• Journal of Internet Computing and Services
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• v.13 no.3
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• pp.31-47
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• 2012

The mobile Virtual Private Network (MVPN) of Internet Engineering Task Force (IETF) is not designed to support NEwork MObility (NEMO) and is not suitable for real-time applications. Therefore, an architecture and protocol which supports VPN in NEMO are needed. In this paper, we proposed the cost-reduced secure mobility management scheme (CR-SeMMS) which is designed for real-time applications in conjunction with VPN and also which is based on the session initiation protocol (SIP). Our scheme is to support MVPN in NEMO, so that the session is well maintained while the entire network is moved. Further, in order to reduce the authentication delay time which considers as a delaying factor in hands-off operations, the signaling time which occurs to maintain the session is shortened through proposing the hands-off scheme adopting an authentication method based on HMAC based One Time Password (HOTP). Finally, our simulation results show the improvement of the average hands-off performance time between our proposed scheme and the existing schemes.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.1
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• pp.47-52
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• 2017

This study is BIT(: Bus Information Terminal) features that take advantage of KEPCO eIoT(: energy Internet of Thing) platform, and it's Network configuration is composed of display terminal device, gateway, platform, and the service server. The key features are parts for processing protocol data between the gateway and the device using LoRa(: Long Range) technology, Intelligent applications and SIP(: Session Initiation Protocol) data handling connected to the Taxi reservation system. And the resource tree provided BIT for the service, which commonly used in the application server and the device.

• 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.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.1
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• pp.130-140
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• 2010

SIP (Session Initiation Protocol) is a signaling protocol to provide IP-based VoIP (Voice over IP) service. However, many security vulnerabilities exist as the SIP protocol utilizes the existing IP based network. The SIP Malformed message attacks may cause malfunction on VoIP services by changing the transmitted SIP header information. Additionally, there are several threats such that an attacker can extract personal information on SIP client system by inserting malicious code into SIP header. Therefore, the alternative measures should be required. In this study, we analyzed the existing research on the SIP anomaly message detection mechanism against SIP attack. And then, we proposed a Co-occurrence based SIP packet analysis mechanism, which has been used on language processing techniques. We proposed a association rule generation and an attack detection technique by using the actual SIP session state. Experimental results showed that the average detection rate was 87% on SIP attacks in case of using the proposed technique.

• 한국정보통신설비학회:학술대회논문집
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• 2003.08a
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• pp.223-225
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• 2003

Session Initiation Protocol(이하 SIP)은 H.323 network에서 제기된 문제점을 해결하고 다양한 multimedia service를 제공하기위해 IETF에서 정의된 protocol로서, 3GPP의 IP Multimedia Subsystem(이하 IMS)의 기반 protocol로 채택되었다. IMS는 multimedia service를 제공하기 위한 infrastructure로서 CSCF, HSS, MRF, MGCF, IM-GW등 기능적으로 분리된 여러 entity로 구성되며, 향후 추가될 다양한 부가 service에 유연하게 대처할 수 있는 Application Server(이하 AS) mechanism을 포함하고 있다. 현재 AS mechanism을 이용하여 3GPP규격 상에 정의된 대표적 서비스는 presence service이며, presence service network은 presence server, presence agent, presence proxy, watcher등으로 구성된다. 마지막으로 상기에 언급한 규격을 바탕으로 실제 구축된 Instant messaging/presence service network을 제시한다. (본문 참조)

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.286-288
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• 2002

SIP는 인터넷 텔레포니 신호(Internet Telephony Call)와 같은 멀티미디어 세션을 성립, 변경, 종료시킬 수 있는 응용계층의 호 제어 프로토콜이다. 또한, SIP 프록시 서버(Proxy Server)라는 네트일 호스트로 구축된 인프라를 통해 멀티미디어 통신을 원하는 네트윅상의 사용자를 찾아 멀티미디어 세션(Session)이 성립될 수 있도록 도와준다. SIP는 Request-Response 방식의 프로토콜이기 때문에 요청(Request)과 응답(Response)을 주고받는 일련의 트랜잭션(Transaction)으로 하나의 Task를 완성한다. 본 논문에서는 rfc2543을 기반으로 기본적인 SIP 트랜잭션을 위한 SIP 스택(Stack)의 설계 및 그 구현 방안을 제지한다.

• Journal of Communications and Networks
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• v.9 no.4
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• pp.511-523
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• 2007

In broadband convergence network(BcN), heterogeneous broadband wired & wireless subnetworks and various terminal equipments will be interconnected. In order to provide end-to-end realtime multimedia services on such heterogeneous networking environment, as a result, two major problems should be resolved: i) Multimedia session establishment & negotiation that adjusts the differences in the capability of multimedia data processing at the end terminal nodes, ii) quality of service(QoS)-guaranteed connection establishment or resource reservation with connection admission control(CAC) in each heterogeneous subnetworks along the path. The session layer signaling(e.g., SIP/SDP) should be extended for QoS negotiation, and must be tightly cooperating with network layer signaling or resource reservation with CAC function. In this paper we propose a session and connection management architecture for the QoS-guaranteed realtime multimedia service provisioning on BcN, with Q-SIP/SDP, resource reservation protocol with traffic engineering extension, and CAC functions. The detailed interaction scenario and related algorithms for QoS-guaranteed realtime multimedia session, resource reservation and connection establishment are explained and analyzed. From the experimental implementation of the proposed scheme on a small scale BcN testbed, we verified that the proposed architecture is applicable for the realtime multimedia service provisioning. We analyze the network performance and QoS parameters in detail.

• Journal of the Korea Society of Computer and Information
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• v.17 no.1
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• pp.133-140
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• 2012

In SIP(Session Initiation Protocol) service, users should send SIP SUBSCRIBE messages to the SIP server to get subscription informations of the other parties, and total amount of SIP messages becomes large in network as refresh messages and notification messages are generated to each SIP subscription. In this paper, a new resource lists architecture using dynamic notification control method has been proposed to solve these problems. In this architecture, service users can control event notification generation rate dynamically from the server according to the operation circumstances, and users can control generation of notification messages during refresh for maintaining SIP session. Furthermore users can also control generation of notification messages according to session connection conditions. A new resource lists MIME type has been designed, and several parameters for dynamic notification message control have been added to the extended SIP message headers. In this system new resource lists MIME type can be used for processing of the event package. Therefore total amount of SIP messages in network can be largely reduced, and load of the resource lists server for processing messages can be reduced also. The performance of the proposed method has been evaluated by experiments.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod 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 SIP 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 Advanced Navigation Technology
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• v.17 no.4
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• pp.421-428
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• 2013

Voice over IP refers to technology that enables routing of voice conversations over the Internet or a TCP/IP network. VoIP communication costs cheaper than traditional analog phone. Phone calls can be made to anywhere / anyone: Both to VoIP numbers as well as people with normal phone numbers. VoIP protocol equipment available today follows the SIP standard. Older VoIP equipment though would follow H 323, MGCP, Megaco/H.248. A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the TCP/IP network. A SIP server is also referred to a Asterisk IP-PBX. A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or PC by using App store. A VoIP telephone can be a simple software-based softphone. However, the SIP Server and the program is vulnerable to VoIP attacks. In this paper, eavesdropping attacks tested by using the Asterisk SIP server. Eavesdropping attacks and TLS security methods apply to VoIP system. TLS can be applied to determine whether the eavesdropping available for VoIP Environments.