• 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.13 no.1
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• pp.67-74
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• 2009

This paper presents a method of improving the service availability by distributing the traffic of voice/data calls over the multi-layer cells in a mobile communication network. The traffic model is described and the call handling performance is analyzed. In our method, a fast moving call is moved to and serviced in the upper layer cell. A call is also moved upward when an overflow occurs. But unlike other methods, the call that is moved upward in the overflow case is the one which has the longest sojourn time in the cell. Moreover, when the call that was moved upward due to overflow condition stays longer than a certain period of time in the upper layer cell, the system moves the call back to the lower layer in order to save the more expensive resources of the upper layer cell. Call handling performance of this method evaluated from M/M/C/K models shows clear improvement with respect to call blocking probability and forced termination probability.

• The Journal of the Korea Contents Association
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• v.11 no.12
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• pp.570-577
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• 2011

In this paper, Policy-based QoS in 3GPP service network management techniques are not applied to the DiffServ technology is applied to the first interpretation. The next PBNM and DiffServ associated technologies by applying QoS performance improvement is verified. In this case that PBNM and DiffServ technology is applied, the amount of voice traffic reduced about 1 msec while best-effort traffic occurs 75 percent of the output link capacity. Also, video traffic which is the same as data traffic showed a decreased $0\sim10^{-4}$ packet loss rate than the case that DiffServ technology is applied. We apply the appropriate policy PBNM and DiffServ QoS mechanisms of the existing set of policies is not affected, just by using the appropriate 3GPP Service QoS level to suit the network operation, management can do that was found. This analytical method based on the University of California at Berkeley through NS-2 DiffServ technology into existing systems and next-generation networks mandated PBNM and DiffServ technology is applied to performance evaluation for the case.

• Journal of Korea Multimedia Society
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• v.12 no.2
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• pp.257-271
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• 2009

In this paper, we propose a protocol compression scheme for enhancing the performance of call processing of PTT(Push-to-Talk) which is one of the important services in IMS(IP Multimedia Subsystem), a next generation integrated wired/wireless packet communication network. To service the PTT on an IMS network, it should use the same call setup procedure as legacy Mobile and TRS(Trunked Radio System) networks and have a fast call setup time and enough communication bandwidth because a number of terminals should be able to exchange same data in real time. The proposed A+SigComp scheme reduces the initial call setup delay of SIP by about 10%, which is used by PTT service for call setup. In addition, the A+ROHC scheme is proposed to compress the header of RTP packets transferred during PTT voice transmission and, as a result, about 5% of increase in communication efficiency is observed.

• Journal of Korea Multimedia Society
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• v.10 no.7
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• pp.902-911
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• 2007

GSM platform is a hugely successful wireless technology and an unprecedented story of global achievement. In less than ten years since the first GSM network was commercially launched, it became the world's leading and fastest growing mobile standard, using over 1 billion GSM subscribers across more than 200 countries of the world. GSM platform evolved into 3th generation mobile communication which includes not only voice call services but also the international roaming and various kinds of the multimedia services. GSM is an essential element techniques a safe data transmission and a personal private protection while support services. However, a crypto algorithm and a secure protocol for a safe data communication using GSM are indicating various kinds of problems. In this paper, we propose a more safer and more efficient authentication protocol in 3th generation network through analysis of GSM security mechanism of 2th/2.5th generation. This security protocol offers enforced security efficiency by using user verification between SIM/ME and reduction of authentication and key agreement step between SIM/ME/AuC.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.652-658
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• 2009

As the rate of personnel expenses of contact center operating costs is the highest, we can reduce operating costs, if we answer a automatic phone call wholly or partially. The MCS(Managed Contact Services) play an important role in cost reduction and work efficiency related with voice self-services. The excellent functions of the MCS are able to implement self-service applications in network connected with effective network routing of the enterprise. The MCS makes a proper consultant rapidly response and process customer calls, improve customer processing services, and consequently increase customer satisfaction. The increase of customer satisfaction lead to improve profits and reduce the cost of building contact center infrastructure. In this paper, we analyze the contact center line capacity and PBX capacity based on the MCS. We can reduce communication costs and personnel expenses by reducing the call shifting need between consultants and rapidly solving customer questions with the MCS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.2
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• pp.381-389
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• 1994

In the communication network, multimedia service such as high quality voice, high speed data, image etc. will be added to the existing service. This service generates new requirements for the communication networks. The priority control mechanism can be used to control multimedia traffics generated by many communication systems. The priority mechanism which assigns prioirities to generated cells according to service quality is one of the traffic control. The priority assignment can be divided by priority criterion for each traffic characteristics such as loss sensitivity and delay sensitivity. In this paper, we alnalyzed the partial buffur sharing (PBS) mechani느 as a traffic control reducing the cell loss, and proposed analysis method. We analyzed the PBS mechanism using classical approach as a Markov chain. In order to validata proposed analysis method, simulation is performed using simulation package SIMSCRIPT 11.5. From this results, we confirmed that proposed analysis method can be verified. Also, we presented cell loss probability of ATM network when this results are to be applied to ATM networks.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.5
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• pp.85-89
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• 2017

In this papers, we apply high performance hardware and machine learning algorithm to build an advanced VAD algorithm for speech enhancement. Since speech is made of series of phoneme, using recurrent neural network (RNN) which consider previous data is proper method to build a speech model. It is impossible to study every noise in real world. So our algorithm is builded by phoneme based study. we detect voice present frames in noisy speech signal and make enhancement of the speech signal. Phoneme based RNN model shows advanced performance in speech signal which has high correlation among each frames. To verify the performance of proposed algorithm, we compare VAD result with label data and speech enhancement result in various noise environments with previous speech enhancement algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.1-8
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• 2011

Power Line Communication (PLC: Power Line Communication) to propagate the current power lines are in every household of the existing infrastructure is the most extensive network configuration. In addition, the cost required for network configuration, the advantage of almost zero for the investors and is sufficient to attract the attention of operators. The PLC is supply power to power lines used the voice and data communication technologies put it on KHz ~ tens of hundreds of high-frequency signal MHz. But because uses power lines as existing wireless communications systems will occurs interference. The notch filters of a common way to eliminate the interference are used. In this paper, a dynamic high-performance notch filter applying CIC filter performance was verified through MATLAB and was implemented using a TI's TMS320C6416T DSP board.

• Journal of Korea Multimedia Society
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• v.3 no.6
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• pp.611-616
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• 2000

By the most important media to deliver the contents on a remote instruction we commonly use audio and documents. A number of remote instruction system are trying to offer the video, but they did not acquire satisfiable results because of the limited network and width. Also, they use the general web browsers that have a lot of unspecific users access the contents. Also, they use the general web browsers that have a lot of unspecific users access the contents. Like this most systems that use the continuous media have not offer the satisfiable contents because of the network limitation. Moreover, because they use the web browser, they offer the contents having documents(web pages) only. In this paper, we propose a web-based remote instruction system on real-time using audio and documents which are the most important media for a information delivery. In addition to the system, we use a action synchronization mechanism between the web browsers. If the instructor uses web pages on his computer and explains the contents of them, the learners see the same web pages as the instructor's and listens to his voice.