• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2150-2164
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• 1997

Two distinct features required in transport service and protocol to meet new multimedia applications requirements are multicast transmission capability and enhanced QoS facility. In this paper, we presents major functions and facilities of enhanced communications transport service definition supporting these requirements. At first, major fundamental issues, that should be handled to design multi-peer communcation service and protocol, are resoved and applied to design service definition. Secondly, we proposed several new functions for enhanced transport service such as heterogeneous TC establishment, invitation capability, TC-ownership transfer, restriction of transmit concurrency. The detils of proposed enhanced transport service are TC creation, TC invitation, TC join, multicase data transfer, pause of data transfer, resulme of data transfer, report of specific status, TC leave, TC termination, TC-ownership transfer, Token transfer services. The proposed transport service was submitted and accepted as a draft text for committee draft of international standard by the international standradization body such as JTC1/SC6 and ITU-T SG 7.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.2
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• pp.104-113
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• 2009

Recently, high-speed Internet users and broadband network infrastructure are increasing. However, the TCP protocol widely used in the Internet is an appropriate transport only for the legacy Internet, it is inefficient for traffic transport for network environments with large bandwidth-delay product. In order to remedy this problem, there have been many researches about improved congestion control algorithms for broadband networks. In these studies, most of performance evaluation has been done by simulations. In this paper, after we reviewed the proposed high bandwidth transport layer protocols, we implemented a real testbed, measured the performance and analyzed the problems of high-speed transport protocols through the international research network TEIN2 between three nations, Korea, China, and Germany.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1828-1840
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• 1993

In this paper, a centralized protocol model for videoconference service over a wide-area-network is presented. the model is comprised of three distinct components : clients, server and mixer. The server handles all can management functions, and the mixer performs realtime traffic mixing. The control and data are separated, flowing over separated connections in the model. A set of new protocols are defined : call management protocol, media transport protocol, and multimedia synchronization protocol. This paper also presents the functional design of the mixer.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.13-16
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• 2000

Recently, with the advent of satellite communication technologies, the network access service through satellites has emerged as an alternative to existing ones. Of satellite access services including multimedia services, the high speed internet access service will be one of the important services. Particularly, the efficient transportation of real time internet traffic seems very essential. The aim of this research is to examine the applicability of the IETF RTP protocol in the satellite access network. The RTP has been developed for the QoS-met transportation protocol of real time traffic over the existing best-effort internet. This paper summarizes the RTP protocol, and describes the architecture of the implementation of the RTP. And also, the compressed RTP header protocol is discussed for the applicability to the satellite link.

• Journal of Communications and Networks
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• v.11 no.1
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• pp.72-83
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• 2009

We have studied the effect of head of the line blocking (HOLB) on session initiation protocol (SIP) session establishment delays. Our results are based on experiments performed in a test bed and on the public Internet. We used the stream control transmission protocol (SCTP) as a transport for SIP because SCTP can be configured to suffer or to avoid HOLB. Our experiments show that the effect of HOLB on session establishment delays generally starts to be significant starting at fairly low packet loss rates. However, there are scenarios where network conditions are good enough to make the effect of HOLB insignificant.

• Journal of the Korea Society of Computer and Information
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• v.21 no.1
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• pp.79-84
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• 2016

In this paper, we propose reliable message transmission scheme based on CoAP, considering the constrained feature of IoT device, such as low power, the limited memory size and low computing capacity. Recently, the various kinds of application protocol has been studied to support IoT environments. In particular, CoAP protocol was developed as application protocol for IoT at the IETF core WG. However, because CoAP protocol is deigned to be used in constrained node, this protocol uses UDP at transport layer. Thus, data loss may occur frequently in network congestion environments. The proposed scheme, in this paper, is to overcome the problem of frequent data loss with low overhead. Also it includes the function which is to minimize the data loss in sleep mode of IoT device.

• International Journal of Internet, Broadcasting and Communication
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• v.11 no.3
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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.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.811-813
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• 2002

WAP은 WAP Forum에서 제정한 무선 환경에서의 데이터 통신을 위한 표준 프로토콜이다. WAP에서는 보안 통신을 위한 프로토콜로서 WTLS(Wireless Transport Layer Security)를 제안하고 있다. 이것은 TCP/IP 상의 TLS(Transport Layer Security)를 바탕으로 무선 환경에 맞게 최적화한 것이다. 그러나, WAP은 기본적으로 게이트웨이 모델에 따른 프로토콜이라는 점과 무선 구간에서의 전송 효율을 높이기 위한 인코딩 기능 때문에 게이트웨이에서 클라이언트와 서버 사이에 교환되는 정보가 노출되는 이른바 종단간의 보안 (End-to-End Security)문제가 존재한다. 이러한 이유로 유선에서와는 달리 안전한 종단간 보안을 제공하지 못하고 있다. 이에 본 논문에서는 기존 TLS와 WTLS를 합친 새로운 TLS(Transport Layer Security)-Plus 프로토콜을 제안하여 무선 환경에서 무선 단말기에 부담을 주지 않는 안전한 종단간 보안을 제공하려고 한다.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.241-246
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• 2009

Advances in mobile and wireless networking technologies have enabled mobile ad hoc networks applicable to a wide range of areas. Many cases of even such ad hoc networks demand to be accessible to the global network. Owing to be diversified in features depending on applications, however, some those networks may consist of devices or nodes which do not facilitate all the same transport protocols. This results in unreachable situations of establishing ad hoc network: such as non-existence of all required access points, faults or contention in a path of particular protocol communication. Interworking between different transport protocols can alleviate such problems. This paper proposes an interworking scheme to improve data transport performance and network accessibility, especially in such a mobile ad hoc network that is applicable to data communications among ships or user's convenient u-health services. Modeling and simulation analysis are used to verify the improvement.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.281-284
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• 2004

This paper presents the design and implementation of a QoS based enhanced collaboration system in MDO. This is an efficient distributed communication tool between designers. It supports text communication, audio/video communication, file transfer and XML data sending/receiving. Specially, this system supports a dynamic QoS self-adaptation by using the improved direct adjustment algorithm (DAA+). The original direct adjustment algorithm adjusts the transmission rate according to the congestion level of the network, based on the end to end real time transport protocol (RTP), and controls the transmission rate by using the information of loss ratio in real time transport control protocol (RTCP). But the direct adjustment algorithm does not consider when the RTCP packets are lost. We suggest an improved direct adjustment algorithm to solve this problem. We apply our improved direct adjustment algorithm to our of QoS (Quality of Service) [1] based collaboration system and show the improved performance of transmission rate and loss ratio.