Samsung Electronics Co., Ltd.

Eung-Moon Yeom

Jung-Gi Kim

Si-Baek Kim

An apparatus and a method for determining a state of a counterpart without executing a separate program executed in order to determine a current state of the counterpart in a communication apparatus are provided. The apparatus includes a state information server and a communication apparatus. When receiving state information, the state information server provides the received state information to a user who has registered the communication apparatus that has provided the state information. When receiving state information from the state information server, the communication apparatus outputs the user's state information corresponding to the state information.

Apparatus and method for providing presence service in communication device

A voice/data combination system and a method that manages a bandwidth in the system. A determination is made as to whether a bandwidth for real-time data service is available, and the bandwidth, when not available, is reserved, so that the real-time data service is performed when the bandwidth is available. The system reserves predetermined bandwidths for real-time packets when bandwidth of real-time packets is not available, and process the real-time packets in their corresponding reserved bandwidth when bandwidth is available. Therefore, real-time packets and general packets are always processed in their corresponding bandwidths.

Voice/data combination system and method for managing bandwidth in the system

A call processing method in an Internet Protocol (IP) converged system includes: requesting an incoming call to be routed through an IP network; checking a data traffic-processing state of a traffic manager in response to the request; and rerouting the call through the IP network or rerouting the call through a Public Switched Telephone Network (PSTN) according to the checked data traffic-processing state.

IP converged system and call processing method thereof

A Quality-of-Service (QoS) control system and method of a Voice over Internet Protocol (VoIP) packet received from a broadband port in a router/gateway integrated VoIP system, which can process an incoming VoIP call by detecting in real-time an available bandwidth of the VoIP packet through interaction with a QoS module, determining whether to allow the VoIP call based on the result of the detection, and responding to the VoIP call based on the result of the determination. The QoS can be ensured according to the size of a VoIP media packet received through a broadband port.

QoS control system and method of VoIP media packet received from broadband port in router/gateway-integrated VoIP system

An IP converged system includes a VoIP ALG module and a policer module. The VoIP ALG module acquires dynamically changing RTP IP/port information of a packet by parsing a VoIP SIP message, and transmits the RTP IP/port information to the policer module. The policer module sets IP/port, which provides a real-time data service, by referring to the information from the VoIP ALG module, and discriminatively sets a packet processing condition for a non-real-time data service and a packet processing condition for the real-time data service. The VoIP ALG module and the policer module share RTP IP/port information, dynamically determined by the negotiation between VoIP gateways or VoIP terminals, in call setup/release, so that the policer can discriminately drop or mark VoIP packets by referring to the RTP IP/port information.

IP converged system and packet processing method therein

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.

Processing session initiation protocol signaling in voice/data integrated switching system

A dynamic network security system and a control method thereof in a router where an Intrusion Detection System (IDS) and a Voice over Internet Protocol Application Level Gateway (VoIP ALG) are integrated, system including: a VoIP ALG module for acquiring VoIP IP/port information of a counterpart unit in use for determining whether or not to perform intrusion detection on a packet received via VoIP signaling with the counterpart unit; an intrusion detection module for comparing the received packet with a preset intrusion detection log entry to perform intrusion detection on the received packet, and based on a result of the intrusion detection, determining whether or not to allow passage of the received packet; and an IP/port check module for checking VoIP IP/port information of the received packet according to the VoIP IP/port information of the counterpart unit provided from the VoIP ALG module to determine whether or not to perform the intrusion detection, and providing result information on the determination whether or not to perform the intrusion detection to the intrusion detection module.

Dynamic network security system and control method thereof

In a dynamic network security system and method for networks using a firewall, in judging whether packets using unfixed IPs and ports, such as VoIP (Voice over Internet Protocol) packets, are to be passed or blocked by a firewall, a current communication condition is reflected, thereby enhancing QoS (Quality of Service) of networks.

Dynamic network security

An apparatus and method for voice processing of VoIP are disclosed. The method includes: allowing a call server to perform a VoIP application level gateway (ALG) function and to set a public media gateway address corresponding to a private media gateway address of the call server in a packet data unit (PDU) of a VoIP packet to perform VoIP signaling; allowing the call server to set a private signaling address in an IP header of the VoIP packet and to transmit the corresponding VoIP packet to a data server; and allowing the data server to convert the private signaling address set in the IP header of the VoIP packet received from the call server into a public signaling address and to transmit the converted address to a destination through the Internet. Since the data server does not perform the VoIP ALG function but performs only a network address translation/port translation (NAPT) task for the IP header, a task load due to correction of data set in the PDU may be omitted, so that a quality of the VoIP service may be enhanced.

Apparatus and method for voice processing of voice over internet protocol (VoIP)

In a bandwidth management system and method for guaranteeing quality of service (QoS) in a Voice over Internet protocol (VoIP) network, bandwidth use information used to establish or terminate a call is provided, a bandwidth is reset when the bandwidth use information is provided, and available bandwidth information which is changed according to the reset bandwidth is reported.

Bandwidth management system and method for guaranteeing quality of service in voice over internet protocol network

A routing service method in a public switched telephone network (PSTN) system uses a stand-alone gateway interworking with disparate central office lines. The stand-alone gateway checks whether a call service can be provided over the Internet with reference to an internal database having periodically updated call service state information in response to a call service request from a calling private branch exchange (PBX) connected to the stand-alone gateway via a central office line. If the call service cannot be provided over the Internet, the stand-alone gateway transmits a call service request, together with a public switched telephone network (PSTN) rerouting control request, to the calling PBX. The calling PBX transfers the call to a called party over the PSTN in response to the PSTN rerouting control request.

Routing service method in voice over internet protocol system

A method for establishing VoIP connection between a caller and a called party. Upon receipt of a first dial tone, the caller enters only the access code of the called party's central office. Then, VoIP call setup and a speech path are established between a remote gateway and the caller via a local gateway. A second dial tone is generated by the central office of the PSTN of the called party and this dial tone is relayed to the caller which prompts the caller to enter the telephone number of the called party to thereby establish communication between the caller and the called party.

Method for providing voice over internet protocol call service

A method for call processing and link test in a Voice over Internet Protocol (VoIP) Gateway and system thereof. When a Gateway with Gatekeeper is operated in manual Registration/Admission/Status (RAS) mode, the link test is selectively performed depending on the status of the Gatekeeper according to an access code, whereby traffic of the Gatekeeper can be dispersed. In addition, when the Gateway with Gatekeeper is operated in manual RAS mode, depending on the status (i.e. enable or disable) of the Gatekeeper according to the access code, VoIP call processing or call routing service is divided into public VoIP service for charging a VoIP call and local VoIP service with free of charge. In case there is no routable Gateway during the VoIP call service, the call is detoured to Public Switched Telephone Network (PSTN) network for processing. As a result, service reliability of VoIP network is greatly improved.

Method for call processing and link test in a VoIP gateway and system thereof

A method for controlling bandwidth in a Voice over Internet Protocol (VoIP) system provided with gateways that are capable of performing real time bidirectional communication between endpoint terminals over the Internet, where the endpoint terminals include voice and image and/or data communication, includes the steps of calculating total bandwidth usage of each gateway upon receiving a call setup request from an endpoint terminal in consideration of the number of calls currently being processed and the bandwidth usage according to coder/decoder (codec) usage for each call, and transferring the calculated total bandwidth usage to other gateways in the VoIP system.

Method for controlling bandwidth in a voice over internet protocol system

A method for sending a dual-tone multi-frequency (DTMF) signal using a voice over Internet protocol (VoIP). A user datagram protocol (UDP) port is set up for transfer of DTMF data between a VoIP gateway and a VoIP client. If a caller inputs a numeral key in a VoIP-based call connection state, the VoIP client requests the VoIP gateway to send information regarding a digital signaling processor (DSP) channel currently established therein. The VoIP gateway retrieves the currently established DSP channel information on the basis of an IP address of the VoIP client and sends the retrieved DSP channel information to the VoIP client. The VoIP client sends DTMF data corresponding to the inputted numeral key to the VoIP gateway through the set-up UDP port. The VoIP gateway receives the DTMF data from the VoIP client through the UDP port, generates a DTMF signal corresponding to the received DTMF data and sends the generated DTMF signal to the VoIP client's counterpart. Therefore, the DTMF signal can be accurately sent to the counterpart, and routing information for transfer of the DTMF data is provided by sending DSP channel information of a current VoIP call to the VoIP client and a Web call server.

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Suwon-si, South Korea

Eung-Moon Yeom