• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.43-48
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• 2004

RTP that is proposed supplement of real-time services on internet environment, as Real-time Transport Protocol, is the protocol that for the purpose of sending data of stream type. RTP and RTCP(Real-time Transport Control Protocol) basically work at the same time, RTCP serves with state information of network at present. RTP has important properties of a transport protocol that runs on end-to-end systems and provides demultiplexing. It also offer reliability and protocol-defined flow／congestion control that transport protocol like TCP can not provides. In this paper, we look around concept and construction of Differentiated sen1ice tint run on RTP and by setting parameters of packet transfer method be used CBQ(Class-Based Queuing) for packet transfer on Differentiated service, each service queue controls properly through packet scheduling method, such as WRR(Weighted Round Robin) and PRR(Packet-by-packet Round Robin) all service classes do not experience the starvation and confirm the performance through computer simulation to achieve fairly scheduling.

• 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 the Korea Institute of Information and Communication Engineering
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• v.5 no.6
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• pp.1104-1109
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• 2001

In this paper, we describe a method for real-time transmitting video streams based on RTP. In order to guarantee synchronous video streams, we propose a method to grasp network situations by analyzing end-to-end network traffic. In addition, we present an algorithm for satisfying QoS requirements of real-time multimedia transmission and maintaining continuity of transmission. It describes a buffering method for overcoming bandwidth limitations and an analyzing method based on RTCP for grasping network traffic situation to resolve the problem of real-time transmission of video streams.

• Journal of the Korea Society of Computer and Information
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• v.8 no.3
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• pp.144-149
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• 2003

Encryption/decryption delay is minimized, because there are constraints in transporting a multimedia data through the Internet. Therefore, encryption algorithm is needed which is changed with considering network traffic and load. And during many users participate in the same multimedia service, an user who already left the service can receive and the method which decrypt the RTP payload is needed because of knowing the encryption key. Therefore in this paper, Security Control Protocol for RTP is designed and implemented for changing the encryption algorithm and the key.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.255-258
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• 2003

TCP is taking over 95％ among the Internet traffics. Recently the demands of multimedia services in the Internet has been increasing. These multimedia services mostly need real-time deliverly, and then RTP has been a de facto to transmission protocol for these real-time multimedia services. RTP uses UDP as its underlying transport protocol, and thus it does not support any rate and congestion control. Thus, for fair use of the Internet bandwidth with TCP traffics. RTP also needs a rate control. One constraint of RTP is that the feedback information(delivered by, RTP's twin protocol, RTCP) is recommended to be sent no less than 5 seconds. In this paper, we propose a TCP-friendly RTP rate control which use only RTCP feedback information at every 5 seconds. The experiment results show that our proposed algorithm works. But, it is found that we need more time to test the effects of parameters and policies of the algorithms, which will be reported later.

• Convergence Security Journal
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• v.2 no.2
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• pp.1-8
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• 2002

Real-Time Protocol (RTP) is used for multimedia information transmission and RTP payload must be encrypted for providing multimedia information security. Encryption/decryption delay is minimized, because there are constraints in transporting a multimedia data through the Internet. Therefore, encryption algorithm is changed with considering network traffic and load. Doting many users participate in the same multimedia service, an user who already left the service can receive and decrypt the RTP payload because of knowing the encryption key. In this paper, Security Control Protocol for RTP is designed and implemented for changing the encryption algorithm and the key.

• Convergence Security Journal
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• v.2 no.1
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• pp.49-57
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• 2002

Real-Time Protocol (RTP) is used for multimedia information transmission and RTP payload must be encrypted for providing multimedia information security. Encryption/decryption delay is minimized, because there are constraints in transporting a multimedia data through the Internet. Therefore, encryption algorithm is changed with considering network traffic and load. During many users participate in the same multimedia service, an user who already left the service can receive and decrypt the RTP payload because of knowing the encryption key. In this paper, Security Control Protocol for RTP is designed and implemented for changing the encryption algorithm and the key.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.169-172
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• 2001

In this paper, we describe a method for real-time transmitting video streams based on RTP In order to guarantee synchronous video streams, we propose a method to grasp network situations by analyzing end-to-end network traffic. In addition, we present an algorithm for satisfying QoS requirements of real-time multimedia transmission and maintaining continuity of transmission. It describes a buffering method for overcoming bandwidth limitations and an analyzing method based on RTCP for grasping network traffic situation to resolve the problem of real-time transmission of video streams.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.6
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• pp.326-336
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• 2006

Due to the evolution of wired and wireless communication technologies and the Internet, multimedia services such as Internet broadcast and VOD have been prevalent recently. RTP is designed to be suitable for transmission of real-time multimedia data on the Internet by IETF While a variety of applications have utilized different RTPs implemented as a library, embeddedRTP is RTP-based kernel-level protocol that resolved performance issues of this kind of RTPs. This paper proposes the ExtendedERTP protocol based on existing embeddedRTP. This new protocol resolves a couple of issues such as packet processing overhead and buffer requirement and combines its APIs with BSD socket APIs which have been widely utilized in network applications. This paper demonstrates that this integration makes it possible to transmit real-time multimedia data through the accustomed interface of BSD socket APIs with nominal extra overhead. This paper also proposes a scheme for improving packet processing time by 15$\sim$20% and another scheme for reducing memory requirement for packet processing to about 3.5%, comparing with those of embeddedRTP.