• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2321-2328
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• 2006

Today, SIP standard (The signalling protocol for the Internet phone service) raises to be the standard technique because the expandability is high and complexity is low. It is widely investigated and actively advocated to use Si81a1 ring protocol for SIP in VoIP service. SIP service can be applied even outside the Internet phone service; instance messaging and various multimedia technology are just an example. This paper proposed an embodiment proxy server for rambling support to use JAIN SIP API. It provides standard interface for testing the Proxy server for SIP and embodiment of user agent that transfer instant massaging and voice communication.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.5
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• pp.967-973
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• 2009

Recently, as 3rd-generation mobile communication services using high-speed data rate system are widely employed, the demand for a variety of real-time data services such as VoIP service are also increased. Unlike typical data packets, VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that schedule two deferent kinds of packets efficiently, considering the characteristics of VoIP. Basically this algorithm considers both time delay and channel condition and gives priority depending on the time delay. Simulation results show that the proposed algorithm works more efficiently than conventional algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.2
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• pp.293-300
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• 2007

As the development of the various IP-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN-based telephony service. But there are many issues resolved to spread the Internet telephony service. One of them is to support the emergency services in the Internet telephony. In the case of USA, it has been regulated that 911 services should be supported in the Internet telephony services using VoIP on the similar performance level to PSTN 911 service. According to the regulation, basic VoIP 911 calls should be routed to the general access line of LEA without the location information or the callback number, but the enhanced VoIP 911 calls with the location information and callback number should be routed on the dedicated 911 network and destined to the local 911 distribution center such as PSAP. But, in the current VoIP-based Internet telephony network, the emergency call service has not been handled as one of the special services as well at has a worse performance in comparison to it on PSTN. Moreover, the service has a critical problem that it can not be destined to the nearest PSAP because of the insufficient information about the location information and the call back number. In this paper, we suggest the SIP-based emergency call service mechanism in order to resolve the problems above mentioned. This suggested mechanism is implemented to show its effectiveness and efficiency.

• The KIPS Transactions:PartC
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• v.15C no.2
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• pp.103-110
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• 2008

Internet telephone uses VoIP as its communication protocol, where it provides various additional services with inexpensive fare when the terminal is connected to the Internet. The existing auto provisioning systems that manage Internet telephone terminals are targeting large size VoIP service providers or enterprises, and has few consideration on the convenience of management for small size environments. The difficult installation and management procedures of the current auto provisioning systems prevents Internet telephone system from getting popular. Easy installation and management system is needed in order to spread out Internet telephone system. In this paper, we design and implement the auto provisioning system that provides easy installation and management services in small size environments such as SOHO environments.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.4
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• pp.493-498
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• 2011

In this paper, Transmission performance over MANET(Mobile Ad-hoc Networks) under DDoS Intrusions is evaluated. Intrusion counterplan requirement, which have to be used for MANET under DDoS intrusions, is suggested through this evaluation. VoIP simulator based on NS-2 network simulator is used for performance measurement. MOS, network delay, packet loss rate and call connection rate is measured with this simulation. Finally, requirement of intrusion continuing time shorter then 10 seconds is suggested for VoIP service over MANETs under DDoS intrusions.

• 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.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

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

• Electronics and Telecommunications Trends
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• v.19 no.3 s.87
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• pp.136-144
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• 2004

인터넷의 발달 및 VoIP의 보급으로 인해 VoIP 서비스의 품질에 대한 관심이 증가하고 있다. 그 동안은 망사업자 관점에서 망의 품질을 개선하기 위한 MPLS, Diffserv, RSVP 등의 연구가 진행되어 왔으나, 실제로 서비스 품질은 망뿐만 아니라 단말 등의 품질에도 영향을 받기 때문에 망 사업자의 관점에서 보는 서비스 품질 기준이 아닌, 고객의 관점에서 인식 가능한 수준에서의 종단간 서비스 품질을 다룰 필요가 있다. 본 고에서는 서비스 품질이란 무엇인지 살펴보고, 국제표준단체의 서비스 품질 관련 연구 및 VoIP 서비스를 위한 음성 품질 평가 기술에 대하여 살펴본다.

• The KIPS Transactions:PartC
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• v.18C no.3
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• pp.143-150
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• 2011

The adoption of VoIP is continuously increasing in public institutions, private enterprises and households due to cheaper cost and various supplementary services. Also, it is expected to spread widely the use of VoIP in mobile environment through the increasing use of smartphone. With the growing concern over the incidents of VoIP service while the VoIP service has become increasingly. Especially eavesdropping, it is possible to invade user privacy and drain the secret of company. So, it is important to adopt the protocols for VoIP secure communication. VoIP security protocols are already adopted in public institutions, but it is not adopted in private enterprises and households. In addition, it is necessary to verify whether the VoIP security protocol could be adopted or not in mobile VoIP due to its limited computing power. This paper compared the VoIP security protocol under fixed network and mobile network through performance evaluation. Finally, we found that it is possible to adopt the VoIP security protocols in mobile network.