• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.414-418
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• 2003

VoIP is a core technology used to transmit both voice and data in an integrated packet' form. Within this technology, SIP is the signaling protocol used for 'real time' call services; particularly those where H323 is used. Yet, when considering the needs of mobile users, it is essential we integrate VoIp within the mobile technology so the mobile host is able to receive the 'packet' transported and by, and connected to, any available internet-address. For all this to occur, we need to improve Network Delay by reducing transmission problems associated with mobile services. If we are to obtain an optimal service then we must reduce any network delays which may arise from joining Mobile IP and VoIp services. This paper, therefore, considers how, unlike previous research, these delays may be improved through the use of the signaling technology\ulcorner SIP. It also considers how this research may be introduced into current wired and wireless integrated services enabling them to use the IP 'packet'.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.5
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• pp.1131-1136
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• 2015

VoIP and IP Telephony telephony technology development is a growing by easy to using IP-PBX by using phone from using existing lines rather than the internet. IP-PBX do not use the phone line from phone work for many companies and institutions of management costs reduce as provides similar to regular phone line quality. But IP-PBX to introduce for need to be the initial cost on is should buy for expensive hardware equipment or commercial software. In this paper, suggest way to introduce IP-PBX do not buy expensive hardware equipment or commercial software. Suggest IP-PBX on designed and implement for IP-PBX server using Raspberry Pi and Asterisk. And verification treatise on the suitability of conducted by voice calls based on IP-PBX between PC and a Smartphone

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.1-8
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• 2006

Currently, there are many efforts underway to provide Internet service on integrated wireless and wired networks. Supporting IP mobility is one of the major issues to construct IP based wireless network. Mobile IP has been proposed to solve the IP Mobility problem. But, in processing frequent handoffs in cellular based wireless access network, Micro mobility protocols have been proposed to solve these problems. Micro mobility protocols proposed the Cellular IP, HAWII, and Hierarchical Mobile IP. Cellular IP attracts special attention for it's seamless mobility support in limited geographical areas. New BS must be known to occur begging of handoff in Cellular IP indirect handoff. Therefore during perceiving of hanoff, packet loss or packet duplication still can occur in Cellular IP indirect handoff, which results in the degradation of UDP and TCP performance. In this paper, we propose a enhanced indirect handoff scheme for Cellular IP. Proposed handoff scheme is using a crossover node to minimize the signalling procedure and using a buffering to minimize the packet loss or packet duplication.

• Journal of Digital Convergence
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• v.18 no.1
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• pp.309-314
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• 2020

In this study, look into the future prospects of mobile games produced using IP, in the domestic market and to present more effective IP selection criteria when producing new mobile games using IP. To this end, classified IP-enabled mobile games and analyzed IPs used for IP-enabled mobile games by each type for games that are located in the top 200 in sales ranking of Google Play Store. According to the analysis, the mobile game using IP has shown significant performance in terms of sales performance, and the trend will continue in 2020. It was analyzed that 'Game' types of IP using games(based on domestic games) can achieve the highest sales results. Since the success of a game affects a variety of factors other than IP, further research is needed in more diverse ways on the impact of IP on the success.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.575-584
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• 2009

In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11C
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• pp.1-9
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• 2001

The Mobile IP protocol allows IP hosts to move between different networks without changing their IP addresses. The Mobile IP systems supporting the local registration were introduced to reduce the number of times when a home registration with the remotely located Home Agent is needed. The local registration Mobile IP protocol enhanced the performance by processing the MN\`s registration requests at a local agent. However. the local registration approach may consider other aspects of the Mobile IP systems as the FA fault tolerance. In this paper, we will briefly review previous protocols to support the FA fault tolerance in hierarchical local registration Mobile If system and will propose a fault tolerance protocol with backup FA in hierarchical local registration mobile IP t() enhance the efficiency of such systems against Foreign Agent failures.

• Journal of the Korea Society of Computer and Information
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• v.17 no.3
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• pp.113-120
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• 2012

VoIP provides voice data service using existing IP networks and has received much attention recently. VoIP service has a variety of security vulnerabilities. Types of main attacks on VoIP service are tapping/interception, DoS attacks, spam, misuse of service attacks and the like. Of these, confidential information leak because of tapping/interception has been considered as a critical problem. Encryption techniques, such as SRTP and ZRTP, are mostly used to prevent tap and intercept on VoIP media information. In general, VoIP service has two service scenarios. First, VoIP service operates within a single private network. Second, VoIP service operates between different private networks. Both SRTP and ZRTP for VoIP media information within a single private network can perform encryption. But they can not perform encryption between different private networks. In order to solve this problem, in this paper, we modify SRTP protocol. And then, we propose an encryption method that can perform encryption of VoIP media information between the different private networks.

• Journal of Advanced Navigation Technology
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• v.12 no.1
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• pp.28-33
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• 2008

VoIP provided voice service using Internet is receiving footlights when it escapes an initial curiosity. VoIP interest became larger, because it can transfer existing phone service and deliver voice data through internet technology. Is inheriting as it is threats of IP base because a VoIP technology provides audiocommunication service taking advantage of an existing IP technology, and have new threats that happen from new know-hows for VoIP service offer. In this paper, presented about problem and consideration and countermeasure of examines about security threat of VoIP service, and applies security.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1109-1116
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• 2010

In this paper, some performance characteristics of VoIP(Voice over Internet Protocol) for MANET(Mobile Ad-hoc Networks) with simulation is studied and appropriate condition for implementation of VoIP service is suggested. VoIP simulator is implemented with NS(Network Simulator)-2. VoIP traffic for simulation is generated with some codecs of G.711, G.723.1, G.726-32, G.729A, GSM.AMR and iLBC. As simulation results for traffic transmission under $670{\times}670m$ 50node MANET environment, performance data for MOS(Mean Opinion Score), network delay, packet loss rate and transmission bandwidth are measured. Normalized analysis about measured results shows that maximum VoIP connection satisfying VoIP service quality condition is 15.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.7B
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• pp.464-468
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• 2005

The DVR industry has been grown with their own, proprietary systems without the need of standards because the security products are independently installed in small scale and the communication needs are very limited. The industry might have tighter security than having standardized products, but the appearance of IP surveillance and the need of communications with IP products such as IP Phones and IP-PBX made the standardization unavoidable. In this paper, we built an IP surveillance system with SIP based session control and investigated the applications of it other than the security itself.