• Journal of the Korea Society of Computer and Information
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• v.16 no.4
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• pp.111-117
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• 2011

In this paper, we propose an QoS aware multi-layer MAC(QAML-MAC) protocol in a wireless sensor networks. Since the proposed protocol is based on the sleep-awake architecture, which save node's energy to prolong the entire network lifetime. For this purpose the QAML-MAC first classifies incoming data according to their transmission urgency and then saves them. The protocol also adapts the cross-layer concept to re-arrange the order of transmission with the same destination. So the delay can be decreased, which can not be obtained with the previous related protocols. And high priority data such as real-time multimedia or critical value in the field monitoring applications can be transmitted quickly, Furthermore the proposed protocol has advantage of decreasing transmitted data collisions using multiple layers of idle listening when there is no high-priority data. So energy consumptions of sensor nodes can be saved and the network lifetime can be prolonged.

• Journal of Korea Multimedia Society
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• v.7 no.2
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• pp.187-193
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• 2004

Wireless networks play a very important role in communications today. For example, wireless networks today provide from the basic services like e-mail and FTP to the multimedia applications like Web service. It is obvious that QoS requirements to these diverse applications over wireless networks will continue as in wired networks. Much research has been done to develop QoS supporting algorithms on Internet. But due to the limited bandwidth and varying channel states of the wireless networks, it is difficult to support differentiated service over wireless networks. In this paper we propose the modified wireless token ring protocol supporting QoS to the real-time traffic service node over Internet environments in which non-real-time and real-time traffic service nodes coexist. In the proposed algorithm, the real-time traffic service node gets the priority to take token over the non-real-time traffic service node. So the proposed algorithm support quick transmission of the real-time traffic service node. And this advantage can be obtained with minor modification of the legacy wireless token ing protocol to support QoS. We also consider the lost token recovery mechanism.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.11b
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• pp.194-197
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• 2002

본 논문은 리눅스 환경에서 IEEE1394의 동시 전송 프로토콜을 이용하는 멀티미디어 스트리밍 시스템의 구현에 대해 기술한다. 구현된 시스템은 DV 데이터의 스트리밍과 재생, 저장은 물론 로컬 재생과 장치 제어 기능을 갖는 사용자 프로그램과 dv1394 디바이스 드라이버로 구성된다. 스트리밍을 위한 메모리할당, 버퍼관리, 전송 데이터 형식 선언, IEEE1394 헤더 및 CIP 처리기능을 가진 dv1394 드라이버를 사용하여 드라이버와 OHCI1394 카드 사이의 스트리밍에 대한 투명성을 제공함으로써 기존의 스트리밍 시스템이 가졌던 응용 프로그램 개발의 복잡성을 줄였다. 이 시스템은 실시간 원격 모니터링이나 화상 회의 시스템, 그리고 비선형 비디오 편집 시스템에 응용될 수 있다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.10
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• pp.18-23
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• 2001

In this paper, we propose stabilized MAC protocol to resolve the bistable problem of the slotted ALOHA. The proposed protocol controls the random access rates according to the characteristics of traffic classes. The mathematical model and the experimental results show that the proposed protocol can guarantee the stable operation of random access procedure.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.10
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• pp.1225-1236
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• 1999

기존의 오류 제어 방법들은 실시간 특성과 신축적인 신뢰성을 가지는 멀티미디어 통신 응용의 특징을 충분히 반영하지 못하고 있다. 본 연구는 기존의 재전송을 기반으로 하는 방법에 두 가지 기법을 추가하여 이 문제에 접근하였다. 첫째, 실시간 데이타의 마감 시간을 고려하여 미리 재전송을 실시하는 사전 재전송과 시간적으로 불가능하다고 판단될 때 복구를 포기하는 재전송 포기를 통한 적시 재전송 기법이다. 둘째, 요구되는 신뢰성에 근접하도록 오류가 많이 발생하였을 때만 재전송을 하는 선택적 오류 제어 기법이다. 본 연구는 두 기법을 적용한 오류 제어 프로토콜을 구현하고 네트워크 환경을 바꾸어 가며 성능을 검사하였다. 실험 결과, 제안된 오류 제어 방법은 전체적인 오류 제어 부하는 감소시키면서 오류 복구율을 높인다는 점을 확인하였다.Abstract The current error control schemes do not fit well to the characteristics of multimedia communications: real-time transmission and flexible reliability requirements. This research proposes a new error control scheme, which extends the retransmission-based error control with the following two mechanisms. First, error recovery is performed in two timely fashions: one is retransmission in advance where a retransmission is performed early enough for the deadline if an error is suspected, and the other is retransmission abort where a retransmission is given up if its recovery within the deadline seems to be impossible. Second, error control is selectively performed only when an actual error rate approaches to the threshold on a given reliability requirement. The proposed scheme has been implemented and experimented in various network environments. The performance results show that it has lower control overhead and higher error recovery than the existing schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.5
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• pp.1249-1259
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• 1998

This paper proposes an efficient channel assignment method for real-time service in the IEEE 802.11 Wireless LAN system, and analyzes the supportability to real-time service and the channel utilzation of that through computer simulations. The improved MAC(Medium Access Control) protocol employing the proposed channel assignment method achieved good throughput over contention period as well as contention-free period by making real-time terminals transmit the information about their desiring value before starts of data transmissions. For performance evaluation of the proposed method, a simulation analyzing the variation of allocated bandwidths to terminals using real-time service and their buffer sizes wasexecuted. Real-time data traffics was modeled s 7-stae transition VBR sources, and asynchronous data traffics modeled as Poisson processes. In the simulation, the proposed method showed smaller variation of the bandwidth of each terminal and samller buffer size than the existing method, and also recorded good throughput over contention period.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.304-309
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• 2004

Recently, the needs for multimedia stream such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic thorough the Internet, does not apply any congestion control mechanism to regulate the data flow thorough the shared network And it leads to congestion collapse of the Internet and results in long-term transmission delay. To avoid any adverse effect on the current Internet functionality, a now protocol of modification or addition of some functionality to perform congestion control and to reduce huge transmission delay in transmitting of multimedia stream are in study. TCP-friendly congestion control mechanism is classified into two : one is window-based congestion control scheme using general window management functionalities, the other is rate-based congestion control scheme using TCP modeling equation. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and late-based congestion control for multimedia stream. And we also simulate the performance of improved TEAR implementation using NS. With He simulation results, we show that the improved TEAR can provide better fairness and lower rate fluctuations than TCP.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.4
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• pp.684-690
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• 2003

In this paper, a contention-based reservation access control algorithm and a slot allocation algorithm are proposed for non-real-time bursty traffic services in the wireless ATM networks. In order to reduce the collisions of the reservation request packets, the base station scheduler calculates the transmission probability using the number of RAS minislots and the estimated load of the reservation request packets. And it broadcasts the calculated transmission probability over the downlink control channel. Mobile terminals, which have traffic burst, select randomly a RAS minislot and transmit the reservation request packet by the received transmission probability. The reservation request packet contains the number of packets in the buffer. The slot allocation is based on the buffer status of the received reservation request packets. The base station scheduler allocates slots in proportion to the total number of slots that are requested by each terminal.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.232-234
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• 2003

TCP(Transmission Control Protocol)는 신뢰성 있는 전송계층 프로토콜이다. 이것은 데이터 스트림을 TCP 로 전달하는 응용프로그램이 전체 스트림을 순서에 맞고 오류 없이 전달하는 것을 의미한다. TCP 는 오류 제어를 이용하여 신뢰성을 제공하는데, 오류제어는 손상 세그먼트, 손실 세그먼트, 순서가 어긋난 세그먼트, 그리고 중복 세그먼트를 감지하는 메커니즘이 포함되며 특히 타이머(timer)를 이용한 오류제어를 본 내용에서 설명하고 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.44-52
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• 2005

This paper proposes a neural network prediction model to improve the valid packet transmission rate for the QoS(Quality of Service) of multimedia transmission. The Round Trip Time(RTT) and Packet Loss Rate(PLR) are predicted using a neural network and then the transmission rate is decided based on the predicted RTT and the PLR. The suggested method will improve the transmission rate since it uses the rate control factors corresponding to time of data is being transmitted, while the conventional one uses the transmission rate determined based on the past informations. An experimental set-up has been established using a Linux PC system, and the multimedia data are transmitted using UDP protocol in real time. The valid transmitted packets are about 5% higher than the one in the conventional TCP-Friendly congestion control method when the suggested algorithm was applied.