摘要:
It is made possible for a DTMF tone signal associated with a button depressed on an IP telephone terminal to arrive at the opposite party even if an IP telephone terminal does not have a function of sending a DTMF tone signal. If a DTMF tone signal transmission request signal detection unit in a proxy detects a DTMF tone signal transmission request signal inserted in an RTP packet or a SIP packet received from an IP telephone terminal or detects a DTMF digit signal contained in the DTMF tone signal transmission request signal, a DTMF tone signal generation unit generates a DTMF tone signal associated with the DTMF digit signal. A voice encoder encodes the DTMF tone signal in the same way as a voice signal. An RTP payload replacement unit replaces a voice signal in a payload of the RTP packet with the encoded DTMF tone signal.
摘要:
Voice is input to a voice input and output portion 11. Thereafter, a digital-analog portion converts analog signal of voice into digital data. An RTP port confirming portion confirms whether or not a communication according to the RTP can be made. When the communication according to the RTP cannot be made, with reference to a history table, a port through which a voice data communication can be made is determined. Digitized voice is sent to a call controlling portion. The call controlling portion performs a call control for the voice communication. When the communication according to the RTP can be made, the voice data is sent to an IP communication portion. The IP communication portion paketizes the voice data. Packetized voice data according to the IP is sent to a LAN interface portion. The LAN interface portion outputs the packetized voice data to a network.
摘要:
An IP telephony method facilitating transferring an IP telephony communication inside a LAN is realized. A telephone conversation takes place between a first IP telephone terminal and a conversation partner's IP telephone terminal through a router and a proxy. The proxy relays the telephone conversation by referring to a proxy table that holds a correspondence among a global IP address and a receive port of the conversation partner's IP telephone terminal, and a local IP address and a receive port of the first IP telephone terminal during relay. To perform transfer from the first IP telephone terminal to a second IP telephone terminal, the local IP address and the receive port of the first IP telephone terminal in the proxy table are changed to a local IP address and a receive port of the second IP telephone terminal, respectively.
摘要:
An IP telephony method facilitating transferring an IP telephony communication inside a LAN is realized. A telephone conversation takes place between a first IP telephone terminal and a conversation partner's IP telephone terminal through a router and a proxy. The proxy relays the telephone conversation by referring to a proxy table that holds a correspondence among a global IP address and a receive port of the conversation partner's IP telephone terminal, and a local IP address and a receive port of the first IP telephone terminal during relay. To perform transfer from the first IP telephone terminal to a second IP telephone terminal, the local IP address and the receive port of the first IP telephone terminal in the proxy table are changed to a local IP address and a receive port of the second IP telephone terminal, respectively.
摘要:
It is made possible for a DTMF tone signal associated with a button depressed on an IP telephone terminal to arrive at the opposite party even if an IP telephone terminal does not have a function of sending a DTMF tone signal. If a DTMF tone signal transmission request signal detection unit in a proxy detects a DTMF tone signal transmission request signal inserted in an RTP packet or a SIP packet received from an IP telephone terminal or detects a DTMF digit signal contained in the DTMF tone signal transmission request signal, a DTMF tone signal generation unit generates a DTMF tone signal associated with the DTMF digit signal. A voice encoder encodes the DTMF tone signal in the same way as a voice signal. An RTP payload replacement unit replaces a voice signal in a payload of the RTP packet with the encoded DTMF tone signal.
摘要:
Voice is input to a voice input and output portion 11. Thereafter, a digital-analog portion converts analog signal of voice into digital data. An RTP port confirming portion confirms whether or not a communication according to the RTP can be made. When the communication according to the RTP cannot be made, with reference to a history table, a port through which a voice data communication can be made is determined. Digitized voice is sent to a call controlling portion. The call controlling portion performs a call control for the voice communication. When the communication according to the RTP can be made, the voice data is sent to an IP communication portion. The IP communication portion paketizes the voice data. Packetized voice data according to the IP is sent to a LAN interface portion. The LAN interface portion outputs the packetized voice data to a network.