• 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.

• Journal of KIISE:Computing Practices and Letters
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• v.5 no.5
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• pp.619-627
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• 1999

To support multimedia services between network domains with different environments, it is required to map the functionalities in many aspects. In this paper, we implemented interworking gateway which provides protocol conversion and QoS(Quality of Service) adaptation to interwork DAVIC services based on ATM(Asynchronous TRansfer Model )network and Internet AV services. The interworking gateway converts RTSP(Real-Time Streaming Protocol ) message into DSM-CC(Digital Storage Media Command & Control) messages to control the stream that is served in ATM network, and transmits data stream by using RTP(Real-Time Transport Protocol) The interworking gateway provides QoS adaptation functionalities by QoS monitoring and MPEG filtering to meet the variation of network bandwidth.

• Information and Communications Magazine
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• v.14 no.6
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• pp.123-132
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• 1997

• Proceedings of the Korean Information Science Society Conference
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• 2002.10c
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• pp.439-441
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• 2002

Real-time Transport Protocol (RTP) is widely used in VoIP stacks charging the multimedia data delivery. Concerning with payload protection of RTP packets, Secure RTP has been discussed in IETF AVT group to provide confidentiality and authentication features using block ciphering and message authentication coding. However, Secure RTP only concentrates on payload protection. Signcryption is a good candidate for key agreement. This paper proposes a key establishment protocol using Signcryption and shows example implementation of a secure VoIP application based on Secure RTP with the proposed scheme.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.13 no.4
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• pp.177-186
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• 2003

DRM(Digital Rights Management) is a technology that manages secure distributions and copyrights of digital contents on the Internet. It is general giving the rights to use the encrypted contents that are downloaded by a simple authorization process in the existing DRM system. Once this is done you are allowed to access. In this paper, we use RTP(Real-time Transport Protocol) for end-to-end real-time data transmission. And the system is designed to make it Possible to Protect copyrights and to distribute contents with safety through periodic authentication. We implemented DRM system to stand this basis. The proposed system vests only authorized users with authority to access the license. Hence it prevents contents to be distributed and copied illegally on networks.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04b
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• pp.393-395
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• 2000

CSCW 시스템은 효율적인 공동작업의 인지를 위해 텔리포인팅, 스냅샷, 문서/프로젝트 추적등과 같은 기능들을 제공한다. 이 중 텔리포인팅은 지역 시스템의 공유윈도우에서 텔리포인터를 통해 발생한 이벤트를 원격지 시스템의 공유윈도우에 동기적으로 표현하는 기법이다. 텔리포인팅은 여러 연구들에서 응용된 바 있으나, 텔리포인팅의 프로토콜에 관한 연구는 미흡하다. 본 논문은 RTP(Real-Time Transport Protocol) 기반의 텔리포인팅을 위한 RTP 페이로드 형식을 정의한다.

• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.147-150
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• 2003

In this paper, we describe a remote surveillance system to monitor the states of remote processes through Internet on Web browser The images captured from multiple remote cameras are transferred to data server exploiting RTP (Real-time Transport Protocol) and displayed up to 4 channels simultaneously. The angle of cameras is moved vertically an horizontally through RS-232C communication line by motor control board e벼ipped with AT90S2313 CPU.

• 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.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.