by the old bridge station are dropped and TCP transmission performance of a WLP device in the congested bridge station degrades due to increased congestion by those forwarded burst packets. In this paper, a PBM(Packet Bridge Marker) is proposed for preventing buffered out-of-profile(OUT) packets from reducing the throughput of in-profile(IN) packets of an Assured Service WLP device. From this operation, the packet losses of buffered OUT packets are avoided and the throughput of IN and Total packets of an AS WLP device are increased.

• Journal of information and communication convergence engineering
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• v.8 no.2
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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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• 2004.08a
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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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• v.3 no.2
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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.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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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.

• Journal of Korea Multimedia Society
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• v.8 no.6
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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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.5C
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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.

• Proceedings of the IEEK Conference
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• 2008.06a
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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%.

• Proceedings of the IEEK Conference
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• 2003.07a
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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.