• The KIPS Transactions:PartC
• /
• v.11C no.4
• /
• pp.539-544
• /
• 2004

This paper presents the implementation of the bluetooth headset and the audio gateway connected to the mobile Phone in the embedded environment. The bluetooth module includes the BC02 processor chip, the BCSP02 firmware and the bluelab software Including bluetooth protocol stack. The above components in the bluetooth module developed at CSR company are used as the development environment. The application program using API functions supported by bluelab is coded by C language and loaded on the flash ROM of the bluetooth module. The cail processing capacity measuring the call setup time and the clearing time between the audio gateway and the headset is considered as the performance parameter of the developed systems. As a call setup and clearing time between the audio gateway and the headset is about 88.8ms, the call processing capacity is about 11 calls per second. Therefore the performance result is satisfied in the aspect of the call processing time.

• The Journal of the Korea Contents Association
• /
• v.8 no.2
• /
• pp.67-73
• /
• 2008

Recently, due to the rapid advances of mobile communication, multimedia service can be provided by mobile devices. Cellular network tends to uses SIP as a call setup protocol in order to provide various multimedia service to consumers. Cellular network holds the characteristic of higher BER(Bit Error Rate), narrower bandwidth than the wired network. The value of SIP RTI(Retransmission Time interval) based on the wired network decreases a network efficiency and increases a call setup delay over cellular network. By using NS-2 simulator, we show new SIP RTI value adequate over cellular network. We design and implement home network by using the modified SIP that is suitable for cellular network.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.5A
• /
• pp.660-667
• /
• 2000

This paper aims at analyzing the traffic of CDMA base station. RTD(Round Trip Delay) method, used for the study, is one of the developed tool for calculating the call setup time between the BTS(Base Station TransceiverSubsystem) and mobile station.We compare the calculated call setup time in air with the field experiments.And we suggest the RTD method for dividing the traffic of the connected repeater from that of the BTS, andwe can testify it by the experiment which analyze the difference of the received time between the base stationand the repeater including the forced delay elements.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.7
• /
• pp.1437-1447
• /
• 1997

This paper proposes a cluster-based PCS(Personal Communication Services) network architecture to support increased signaling and control traffic. The procedures are presented to serve the mjobility management and the efficient procesing of the call-control functional using distributed servers. Especially when call-setups are made within a cluster, this distributed servers. Especially when call-setup time and the amount of control traffic of H-HLS (High level Home Location Server) in the network as L-HLS(Low lever Home Location Server) serves the call on behalf of H-HLS. Performance and reliability of the architecture have been also cairried out.

• Journal of the Korea Society of Computer and Information
• /
• v.9 no.3
• /
• pp.163-169
• /
• 2004

In this paper, a conventional key exchange method simply performs the key exchange setup step based on discrete algebraic subjects. But the mutual-authentication procedure of wireless PKI for reducing authentication time uses an elliptical curve for a key exchange setup step. Proposed handover method shows reduced handover processing time than conventional method since it can reduce CRL retrieval time. Also, we compared proposed authentication structure and conventional algorithm. and simulation results show that proposed authentication method outperforms conventional algorithm in all environment regardless of call arrival rate. queue service rate. queue size.

• The KIPS Transactions:PartC
• /
• v.9C no.6
• /
• pp.927-934
• /
• 2002

In order to provide various multimedia services in a wireless network, the call admission control for wireless channels should be resolved at the time of call setup and handoff by moving mobile terminal. In this paper. we propose a reliable DCAC( Distributed Call Admission Control)scheme using virtual cluster concept. The proposed DCAC scheme considers the state of $1^{st}$ and $2^{nd}$ adjacent cells to provide a reliable call handling. The proposed scheme is analyzed by simulations and mathematical methods.

• Journal of Information Processing Systems
• /
• v.18 no.1
• /
• pp.1-11
• /
• 2022

Online video telephony systems have been increasingly used in several industrial areas because of coronavirus disease 2019 (COVID-19) spread. The existing session initiation protocol (SIP)-based video call system is being usefully utilized, however, there is a limitation that it is very inconvenient for users to transmit additional information during conversation to the other party in real time. To overcome this problem, an enhanced scheme is presented based on augmented real-time reality (AR). In this scheme, augmented information is automatically searched from the Internet and displayed on the user's device during video telephony. The proposed approach was qualitatively evaluated by comparing it with other conferencing systems. Furthermore, to evaluate the feasibility of the approach, we implemented a simple network application that can generate SIP call requests and answer with AR object pre-fetching. Using this application, the call setup time was measured and compared between the original SIP and pre-fetching schemes. The advantage of this approach is that it can increase the convenience of a user's mobile phone by providing a way to automatically deliver the required text or images to the receiving side.

• ETRI Journal
• /
• v.25 no.1
• /
• pp.34-40
• /
• 2003

This paper considers a TMN-based management system for the management of public ATM switching networks using a four-level hierarchical structure consisting of one network management system, several element management systems, and several agent-ATM switch pairs. Using Jackson's queuing model, we analyze the effects of one TMN command on the performance of the component ATM switch in processing local calls. The TMN command considered is the permanent virtual call connection. We analyze four performance measures of ATM switches- utilization, mean queue length and mean waiting time for the processor directly interfacing with the subscriber lines and trunks, and the call setup delay of the ATM switch- and compare the results with those from Jackson's queuing model.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2001.10b
• /
• pp.1533-1536
• /
• 2001

This paper considers the TMN system for management of public ATM switching network which has the four-level hierarchical structure consisting of one network management system, a few element management system and several agent-ATM switch pairs, respectively. The effects of one TMN command on the local call processing performance of the component ATM switch an analyzed using Jackson's queueing model. The TMN command considered is the permanent virtual call connection, and the performance measures of ATM switch are the utilization, mean queue length and mean waiting time for the processor interfacing the subscriber lines and trunks directly, and the call setup delay of the ATM switch.

• ETRI Journal
• /
• v.36 no.5
• /
• pp.865-875
• /
• 2014

Voice phishing (vishing) uses social engineering, based on people's trust in telephone services, to trick people into divulging financial data or transferring money to a scammer. In a vishing attack, a scammer often modifies the telephone number that appears on the victim's phone to mislead the victim into believing that the phone call is coming from a trusted source, since people typically judge a caller's legitimacy by the displayed phone number. We propose a system named iVisher for detecting a concealed incoming number (that is, caller ID) in Session Initiation Protocol-based Voice-over-Internet Protocol initiated phone calls. Our results demonstrate that iVisher is capable of detecting a concealed caller ID without significantly impacting upon the overall call setup time.