Abstract: A method for transmitting a Voice over Internet Protocol (VoIP) by a wireless LAN Access Point (AP) in a communication system includes when a VoIP frame is input from a terminal, determining whether a buffer for eliminating a jitter generated in a wireless network is used, and storing the VoIP frame in a queue, determining whether the buffer is used, and when the buffer is used, determining a time interval for transmitting the VoIP frame stored in the queue and a service start time, and transmitting the VoIP frame stored in the queue in a wired fashion based on the determined time interval and service start time.

Abstract: A method for transmitting a Voice over Internet Protocol (VoIP) by a wireless LAN Access Point (AP) in a communication system includes when a VoIP frame is input from a terminal, determining whether a buffer for eliminating a jitter generated in a wireless network is used, and storing the VoIP frame in a queue, determining whether the buffer is used, and when the buffer is used, determining a time interval for transmitting the VoIP frame stored in the queue and a service start time, and transmitting the VoIP frame stored in the queue in a wired fashion based on the determined time interval and service start time.

Abstract: A Voice over Internet Protocol (VoIP) system includes at least one Internet Protocol (IP) terminal setting up a call through a switching system according to a signaling protocol, and at least one computer terminal capable of remotely accessing the IP terminal. The IP terminal sets up the call by exchanging a signaling message with a counterpart IP terminal when the computer terminal has remotely accessed the IP terminal, by transmitting a packet received through the call to the computer terminal, and by transmitting a packet received from the computer terminal to the counterpart IP terminal. A subscriber can receive and originate a call by accessing the subscriber's IP terminal irrespective of the subscriber's location.

Abstract: A system and a method providing UMS services using VoIP (Voice over Internet Protocol), upon transmitting voice data to a VoIP gateway, a VoIP-UMS having a VoIP interface transmits the voice data in a burst mode using the UMS's characteristic of sending the voice data that has been already recorded, and the VoIP gateway receives voice data that will listen to a user, stores the voice data in a buffer in advance, and reproduces the voice data at constant intervals so that the user listens to the voice, thereby eliminating noises due to delay and jitter.

Abstract: A network system and method of testing the terminal status so as to acquire the link status of the wired/wireless terminals associated to the VoIP-based network includes at least one user terminal which transmits a status message in response to a SIP-based Options message and a soft switch which transmits the Option message, determines whether or not the status message has been received, and recognizes whether the user terminal is in a Busy state or an Idle state based on a result of the determination. Since the terminal status can be accurately determined, it is possible to process calls, manage the terminal, and fairly charge the utilization costs to improve the stability and QoS of the network system.

Abstract: A Voice over Internet Protocol (VoIP) system includes at least one Internet Protocol (IP) terminal setting up a call through a switching system according to a signaling protocol, and at least one computer terminal capable of remotely accessing the IP terminal. The IP terminal sets up the call by exchanging a signaling message with a counterpart IP terminal when the computer terminal has remotely accessed the IP terminal, by transmitting a packet received through the call to the computer terminal, and by transmitting a packet received from the computer terminal to the counterpart IP terminal. A subscriber can receive and originate a call by accessing the subscriber's IP terminal irrespective of the subscriber's location.

Abstract: A Voice over Internet Protocol (VoIP) system having a dynamic gain control function and a method for providing a gain dynamically using the same system wherein, in the process that Pulse Code Modulation (PCM) data are transformed into a VoIP packet or vice versa, a gain value is dynamically assigned according to the type of terminal equipment utilized and the PCM voice data can be amplified and outputted according to the assigned gain value. The system includes a private exchange having a gain table and a VoIP gateway for the receiving the gain value from the private exchange in order to amplify the PCM voice and transform the amplified PCM voice data into the VoIP packet or convert VoIP packets to PCM voice data and amplifying the PCM voice data.

Abstract: A communication terminal includes a controller to, when a call set up between the communication terminal and a second communication terminal in an internet protocol network is released, encapsulate in a call release request message call status information relating to at least one status of the call and transmit the call release request message. The communication terminal further includes an output element coupled to the controller to transmit the call release request message. The call release request message includes the call status information and information indicating the call release.

Abstract: A network system and method of testing the terminal status so as to acquire the link status of the wired/wireless terminals associated to the VoIP-based network includes at least one user terminal which transmits a status message in response to a SIP-based Options message and a soft switch which transmits the Option message, determines whether or not the status message has been received, and recognizes whether the user terminal is in a Busy state or an Idle state based on a result of the determination. Since the terminal status can be accurately determined, it is possible to process calls, manage the terminal, and fairly charge the utilization costs to improve the stability and QoS of the network system.

Abstract: A call processing apparatus and a related method in a Wireless Fidelity (WiFi) network system construct an Internet protocol (IP) table for enabling intercommunication between WiFi terminals. Upon receiving a destination terminal telephone number from a user, the apparatus and method determine whether a destination terminal corresponding to the input telephone number has an intercommunication option set in the IP table. If the destination terminal has an intercommunication option set, a call connection request message is transmitted directly to the destination terminal via an access point according to an IP address and a port number corresponding to the input telephone number, thereby establishing a call connection without having to pass through the soft-switch.

Abstract: A Voice over Internet Protocol (VoIP) system having a dynamic gain control function and a method for providing a gain dynamically using the same system wherein, in the process that Pulse Code Modulation (PCM) data are transformed into a VoIP packet or vice versa, a gain value is dynamically assigned according to the type of terminal equipment utilized and the PCM voice data can be amplified and outputted according to the assigned gain value. The system includes a private exchange having a gain table and a VoIP gateway for the receiving the gain value from the private exchange in order to amplify the PCM voice and transform the amplified PCM voice data into the VoIP packet or convert VoIP packets to PCM voice data and amplifying the PCM voice data.

Abstract: A system and a method providing UMS services using VoIP (Voice over Internet Protocol), upon transmitting voice data to a VoIP gateway, a VoIP-UMS having a VoIP interface transmits the voice data in a burst mode using the UMS's characteristic of sending the voice data that has been already recorded, and the VoIP gateway receives voice data that will listen to a user, stores the voice data in a buffer in advance, and reproduces the voice data at constant intervals so that the user listens to the voice, thereby eliminating noises due to delay and jitter.