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

• Journal of Digital Convergence
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• v.18 no.5
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• pp.249-256
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• 2020

The session initiation protocol (SIP) is a signaling protocol, which is used to controlling communication session creation, manage and finish over Internet protocol. Based on it, we can implement various services like voice based electronic commerce or instant messaging. Recently, Qiu et al. proposed an enhanced password authentication scheme for SIP. However, this paper withdraws that Qiu et al.'s scheme is weak against the off-line password guessing attack and has denial of service problem. Addition to this, we propose an improved password authentication scheme as a remedy scheme of Qiu et al.'s scheme. For this, the proposed scheme does not use server's verifier and is based on elliptic curve cryptography. Security validation is provided based on a formal validation tool ProVerif. Security analysis shows that the improved authentication scheme is strong against various attacks over SIP.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.225-228
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• 2001

There are two standards currently compete for the dominance of IP telephony architecture. Those are H.323 protocol suit by International Telecommunication Union Sector T(ITU-T) and Session Initiation Protocol/Session Description Protocol(SIP/SDP) by International Engineering Task Force(IETF). This paper has been studied a adaption VoIP in home networking, Especially SIP-based call screening service in home gateway. And then this paper has designed SIP-based call screening service in home gateway working protocol, verified them.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 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 Korea Society for Simulation
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• v.19 no.4
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• pp.43-50
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• 2010

Recently, the Internet telephony has been used rather than the traditional telephony by many Internet users, with low cost. Session Initiation Protocol(SIP) is the standard of application layer protocol for establishment and disconnection of the session for Internet telephony. SIP mainly runs over the UDP for transport. So in case of the loss of the INVITE request message, the message is retransmitted by an appropriate timer for reliable transmission of the UDP message. Though the retransmission is useful for ensuring the reliability of SIP messages sent by the users, it may cause the overload traffic in the SIP proxy server. The overload in SIP proxy servers results in the loss of many input messages. This paper presents a non-linear overload control algorithm to resolve the overload condition of the server. we simulate our proposed algorithm using the network simulator ns-2. The simulation results show that the throughput of the server with the proposed algorithm have been improved about 12% compared to the existing linear control algorithm.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.298-300
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• 2001

VoIP 기술은 사용자 사이에서 call을 설정, 변경, 종료할 수 있는 signaling 프로토콜이 필수적이다. 이러한 점에서 인터넷 프로토콜 네트워크 상에서의 call과 멀티미디어 세션의 실시간 제어를 목적으로 IETF의 MUSIC WG에서 말들어진 새로운 signaling 프로토콜이 SIP(Session Initiation Protocol)이다. 또한, IETF에서 SIP WG이 생겨나면서 이 프로토콜에 대한 연구가 더욱 활발해지고 있다. 본 노문에서는 SIP에 대한 개략적인 내용과 SIP 프로토콜의 기능들을 간략하게 소개하였고, SIP session initiation의 과정과 registration 과정 등의 동작에 대하여 설명하였다. 그리고 User Agent의 기본적인 동작을 설명하고, VOVIDA에서 개발한 SIP의 리눅스용 공개 소스를 통해서 User Agent의 실행되는 과정을 분석하였다.

• The Magazine of the IEIE
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• v.28 no.6
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• pp.80-87
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• 2001

• Proceedings of the Korea Contents Association Conference
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• 2004.05a
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• pp.342-347
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• 2004

The VoIP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easily accept the various of multimedia services from internet. Previous connection method of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group Is in use recently. Therefore, we need new interworking methodology between H.323 and SW products. In this thesis, the progress interworking method between H.323 and SIP are propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Journal of the Korea Society of Computer and Information
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• v.17 no.10
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• pp.71-77
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• 2012

In this paper, We have proposed an efficient SIP(Session Initiation Protocol) messages control which can be used to reduce conference servers load in distributed conference model. Each conference participant can control the amount of event notification messages from the conference server in real time according to its processing capacity and network circumstances. And the amount of SIP messages needed to maintain session with the conference servers can also be controlled actively in our system. The amount of SIP messages which should be processed by the conference servers can be reduced largely by using our suggested method. Therefore the conference system can be more easily expandable and total network traffic can be reduced largely in SIP environment. We have suggested new conference event packages and conference information data model with some added elements that are necessary for efficient SIP messages control. The performance of the proposed system has been evaluated by experiments.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.11a
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• pp.473-473
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• 2004