• 대한전자공학회논문지TC
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• 제41권10호
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• pp.1-9
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• 2004

CDMA 2000 서비스의 주요 성공여부는 사용자 측면에서 응용서비스의 성능에 좌우한다고 할 수 있다. 본 논문에서는 CDMA 네트워크에서의 단대단 성능을 평가하기 위하여 단대단 시뮬레이션 모델을 개발하고 이를 사용하여 음성과 데이터 서비스의 사용자 측면에서 응용서비스의 성능을 분석하였다. 시뮬레이션 성능분석 모델들은 응용서비스 트래픽 특성 모델, 네트워크 구조 (architecture) 모델, 네트워크 구성요소 모델, 각 계층별 프로토콜 모델들을 포함한다. 다양한 전송계층의 구조에 따른 사용자 측면의 응용계층 성능을 평가하기 위하여 액세스망(RAN: Radio Access Network)과 코어망(CN: Core Network)에서 ATM 과 IP 전송방식을 고려하였다. 음성 서비스 분석 결과로 보코터 바이패스를 사용하는 경우 다른 방식에 비하여 30%의 성능 개선을 보이는 것을 알 수 있었다. 데이터 서비스 성능(웹 서비스)의 경우, 연속적 TCP연결 (persistent TCP connection)과 파이프라이닝 효과에 기인하여 HTTP v.1.1이 HTTP v.1.0에 비하여 성능이 우수하였다. 또한, RAN 구조에서 웹 브라우징 서비스의 경우 IP 전송기술이 ATM 에 비하여 데이터 패킷에 대한 패킷 오버헤드가 상대적으로 적어 우수함을 보였으며, 패킷크기가 작은 음성 서비스의 경우 반대의 효과를 볼 수 있었다. 비록 시뮬레이션 결과를 통하여 3G-lX EV 시스템이 3G-lX RTT에 비하여 우수한 패킷지연 성능을 보여주었지만, 기술적 진화의 모든 단계에서 단대단 사용자 단계의 성능에 영향을 주는 다양한 네트워크의 구성요소 및 계층에 대하여 고려하여야 할 것이다.

• 한국ITS학회 논문지
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• 제4권1호
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• pp.107-115
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• 2005

For effective IP based service implementation on the wireless network environments, wireless network including the ITS network have to support QoS guaranteed protocol such as a RSVP. RSVP is a resource reservation protocol for Internet environment, and its scalability makes easy to implement RSVP over the various IP transport technologies. But for the If based ITS wireless network environment, RSVP is not suitable, since by its path setup procedure characteristic. In the wireless access network for ITS service, when a mobile node moves to other domain it must perform registration procedure. But the registration procedure is time consuming steps, so if a RSVP session was already established in the previous domain, the RSYP session may be disconnected and the time to re establish a new RSVP session is long enough to cause serious packet loss. In this paper, we propose a pre-path reservation mechanism for applying the RSVP in wireless access networks for ITS. In the pre-path reservation mechanism, the resource reservation procedure occurred during a mobile node's handoff time. An access point in wireless access network performs this procedure when the mobile node attempts handoff The access point executes pre-path reservation procedure as a proxy, since the mobile node does not have IP address until the address allocation procedure is finished in a new domain.

• International Journal of Internet, Broadcasting and Communication
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• 제11권3호
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• pp.59-63
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• 2019

VoIP technology is a technology for exchanging voice or video data through IP network. Various protocols are used for this technique, in particular, RTP(Real-time Transport Protocol) protocol is used to exchange voice data. In recent years, with the development of communication technology, there has been an increasing tendency of services such as "Kakao Voice Talk" to exchange voice and video data through IP network. Most of these services provide a service with security guarantee by a user authentication process and an encryption process. However, RTP protocol does not require encryption when transmitting data. Therefore, there is an exposition risk in the voice data using RTP protocol. We will present the risk of the situation where packets are sniffed in VoIP(Voice over IP) communication using RTP protocol. To this end, we configured a VoIP telephone network, applied our own sniffing tool, and analyzed the sniffed packets to show the risk that users' data could be exposed unprotected.

• Journal of Communications and Networks
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• 제1권3호
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• pp.145-152
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• 1999

A new switch architecture is presented to enhance out-put queuing in photonic packet switches. Its appkication is for a packet switching enviroment based on the optical transport of fixed length packets. This architecture consists of a couple of cas-cading switching elements with output queuing, whose buffer ca-pacity is limited by photonic technology. The introduction of a suitable buffer management allows a very good and balanced ex-ploitation of the available optical memories, realized with fiber de-lay lines. In particular, packet loss performance is here evaluated showing the improvement with respect to the single switch and a way to design large optical switches is shown in order to meet broadband network requirements.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 통신소사이어티 추계학술대회논문집
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• pp.145-148
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• 2003

A lot of institutes have proposed RMT (Reliable Multicast Transport) techniques over IP multicast environment which provides only best-effort services. The ECTP (Enhanced Communications Transport Protocol) is a kind of RMT technique and an international standard of ITU-T and JTC1/SC6. It is a protocol that provides reliability fer IP multicast environment. In this paper, we tested ECTP applications which are MFTP (Multicast file transport protocol) and WB (white Board) on the APAN (Asia-Pacific Advanced Network) with the Kyushu University in Japan. We measured ECTP traffic using a packet capture tool that programmed by Pcap library that is developed in Lawrence Berkeley institute. In according to the result, we think that ECTP is an efficient protocol in transporting files to many receivers simultaneously.

• 디지털산업정보학회논문지
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• 제18권2호
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• pp.1-8
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• 2022

The message propagation between vehicles must be efficiently performed to quickly transmit information between vehicles in vehicle ad hoc networks (VANETs). Broadcasting can be the most effective solution for propagating these messages. However, broadcasting can cause broadcast storm problems, which can reduce the performance of the overall network. Therefore, rapid information delivery in VANET requires a method that can propagate messages quickly without causing the broadcast storm problems. This paper proposes a lightweight and opportunistic broadcast (LOB) protocol that leverages the features of opportunistic routing to propagate messages quickly while minimizing the load on the network in VANETs where the network topology changes frequently. LOB does not require any routing information like greedy forwarding scheme, and neighboring nodes at the farthest distance within the range of transmission nodes are likely to be selected as forwarding nodes, and multiple forwarding candidate nodes like opportunistic routing scheme can increase packet transmission rates. Simulation results using ns-2 showed that LOB outperformed existing broadcast protocols in terms of packet rate and packet delay.

• 한국항행학회논문지
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• 제14권2호
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• pp.253-259
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• 2010

무선 메쉬 네트워크는 Base Station 기반과 더불어 애드혹 네트워크나 블루투스와 같이 유연성을 가진 네트워크이다. 하지만 무선 메쉬 네트워크는 높은 패킷 손실률을 보이고 TCP(Transport Control Protocol) 알고리즘은 무선 메쉬 네트워크의 패킷손실 원인을 네트워크 내의 혼잡으로 인식하기 때문에 TCP 혼잡제어 알고리즘(Congestion Control Algorithm)을 실행하게 된다. 이러한 TCP 동작은 혼잡 손실이 아닌 패킷 손실로 발생 할 경우, 상당한 성능 저하를 초래하게 된다. 본 논문에서는 무선 메쉬 네트워크에서 TCP 혼잡제어 알고리즘의 성능을 개선시키기 위해 혼잡 윈도우를 무선망에서도 적응력 있게 조절하는 연구를 제안하였다.

• 한국ITS학회 논문지
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• 제9권2호
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• pp.104-109
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• 2010

본 논문에서는 차량의 무선단말을 이용하여 유선 네트워크와 연결된 정보통신 시스템 서비스를 이용하는 텔레메틱스 환경에서 제3자의 도청이나 공격을 막기 위한 사용자 인증기술을 제안한다. 제안된 사용자 인증방식에서는 차량안의 사용자의 음성신호로부터 생성된 음성 바이오 키를 이용하여 패킷 음성데이터를 암호화하여 정보통신 시스템에 전송하고, 정보통신 시스템 서버에서는 암호화된 패킷 음성데이터로부터 사용자의 음성특징을 복원하여 미리 등록된 사용자의 음성 바이오 키와 비교하여 실시간으로 사용자를 인증한다. 실험을 통해 다양한 공격으로부터 제안된 방식에 대한 안정성을 분석하였다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.385-389
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• 2009

Multi-hop wireless mesh networks (WMNs) suffer from significant packet losses due to insufficient available bandwidth and high channel error probability. To conquer packet losses, end-to-end (E2E) error control schemes have been proposed. However, in WMNs, E2E error control schemes are not effective in adapting to the time-varying network condition due to large delay. Thus, in this paper, we propose a network-adaptive error control for video streaming over WMNs that flexibly operates E2E and hop-by-hop (HbH) error control according to network condition. Moreover, to provide lightweight support at intermediate nodes for HbH error control, we use path-partition-based adaptation. To verify the proposed scheme, we implement it and evaluate its transport performance through MPEG-2 video streaming over a real IEEE 802.11a-based WMN testbed.