Abstract:
PURPOSE: A mobile multicast method for supporting network based portability is provided to offer a multicast service such as IPTV considering restricted bandwidth in IP based mobile network. CONSTITUTION: An MICS(Mobility Information Control Server) transmits a membership report message to HCA#3(273). The MICS transmits a location information response message to HCA#1(275). The HCA#1 configures a multicast packet forwarding entry on an internal interface through a multicast group address(276). The HCA#1 and the HCA#3 are generated through a tunnel(280). The multicast packet is transmitted to the HCA#1 through the tunnel(281, 282).
Abstract:
PURPOSE: An apparatus and a method for registering an enabler in a service transmission platform are provided to extend a function of a service transmission platform by newly adding interface which is not supported in a current enabler. CONSTITUTION: An authentication processing unit(200) authenticates a service user(100) which uses an enabler and provides the information of the enabler to the service user which is allowed to the information according to a service convention. The service user registers a new enabler. The authentication processing unit re-authenticates the registration for preventing the indiscreet registration of the enabler. An enabler management unit(201) provides and manages information about the enabler provided from the service transmission platform. An interface management unit(202) manages an interface provided from the enabler.
Abstract:
본 발명은 VoIP 망에서의 통화품질 측정 및 관리 장치와 그 방법을 제공한다. 본 발명은 통화세션 및 통화품질을 관리 단계에 따라 단말과 통화품질 측정/관리 장치 간의 메시지를 송수신하여 처리함으로써, 서비스 가입자들에게 종단 간 정확게 QoS가 보장된 고품질의 VoIP 서비스를 제공할 수 있으며, VoIP 서비스의 안정적 품질관리 업무에 있어서 효율성을 증대시킬 수 있다. VoIP, 품질관리, 통화세션
Abstract:
본 발명은 VoIP 서비스를 위한 잡음 제거 장치 및 방법에 관한 것으로, 입력 신호의 SNR 가중치를 반영하여 잡음을 제거하기 위하여, 잡음 음성 신호에 대한 음성 파워 스펙트럼과 잡음 파워 스펙트럼을 예측하는 파워 스펙트럼 예측기와, 상기 음성 파워 스펙트럼과 잡음 파워 스펙트럼으로부터 SNR를 계산하고, 상기 계산된 SNR에 대응되는 SNR 가중치를 획득한 후, 상기 SNR 가중치를 반영하는 위너 필터 함수를 설계하는 위너 필터 설계부와, 상기 위너 필터 함수를 이용하여 상기 잡음 음성 신호에 포함된 잡음 신호를 제거하는 위너 필터부를 포함하여 구성되며, 이에 의하여 VoIP 서비스에서 보다 향상된 통화 음질을 제공할 수 있도록 한다. VoIP 서비스, 잡음 제거, SNR 가중치
Abstract:
A method and system for bundled authentication of wired or wireless terminal between the service and access networks in NGN environment are provided to reduce the curriculum of additionally inputting password by performing the NACF-SCF bundle authentication regardless of wireless and wired access. An NGN terminal unit(10) comprises a WLAN(Wireless LAN) terminal(101), a WiBro terminal(102), and a cable terminus(103) and a ADSL (Asymmetric Digital Subscriber Line) terminal(104). A connection unit(11) is used for the packet connection of the NGN terminal. An AP (Access Point)(111) is connected to the WLAN terminal. A RAS (Remote Access Server)(112) is connected to the WiBro terminal. A CM (Cable Modem) part(113) is connected to the cable terminus. The CM part is formed with CM and CMTS(Cable Modem termination System). The ADSL terminal is connected to wired and wireless network(16) through the ADSL circuit. A NACF(Network Attachment Control Functions)(12) assigns the connection internet protocol address. The NACF part performs the access authentication. A SCF(Service Control Function)(13) which is the service control network performs the service routing and service certification. A bundle authorization control server(14) performs the exchange of the authentication information between the NACF part and the SCF part.
Abstract:
A method and system for bundled authentication of wired or wireless terminal between service and access networks in NGN environment are provided to perform the NACF-SCF(Network Attachment Control Functions-Service Control Function) bundle authentication in NGN regardless of wireless and wired access. The NGN(Next Generation Network) terminal unit(10) comprises a WLAN(Wireless LAN) terminal(101), a WiBro terminal(102), and a cable terminus(103) and an ADSL (Asymmetric Digital Subscriber Line) terminal(104). A connection unit(11) comprises an AP(Access Point)(111) connected to the WLAN terminal, an RAS(Remote Access Server)(112) connected to the WiBro terminal, and a CM(Cable Modem) part(113) connected to the cable terminus. The ADSL terminal is connected to wired and wireless network(16) through the ADSL circuit. A NACF(Network Attachment Control Functions) part(12) which is the access control network performs the IP address assignment and the access authentication. The SCF(Service Control Function) part(13) which is the service control network performs the service routing and service certification. The bundle authorization control server(14) exchanges the authentication information between the SCF part and the NACF part. The service user ID corresponding to the subscriber identification is stored in the bundle authorization control server.
Abstract:
An apparatus and a method of measuring and managing real-time speech quality in VoIP network are provided to increase efficiency of the VoIP service by monitoring the speech status in real time through a report of real time quality measure parameters, and accurate management of quality measure values. A system of measuring and managing real-time speech quality in VoIP network comprises a terminal, a soft switch(40) and a quality measure server(50). The terminal comprises a soft phone(10) and a general telephone(20) which are able to perform internet communication. The soft phone performs the internet communication through one port gateway and a modem. The general telephone performs the internet communication through the VoIP(Voice over Internet Protocol) gateway. Each terminal collects speech session status information and various data which is necessary to measure the quality, and reports the information and the data to a quality measure server periodically. Each terminal transmits and receives the speech session control message by using a UDP(User Datagram Protocol), and transmits the collected quality measure data to the quality measure server by using a real time transport protocol. The quality measure server measures and controls the speech quality between terminals by receiving quality measure parameter values and the speech session status information from each terminal.
Abstract:
A data grid-based profile management apparatus in a broadband integrated network is provided to perform easily network interoperation in a wired/wireless integrated network by sharing information of subscribers, and provide an effective service in network combining and roaming. When a user profile is requested, a service server, which includes an HSS(Home Subscriber Server), an AAA(Authentication, Authorization, Accounting), an SLF(Subscriber Locator Function) and an HLR(Home Location Register), transfers the user profile to a CUPS(Converted User Profile Server) through a WS-RF(Web Service-Resource Framework) module, and provides obtained results through the CUPS. The CPUS searches each profile database through the WS-RF module according to the user profile request from the service server, and transfers obtained results.
Abstract:
본 발명은 VoIP(Voice over Internet Protocol)망을 통해 송신자와 수신자가 실제 음성 통화를 하고 있는 상태에서 종단간의 통화품질을 정확하게 측정함으로써 효율적인 VoIP 망 구성 및 운영을 지원할 수 있는 VoIP망에서의 양방향 대화형 통화품질 측정 방법에 관한 것으로, VoIP 서비스를 이용하는 송신측과 수신측의 종단간 통화품질 관리를 위하여 객관적 품질 측정 방법인 PESQ(Perceptual Evaluation of Speech Quality)와 E-Model을 통합하여 새로운 방식의 E-Model을 적용하는 한편, 패킷 지연의 불규칙성으로 발생하는 지터값을 새로운 통화품질 측정 파라미터로 E-Model에 적용함으로써, VoIP 서비스망에서 종단간의 통화품질을 정확하게 측정할 수 있는 것을 특징으로 한다. 본 발명에 따르면 종단간 통화품질 측정시 VoIP 네트워크망에서의 패킷 손실, 지연 및 지터에 의한 통화품질 저하 요인이 모두 반영된 통화품질 측정이 가능하게 되므로, 종래의 단일망 기반의 통화품질 측정 방법에 비해 xDSL, HFC, 전용회선, FTTH와 같은 다양한 망들로 결합된 환경에서도 사용자간의 종단간 통화품질을 정확하게 측정할 수 있게 된다. 양방향, 대화형, 통화품질, VoIP, PESQ, E-Model
Abstract:
본 발명은 SIP를 이용하여 VoIP 서비스를 제공하는 경우 비정상적인 장기호를 빠른 시간안에 찾아 제거하는 방법에 관한 것이다. 본 발명은, 제1 SIP 단말기가 제2 SIP 단말기로부터 수신되는 데이터가 있는지를 모니터링하는 체크주기(T1) 및 상기 제1 SIP 단말기에서 제2 SIP 단말기로 특정 메시지를 전송하여 응답을 수신할 때까지 유효한 시간(T2)을 설정하는 시간설정단계; 상기 제1 SIP 단말기 및 제2 SIP 단말기 간에 호 연결을 완료하는 호연결단계; 상기 제1 SIP 단말기에서 상기 T1주기 동안 상기 제2 SIP 단말기로부터 수신되는 데이터를 모니터링하는 모니터링단계; 상기 T1주기 동안 RTP 데이터가 수신되었는지를 판단하는 판단단계; 상기 T1주기 동안 RTP 데이터가 수신되지 않았으면 제1 SIP 단말기에서 감사(audit)과정을 수행하는 감사수행단계; 및 상기 감사과정 수행결과가 비정상적이면 단말기 호 종료 과정을 수행하는 호종료수행단계를 포함한다. 호, SIP, VoIP, 감사(audit), 장기호(long call)