• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.16 no.11
• /
• pp.2405-2410
• /
• 2012

The 4th generation mobile communication system, LTE, does not support an additional core network to provide voice service, and it is merged into a packet network based on all IP. Although Voice service over LTE can be supported by VoIP, it will be provided by the existing 3G networks because of the discontinuity of LTE coverage. However, it is inevitable to adopt VoIP over LTE to provide high quality voice service. In this paper, we investigate the capacity of VoIP over LTE. Our results indicate that spectral efficiency can be significantly improved as channel bandwidth increases in terms of VoLTE capacity. In addition, we can achieve higher VoLTE capacity without decreasing control channel capacity.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.10
• /
• pp.2292-2302
• /
• 1997

In this paper, we propose a wireless MAC protocol named APRMA, which is capable of supporting the ABR type data service and maximizing channel utilization. In PRMA protocol, data terminals with random data packets cannot reserve slot. That is, slot reservation is applicable to the time constraint voice packet exclusively. But the reservation scheme has to be performed for loss sensitive data packet, and so data packets can get their quality of service. Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. But in service aspects, if a fixed bandwidth is allocated to data terminals, time constraint voice packets may have a low efficiency. So in this study, the terminal which wants to request for ABR tyupe service, acquires a minimum bandwidth from system for thefirst time. If the system has extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot. As a result, APRMA protocol cansupporty the data service with loss sensitivity and maintain their channel utilization high. Also high priority services like voice can be satisfied with their QoS by APRMA.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.2A
• /
• pp.174-185
• /
• 2010

A seamless handover between the 3G LTE and legacy 3G system is required for the smooth deployment of the 3G LTE system which is the next generation cellular network system. Especially on voice call handover, the service interruption time is very sensitive for user's satisfaction and therefore, a seamless voice call handover scheme is necessarily required. However, handover between the 3G LTE and the 3G CS system is hard to be achieved due to the lack of interface between two systems and the restriction of radio resource. In this paper, a new network entity called SCSE is proposed and inter-working between the 3G LTE and the 3G CS systems is enabled. Also contributed to the feature of the SCSE, the handover procedure is simplified and the service interruption time is minimized as a consequence. The evaluation result shows that the proposed SCSE scheme exclusively meets the service interruption time requirement which is smaller than 300 ms.

• Journal of Internet Computing and Services
• /
• v.10 no.6
• /
• pp.27-36
• /
• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

• Journal of the Institute of Convergence Signal Processing
• /
• v.9 no.3
• /
• pp.208-212
• /
• 2008

This paper proposes a model of a voice communication using bluetooth and a mobile, and implements a hands-free for this model. A Proposed model uses bluetooth for a voice network, and is possible to share a voice with a great number of people using preemptive algorithm proposed in this paper. There are two types of models. One is a model only using a hands-free, the other is a model using a mobile. Second model uses a scatternet and a call forwarding service for a call on a voice communication. In case of using a scatternet, scatternet is composed of a piconet of a voice communication and a piconet of a call. In case of using a call forwarding service, the mobiles share information of each others before formation of a network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2002.11a
• /
• pp.758-761
• /
• 2002

Voice of U(VoIP) technology Provides voice service as well as data service via Internet. It has been a promising technology as Internet grows fast and the requirements are increasing. Recently, serveral protocols have been created to allow telephone calls to be made over IP networks, notably, SIP and H.323. Due to introducing SIP and H.323, There are many change at internet telephony service. Internet telephony enables a wealth of new service possibility Users can control telephony service directly. In this paper, we design and implementation CPL client based on SIP system.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.7 no.5
• /
• pp.949-954
• /
• 2003

In this study, Automatic Speech Recognition Application System is designed and implemented to realize transformation from present GUI-centered web services to VUI-centered web service. Due to ASP's restriction with web in reusability and portability, in this study, Automatic Speech Recognition Application System with Javabeans Component Architecture is devised and studied. Also the voice web browsing which is able to transfer voice and graphic information simultaneously is studied using Remote AWT(Abstract Windows Toolkit).

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.9A
• /
• pp.737-748
• /
• 2005

Two flexible admission control schemes for integrated voice and stream-type data services are proposed in DS-CDMA systems. Most Previous studies on admission control have focused on integration of short, bursty Packet-type data services and conventional voice services. However, stream-type data services with a relatively long service holding time are expected to be a considerable portion of data traffic in future generation cellular systems. Scheme I is a basic scheme that accommodates both voice and data services with full bandwidth. However, voice services are given priority over data services using the duration difference between the holding times for these services. Scheme ll uses a different method to efficiently give priority to voice services over stream-type data services. An additional interference margin for voice services is provided by suppressing interference from stream-type data services according to voice access requests and a varying interference status. Performance of the two schemes is evaluated by developing Markovian models. Numerical results show that the voice capacity is highly sensitive to the service holding time of data services while the performance measures of data services are not highly sensitive. Scheme H is a significant improvement over Scheme I for accommodating voice and stream-type data services

• The KIPS Transactions:PartC
• /
• v.18C no.2
• /
• pp.79-88
• /
• 2011

It is necessary for us to enhance the security service on smart phone by using voice data on authentication procedure. In this study, a voice data based one-time password generation mechanism is designed and implemented for enhancing user authentication on smart phone. After receiving a PIN value from the server, a user inputs his/her own voice biometric data using mike device on smart phone. And then this captured a voice biometric data will be used to generate one-time token on server side after verification procedures. Based on those mutual authentication steps, a voice data based one-time password(V-OTP) will be generated by client module after receiving the one-time token from the server finally. Using proposed voice one-time password mechanism, it is possible for us to provide more secure user authentication service on smart phone.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.7
• /
• pp.101-108
• /
• 2001

In this paper we focus on dynamic reservation slot allocation scheme for supporting QoS of a voice traffic in WATM MAC. Especially, voice traffic is the most important real-time object, and so we propose a new MAC protocol for voice traffic over WATM networks in the multimedia environment. According to the characteristics of voice traffic which is repeatedly in silent state and active state, new protocol allocates reservation slots dynamically with respect to the number of silent voice source of which starting time is stored to the state table in base station (BS). The simulation results show that the proposed protocol has better performance than slotted ALOHA in average access delay, collision rate, better than NC-PRMA(Non Collision Packet Reservation Multiple Access) in band width efficiency, and can provide a certain level of QoS requirement by the given slot assignment even though the number of voice terminals is increased.