• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.4
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• pp.99-104
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• 2015

Service platform which can offer various multimedia communication as the video, audio, voice and data is IMS(IP Multi-Media Subsystem). It is effective in the company is introducing only such convergence IMS services to be required to provide various multimedia services at the lowest cost and existing communication environment while keeping the maximum Therefore, in this study, we had developed IMS 코어 main system that not more than 1,000 employees of companies can effectively establish IMS solutions. This system is located at the middle between IMS terminal and CSCF(Call Session Control Function) in line with IMS services and provides CSCF in response to the IMS terminal and IMS terminal in response to the CSCF. As well, corded telephone and SIP phone which were used as terminal is linked with gateway.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.5
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• pp.941-950
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• 2001

In this paper, we present the integrated network architecture for supporting mobile IP in third generation mobile communication networks, and propose the end-to-end QoS control mechanisms of DiffServ and QoS control functions of each network element in the proposed network. First, the QoS supporting schemes of IMT-2000 are described. Second, the necessities of integrating the networks are discussed and the integrated architecture are proposed. Third, the mapping between wireless channels and DiffServ classes are presented. Finally, the end-to-end QoS control mechanisms are proposed. We also analyze the link level protocols with QoS provisioning for mobile multimedia assuming that the system support voice and data traffic simultaneously. In case of data traffic, the delay and throughput of SREJ ARQ and Type-1 Hybrid ARQ scheme are compared, and In case of voice traffic, the packet loss rate of BCH coding is analyzed according to the varying data traffic loads. The results indicate that the adaptive link level protocols are efficient to meet the QoS requirements while the complexities are increased.

• Journal of KIISE:Information Networking
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• v.37 no.4
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• pp.263-271
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• 2010

SIP/RTP-based VoIP services are being popular. Recently, however, VoIP anomaly traffic such as delay, interference and termination of call establishment, and degradation of voice quality has been reported. An attacker could intercept a packet, and obtain user and header information so as to generate an anomaly traffic, because most Korean VoIP applications do not use standard security protocols. In this paper, we propose three VoIP anomaly traffic generation methods for CANCEL;BYE DoS and RTP flooding, and a detection method through flow-based traffic measurement. From our experiments, we showed that 97% of anomaly traffic could be detected in real commercial VoIP networks in Korea.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Journal of the Korea Society of Computer and Information
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• v.10 no.4 s.36
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• pp.143-148
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• 2005

Video conference system it is various at internet and uses the reading is become accomplished. Research of like this portion synchronization of audio, the compression technique and multimedia data, supports the video conference the research of the Mbone of the IP multicast for being active. being become accomplished the multimedia service which is various an video from internet, the line speed of communication becomes high-speed anger and to follow leads is become accomplished. The video conference from opening elder brother dispersion internet network environment the problem against the image which is an image conference data and a voice security is serious and it raises its head. To sleep it presents the security method which from the video conference it follows in quality of multimedia data from the dissertation which it sees and it does.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.6
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• pp.1313-1319
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• 2011

VoIP service by taking advantage of the current PSTN network and internet over the existing telephone network at an affordable price allows you to make voice calls to the service is being expanded. However, the security of public must be maintained for security vulnerabilities in Smart Phone VoIP case problems arise, and is likely to be attacked by hackers. In this paper, the Internet, using wired and Smart Phone VoIP services may occur during analysis of the type of incident and vulnerability analysis, the eavesdropping should conduct an attack. Smart Phone VoIP with institutional administration to analyze the vulnerability OmniPeek, AirPcap the equipment is installed in a lab environment to conduct eavesdropping attack. Packet according to the analysis and eavesdropping attacks, IP confirmed that the incident as an attack by the eavesdropping as to become the test proves. In this paper, as well as Smart Phone VoIP users, the current administration and the introduction of Smart Phone service and VoIP service as a basis for enhanced security will be provided.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.5
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• pp.101-105
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• 2013

This paper addresses QoS functions in mobile backhaul network to accommodate the diverse traffics in cell site. The traffics assigned to the switching function in RAN system, such as Ethernet frame, IP packet, and ATM cell, are segmented, and then encapsulated to transfer then to the mobile backhaul network. ISP can converge all generation traffics, such as voice, HSPA, over all-IP RAN through standard pseudowire encapsulation. These can be enhanced with diverse QoS methods as well as comprehensive monitoring and diagnostic capabilities. Therefore in this paper, QoS functions under theses operations is simulated according to the encapsulation functions.

• Proceedings of the KIEE Conference
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• 2007.10a
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• pp.341-342
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• 2007

인터넷 전화가 보편화되면서 인터넷 전화를 이용한 많은 응용 기술이 발전하고 있다. 본 논문에서는 인터넷 전화를 이용하여 로봇을 제어하는 시스템을 소개하고자 한다. VoIP(Voice over Internet Protocol)를 이용하여 가정의 모든 가전기기를 제어한다. 특히 로봇이 전화를 받을 수 있으며, 기존의 텍스트 기반 제어 방식을 음성 기반으로 제어함으로써 시스템의 사전 학습이 없이도 쉽게 사용이 가능하다는 장점이 있다. 이 시스템은 두 종류의 로봇에 적용되어 실험하였으며, 대회에 출전하여 수상을 함으로써 객관적인 평가를 마쳤다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.47-54
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• 2013

In this paper, analyze the technical elements of the CTI system to design the proposed system, understand the requirements of CTI IP based system. in the Hearing-Impaired designed a CTI system of mobile phone-based services available to the CTI call center system based on this. Furthermore, we implemented voiceXML scenario data analysis modules using the JAVA language to implement the system was designed, the server provides. And an implementation of the CTI system of mobile phone base for the Hearing-Impaired by integrating the modules that have been implemented. Finally, create a scenario that uses the CTI system for mobile base to test and evaluation, based on the test scenario each functional, we conducted repeated tests. It was possible to confirm the results of time for the acquisition of the test result information has been reduced about 20 seconds on average than the audio system based on conventional.