• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.291-294
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• 2004

In this paper, we consider the broadcast switch architecture for hish performance multicast packet switching. In input and output buffered switch, we propose a new switch architecture which supports high throughput in broadcast packet switching with switch planes of single input and multiple output crossbars. The proposed switch architecture has a central arbiter that arbitrates requests from plural input ports and generates multiple grant signals to multiple output ports in a packet transmission slot. It provides high speed pipelined arbitration and large scale switching capacity.

• Journal of KIISE:Information Networking
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• v.28 no.3
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• pp.389-405
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• 2001

In these days, we have increasing demands on the real-time services, especially for the multimedia data transmission in both of wired and wireless environments and thus efficient and stable ways of transmitting realtime data are needs. Although RTP is widely used for internet-based realtime applications, it cannot avoid packet losses, due to the use of UDP stack and its underlying layers. In the case of mobile computing applications, the packet losses are more frequent and consecutive because of the limited bandwidth. In this paper, we first statistically analyze the characteristics of packet losses in the wired and wireless communications, based on Gilbert model, and a new packet recovery scheme for realtime data transmission is presented. To reflect the transmission characteristics of the present network environment, our scheme makes the sender to dynamically adjust the amount of redundant information, using the current packet loss characteristic parameters reported by the receiver. Additionally, we use relatively large and discontinuous offset values, which enables us to recover from both of the random and consecutive packet losses. Due to these characteristics, our scheme is suitable for the mobile computing environment where packet loss rates are relatively high and varies rapidly in a wide range. Since our scheme is based on the analytic model form statistics, it can also be used for other network environments. We have implemented the scheme with Mobile IP and RTP/RTCP protocols to experimentally verify its efficiency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6C
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• pp.614-622
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• 2006

In this paper, we propose an efficient packetization method to reduce the packetization overhead. For the purpose, we firstly verify the relationship between packet length and packet loss rate. The empirical results show that as the packet length is larger than the path MTU, the packet loss rate is drastically increased, producing poor visual quality at the receiver side. However, as the length of the packet is reduced, we should transmit more packets per frame and the packetization overhead will be increased. This increase in the packetization overhead reduces the number of bits allocated to the video data, resulting in the low visual quality. Therefore, each packet should be packetized to have the packet length close to the path MTU. In this paper, we show that the this process of the packetization with the constraint on the packet length is very similar to the dynamic storage allocation in the operating system. We had thoroughly surveyed the dynamic storage allocation methods used in the recent operating systems and propose to use the allocation methods for the video packetization. We empirically show that the proposed method can reduce the packetization overhead upto 28.3%, compared with the conventional sequential packetization method which have been widely used in Internet video transmission.

• ETRI Journal
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• v.26 no.6
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• pp.653-656
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• 2004

This paper presents an effective algorithm for inserting an MPEG-2 transport stream (TS) into a Digital Audio Broadcasting (DAB) ensemble without any bandwidth waste in a Digital Multimedia Broadcasting (DMB) transmission system. The key technologies of this algorithm include packet rate control and program clock reference correction, which are important for TS processing. The proposed algorithms are applied to the various DMB transmission systems based on Eureka-147, and the performance of the proposed algorithm is confirmed through the experimental DMB broadcasting.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.3
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• pp.448-453
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• 2003

Use of wireless LAN limited to special fields in restricted space as enterprises and university networks becomes gradually generalized and its needs are gathering strength. This paper shows transfer modes and structures of 2.400 wireless LAN spreaded at present. And we made experiments on an influence of transmission characteristics and rates of Packet signals in various User's environments and analyzed optimum transmission levels of signals fitted in User's environments.

• 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%.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.6
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• pp.2333-2353
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• 2020

This paper proposes a novel data prioritization and transmission mechanism to minimize the number of packets transmitted and reduce network overload using the constrained application protocol (CoAP) in resource-constrained networks. The proposed scheme adopts four classification parameters to classify and prioritize data from a sensor. With the packet prioritization scheme, the sensed data having the lowest priority is only delivered using the proposed keep-alive message notification to decrease the number of packets transmitted. The performance evaluation demonstrates that the proposed scheme shows the improvement of resource utilization in energy consumption, and bandwidth usage compared with the existing CoAP methods. Furthermore, the proposed scheme supports quality-of-service (QoS) per packet by differentiating transmission delays regarding priorities.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.16 no.9
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• pp.3124-3137
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• 2022

We propose a spectral-efficient coordinated direct and relayed transmission (CDRT) scheme for a relay-assisted downlink system with two users. The proposed scheme is based on backscatter communication (BC) and non-orthogonal multiple access (NOMA) technique. With the proposed BC-NOMA-CDRT scheme, both users can receive one packet within one time slot. In contrast, in existing NOMA-CDRT schemes, the far user is only able to receive one packet in two time slots due to the half-duplex operation of the relay. We investigate the outage of the BC-NOMA-CDRT scheme, and derive the outage probability expressions in closed-form based on Gamma distribution approximation and Gaussian approximation. Numerical results show that the analytical results are accurate and the BC-NOMA-CDRT scheme outperforms the conventional NOMA-CDRT significantly.

• Journal of IKEEE
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• v.7 no.1 s.12
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• pp.97-106
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• 2003

In ARQ based error control, imperfect error detection leaves error remains on a packet. Aiming for a reduction of error remains in multicopy transmission ARQ system, we propose a rule of requesting a retransmission and deciding a correct copy, (identified as $(m, \;{\sigma})$rule). While the probability of error remains is reduced by the employment of the $(m, \;{\sigma})$ rule at multicopy transmission ARQ, delay and throughput performance may be degraded in comparison with those of conventional single copy transmission ARQ. Thus, we develop an analytical method to evaluate the performance trade-off in multicopy transmission ARQ following the $(m, \;{\sigma})$ rule. From the numerical results obtained by the analytical method, we investigate the effect of channel characteristics on the performance of error remains, packet loss, throughput, and packet delay, and confirm that the adaptability of the $(m, \;{\sigma})$ rule to conform to various QoS requirements with ease.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.8A
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• pp.858-870
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• 2007

TCP is popular protocol for reliable data delivery in the Internet. In recent years, wireless environments with transmission errors are becoming more common. Therefore, there is significant interest in using TCP over wireless links. Previous works have shown that, unless the protocol is modified, TCP may perform poorly on paths that include a wireless link subject to transmission errors. The reason for this is the implicit assumption in TCP that all packet losses are due to congestion which causes unnecessary reduction of transmission rate when the cause of packet losses are wireless transmission errors. In this paper, we propose a new LDA that monitors the network congestion level using ROTT. And we evaluate the performance of our scheme and compare with TCP Veno, Spike scheme with NS2(Network Simulator 2). In the result of our experiment, our scheme reduces the packet loss misclassification to maximum 55% of other schemes. And the results of another simulation show that our scheme raise its transmission rate with the fairness preserved.