• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.784-786
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• 2001

기하 급수적으로 증가하는 인터넷 가입자의 새로운 서비스에 대한 관심과 요구가 높아지고 있다. 이러한 때에 인터넷을 통한 VoIP 기술이 각광받게 되었는데, 기존에는 ITU-T의 H.323 기술로 구현된 VoIP 서비스가 대부분이었지만 H.323이 서비스 추가에 상당한 제약을 가지므로 차세대 VoIP 기술로 SIP가 떠오르고 있다. 국내에서 SIP 기술이 확장되고 있는 가운데 다양한 어플리케이션들이 나오고 있지만 각각의 SIP의 표준 적합성을 테스트할 수 있는 장비 또는 판단 기준이 아직 미흡하다. 따라서 본 논문에 시는 SIP 상호운용성 시험을 위만 테스터 설계와 시험방식과 시험 시나리오를 기술하였다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.6
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• pp.63-69
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• 2010

Today, Internet Telephony Services based on VoIP are gaining tremendous popularity for general user. Therefore a various demands of the user keep up increase, the most important requirements of these is voice security about telephony system. It is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed VPN IP-PBX for the voice security and measured conversation quality by adopting VPN IPsec based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of voice data. This VPN IP-PBX that is connected Soft-phone provide various optional services.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.146-149
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• 2001

In order to provide VoIP service that using SIP on the Internet, users must register their current location to receive call from other cal]or. 13y using REGISTER method, users register his or her SIP URL address and currently contactable addresses to registrar server. The registered addresses are not only SIP URI but also mai]to, http, tel, and so on. In these procedures, we must consider authentication mechanism whether he/she has authority to handle the record or not. In this paper, we will briefly describe authentication mechanism that used in SIP and design of registrar server that support the 3rd party registration.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.7-16
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• 2012

In this paper we describe a VoIP transcript system which is able to record call conversation between counselor and customer in an IP contact center based on IP telephony environment. The transcript system, designed and implemented in this paper, uses packet sniffering to capture packets without imposing network overhead on overall system. It can decode H.323 and SIP which are used to setup call sessions in VoIP environment and captures voice data and record without any loss of contents. Implemented transcript system can be integrated with CTI system in that it can manage and record call more effectively. It is designed generically so that it is implemented both on Windows and Linux environment.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.413-416
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• 2013

스마트폰의 보급으로 인하여 무료로 메시지를 보내거나 전화 통화를 할 수 있게 되었다. 그러나 기업 사용자를 위한 VoIP 앱은 거의 없는 것이 현실이다. 이에 본 논문에서는 기업 사용자들을 위한 SIP 기반의 VoIP 구내교환망 시스템과 이를 접근하기 위한 안드로이드 클라이언트 앱을 제안한다. 제안된 시스템에서는 기존 전화 앱에서 비즈니스 사용자를 위한 기능의 부재를 보완하기 위해 Push 기능을 활용한 자동 주소록 싱크 기능과 Push형 컨퍼런스 콜을 기능이 제공된다. 본 시스템을 통하여 회사의 직원들이 보다 사내 환경에 맞는 실용적이고 편리한 음성 커뮤니케이션을 할 수 있을 것으로 기대한다.

• The KIPS Transactions:PartC
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• v.15C no.4
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• pp.245-252
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• 2008

An emergency service over Voice over IP (VoIP) network is an essential condition, like the existing telecommunication services. To support for the emergency services, standardization works have been performed. The National Emergency Number Association (NENA) has been developing the framework and procedures for an emergency service for Non-IP based network, rather than protocols. In contrast, the Internet Engineering Task Force (IETF) has been only focused on end-to-end IP-based emergency calls. The NENA architecture is incompatible with the IETF protocols. To solve the problem, we design and implement a SIP-based VoIP emergency system by adopting the NENA architecture and by applying IETF protocols, for both IP-based Pubic Safety Answering Point (PSAP) and PSTN-based PSAP. It is implemented and tested under UNIX environment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.127-133
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• 2009

It has been 10 years long since the emergence of Dialpad, the first commercial VoIP service from Saerome Technology launched on 1999. In spite of so many research papers and implementations of the VoIP and commercial services with hundreds of VoIP-related products vendors and tens of VoIP service providers during the last decade, the resulting market share of the VoIP callings at the year 2007 in Korea is just about 0.3%, which is extraordinary lower one than expected. There have been proposed several facts for the reason including the QoS issue and the incomplete governing system. In this paper we suggest that the reason is based on the sociocultural issue and further we assert and verify that it can never replace the existing legacy telephone system for the current VoIP service model, as long as the telecommunication companies continue the legacy telephone service. We also suggest the another choice, to upgrade the concept of the current VoIP service to a new paradigm based on SIP and presence service, while letting the legacy telephone system be as it is.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.

• 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.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.05b
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• pp.1453-1456
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• 2003

인터넷의 발달로 인하여 기존 전화망을 인터넷 망으로 적은 비용으로 사용하고자 VoIP 가 연구되기 시작하였다. VoIP 의 기존전화망과의 품질의 차이에도 불구하고 대폭적인 통화비용의 절감과 다양한 이점으로 많은 연구가 되고 있다. 이에 통화품질의 향상을 위하여 QoS에 대한 연구가 활발히 진행되고 있다. 여기서는 QoS를 지원하는 Q-SIP 서버와 Policy Based 의 COPS 를 이용하여 Pre-COPS update message 와 Reservation COPS cache로 정책결정과 QoS를 지원하고자 한다.