• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.1
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• pp.75-82
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• 2003

We can foresee the rapidly growing of users by development of VoIP service what can transmit a audio traffic with low cost among a lots application using internet. VoIP needs a standardized protocol that is able to do signaling for offering high quality of services such as mobility, universal number, multiparty conference, voice mail, automatic call distribution. At the present time, a base composition elements of SIP(Session Intiation Protocol) are developing for offering VoIP based SIP in the inside and outside of the country, because SIP of IETF which has a strength from 'fast connection', 'parsing' & 'easy to compile' points of view. This paper suggests a type of Hybrid SIP Server for providing some services as 'a reducing load of SIP server that process a request method from users', 'efficiency of managing networks', 'offering services to many users'.

• 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.

• Proceedings of the International Microelectronics And Packaging Society Conference
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• 2003.09a
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• pp.109-119
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• 2003

o In order to start SIP Project .Marketing (& ASIC team) should present biz planning, schedule, device/SIP specs., in SIP TFT prior to request SIP development for package development project. .In order to prevent (PCB) revision, test, burn-in, & quality strategy should be fixed by SIP TFT (PE/Test, QA) prior to request for PKG development. .Target product price/cost, package/ test cost should be delivered and reviewed. o Minimum Information for PCB Design, Package Size, and Cost .(Required) package form factor: size, height, type (BGA, QFP), Pin count/pitch .(Estimated) each die size including scribe lane .(Estimated) pad inform. : count, pitch, configuration(in-line/staggered), (open) size .(Estimated) each device (I/O & Core) power (especially for DRAM embedded SIP) .SIP Block diagram, and net-list using excel sheet format o Why is the initial evaluation important\ulcorner .The higher logic power resulted in spec. over of DRAM Tjmax. This caused business drop longrightarrow Thermal simulation of some SIP product is essential in the beginning stage of SIP business planning (or design) stage. (i.e., DRAM embedded SIP) .When SIP is developed using discrete packages, the I/O driver Capa. of each device may be so high for SIP. Since I/O driver capa. was optimized to discrete package and set board environment, this resulted in severe noise problem in SIP. longrightarrow In this case, the electrical performance of product (including PKG) should have been considered (simulated) in the beginning stage of business planning (or design).

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.5
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• pp.507-526
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• 2009

Since SIP uses a text-based message format and is open to the public Internet, it provides a number of potential opportunities for Denial of Service (DoS) attacks in a similar manner to most Internet applications. In this paper, we propose an effective detection method for SIP flooding attacks in order to deal with the problems of conventional schemes. We derive the upper bound of the possible number of SIP messages, considering not only the network congestion status but also the different properties of individual SIP messages such as INVITE, BYE and CANCEL. The proposed method can be easily extended to detect flooding attacks by other SIP messages.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.3
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• pp.251-265
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• 2009

VoIP service is the transmission of voice data using SIP protocol on an IP-based network. The SIP protocol has many advantages, such as providing IP-based voice communication and multimedia service with low communication cost. Therefore, the SIP protocol disseminated quickly. However, SIP protocol exposes new forms of vulnerabilities to malicious attacks, such as message flooding attack. It also incurs threats from many existing vulnerabilities as occurs for IP-based protocol. In this paper, we propose a new virtual proxy to cooperate with the existing Proxy Server to provide state monitoring and detect SIP message flooding attack with IP/MAC authentication. Based on a proposed virtual proxy, the proposed system enhances SIP attack detection performance with minimal latency of SIP packet transmission.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1587-1590
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• 2002

차세대 네트워크 및 서비스를 위한 프로토콜로 IETF (Internet Engineering Task Force)의 SIP (Session Initiation Protocol)가 각광을 받고 있다. SIP는 PC, PDA, IP Phone과 같은 VoIP (Voice over IP) 단말간에 호 설정 프로토콜로 사용된다. SIP는 기본적으로는 양 단말간 호설정 프로토콜이지만 응용, 인터넷 단말기, 네트워크 장치에 구성요소로 구성할 수 있어 쉽게 적용 가능하기 때문에 모든 응용의 호설정 프로토콜로서 넓게 채택되어지고 있다. 그러나 SIP는 텍스트 기반 프로토콜로서 구현은 쉬우나 실제 표준에 맞게 구현하였는지는 판단하기가 어렵다. 따라서 구현된 SIP 프로토콜이 표준에 맞게 구현하였는지를 시험할 필요가 있다. 이를 해결하기 위해서, 본 논문에서는 SIP 시뮬레이터를 구현하였다. SIP 시뮬레이터는 구현된 SIP 제품을 인터넷상에서 시험할 수 있을 뿐만 아니라 시험 시나리오를 선택할 수 있고, 시험 과정을 그래픽하게 볼 수 있으며, 시험 결과를 확인할 수 있다. SIP 시뮬레이터는 사용자 인터페이스인 Testing User Agent와, 테스트 시나리오를 수행하는 Test Server로 구성된다. 사용자 인터페이스는 모든 플랫폼에 적용 가능한 Java를 사용하였으며, Test Server는 Linux 환경하에서 C++을 사용하여 구현하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4B
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• pp.411-423
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• 2004

This paper defines test methodologies and performance indices for SIP server system, and describes elements that can have influence on the test of SIP network equipments. Moreover, we implement a tester to evaluate the performance of SIP Servers such as Registrar and Proxy server. The performance indices for testing SIP servers are message processing rate, transaction delay, and call success probability. The parameters that can have an effect on the performance of SIP servers are user population, transport protocol, method of database access, method of DNS, call creation pattern, definition of transactions, and size of packets. We tested several SIP servers that act as Registrar, Proxy, and Redirect server using the implemented SIP tester, and, as a result, verified functions of the tester and performance indices and input parameters defined in this paper. Performance indices and methodologies presented in this paper can be used to evaluate SIP servers in NGN

• Journal of Internet Computing and Services
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• v.9 no.6
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• pp.37-48
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• 2008

VoIP service is a transmission of voice data using SIP protocol on IP based network, The SIP protocol has many advantages such as providing IP based voice communication and multimedia service with cheap communication cost and so on. Therefore the SIP protocol spread out very quickly. But, SIP protocol exposes new forms of vulnerabilities on malicious attacks such as Message Flooding attack and protocol parsing attack. And it also suffers threats from many existing vulnerabilities like on IP based protocol. In this paper, we propose a new Virtual Proxy Server system in front of the existed Proxy Server for anomaly detection of SIP attack and stateful management of SIP session with enhanced security. Based on stateful virtual proxy server, out solution shows promising SIP Message Flooding attack verification and detection performance with minimized latency on SIP packet transmission.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.19 no.3
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• pp.73-82
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• 2009

This paper proposes an efficient and secure user authentication and key agreement scheme instead of the HTTP digest and TLS between the SIP UA and server. Although a number of security schemes for authentication and key exchange in SIP network are proposed, they still suffer from heavy computation overhead on the UA's side. The proposed scheme uses the HTIP Digest authentication and employs the Diffie-Hellman algorithm to protect user password against dictionary attacks. For a resource-constrained SIP UA, the proposed scheme delegates cryptographically computational operations like an exponentiation operation to the SIP server so that it is more efficient than the existing schemes in terms of energy consumption on the UA. Furthermore, it allows the proposed scheme to be easily applied to the deployed SIP networks since it does not require major modification to the signaling path associated with current SIP standard.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.526-528
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• 2002

SIP는 간단한 텍스트 기반의 응용계충 프로토콜로서, H.323을 대체할 수 있는 프로토콜이다. SIP는 인터넷 환경에 그대로 접목 할 수 있고, 새로운 기능 및 부가서비스의 제공이 용이하다는 장점을 가지고 있다. SIP 프로토콜은 요청메시지와 그에 대한 응답으로 구성되는 Request-Response방식이다. SIP의 장점은 유일한 개인 ID를 이용하여 장소와 단말기에 구애를 받지 알고 SIP서비스를 제공받을 수 있는 Personal Mobility Service에 있다. 본 논문에서는 User Agent 사이에 이루어지는 SIP 트랜잭션 (Transaction)을 기본적인 Call-Flow를 통해서 살펴보고자 한다.