• 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.

• 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.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.5
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• pp.181-190
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• 2013

PMIPv6(Proxy Mobile IPv6) is a network-based IP mobility management protocol, which can control the mobility without depending on the type of access system or the capability of mobile node. Combining it with SIP mobility, it can establish the route optimization effectively and ensure the terminal mobility and the session mobility. There are many literatures on PMIPv6-SIP in mobility management, but efficient performance analysis and mathematical modeling has not been standardized. For this, a new PMIPv6-SIP architecture is proposed to compare with Pure-SIP in terms of the handover delay and packet loss.

• Molecules and Cells
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• v.21 no.3
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• pp.389-394
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• 2006

Human seven-in-absentia (SIAH)-interacting protein (SIP) is a component of the E3 ligase complex targeting beta-catenin for destruction. Arabidopsis has one SIP protein (AtSIP) with 32% amino acid sequence identity to SIP. To investigate the functions of AtSIP, we isolated an atsip knockout mutant, and generated transgenic plants overexpressing AtSIP. The growth rates and morphologies of the atsip and transgenic plants were indistinguishable from those of wild type. However, atsip plants were more susceptible to Pseudomonas syringae infection, and the transgenic plants overexpressing AtSIP were more resistant. Consistent with this, RNA blot analysis showed that the AtSIP gene is strongly induced by wounding and hydrogen peroxide treatment. In addition, when plants were infected with P. syringae, AtSIP was transiently induced prior to PR-1 induction. These observations show that Arabidopsis AtSIP plays a role in resistance to pathogenic infection.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.6
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• pp.1022-1025
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• 2008

Session Initiation Protocol (SIP) has become the call control protocol of choice for Voice over IP (VoIP) networks because of its open and extensible nature. However, the integrity of call signaling between sites is of utmost importance, and SIP is vulnerable to attackers when left unprotected. Currently a herby-hop security model is prevalent, wherein intermediaries forward a request towards the destination user agent sewer (UAS) without a user agent client (UAC) knowing whether or not the intermediary behaved in a trusted manner. This paper presents an integrated security model for SIP-based VoIP network by combining hop-by-hop security and end-to-end security.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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• pp.653-654
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• 2009

인터넷전화의 활성화와 더불어 SIP 프로토콜의 사용이 증가하고 있다. 최근 DDoS 공격이 주요위협으로 이슈가 되고 있으며, 향후 SIP관련 서비스 대상 서비스 거부 공격 위협도 증가할 것으로 예상된다. 본 논문에서는 SIP 프로토콜의 특성을 이용한 SIP서비스 거부 공격들의 유형을 살펴보고, SIP 프로토콜의 특성을 이용한 서비스거부 공격 탐지를 위한 고려사항을 바탕으로 분석시스템 아키텍처를 제안한다. SIP서비스 거부공격의 유형은 분류 기준에 따라 다양할 수 있으나, 본 논문에서는 SIP 프로토콜의 특성을 이용한 공격 기법을 대상으로 한다. SIP서비스 거부 공격 트래픽 분석을 위해 정보수집, 정보분석, 정보관리 기능을 고려한다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.289-291
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• 2002

본 논문에서는 SIP(Session Initiation Protocol)을 사용하는 VoIP(Voice over IP) 네트워크에서 SIP 트래픽을 감시, 분석하는 관리 도구인 SIPMan 을 설계하고 구현하는 연구를 소개한다. 구현하는 SIPMan 은 실시간으로 SIP 패킷을 캡처하여 call 에 대한 다양한 정보를 분석하여 DB 에 저장할 수도 있다. VoIP 망 관리자는 SIPM무의 web GUI를 사용하여 call detail record, SIP 트래픽 정보 등을 모니터링말 수 있다.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05d
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• pp.631-636
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• 2002

H.323은 ITU-T의 설러 표준들을 조합하여 구성된 프로토콜로서 복잡한 구조를 가지며 인터넷에의 접목이 어려운 단정을 갖고 있다. 이에 반해 SIP는 텍스트 기반의 프로토콜로서 간단한 구조를 갖는다는 점과 인터넷에의 접목과 새로운 부가서비스 창출의 용이하다는 점으로 많은 표준화 기구들이 차세대 네트워크의 호 설정 프로토콜로 채택하고 있다. 비록 SIP가 차세대 네트워크 및 서비스에 이용될 전망이지만 이미 H.323 기반 서비스가 전세계적으로 진행되어 왔으므로 SIP 기반의 서비스와 H.323 기반의 서비스 사이의 호환성이 요구된다. 따라서 본 논문에서는 인터넷 상에서 음성 및 영상 서비스의 제공을 위해 이용되고 있는 서비스의 표준안 인 SIP와 ITU-T의 표준안 인 H.323 간의 연동 기술을 제안한다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.481-483
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• 2003

분산가상서버 구조는 인터넷에서 멀티미디어 통신 제어를 위해 개발된 SIP를 이동 애드혹 네트워크(MANET)에 적용하기 위하여 SIP의 프락시서버, 등록서버(Registrar)를 도메인이름을 기반으로 각 노드들로 분산하여 관리하고 탐색하는 방법을 제공한다. 가상 서버를 통해 이동 단말들은 SIP의 응답속도와 서비스 가용도를 향상시키고, 인터넷에서 사용하던 SIP에 기반한 서비스를 MANET에서 자유롭게 사용할 수 있다.