Abstract: The calculation for DFT(Discrete Fourier Transform) is too much in prior art, so the present invention provides a new method DMFT(Discrete Multi-Frequency Transform) for calculation. For example, in prior art, to determine a DTMF signal “3”, the calculation must be performed for k=96˜92 (frequency 1477 Hz with tolerance ±2.5%) and k=45˜43 (frequency 697 Hz with tolerance ±2.5%), total 8 of X[k] value have to be calculated, but the present invention performs a single calculation of X[k] for each frequency. To determine a DTMF signal “3”, only a high frequency of X[k] and a low frequency of X[k] are required to calculate.

Abstract: The present invention provides a method of image conversion operation for panorama dynamic IP camera. Three wide-angle cameras are used for obtaining a panorama dynamic image, and then to be compressed and transferred through Internet to a PC/smartphone/tablet for being decompressed and image conversion operation, so that a user can use the PC/smart phone/tablet to slide a touch screen thereof for viewing the desired dynamic image.

Abstract: The present invention provides a registration method for managing NAT shutdown. In Internet communication field, a user must perform registrations intermittently to a server through NAT, and increase the time interval of registration step by step. But NAT itself will shutdown if no packet is passed through for a long period. The registration method of the present invention is to adjust the time interval of registration step by step so that the time interval of registration is slightly less than the shutdown time of NAT, and then fix the time interval of registration to assure that all of the Invite packet can pass through without blocking up by the shutdown of NAT.

Abstract: The present invention provides a toning control method for setting a WiFi device by a smart device so that the WiFi device can link with Internet through a WiFi AP. The present invention is different from the prior art that a smart phone uses WiFi signal to set an IP camera, and is also different from the prior art that a smart phone generates QR Code for setting an IP camera. The present invention adopts methods of DTMF (Dual-Tone Multi-Frequency) or FSK (Frequency-Shift Keying) for toning encoding, and sends the toning encoding through a loudspeaker of the smart device to a toning decoder of the WiFi device for toning decoding, then performs the setting and linking of the WiFi device.

Abstract: The present invention provides VoIP authentication in 4G VoLTE, and also provides an extra authentication method of VoLTE for achieving communication security. The key point of the extra authentication method of VoLTE is: when a 4G mobile phone is turned on for the first time, a security registration sequence number will be random generated by the 4G mobile phone, and sent with IMSI and IMEI through TLS to an account assignment server for comparison and storage. Thereafter the account assignment server sends an account and a password in a database corresponding to IMSI to the 4G mobile phone, and then the 4G mobile phone conducts VoIP authentication with a SIP server for standby or communication.

Abstract: The present invention provides an Advanced Domain Name System for implementing method of data transfer between TCP (Transmission Control Protocol) and UDP (User Datagram Protocol) in transport layer for IP protocols in application layer of the Communications Protocol, and also provides a relay server between a PC and a server for transmitting data.

Abstract: A smart phone system includes a smart phone with wireless local area communication function, a home phone, a public switched telephone network (PSTN) or Internet, a radio base station, a smart wireless phone gateway and an outside phone, the system utilizing the wireless local area communication system for the smart phone to register in the smart wireless phone gateway, and utilizing the smart wireless phone gateway for dialing out through the public switched telephone network (PSTN) or Internet.

Abstract: A system for dialing from Internet extension of other SIP proxy server to conventional extension is disclosed. A VoIP gateway or an IP auto attendant is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension.

Abstract: The present invention provides an Advanced Domain Name System for implementing method of data transfer between TCP (Transmission Control Protocol) and UDP (User Datagram Protocol) in transport layer for IP protocols in application layer of the Communications Protocol, and also provides bilateral firewall traversal method between a PC and a server for traversing NAT (Network Address Translator) firewall.

Abstract: In SIP network environment, a general NAT traversal method will become invalid when an NAT with ICMP (Internet Control Message Protocol) is met. The present invention provides four sessions for SIP, i.e. Login Session, Port Prediction Session, Synchronization Session and Media Session, and the SIP network environment includes a first Internet telephone, a second Internet telephone, a first symmetric NAT, a second symmetric NAT and an SIP proxy server. The first symmetric NAT and the second symmetric NAT are ICMP-sensitive. In the Synchronization Session, the first Internet telephone and the second Internet telephone are designed to transmit packets synchronously to avoid port locking.

Abstract: In SIP network environment, a general traversal method for a port restricted NAT will become invalid when other users break in. The present invention provides four sessions for SIP, i.e. Login Session, Port Prediction Session, Multi-Traversal Session and Media Session, and the SIP network environment includes a first Internet telephone, a second Internet telephone, a symmetric NAT, a port-restricted NAT and an SIP proxy server. In the Multi-Traversal Session, the second Internet telephone sends a plurality of identical speech packets to consecutive ports of the symmetric NAT through a fixed port of the port-restricted NAT so as to achieve the NAT traversal.

Abstract: In SIP network environment, a general NAT traversal method will become invalid when an NAT with ICMP (Internet Control Message Protocol) is met. The present invention provides four sessions for SIP, i.e. Login Session, Port Prediction Session, Synchronization Session and Media Session, and the SIP network environment includes a first Internet telephone, a second Internet telephone, a first symmetric NAT, a second symmetric NAT and an SIP proxy server. The first symmetric NAT and the second symmetric NAT are ICMP-sensitive. In the Synchronization Session, the first Internet telephone and the second Internet telephone are designed to transmit packets synchronously to avoid port locking.

Abstract: The present invention provides an improved SIP communication protocol. An NAT (Network Address Translator) traversal method is added before the SIP communication protocol, i.e. a client to client (C2C) module function is added to improve the function of SIP communication protocol, so as to solve the problem that the RTP (Real-time Transport Protocol) packets cannot traverse NAT firewall to achieve C2C communication after SIP (Session Initiation Protocol) is ended in VoIP. The major content of the present invention is to conduct a plurality of detection before SIP communication protocol, so as to predict the allocation rules of the port number by the C2C module, and open the RTP channel for C2C.

Abstract: The present invention provides an NAT (Network Address Translator) traversal method in Session Initiation Protocol (SIP) for solving the problems of SIP in Internet phone (VoIP) under current Internet environment. In other words, the present invention solves the SIP problems caused by NAT (Network Address Translator) that P2P (Peer to Peer) transmission cannot traverse the NAT firewall directly. The major content of the present invention is that the computer conducts multiple registrations before issuing an Invite message in order to detect the rule of the NAT server to assign port number and IP address. Additional commands or parameters beyond the SIP standard are not required for the NAT traversal function. Hence, the capability of SIP for a NAT traversal function is provided without no additional commands or parameters.

Abstract: The present invention provides a “Phone Log” in a telephone switchboard for recording the extension numbers, the phone numbers dialed by the extension numbers and the accumulated times. When there is an outside call, the telephone switchboard will first check the “Phone Log” whether the telephone number of the outside call is on the “Phone Log” or not; if the telephone number of the outside call is on the “Phone Log”, then the outside call (callback) is connected directly to the corresponding extension; otherwise the outside call will be connected to an auto-attendant for connecting. As the accumulated time of each of the records reaches a time limit, the related record will be cleared out. After the callback is picked up by the corresponding extension successfully, the related record is also cleared out.

Abstract: The present invention provides a power saving system for smart mobile communication device in Internet communication. The power saving system comprises two smart mobile communication devices with Internet connection capability, an Internet, a server, a gateway and a mobile communication network. The two smart mobile communication devices cut off the Internet connection usually. When a smart mobile communication device performs a call through the Internet to the other smart mobile communication device, the server, the gateway and the mobile communication network are used to inform the other smart mobile communication device for Internet connection. After both sides stop the Internet communication, the two smart mobile communication devices cut off the Internet connection.

Abstract: A system for dialing from Internet extension of other SIP proxy server to conventional extension is disclosed. A VoIP gateway or an IP auto attendant is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension.

Abstract: A smart phone system includes a smart phone with wireless local area communication function, a home phone, a public switched telephone network (PSTN) or Internet, a radio base station, a smart wireless phone gateway and an outside phone, the system utilizing the wireless local area communication system for the smart phone to register in the smart wireless phone gateway, and utilizing the smart wireless phone gateway for dialing out through the public switched telephone network (PSTN) or Internet.

Abstract: A system for dialing from Internet extension to conventional extension is disclosed. A VoIP gateway or an IP auto attendant is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension.

Abstract: The present invention provides an improved SIP communication protocol. An NAT (Network Address Translator) traversal method is added before the SIP communication protocol, i.e. a client to client (C2C) module function is added to improve the function of SIP communication protocol, so as to solve the problem that the RTP (Real-time Transport Protocol) packets cannot traverse NAT firewall to achieve C2C communication after SIP (Session Initiation Protocol) is ended in VoIP. The major content of the present invention is to conduct a plurality of detection before SIP communication protocol, so as to predict the allocation rules of the port number by the C2C module, and open the RTP channel for C2C.