• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2004년도 춘계종합학술대회
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• pp.475-478
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• 2004

본 논문에서는 패킷 CDMA 망에서 데이터 서비스를 위한 코드 상태 감지 및 예약 기반의 CSSMA/AI MAC 프로토콜을 제안한다 단말기들이 코드 예약을 위한 프리앰블을 전송하기 전에 기지국에서는 매 프레임마다 예약된 코드의 상태를 방송하고, 단말기는 수신한 코드의 상태를 기반으로 임의의 예약되지 않은 코드를 예약을 위한 프리앰블을 전송한다. 프리앰블을 전송한 단말기는 선택한 코드의 하향링크를 통하여 기지국으로부터 응답을 기다린다. 기지국으로부터 코드예약에 대한 응답을 수신한 단말기는 남은 프레임 동안 패킷을 전송한다. 한편 전송한 패킷이 남아있는 경우, 기지국에서는 코드의 상태를 예약상태로 방송하고, 단말기에서는 다음 프레임에서 예약한 코드를 통하여 남은 패킷을 전송한다.

• 한국정보과학회논문지:시스템및이론
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• 제33권10호
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• pp.722-733
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• 2006

계층적 압축 기법을 지원하는 스트리밍 시스템 응용은 제한된 네트워크 자원의 효과적인 활용과 사용자가 느끼는 화질을 최대로 해야 한다. 이를 위해서는 적절한 전송 계층의 선택 및 패킷 인터벌 결정이 이루어져야 한다. 본 논문에서는 계층이 갖는 화질의 영향력을 바탕으로 패킷 인터벌 결정 및 계층 선택 알고리즘 SAPS를 제시한다. 인터-프레임 압축 기법을 사용하는 비디오 스트리밍 시스템에서 패킷 손실의 감소만으로는 재생 화질의 향상을 이룰 수 없고, 재생 화질에 높은 영향력을 가진 패킷의 복원율이 높아질 때, 비로소 재생 화질이 향상된다. SAPS는 패킷의 의존성 그래프를 바탕으로 전송 계층을 결정하며, 이렇게 결정된 전송 계층은 사용자가 느끼는 서비스의 품질을 최대로 만든다. 또한, 선택된 계층에 대한 패킷의 인터벌 조절을 통해 계층 선택에 의한 효과가 유지되도록 한다. 실험을 통해 SAPS 알고리즘이 사용자가 느끼는 서비스 품질의 향상뿐만 아니라, 네트워크 자원 활용도 효과적으로 이루고 있음을 보여준다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅲ
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• 한국멀티미디어학회논문지
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• 제19권5호
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• pp.924-931
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• 2016

In this paper, Various network adaptive MAC scheduling technique is proposed to effectively overcome limits of narrow bandwidth and low transmission speed in underwater. UPFC(Underwater Packet Flow Control) is a technique to reduce both the number of transmission and transmission time using three types (Normal, Blocked, Parallel) of data transmission. In this technique, the load information, in which a transmission node have, is transmitted to destination node using marginal bit in reserved header. Then the transmitted information is referred to determine weighting factor. According to the weighting factor, scheduling is dynamically changed adaptively. The performance of UPFC is analyzed and flow control technique which can be applied to Cluster Based Network and Ad Hoc network as well.

• Journal of electromagnetic engineering and science
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• 제12권3호
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• pp.189-195
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• 2012

In this paper, we propose a novel protocol called a Multi-hop Diversity Transmission protocol in which the retransmission is realized by a relay that is Nearest to a current Source (MDTNS). We derive the mathematical expressions of the packet error rate (PER) and the average number of transmissions over Nakagami-m fading channels, and verify them by Monte Carlo simulations. The simulation results show that the MDTNS protocol improves the performance of the network in terms of PER when compared to the Multi-hop Diversity Transmission protocol in which the retransmission is done by a relay that is Nearest to Destination (MDTND) and to the conventional multi-hop transmission (CMT) protocol.

• Journal of Multimedia Information System
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• 제5권3호
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• pp.163-170
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• 2018

We developed a multicast communication packet radio protocol using a time-sharing tablet system ("wireless token ring") to achieve the efficient exchange of files among packet radio terminals attached to swans. This paper provides an overview of the system and the protocol of the packet communications. The packet device forming the main part of the transceiver developed is the Texas Instruments CC2500. This device consists of one call-up channel and one data transmission channel and could improve error frame correction using FEC (forward error correction) with 34.8 kbps MSK and receiving power of at least -64 dBm (output 1 dBm at distance of 200 m using 3 dBi antenna). A time-sharing framework was determined for the wireless token ring using call sign ordinals to prevent transmission right loss. Tests using eight stations showed that resend requests with the ARQ (automatic repeat request) system are more frequent for a receiving power supply of -62 dBm or less. A wireless token ring system with fixed transmission times is more effective. This communication protocol is useful in cases in which frequency resources are limited; the energy consumed is not dependent on the transmission environment (preset transmission times); multiple terminals are concentrated in a small area; and information (position data and vital data) is shared among terminals under circumstances in which direct communication between a terminal and the center is not possible. The method allows epidemiological predictions of avian influenza infection routes based on vital data and relationships among individual birds based on the network topology recorded by individual terminals. This communication protocol is also expected to have applications in the formation of multiple in vivo micromachines or terminals that are inserted into living organisms.

• 한국멀티미디어학회논문지
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• 제7권1호
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• pp.54-63
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• 2004

인터넷을 기반으로 한 오디오 데이터 전송 시 발생되는 품질의 열화를 극복하여 고품질의 오디오 재생이 가능한, 고품질 오디오 스트리밍 서비스를 위한 복합적 QoS 보장 기법을 제안한다. 오디오 데이터는 연속성과 시간제약성을 가지고 있기 때문에 전송시 통신망의 상황에 따라 그 재생 품질이 결정된다. 이때 필요한 QoS 보장 기법으로, 인터넷의 상황에 따라 동적으로 전송률을 가변 시키면서 손실된 패킷의 재전송과 동적 버퍼 제어 기법을 복합적으로 적용하는 기법을 적용하였다. 클라이언트 내의 버퍼를 현재 망 상황에 따라 제어함으로써 재전송에 필요한 시간을 확보하고, 전송률도 함께 연동 조정하여 재전송에 따른 전송률 저하 현상을 방지한다. 실제 인터넷 환경에서 복합적 QoS 보장기법을 적용하여 고품질로 재생되는 오디오 스트리밍 서비스의 성능을 확인하였다.

• 한국멀티미디어학회논문지
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• 제13권12호
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• pp.1767-1777
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• 2010

In digital TV broadcasting, such as terrestrial, cable, satellite, and IPTV, the head-end of digital TV broadcasting has a more complicated transmission structure than that of analog TV broadcasting. Furthermore, digital TV broadcasting has a feature that supports multiplex models, such as Multiple Program Transport Stream (MPTS). Therefore, the purpose of our work was to design and examine a more efficient new system of emergency alert message transmission to support the digital TV broadcasting environments. Digital TV broadcasting is the IP generation or RF transmission of 8-VSB, QAM, and QPSK modulated through a multiplexer or re-multiplexer multiplexed stream as a MPEG-2 Transport Stream after content encoding. The new system proposed in this paper transmits an emergency alert message without scrambling after replacing the PID and payload of the -packet with the message prototype in the TS stream from the multiplexer. If we need to transmit an emergency alert message under digital TV broadcasting services, then the receiver first checks the PID of each packet in the TS stream for the emergency alert message. Next, if a packet is determined to be an emergency alert message, then the set-top box displays the message on the TV screen using its function of On Screen Display, or the PC based software displays the message on the monitor screen using its function of overlay with user interface if the packet is found to be an emergency alert message. We have designed an emergency alert message protocol and a system model. By experiments and analysis of the system, we concluded that the system achieved efficiency and the ability to send and receive emergency alert messages using the system under different digital TV broadcasting service environments.

• 한국통신학회논문지
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• 제27권7B호
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• pp.706-718
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• 2002

In bluetooth system, there are two kinds of interference. One is the frequency static interference, for example 802.11 direct sequence, the interferer uses fixed frequency band. Another is frequency dynamic interference, for example other piconets or 802.11 frequency hopping, the interferer uses dynamic frequency channel and cant be estimated. In this paper we introduce a novel solution of hybrid method of Listen-Before-Talk (LBT) and Adaptive Frequency Hopping (AFH) to address the coexistence of bluetooth and Direct Sequence of wireless local area network (WLAN). Before any bluetooth packet transmission, in the turn around time of the current slot, both the sender and receiver sense the channel whether there is any transmission going on or not. If the channel is busy, packet transmission is withdrawn until another chance. This is the LBT in Bluetooth. Because of asymmetry sense ability of WLAN and bluetooth, AFH is introduced to combat the left front-edge packet collisions. In monitor period of AFH, LBT is performed to label the channels with static interference. Then, all the labeled noisy channels are not used in the followed bluetooth frequency hopping. In this way, both the frequency dynamic and frequency static interference are effectively mitigated. We evaluate the solution through packet collision analysis and a detail realistic simulation with IP traffic. It turns out that the hybrid method can combat both the frequency dynamic and frequency static interference. The packet collision analysis shows it almost doubles the maximal system aggregate throughput. The realistic simulation shows it has the least packet loss.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(5)
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• pp.83-86
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• 2002

In this paper while a router is routing all packet to the next hop, it inspects whether there is congestion on this current hop router or not and if the router discovers that it has some congestion, it informs that the packet is experienced to congestion. The packet arrived to next hop including some information about the congestion is processed first and it has wider bandwidth than another packet The amount of congestion is recorded to the DS field of IP header by congestion experience level. In the next hop when the packet including the congestion information is routed, the standard packet dropping ratio of the current router is changed in proportion to congestion experience that is recorded in IP header on of that. When the packet that has experienced congestion before is arrived, the router extends the drop threshold value not to drop the packet. It mean that transferring the audio or video stream, if the packet is already experienced the congestion in another hop, the router can provide the better service quality about 15∼25% than another.