• Journal of KIISE:Computing Practices and Letters
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• v.10 no.6
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• pp.460-471
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• 2004

Since the RTP is suitable for real-time data transmission in multimedia services like VoD, AoD, and VoIP, it has been adopted as a real-time transport protocol by RTSP, H.323, and SIP. Even though the RTP protocol stack for embedded systems has been in great need for efficient support of multimedia services, such a stack has not been developed yet. In this paper, we explain embeddedRTP which supports the RTP protocol stack at the kernel level so that it is suitable for embedded systems. Since embeddedRTP is designed to reside in the UBP module, existing applications which rely ell TCP/IP services can proceed the same as before, while applications which rely on the RTP protocol stack can request HTP services through embeddedRTp API. EmbeddedRTP stores transmitted RTP packets into per session packet buffer, using the packet's port number and multimedia session information. Communications between applications and embeddedRTP is performed through system calls and signal mechanisms. Additionally, embeddedRTP API makes it possible to develop applications more conveniently. Our performance test shows that packet-processing speed of embeddedRTP is about 7.5 times faster than that oi VCL RTP for multimedia streaming services on PDA in spite that its object code size is reduced about by 58% with respect to UCL RTP's.

• 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.

• Journal of the Korea Computer Industry Society
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• v.3 no.9
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• pp.1223-1234
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• 2002

RTP payload must be encrypted for providing commercial VOD service or private video conference over the Internet. 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 md 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 Korea Information Processing Society Conference
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• 2001.10b
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• pp.1501-1504
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• 2001

MPEG-4 비디오 스트림을 전송하기 위해 RTP 프로토콜을 사용할 경우 RTP 패킷에 MPEG-4 비디오 스트림을 수용하기에 적절한 페이로드 포맷이 정의되어야 한다. RTP 페이로드에 MPEG-4 비디오 스트림을 수용하기 위한 기존의 방법은 MPEG-4 코덱으로부터 생성된 비디오 스트림의 형태에 따라 많은 종류의 페이로드 포맷이 정의되어야 하는 문제점이 있거나, 혹은 각 계층에서 생성된 스트림을 (in 패킷화하는 방법이 명료하지 않은 문제점 등이 있다. 이 논문에서는 MPEG-4 시스템을 사용 하지 않고, MPEG-4 비디오 기초스트림을 RTP 패킷에 담아 전송하기 위한 새로운 RTP 페이로드 포맷을 제안하였고, 제안된 포맷을 이용하여 MPEG-4 비디오를 전송하는 시스템을 설계 및 구현하였다. 제안된 RTP 페이로드 포맷은 비디오 객체(Video Object)에 대한 RTP 페이로드 포맷만을 정의하여 포맷의 종류를 최소화함으로써 포맷의 종류가 많아짐으로해서 생기는 부가적인 오버헤드를 감소시켰으며, 또한 하나의 RTP 패킷에는 단 하나의 비디오 객체에 대한 정보만을 저장하여 비디오 객체간의 독립성을 유지하였다. 제안된 포맷을 이용하여 구현된 MPEG가 비디오 전송시스템은 RTP 패킷의 크기를 pad-MTU와 같거나 작게 설계함으로써, IP 계층에서 RTP패킷의 단편화 현상이 발생하지 않고 효율적인 전송이 이루어지도록 하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6C
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• pp.471-479
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• 2008

In this paper, we propose a novel algorithm to automatically generate an RTP timestamp value that is required for the RTP packetization in order to transmit SVC video over various If networks such as Internet. Unlike the conventional single layer coding algorithms such as H.263, MPEG-4 and H.264, SVC generates a multi-layered single bitstream which is composed of a base layer and one or more enhancement layers in order to simultaneously provide temporal, spatial, and SNR scalability. Especially, in order to provide temporal scalability based on hierarchical B-picture prediction structure, the encoding (or transmission) and display order of pictures in SVC coding is completely decoupled. Thus, the timestamp value to be specified at the header of each RTP packet in video transmission does not increase monotonically according to the display time instant of each picture. Until now, no method for automatically generating an RTP timestamp when SVC video is loaded in a RTP packet has teen introduced. In this paper, a novel automatic RTP timestamp generation method exploiting the TID (temporal ID) field of the SVC NAL unit header is proposed to accommodate the SVC video transmission.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.4
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• pp.185-194
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• 2010

In this paper, we propose an application layer-based vertical handover management protocol, called multihoming RTP (mhRTP), for real-time applications with seamless mobility across heterogeneous wireless access networks. The proposed multi-homing RTP provides a soft handover by utilizing multiple available wireless access network interfaces simultaneously. The newly available path is dynamically added to the ongoing session by the mhRTP session manager. Also the decision making of QoS-improving or QoS-guaranteed handover is possible based on the estimation of available bandwidth in each candidate network. The performances of the proposed mhRTP have been analyzed through a series of simulations on OPNET network simulator. From the performance analysis, we confirmed that the proposed mhRTP can provide QoS-guaranteed vertical handover with efficient session managements.

• Proceedings of the Korean Society for Technology of Plasticity Conference
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• 2004.10a
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• pp.294-297
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• 2004

RTP(Rapid Thermal Pressing) is to fabricate desired pattern on polymer substrate by pressing patterned mold against the substrate heated around glass transition temperature. For a successful RTP process, the whole process including heating, molding, cooling and demolding should be conducted 'rapidly' as possible. As the RTP process is effective in replicating patterns on flat large surface without causing shape distortion after cooling, it is being widely used for fabricating various micro/bio application components, especially with channel-type microstructures on surface. This investigation finally aims to develop a RTP process machine for mass-producing micro/bio application components. As a first step for that purpose, we intended to examine the technological difficulties for realizing mass production by RTP process. Therefore, in the current paper, 4 kinds of RTP machines were examined and then the RTP process was conducted experimentally for PMMA film by using one of the machines, HEX 03. The micro-patterned molds used for RTP experiment was fabricated from silicon wafer by semi-conduct process. The replicated micro patterns on PMMA films were examined using SEM and the causes of defect observed in the replicated patterns were discussed.

• The Transactions of the Korea Information Processing Society
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• v.6 no.9
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• pp.2512-2523
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• 1999

In this paper, we present the design and implementation results of RTP(Real-time Transport Protocol) which is introduced to provide end-to-end delivery service for multimedia data with real-time characteristics and to provide QoS(quality of service) monitoring services. Conventional RTP communication modules have some problems such as poor reusability and inefficiency. Reusability problem stems from the inherent characteristics of RTP that is framework is deliberately not complete, and so, RTP should have been designed and implemented in application-imbedded from. The RTP communication module, proposed in this paper, is designed and implemented in application-independent from in order to be used in all kinds of high level applications independent of their architectures and functions.

• Journal of the Korea Society of Computer and Information
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• v.18 no.4
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• pp.87-96
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• 2013

In this paper, we suggest redesigned RTP protocol that is able to perform encryption of VoIP media information for single private network and between the different private networks. And we conduct a test for performance analysis. Such as SRTP or ZRTP methods have been used for VoIP media encryption. But, the existing encryption techniques have problem that can not perform end-to-end encryption between different private networks. In order to solve this problem, in this paper, we redesign RTP protocol. Redesigned RTP includes all information for encryption of VoIP media. Therefore the encryption is not affected by modification of SIP and SDP information that occurred in gateway. Also, redesigned RTP includes code for whether or not to apply encryption. By using the code, modification of RTP header from gateway prevents. As a result, redesigned RTP maintain the integrity and the RTP is able to perform encryption between the different private networks. Also, we conduct a test for performance analysis of SRTP, ZRTP and redesigned RTP.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.682-684
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• 2002

본 논문에서는 RTP와 TCP가 공존하는 네트워크에서 RTP 패킷의 크기, interval,전송라인의 대역폭, Queue의 크기, delay의 변화에 따라 throughput의 특징에 대하여 알아보기 위해서 ns(network simulator)를 이용하여 RTP, TCP_Reno, TCP_Vegas로 구성된 네트워크를 구성하고 시뮬레이션을 통해서 throughput의 특징 및 원인을 분석했다.