• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.4B
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• pp.165-173
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• 2008

In this paper, we propose a new reliable broadcast protocol referred to as timer-based reliable broadcast (TRB) for wireless sensor networks (WSNs). The proposed TRB scheme exploits (1) bitmap based explicit ACK to effectively reduce the unnecessary error control messages and (2) randomized timer for ACK transmission to substantially reduce the possibility of contentions. Although it has been argued that 100% reliability is not necessary in WSNs, there should be messages (such as mission-critical message, task assignment, software updates, etc.) that need to be reliably delivered to the entire sensor field. We propose to use the TRB algorithm for such cases. Performance evaluation shows that the TRB scheme achieves 100 % reliability significantly better than other schemes with expense of slightly increased energy consumption.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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• v.17 no.2 s.58
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• pp.38-49
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• 2006

In order to increase the spectrum efficiency, recently, there is the number of studies for CR technology. For instance, IEEE 802.22 WRAN(Wireless Regional Area Network) WG considered the CR technology as a solution of WRAN system to serve the high speed internet service(1.5 Mbps down stream and 384 kbps up stream) in 100 km overall coverage and 54 MHz-746 MHz band. Basically, in MAC point of view, the WRAN system have been standardizing based on the IEEE 802.16 MAC layer features such as Data transmission method, QoS provision and Bandwidth request schemes. Additionally, the WRAN system further include CR nature functions such as incumbent user protection, self coexistence which would be importantly considered. Also, the inherent WRAN functions are added such as channel bonding and fractional bandwidth usage. This paper mainly explained frame structure, IU protection, self coexistence which are key functions of WRAN system. Finally, in this paper, we expressed a prospect of IEEE 802.22 WRAN standardization.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.7
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• pp.9-19
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• 2004

WLAN (Wireless Local Area Networks) needs error recovery and flow control schemes to support reliable multicast protocol. Limited wireless bandwidth, as well as queuing losses caused by the asymmetric wired/wireless interactions, demands more effective approaches for reducing packet losses. Moreover, since the wireless channel is a shared broadcast medium, if sender receives feedback information simultaneously from several receivers, the feedback delays data frame transmission of forward direction by introducing channel congestion and burden at the sender. Therefore, it is important to minimize the amount of feedback information from receivers. In this paper, we propose an ARS(Adaptive Redundancy Scheme) that combines FEC(Forward Error Correction) using channel state estimation and ARQ(Automatic Repeat Request) both to reduce the amount of feedback information and the number of retransmissions and to guarantee high data reliability in a WLAN multicast environment. Performance of the proposed scheme is evaluated by means of analysis and simulations in AWGN and Rayleigh fading channels. The results show that the proposed scheme reduces the amount of feedback information and the number of retransmissions and guarantees high data reliability, while keeping throughput efficiency similarly with the conventional FEC and ARQ scheme.

• Aerospace Engineering and Technology
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• v.9 no.1
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• pp.67-71
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• 2010

In this study, an algorithm for blind modulation detection in EDGE systems is presented. EDGE introduces an 8PSK modulation to provide high-speed data rates in addition to the existing GSM system. A transmitter may switch dynamically the modulation and coding schemes for transmission of data according to the channel quality. To decode the data correctly, the receiver has to detect using only training sequence which modulation is being used. Basically the property of one radio block composed of four bursts to detect effectively the modulation scheme even under severe conditions is used. More specifically, the reference value calculated for received burst is accumulated with previous reference value to minimize statistically the false detection probability in one radio block. Also each burst data having different modulation from the modulation of the fourth burst is set to zero to improve the decoding performance because the reference of the fourth burst has the highest reliability.

• Journal of KIISE:Information Networking
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• v.35 no.3
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• pp.192-206
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• 2008

Multihop is the main communication style for wireless sensor networks composed of tiny sensor nodes. Until now, most applications have treated the periodic small sized sensing data. Recently, the burst traffic with the transient and continuous nature is increasingly introduced due to the advent of wireless multimedia sensor networks. Therefore, the efficient communication protocol to support this trend is required. In this paper, we propose a novel PIGAB(Packet Interval Gap based on Adaptive Backoff) protocol to efficiently transmit the burst data in multihop wireless sensor networks. The contention-based PIGAB protocol consists of the PIG(Packet Interval Gap) control algorithm in the source node and the MF(MAC-level Forwarding) algorithm in the relay node. The PIGAB is on basis of the newly proposed AB(Adaptive Backoff), CAB(Collision Avoidance Backoff), and UB(Uniform Backoff). These innovative algorithms and schemes can achieve the performance of network by adjusting the gap of every packet interval, recognizing the packet transmission of the hidden node. Through the simulations and experiments, we identify that the proposed PIGAB protocol considerably has the stable throughput and low latency in transmitting the burst data in multihop wireless sensor networks.

• Journal of Satellite, Information and Communications
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• v.4 no.1
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• pp.1-7
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• 2009

As the information age comes out, the aspect of future war brings about the many changes in terms of war-fighting environment. Accordingly, information superiority and intelligence-centric warfare have been important and new war-fighting concept such as NCW(network centric warfare) have been turned up. This paper proposed all-weather core-strategy communications systems guaranteeing not only the real-time transmission of the information collected in a battlefield and expansion, automation, and rapidity of a battlefield but also broadband, mobility, survivability, and flexibility. The proposed military satellite communications system is classified into wideband mass capacity link, survivability, and the system supporting OTM(on the move) communication for the real-time transmission of battlefield information. This paper analyzed the essential operation concepts and core schemes of the U.S. Army's next generation system, TSAT(Transformational Satellite Communication System). Base on the analysis results, this paper proposed that the architecture of next generation military satellite communications systems for NCW have to provide the data rate, anti-jamming capability, network control and management capability which are optimally adaptable for the wireless channel environments such as jamming and interference and to support the variety of platforms like high-speed mobile vehicles, micromini devices, super-high speed unmanned aerial vehicles. Finally, this paper also proposed that next generation military satellite communications systems need the technologies such as the adaptable multi-antenna, laser link, and next-generation anti-jamming waveform.

• Journal of KIISE:Information Networking
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• v.30 no.1
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• pp.134-142
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• 2003

We present an ATM switch architectures based on the multistage interconnection network(MIN) for the efficient multicast traffic control. Many of these applications require multicast connections as well as point-to-point connections. Muiticast connection in which the same message is delivered from a source to arbitrary number of destinations is fundamental in the areas such as teleconferencing, VOD(video on demand), distributed data processing, etc. In designing the multicast ATM switches to support those services, we should consider the fairness(impartiality) and priority control, in addition to the overflow problem, cell processing with large number of copies, and the blocking problem. In particular, the fairness problem which is to distribute the incoming cells to input ports smoothly is occurred when a cell with the large copy number enters upper input port. In this case, the upper input port sends before the lower input port because of the calculating method of running sum, and therefore cell arrived into lower input port Is delayed to next cycle to be sent and transmission delay time becomes longer. In this paper, we propose the cell splitting and group splitting algorithm, and also the fairness scheme on the basis of the nonblocking characteristics for issuing appropriate copy number depending on the number of Input cell in demand. We evaluate the performance of the proposed schemes in terms of the throughput, cell loss rate and cell delay.

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.1007-1015
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• 2008

In order to provide real-time data from sensors and instruments at manufacturing processes on web, we proposed a communication service model based on XML(eXtensible Markup Language). HTML(Hyper Text Markup Language) is inadequate for describing real-time data from manufacturing plants while it is suitable for display of non-real-time multimedia data on web. For applying XML-based web service of process data in Intranet environment, real-time performance of communication services was evaluated to provide the system design criteria. XML schema for the data presentation was proposed and its communication performance was evaluated by simulation in terms of transmission delay due to increased message length and processing delay for transformation of raw data into defined format. For transformation of raw data into XML format, we proposed two structures: one is the scheme where transformation is done at an SCC(Supervisory Control Computer) after receiving real-time data from instruments. the other is the scheme where transformation is carried out at instruments before the data are transmitted to the SCC. Performances of two structures were evaluated on a testbed under various conditions such as six packet sizes and offered loads of 20%, 50% and 80%, respectively. Test results show that proposed schemes are applicable to the systems in Ethernet 100BaseT network if total message traffic is less than 7 Mbps.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.7
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• pp.813-826
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• 1999

실시간 멀티미디어 응용의 등장으로 멀티캐스트와 QoS(Quality of Service) 지원이 필수적인 망 서비스로 부각되고 있다. 이에, ATM 기반의 인터넷에서 IP 멀티캐스트의 효율적인 처리를 위하여 MARS(Multicast Address Resolution Server)가 제안되었고, 기존의 최선 서비스 기반의 인터넷에서 QoS(Quality of Service)를 지원하기 위하여 RSVP(Resource Reservation Protocol)가 제안되었다. 본 논문에서는 ATM 망에서 QoS가 지원되는 IP 멀티캐스트 서비스를 제공하기 위하여 MARS 구조에서 RSVP를 지원하는 두 가지 방안을 제안하고, 시뮬레이션을 통하여 그 성능을 분석하였다. 제안하는 두 가지 방법은 각각 'RSVP 전 홉 노드 방식'과 'MARS 서버 방식'이라 명명하였다. RSVP 전 홉 노드 방식은 송신원으로부터 ATM 망으로 진입하는 노드와 수신원을 향하여 ATM 망을 진출하는 노드 간에 각각 일대일 양방향 VC를 설정하여 멀티캐스트 그룹에 속하는 수신원들이 보내는 자원 예약 메시지를 ATM 망에서 전송하는 방안이다. MARS 서버 방식은 ATM 망을 진출하는 노드와 MARS 서버간에 MARS 제어 메시지 교환을 위해 존재하는 ATM VC를 사용하여 RSVP의 자원 예약 메시지를 전송하고, MARS 서버가 RSVP 자원 예약 메시지를 처리하도록 그 기능을 확장함으로써 ATM 망에서 필요로 하는 제어 VC 수를 절약할 수 있는 방안이다. 시뮬레이션을 통하여, MARS 서버 방식은 ATM 제어 VC의 수를 절약할 수 있을 뿐 아니라 경우에 따라 RSVP 자원 예약 메시지 전달 지연을 줄일 수도 있음을 볼 수 있었다. 그러나, MARS 클러스터 내에 동시에 존재하는 RSVP 흐름이 많을 때에는 MARS 서버 방식의 경우 MARS 서버에서의 병목 현상으로 인해 성능이 저하될 수 있다.Abstract Emerging real time multimedia applications require multicast service with a QoS(Quality of Service) support. An overlay service architecture MARS(Multicast Address Resolution Server) is proposed to support IP multicast over an ATM network, and a resource reservation protocol RSVP is proposed to provide QoS support in the Internet which is originally based upon best effort service only. In this paper, we propose two schemes to support IP multicast service with QoS support over ATM networks: 'RSVP Previous Hop Node(RPHN) scheme' and 'MARS server based scheme'. In RPHN scheme, the RSVP reservation messages are transported via one-to-one ATM control VC from the egress nodes to the ingress nodes of the the multicast flow set up between each pair of nodes. The RSVP message processing occurs at the ingress nodes of the multicast flow. Whereas, in the MARS server based scheme, the RSVP reservation messages are transported via the MARS control VCs between the egress nodes and the MARS server. The RSVP message processing burden is imposed at MARS server in this scheme. For MARS server based scheme, no additional ATM VC is required for RSVP reservation message transmission, while the processing burden at the MARS server is high. Simulation results show that the MARS server based scheme, may accomplish RSVP reservation message delivery with smaller delay as well as saving of the number of ATM VCs. When the number of simultaneous RSVP flows in the MARS cluster is large, however, MARS based scheme may suffer performance degradation since MARS server becomes a performance bottleneck.

• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.