• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.47-50
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• 1998

In recent years, it is required that computing support mobile user with computer. The advantage of mobile computing is that users may access all their applications from any location, whether they are in another building or a different state. So, Internet combines with mobile computing technology to make new communication environment for supporting mobility. The research for solving the problem of mobility is actively in progress. This paper describes the implementation of tunneling method for flexible bypass between specific region. Tunneling method provide mobile service to mobile hosts. IP datagram's address tranform method is IP-within-IP encapsulation by which an IP datagram may be encapsulated within an IP datagram. The developed IP-within-IP protocol can provide not only enhanced performance because it is implemented in kernel mode, but also convenience of usage to the application developers because it gives user interface as a dynamic link library. Verification of IP packet tunneling was text file transfer program.

• Proceedings of the Korean Society of Computer Information Conference
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• 2019.07a
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• pp.245-246
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• 2019

본 논문에서는 GUI(Graphical User Interface)를 이용하여 다수의 드론을 동시 제어하는 시스템 구현 결과를 보인다. 네트워크 소켓(Network Socket) 응용 프로그램인 Packet Sender를 이용하여 다수의 드론을 AP(Wireless Access Point)에 연결하였다. Python 응용 프로그램으로 UDP(User Datagram Protocol) 소켓을 통해 AP에 연동된 드론으로 명령을 전송하여 제어한다. Python GUI 모듈인 Tkinter를 이용하여 사용자에게 GUI를 제공함으로써 접근성(Accessibility)을 높인 시스템을 개발하였다.

• ETRI Journal
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• v.43 no.3
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• pp.561-570
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• 2021

This paper presents master key identifier based protocol steganography (MKIPS), a new approach toward creating a covert channel within the Secure Real-time Transfer Protocol, also known as SRTP. This can be achieved using the ability of the sender of Voice-over-Internet Protocol packets to select a master key from a pre-shared list of available cryptographic keys. This list is handed to the SRTP sender and receiver by an external key management protocol during session initiation. In this work, by intelligent utilization of the master key identifier field in the SRTP packet creation process, a covert channel is created. The proposed covert channel can reach a relatively high transfer rate, and its capacity may vary based on the underlying SRTP channel properties. In comparison to existing data embedding methods in SRTP, MKIPS can convey a secret message without adding to the traffic overhead of the channel and packet loss in the destination. Additionally, the proposed covert channel is as robust as its underlying user datagram protocol channel.

• ETRI Journal
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• v.34 no.6
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• pp.827-837
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• 2012

We propose a flow admission control (FAC) for setting up a wire-speed connection for new flows based on their negotiated bandwidth. It also terminates a flow that does not have a packet transmitted within a certain period determined by the users. The FAC can be used to provide a reliable transmission of user datagram and transmission control protocol applications. If the period of flows can be set to a short time period, we can monitor active flows that carry a packet over networks during the flow period. Such powerful flow management can also be applied to security systems to detect a denial-of-service attack. We implement a network processor called a flow management network processor (FMNP), which is the second generation of the device that supports FAC. It has forty reduced instruction set computer core processors optimized for packet processing. It is fabricated in 65-nm CMOS technology and has a 40-Gbps process performance. We prove that a flow router equipped with an FMNP is better than legacy systems in terms of throughput and packet loss.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.3
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• pp.341-357
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• 2010

Forward Error Correction (FEC) techniques have been adopted to overcome packet losses and to improve the quality of video delivery. The efficiency of the FEC has been significantly compromised, however, due to the characteristics of the wireless channel such as burst packet loss, channel fluctuation and lack of Quality of Service (QoS) support. We propose herein an Adaptive Cross-layer FEC mechanism (ACFEC) to enhance the quality of video streaming over 802.11 WLANs. Under the conventional approaches, FEC functions are implemented on the application layer, and required feedback information to calculate redundancy rates. Our proposed ACFEC mechanism, however, leverages the functionalities of different network layers. The Automatic Repeat reQuest (ARQ) function on the Media Access Control (MAC) layer can detect packet losses. Through cooperation with the User Datagram Protocol (UDP), the redundancy rates are adaptively controlled based on the packet loss information. The experiment results demonstrate that the ACFEC mechanism is able to adaptively adjust and control the redundancy rates and, thereby, to overcome both of temporary and persistent channel fluctuations. Consequently, the proposed mechanism, under various network conditions, performs better in recovery than the conventional methods, while generating a much less volume of redundant traffic.

• Journal of Communications and Networks
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• v.10 no.3
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• pp.316-330
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• 2008

This paper proposes a novel method for transporting various types of user traffic effectively over the next generation network called integrated services digital network 3 (ISDN3) (or quantum network) using quantum packets. Basically, a quantum packet comprises one or more 53-byte quanta as generated by a "quantumization" process. While connection-oriented traffic is supported by fixed-size quantum packets each with one quantum to emulate circuit switching, connectionless traffic (e.g., IP packets and active packets) is carried by variable-size quantum packets with multiple quanta to support store-and-forward switching/routing. Our aim is to provide frame-like or datagram-like services while enabling cell-based multiplexing. The quantum packet method also establishes a flexible and extensible framework that caters for future packetization needs while maintaining backward compatibility with ATM. In this paper, we discuss the design of the quantum packet method, including its format, the "quantumization" process, and support for different types of user traffic. We also present an analytical model to evaluate the consumption of network resources (or network costs) when quantum packets are employed to transfer loss-sensitive data using three different approaches: cut-through, store-and-forward and ideal. Close form mathematical expressions are obtained for some situations. In particular, in terms of network cost, we discover two interesting equivalence phenomena for the cut-through and store-and-forward approaches under certain conditions and assumptions. Furthermore, analytical and simulation results are presented to study the system behavior. Our analysis provides valuable insights into the. design of the ISDN3/quantum network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.10b
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• pp.1145-1148
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• 2000

최근 멀티미디어 데이터에 대한 요구는 사용자가 증가함에 따라 데이터 양도 증가하고 있다. 이에 반해 서버시스템이나 네트워크의 대역폭은 이러한 서비스 요구를 충족시키기에 미흡하다. 본 논문에서는 리눅스 환경에서 일반적인 UDP(User Datagram Protocol) 전송 메커니즘을 SIO(Special Input/Output) 시스템 콜을 커널 내부에 추가하였다. UDP전송을 커널 내부에서 수행함으로써, 사용자 모드와 커널 모드사이의 데이터 복사의 횟수와 문맥 교환을 줄였다. 커널 내부에서 수행하게 SIO 시스템 콜을 설계하고 구현함으로써 일반적인 리눅스 환경보다 약 31%의 성능향상을 보았다. 본 논문에서는 SIO와 같은 효과적인 커널 내부의 전송 시스템 콜을 사용함으로써 멀티미디어 관련 서버에 적응할 수 있도록 하였다.

• International Journal of Computer Science & Network Security
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• v.22 no.2
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• pp.327-333
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• 2022

Internet Protocol Television (IPTV) has emerged as a personal entertainment source for home users. Streaming IPTV content over a wireless medium with good Quality of Service (QoS) can be a challenging task as IPTV content requires more bandwidth and Wireless Local Area Networks (WLANs) are susceptible to packet loss, delay and jitter. This research presents the capacity of IPTV using User Datagram Protocol (UDP) and TCP Friendly Rate Control (TFRC) over IEEE 802.11ac WLANs in good and bad network conditions. Experimental results show that in good network conditions, UDP and TFRC could accommodate a maximum of 78 and 75 Standard Definition Television (SDTV) users, respectively. In contrast, 15 and 11 High-Definition Television (HDTV) users were supported by UDP and TFRC, respectively. Performance of UDP and TFRC was identical in bad network conditions and same number of SDTV and HDTV users were supported by TFRC and UDP. With background Transmission Control Protocol (TCP) traffic, both UDP and TFRC can support nearly the same number of SDTV users. It was found that TFRC can co-exist fairly with TCP by giving more throughput to TCP unlike UDP.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.2
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.