• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.526-528
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• 2002

SIP는 간단한 텍스트 기반의 응용계충 프로토콜로서, H.323을 대체할 수 있는 프로토콜이다. SIP는 인터넷 환경에 그대로 접목 할 수 있고, 새로운 기능 및 부가서비스의 제공이 용이하다는 장점을 가지고 있다. SIP 프로토콜은 요청메시지와 그에 대한 응답으로 구성되는 Request-Response방식이다. SIP의 장점은 유일한 개인 ID를 이용하여 장소와 단말기에 구애를 받지 알고 SIP서비스를 제공받을 수 있는 Personal Mobility Service에 있다. 본 논문에서는 User Agent 사이에 이루어지는 SIP 트랜잭션 (Transaction)을 기본적인 Call-Flow를 통해서 살펴보고자 한다.

• The Journal of the Korea Contents Association
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• v.6 no.11
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• pp.95-103
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• 2006

A good storytelling is essential to make a computer game more immersed and appealing as it impress the players emotion. The Storytelling consists of many factors. Among them, the Plot, or the flow of the story (Story-flow) is significant. The 'Call of Duty' is a successful computer game in the aspect of the story-flow. In this paper, the way to apply traditional theories about the story-flow to this game is analyzed. And an applying policy of traditional story-flow theories to computer games is presented.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.30 no.5
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• pp.881-893
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• 2020

CFI(Control Flow Integrity) is a mitigation mechanism that protects programs by verifying control flows. IFCC(Indirect Function Call Checks) and SCS(Shadow Call Stack), CFI supported by LLVM Clang compiler, were introduced to protect applications in Android. IFCC protects function calls and SCS protects function returns. In this paper, we propose attacks to bypass CFI on the application environment with IFCC and SCS. Even if IFCC and SCS were applied to user applications, it was confirmed that there were many code segments not protected by IFCC and SCS in the application memory. We execute code in CFI unprotected segments to construct 1) bypassing IFCC to call a protected function, 2) modulating return address via SCS bypass. We identify code segments not protected by IFCC and SCS in Android10 QP1A. 191005.007.A3. We also implement proof-of-concept exploits to demonstrate that modulation of control flow is possible in an environment where IFCC and SCS are applied.

• Journal of the Korean Operations Research and Management Science Society
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• v.30 no.3
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• pp.119-135
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• 2005

Design of location erea(LA) in a cellular network is to partition the network into clusters of cells so as to minimize the cost of location updating and paging. Most research works dealing with the LA design problem assume that the call. arrival rate and mobile flow rate are fixed parameters which can be estimated independently. In this aspect, most Problems addressed so far are deterministic LA design problems(DLADP), known to be NP hard. The mobile flow and call arrival rate are, however, varying with time and should be treated simultaneously because the call arrival rate in a cell during a day is influenced by the change of a population size of the cell. This Paper Presents a new model on IA design problems considering the time-dependent call arrival and mobile flow rate. The new model becomes a stochastic LA design problem(SLADP) because It takes into account the possibility of paging waiting and blocking caused by the changing call arrival rate and finite paging capacity. Un order to obtain the optimal solution of the LA design problem, the SIADP is transformed Into the DLADP by introducing the utilization factor of paging channels and the problem is solved iteratively until the required paging quality is satisfied. Finally, an illustrative example reflecting the metropolitan area, Seoul, is provided and the optimal partitions of a cell structure are presented.

• Proceedings of the Korea Society for Simulation Conference
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• 2003.11a
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• pp.75-85
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• 2003

With recent advances in technology and the changing nature of business, call center management has become a rapidly growing industry. However theoretical analysis about the call center system is very difficult, and the forecasting of call volume also. In the situation, it is significant that we study call-flow system, design system model and perform simulation. If these are possible, it is able to control the staff schedule and the resource management efficiently. This study introduces the process of applying the call center to simulation. So, it is feasible to break from the intuitive management by a minority manager and analyze it scientifically. The enterprise can reduce unnecessary expense, make an offer high quality to user in a keen competition

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.621-630
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• 2003

In this paper, we implemented SIP protocol that supports IPv6 and differentiated call processing scheme for NGN(Next Regeneration Network). In NGN, SIP processes call signaling among various application services. A softswitch and SIP server must give priority to sensitive services such as Fax, network management and home networking that require a fast call setup time. Also, the support of IPv6 is needed under consideration of All-IP. We proposed differentiated call processing scheme. The differentiated call processing scheme supports differentiated call processing as priority of service class on call processing in SW server We defined three service classes and use the Flow Label field of the IPv6 header for setting service class. Through the performance analysis, we proved that it improves throughput for call message with the high priority. The result of performance analysis demonstrates that differentiated call processing scheme gives better performance for the service requiring a fast session establishment in NGN.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.11b
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• pp.1189-1192
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• 2003

초기 콜 센터는 단순히 고객에게 걸려오는 전화를 응대하는 서비스 수준으로 시작되었지만, 현재는 고객과 마주하는 최 접점으로 서비스 뿐만 아니라 마켓팅, 세일즈를 행하는 기업의 경쟁력을 유지하는데 있어 필수적인 요소로 자리잡아 가고 있다. 특히 개인고객을 많이 접하는 금융기관에서는 콜 센터를 대형화, 지방 분산화 및 백업체제 구축을 위한 Multi 센터를 구축하여 콜만 처리하는 센터로서가 아니라 다양한 채널[전화,email,fax,chating]를 통해 접속되는 고객요구 사항들을 통합된 인프라에서 One Stop 으로 처리하고자 한다. 하지만 회선 교환망을 이용하는 PBX기반 Voice 회선과 IP 환경의 DATA 를 처리하는 이원화된 구조로 Multi 콜 센터를 구축 할 경우, 타 센터의 상담원에게로 콜을 설정시 시간이 많이 소요되어 센터간 Call Load Balancing 및 상담원 Skill Based Routing 을 수행함이 어렵다는 단점이 있다. 본 논문에서는 이러한 단점을 개선할 수 있는 방안으로 Voice 와 Data 를 IP 기반 단일 망으로 구성하고 WAN상에서 음성을 전달할 때 적정한 MOS(Mean opinion score)를 측정 후 객관적으로 필요한 Bandwidth 를 검증하고 Muti 콜 센터에서 효과적인 콜 load Balancing 및 Routing을 할 수 있는 VOIP기반 Convereed Virtual Call Center Architecture 설계 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1555-1558
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• 2002

SIP는 간단한 텍스트 기반의 응용계층 프로토콜로서, H.323을 대체할 수 있는 프로토콜이다. SIP는 인터넷 환경에 그대로 접목 할 수 있고, 새로운 기능 및 부가서비스의 제공이 용이하다는 장점을 가지고 있다. SIP 프로토콜은 요청메시지와 그에 대한 응답으로 구성되는 Request-Response방식이다. SIP의 장점은 유일한 개인 ID를 이용하여 장소와 단말기에 구애를 받지 않고 SIP서비스를 제공받을 수 있는 Personal Mobility Service에 있다. 본 논문에서는 User Agent와 프록시 서버(Proxy Server) 사이에 이루어지는 SIP 트랜잭션(Transaction)을 Call-Flow를 통해서 살펴보고자 한다.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.698-712
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• 2004

DiffServ network architecture does not define a call admission control procedure explicitly. In this paper, a new DiffServ QoS control mechanism is suggested which, after the call admission control, can execute packet process according to the class while, at the same time, executing on the flow based call admission control in the DiffServ network. Routers on the path from the source to the destination estimate the aggregated class traffic of the existing flows and the new incoming flow and then, perform a call admission control in accordance with the type of classes efficiently based on the required bandwidth per each class that can meet the user's QoS requirements. In order to facilitate the packet process according to the class after the flow based call admission control, a mechanism is suggested that can adjust the network resources to classes dynamically. The performance analysis on this mechanism is carried out through a simulation.

• IE interfaces
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• v.15 no.4
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• pp.391-400
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• 2002

The internet call systems using CTI(Computer Telephony Integration) functions are designed with system configuration, DFD(Data Flow Diagram) and ERD(Entity Relationship Diagram) in this paper. The internet call systems are constructed to cooperate with conventional CTI call center. The internet phone calls occurred from the web browser of customer can be connected throughout VoIP gateway and PBX to many counselors. The internet call systems can provide various services; customer information service, escorted browsing service, text chatting service, text sharing service, conference service, and statistical analysis service.