Abstract: A call routing method in VoIP (Voice over Internet Protocol) capable of performing voice service assuring best voice quality is obtained by predicting voice quality using network parameters that affect the voice quality, and then performing a call routing to a gateway with a maximum MOS (Mean Opinion Score) value out of a gateway group. Also, actively responding to organically changing network environment and providing a best call routing path for the network environment before setting up a call, best voice quality can be provided.

Abstract: A secure communication system comprises at least one virtual private network (VPN) gateway for managing tunnel information of other VPN gateways that are connected via tunnels for secure communication, and for transmitting the tunnel information to each terminal connected via a secure network. When a packet received from the terminal is destined for the secure network, the VPN gateway encrypts the packet, adds a tunnel header to the packet, and transmits the resultant packet to the corresponding tunnel. At least one terminal stores the tunnel information received from the gateways.

Abstract: In a symmetric network address translation system using a Simple Traversal of UDP over NAT (STUN) technique and a method for implementing the same, a voice over Internet protocol (VoIP) network includes a STUN server for transmitting, to a private network terminal, a public Internet Protocol (IP) address and first port information of a router which is used for a VoIP call. The private network terminal transmits a session setup request message, including the public IP address and the first port information of the router, through its private IP address and a second port, and the router maps and stores the public IP address and the first port information of the router, and the private IP address and the second port of the private network terminal, and routes a packet received through the public IP address and the first port to the private IP address and the second port. Thus, a VoIP call is performed using the symmetric network address translation system to which the STUN technique is applied.

Abstract: In a system and method for providing Push To Talk (PTT) service according to a state of the user, when a PTT terminal user is absent, in conference or busy, the PTT system converts the PTT voice message directed to the PTT terminal to a text message, and displays it on a screen of another terminal or the PTT terminal. As a result, the user can immediately check the PTT message when he/she is in absence or conference state, and is busy without communication interference.

Abstract: Disclosed are an apparatus for providing additional information by using an extension subtitle file including an inverse multiplexing unit to separate video and audio data from a multimedia file input by a data input unit, an XML parser to parse a caption data file and an additional information data file input by the data input unit, video and audio data processing units to decode the video and audio data input by the inverse multiplexing unit, and convert the decoded video and audio data into predetermined video and audio signals, a caption processing unit to process the caption data parsed and input by the XML parser, an additional information data processing unit to process the additional information data file parsed and input by the XML parser, and to generate user interface information based on the processed additional information, and a data output unit to display the processed video, audio, caption and user interface information.

Abstract: A network system for recovering from a link failure in a Voice over Internet Protocol (VoIP) service comprises: a first media gateway having a current link setting and transmitting a corresponding packet upwardly for the VoIP service; a second media gateway which is link-connected to the same remote terminal as the first media gateway for the VoIP service; a media gateway controller for allocating the second media gateway for link resetting, and for transmitting allocation information when receiving failure information of a link set through the first media gateway; and a router for determining whether a packet transmitted from the first media gateway is received within a set time to determine whether there is a failure of the set link, for transmitting the link failure information to the media gateway controller, and for resetting the link to the second media gateway according to the allocation information transmitted from the media gateway controller when there is a link failure.

Abstract: An apparatus for providing a session initiation protocol (SIP) service comprises: a gateway installed in a boundary between a private network and a public network, the gateway detecting identifier information and address information of SIP terminals located in the private network using an SIP message transmitted between the private network and the public network; and a storage device for storing a mapping table which manages the identifier information and the address information.

Abstract: A method and system to rout a Voice over Internet Protocol (VoIP) call based on a service class of a subscriber enables efficient utilization of bandwidth by providing service corresponding to the subscriber service class, while enabling use of a Virtual Private Network (VPN) which provides security to a VoIP service.

Abstract: A system and method to change codec information to provide a coloring service in a Voice over Internet Protocol (VoIP) terminal uses different compression methods depending on a calling state and a call connecting state between the VoIP terminals so that a more efficient coloring service is provided. The system for changing codec information includes a gateway adapted to compress ring back tone data in a calling state and to compress voice signal data in a call connecting state between communication terminals according to preset different compression information and to transmit both data to a receiving terminal.

Abstract: Processing voice data in a packet communication network with encryption for efficient use of a bandwidth in a Virtual Private Network (VPN) includes: confirming, by a terminal at a transmitting side, a destination address of a call connection packet; when the destination address is directed to a private network, storing call connection information within the call connection packet and registering the call connection information with an address translation table; encrypting, by the terminal at the transmitting side, the call connection packet and transmitting the encrypted call connection packet to a receiving side; storing, by a terminal at the receiving side receiving the call connection packet, the call connection information within the call connection packet therein; encrypting, by the terminal at the receiving side, a call connection response packet responsive to the call connection packet and transmitting the encrypted response packet to the terminal at the transmitting side to establish a communication

Abstract: A method for guaranteeing Quality of Service (QoS) using bandwidth reservation in a switch includes: establishing a MAC table including MAC addresses of terminals connected to each port of the switch and an identifier indicating whether or not the corresponding terminal is a VoIP terminal; calculating a VoIP bandwidth necessary for each port based on information of the MAC table, calculating a ratio of an entire bandwidth to a bandwidth equal to the entire bandwidth minus the VoIP bandwidth, and setting the ratio as a port ratio of the switch; determining whether or not the corresponding packet is a VoIP packet by filtering the MAC address of the corresponding packet upon an arbitrary packet of the switch being input; and passing the corresponding packet upon a determination that the corresponding packet is a VoIP packet, and passing the corresponding packet only when the bandwidth of the corresponding packet is within a permissible limit of the set port ratio by performing port ratio control upon a determina

Abstract: An apparatus and a method that manage multiple inputs using a management module are disclosed. The input management apparatus and method is applied to a system that can handle an input made by at least one input device and an input made by executing at least one program. The input management apparatus according to the present invention, comprising: an event generation module, which generates a first event corresponding to the input; an input management information storing unit, which stores management information of the input; and an event management module, which, if the first event is generated, generates a second event referring to the input management information storing unit.

Abstract: A method for processing an outgoing call in a VIP gateway system includes the steps of setting a port of an ITM card of a VIP gateway when an outgoing telephone number is inputted, recognizing an area code of the inputted number as an access code of an ITM card of a VIP gateway of the called party, adding a trunk access code matching the ITM access code to the telephone number to obtain an augmented number, and transferring the augmented number to the ITM card of the VIP gateway of the called party.

Abstract: A call routing method in VoIP (Voice over Internet Protocol) capable of performing voice service assuring best voice quality is obtained by predicting voice quality using network parameters that affect the voice quality, and then performing a call routing to a gateway with a maximum MOS (Mean Opinion Score) value out of a gateway group. Also, actively responding to organically changing network environment and providing a best call routing path for the network environment before setting up a call, best voice quality can be provided.

Abstract: A voice over Internet protocol (VoIP) gateway includes a foreign exchange office (FXO), a foreign exchange station (FXS), and a VoIP processor. A controller of the VoIP gateway sets the VoIP gateway to either a TANDEM (trunk and ENM (ear and mouth)) mode or a standalone mode. In the TANDEM mode, the VoIP gateway transmits an incoming call from the VoIP processor to the FXO and an outgoing call from the FXS to the VoIP processor. In the standalone mode, the VoIP gateway transmits the incoming call from the VoIP processor to the FXS and the outgoing call from the FXS to the VoIP.

Abstract: A method for processing an outgoing call in a VIP gateway system includes the steps of setting a port of an ITM card of a VIP gateway when an outgoing telephone number is inputted, recognizing an area code of the inputted number as an access code of an ITM card of a VIP gateway of the called party, adding a trunk access code matching the ITM access code to the telephone number to obtain an augmented number, and transferring the augmented number to the ITM card of the VIP gateway of the called party.