• Title/Summary/Keyword: IP Telephony

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Development of Domestic Standard for Internet Telephony Supplement Service (국내 인터넷 전화 부가 서비스를 위한 호 처리 언어(CPL) 표준 개발)

  • Lee, Il-Jin;Yi, Jong-Hua;Kang, Shin-Gak
    • Proceedings of the Korea Information Processing Society Conference
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    • 2003.11b
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    • pp.1217-1220
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    • 2003
  • 인터넷의 발전과 더불어 음성 서비스를 인터넷에서 제공하는 인터넷 전화 서비스가 지속으로 증가하고 있다. 이러한 서비스를 제공하는 인터넷 전화 기술들은 H.323 과 SIP 따라 서비스를 제공하고 있다. 인터넷 전화의 활성화에 의해 사용자의 요구 사항 또한 기존 PSTN 에서 제공되던 부가서비스는 물론 그 보다 다양한 부가서비스에 대한 욕구가 증대하고 있다. 호 처리언어는 사용자가 직접 자신의 부가 서비스를 생성 및 제어할 수 있는 표준 언어로써 기존 PSTN 에서 제공되는 부가서비스외에 사용자의 편의에 따라 여러 부가서비스를 창출할 수 있도록 만들어진 언어로써 IETF 에서 현재 표준화가 진행중이다. 본 논문에서는 국내 VoIP 포럼을 중심으로 개발되고 있는 인터넷 전화 부가 서비스를 위한 호 처리 언어(CPL) 표준의 주요 내용을 기술한다.

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A Caching Scheme to Support Session Locality in Hierarchical SIP Networks

  • Choi, KwangHee;Kim, Hyunwoo
    • Journal of Korea Society of Industrial Information Systems
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    • v.18 no.1
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    • pp.1-9
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    • 2013
  • Most calls of a called user are invoked by the group of calling users. This call pattern is defined as call locality. Similarly Internet sessions including IP telephony calls have this pattern. We define it session locality. In this paper, we propose a caching scheme to support session locality in hierarchical SIP networks. The proposed scheme can be applied easily by adding only one filed to cache to a data structure of the SIP mobility agent. And this scheme can reduce signaling cost, database access cost and session setup delay to locate a called user. Moreover, it distributes the load on the home registrar to the SIP mobility agents. Our performance evaluation shows the proposed caching scheme outperforms the hierarchical SIP scheme when session to mobility ratio is high.

Proposal of Traffic Management Strategy between Hierarchical Mobile-WiMAX/WLAN Networks (계층적 Mobile-WiMAX/WLAN 네트워크에서의 트래픽 관리 전략에 관한 연구연구)

  • Moon, Tae-Wook;Kim, Moon;Cho, Sung-Jun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.1
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    • pp.153-160
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    • 2009
  • A hierarchical network between Mobile-WiMAX and WLAN systems is to make it possible to utilize wireless internet services including time sensitive applications such as VoIP, VOD, visual telephony etc. During the process of vertical handoff decision from Mobile-WiMAX to WLAN hotspot, vertical handoff delay causes user dissatisfaction because it doesn't provide the seamless wireless internet service. We make use of type of service (ToS) parameters in IEEE 802.16e specification as the criterion parameter of vertical handoff decision process in hierarchical Mobile-WiMAX/WLAN networks. In this paper, we propose the process of vertical handoff decision for seamless wireless internet service which is sensitive to time delay. If type of service is time sensitive application, the decision of vertical handoff is withdrawn until the service is terminated. In focus on user satisfaction, if the proposed traffic management strategy in hierarchical Mobile-WiMAX/WLAN networks is used, user will utilize seamless wireless internet services including time sensitive applications.

Design and Implementation of SIP Internet Call-setup System using Seven States (7가지 상태를 이용한 SIP 인터넷 전화연결 시스템 설계 및 구현)

  • Shin, Yong-Kyoung;Kim, Sang-Wook
    • Journal of KIISE:Computing Practices and Letters
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    • v.13 no.5
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    • pp.300-310
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    • 2007
  • The Session Initiation Protocol (SIP) is one of the major protocols used in call-setup over IP telephony. The SIP-signaled calls use many-sided states according to a request of user. In this paper, we suggest seven states and some events that help developers to design and implement new applications efficiently. And they enable an object-oriented design of the system. If you design the call-setup procedure only by the processing model suggested in RFC 3261 over commercial network, a fatal error may occur in the system because of heavy data traffic or unpredicted exception cases. However, according to the suggested seven states, if they are predefined events in the current system state, the standardized processing routine is executed. Otherwise, they can be processed by the exception routine in system. All event processing routines are designed and implemented using Finite State Machine (FSM).

Web Server Implementation via OpenWRT-based Wired/Wireless Router (OpenWRT 기반 유무선 공유기를 통한 웹 서버 구축)

  • Ha, Seung-Eup;Min, Jun-Ki;Ban, Tae-Hak;Jung, Hoe-Kyung
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2013.10a
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    • pp.1055-1057
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    • 2013
  • Routers currently used in homes and institutions have a function of multiple computers to connect to the Internet with a single IP address. It is possible only through setting or establishing a separate operation in a used PC or server to function as a web/printer server, web hard, and P2P. In this paper, by utilizing the OpenWRT Linux-based through the construction of the Web server through a router OpenWRT of (Open Wireless Router) base, to build a server simple compact, low power consumption, low cost, to operate by reliability to provide Internet telephony and multimedia services high securely over the server security, which is based on the setting of policy and various firewall at the same time the configuration of the network utilizing OpenWRT router that can be enhanced, to construct a Web server through a router multi functional can receive the provision of services using the program PC and mobile APP.

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IPTV Service Provider over FTTH (광가입자망을 통한 IPTV 서비스 제공)

  • Park In-Gyu
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.43 no.5 s.347
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    • pp.7-16
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    • 2006
  • IPTV is referred to the service which provide integrated IPTV services for providing video, 10/100-Mbit/sec Internet, voice, video-on-demand (VOD), and other broadband applications including home security, video conferencing, and telemedicine. All services are integrated into an IP (Internet Protocol) architecture designed specifically for Gigabit Ethernet FTTH systems, HFC or xDLC. It is absolutely necessary that telecon operators provide IP video delivery platforms that enable service providers to transform their business. With their own products, they can better manage their existing services and generate new revenues from broadcast TV, movies on demand and multimedia. Triple-play is a combination of broadcast, telephony and broadband services offered through IPTV networks. With cable operators allowed to offer a triple-play bundle, the nation's telecom operators are beginning to get a little anxious. Cable operators assert that triple-play is a must-have and natural extension of the cable service bundle. The Korean Cable TV Association asserts that the triple-play model is of paramount importance to the cable industry's future growth. But the telecom sector considers itself unfairly disadvantaged, saying they cannot compete until regulatory issues are resolved. The start of web-based television in Korea may still be some time off with a confrontation between the nation's IT regulator and broadcasting sector over the service's legal boundaries shows no signs of being resolved my time soon. korea should be is the fastest-growing provider of IPTV solutions in the industry, with over worldwide customers.

The Overload Control Scheme Using a Delay Queue in the SIP Signalling Networks (SIP 시그널링 네트워크에서 지연 큐를 이용한 과부하 제어 방법)

  • Lee, Jong-Min;Jeon, Heung-Jin;Kwon, Oh-Jun
    • Journal of Korea Multimedia Society
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    • v.15 no.8
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    • pp.1038-1047
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    • 2012
  • The SIP(Session Initation Protocol) is an application layer protocol that is used to establish, release, and change the call session of the IP telephony. In the SIP signalling networks, when the number of the UA(User Agent) requested the call session increase, the number of messages to be processed by SIP proxy server increase. It often will be caused the overload of the SIP proxy server. In this paper, we proposed the overload control method with a normal queue and a delay queue in the SIP proxy server. When it is estimated the overload of the server by the excess of the high threshold in the normal queue, new INVITE messages will be put into the delay queue to reduce the load of the server. It results in some delay of the call session from the INVITE message. Subsequently when the number of messages in the normal queue is reduced below the low threshold, the INVITE messages in the delay queue is processed. The simulation results showed that the number of the retransmission messages by our proposed method was 45% less than the one by the method with single queue. The results also showed that the average call success rate by the proposed method was 2% higher than the one by the method with single queue.