Abstract:
본 발명은 통신 장치에서 상대방의 현재 상태를 확인하기 위하여 수행하는 별도의 프로그램 실행없이 상대방의 상태를 확인하기 위한 장치 및 방법에 관한 것으로, 상태 정보를 수신할 경우, 상기 상태 정보를 제공한 통신 장치를 등록한 사용자로 상기 수신한 상태 정보를 제공하는 상태 정보 서버와, 상기 상태 정보 서버로부터 상태 정보를 수신할 경우, 상기 상태 정보에 해당하는 사용자의 상태 정보를 출력하는 통신 장치를 포함하는 것을 특징으로 한다.
Abstract:
PURPOSE: An apparatus and a method for a presence service in a portable communication system are provided to confirm the state of the other party without an additional program. CONSTITUTION: A state information management unit(110) registers the other party communication device to confirm the state information. If the state information of the registered user is received without the additional application program, the state information management unit outputs the state information corresponding to the received state information. The state information management unit outputs the state information through a speaker and/or an LED(Light Emitting Diode), and outputs the state information by using the defined LED light emitting information for each state of the other party's communication device.
Abstract:
본 발명은 호를 설정하거나 해제하는 경우 사용되는 사용 대역폭 정보를 제공하여 사용 대역폭 정보가 제공될 때 대역폭을 재설정하여 변경되는 가용 대역폭 정보를 통보하는 브이오아이피 네트워크에서 큐오에스 보장을 위한 대역폭 관리 시스템 및 그 방법에 관한 것이다.
Abstract:
In a Voice over Internet Protocol (VoIP) call processing system and method, a first unit is adapted to setup address information, for a terminal requesting a call connection during VoIP call signaling, as Quality of Service (QoS) guarantee information for a VoIP service, and to delete the QoS guarantee information upon releasing the VoIP call; and a second unit is adapted to guarantee a QoS for the VoIP service according to the QoS guarantee information during the VoIP call.
Abstract:
본 발명은 원격 게이트웨이의 상태 관리 방법에 있어서, 원격 게이트웨이의 억세스 코드 테이블과, IP 어드레스 리스트를 나타내는 IP 테이블 및 상태를 나타내는 상태 테이블에 대한 데이터베이스를 미리 구비하는 과정과, 자체 게이트웨이의 상태에 대한 정보를 전송하고 다른 게이트웨이들의 상태 정보를 수신하여 상기 데이터베이스의 저장 및 갱신하며, 이를 통해 각 게이트웨이의 상태를 관리하는 과정을 가진다.
Abstract:
Processing Session Initiation Protocol (SIP) signaling in a voice/data integrated switching system includes: transceiving a message and data between a terminal and a system using a Voice over Internet Protocol (VoIP) SIP “INFO” method upon communication being performed between terminals via an Internet Protocol (IP)network and using a VoIP SIP “MESSAGE” method in an idle state upon communication not being performed between the terminals; processing proprietary signaling of a legacy voice switching system as standard signaling using standard VoIP SIP signaling, and simultaneously and separately processing the standard SIP signaling and the proprietary signaling of the legacy voice switching system in a system using the VoIP SIP signaling.
Abstract:
PURPOSE: A VoIP(Voice over IP) gateway and a method for processing a call in a Voip system and performing link test are provided to decide whether a gatekeeper is used according to an access code, and to process a link test and a VoIP call, thereby distributing traffic of the gatekeeper and improving QoS. CONSTITUTION: A gateway decides whether a link test cycle comes for remote gateways(S10). If so, the gateway searches a link test target gateway, and decides whether a gatekeeper is used according to an access code(S20). If not, the gateway searches a remote gateway to perform a link test, and transmits link test data to the remote gateway(S21). If a response signal for the link test data is received, the gateway carries out an updating process according to state information of the remote gateway(S22). The gateway decides whether the link test is to be further carried out for a gateway IP(S30).
Abstract translation:目的:提供一种VoIP(Voice over IP)网关和一种在Voip系统中处理呼叫并执行链路测试的方法,以决定是否根据接入码使用网守,并处理链路测试和VoIP呼叫 从而分配关守的流量并提高QoS。 规定:网关决定远程网关是否有链路测试周期(S10)。 如果是,则网关搜索链路测试对方网关,并根据接入码判断是否使用网守(S20)。 如果不是,则网关搜索远程网关进行链路测试,并将链路测试数据发送到远程网关(S21)。 如果接收到链路测试数据的响应信号,则网关根据远端网关的状态信息进行更新处理(S22)。 网关决定是否对网关IP进一步执行链路测试(S30)。
Abstract:
PURPOSE: A method of performing a VoIP(Voice over Internet Protocol) call routing of a VoIP gateway is provided to adaptively configure a VoIP call routing DB to sequentially store IP addresses in IP address fields of IP tables, and to perform a VoIP call routing by using the DB, thereby efficiently using the IP tables. CONSTITUTION: When an outgoing call request enters a VoIP gateway(300), a VoIP processor checks a destination number(302). The VoIP processor confirms whether an access code of an originator is included in a VoIP call routing table, and is valid(304). If the access code is included in the table and is valid, the VoIP processor retrieves IP terminals or IP addresses of a VoIP gateway up to a finish point from a start point of an IP table corresponding to the access code(306). If a link test and a channel status are transceived with a remote VoIP gateway and a VoIP terminal, the VoIP processor checks whether a VoIP call is available to the VoIP gateway or the VoIP terminal(308). If so, the VoIP processor processes a normal outgoing call(310) and completes the procedure(314). If not, the VoIP processor performs a call release procedure(312).
Abstract:
PURPOSE: A method for managing bandwidths in a VoIP system is provided to control bandwidths in order to efficiently execute call routing in a VoIP service network not having a gate keeper. CONSTITUTION: A local node(30), a gateway in a calling subscriber side, and a remote node(40), a gateway in a called subscriber side, comprise a plurality of gateway modules(GW(A)-GW(N),GW(1)-GW(n)) respectively. Each of the gateway modules supports dozens of channels. The gateway modules(GW(A),GW(1)) among the gateway modules(GW(A)-GW(N),GW(1)-GW(n)) respectively installed in the local node(30) and the remote node(40) are total bandwidth control gateway modules to control the total bandwidth.