Abstract: A method and system for transmitting a call over a packet switched network. A gateway server receives a telephone call and converts analog voice signals associated with the telephone call to a stream of digital data. A first part of the digital data is processed using a first codec algorithm and transmitted over the packet switched network. A change in network conditions is detected. A second part of the digital data is processed using a second codec algorithm and transmitted over the packet switched network.

Abstract: A method and system for transmitting a call in a client/server architecture. A client device initiates a telephone call and converts first analog voice signals associated with the telephone call to digital signals. The digital signals are then transmitted over the Internet to a first gateway server. The first gateway server processes the digital signals using a codec algorithm and transmits the processed digital signals over the Internet to a second gateway server. The second gateway server converts the processed digital signals to second analog voice signals and transmits the second analog voice signals over a public switched telephone network.

Abstract: A method and system for transmitting a call over a packet switched network. A gateway server receives a telephone call and converts analog voice signals associated with the telephone call to a stream of digital data. A first part of the digital data is processed using a first codec algorithm and transmitted over the packet switched network. A change in network conditions is detected. A second part of the digital data is processed using a second codec algorithm and transmitted over the packet switched network.

Abstract: The invention is concerned with an Internet telephone system having a client/server architecture and providing voice communication between client stations over the Internet through gateway servers. The system includes a plurality of software modules within each of the gateway servers for performing digital signal processing (DSP), and an account manager placed at an arbitrary location on the Internet for monitoring transactions between client stations to produce billing information. The system is characterized by low latency, full duplex voice communication, and permits telephone to telephone or PC to telephone connections.

Abstract: A method and system for transmitting a call in a client/server architecture. A client device initiates a telephone call and converts first analog voice signals associated with the telephone call to digital signals. The digital signals are then transmitted over the Internet to a first gateway server. The first gateway server processes the digital signals using a codec algorithm and transmits the processed digital signals over the Internet to a second gateway server. The second gateway server converts the processed digital signals to second analog voice signals and transmits the second analog voice signals over a public switched telephone network.

Abstract: The invention is concerned with improvements in full duplex Internet telephone systems with a system architecture having low latency and permitting voice communication with telephone to telephone or PC to telephone connections. The architecture permits dynamic packet-to-packet change in codec to adjust for Internet conditions. The voice port creates self-describing packet conditions so that the higher-level software of the system is independent of codec selection. In addition to adjusting the codec, the voice port has the capability of dynamically and concurrently selecting other factors such as the level of error correction redundancy, the packet size and packet bundling on a packet-to-packet basis. The invention further includes a technique to eliminate dead air spaces in the voice data transmission stream by speeding up or slowing down the data rate in the buffer while maintaining a constant pitch of speech.

Abstract: The invention is concerned with an Internet telephone system having a client/server architecture and providing voice communication between client stations over the Internet through gateway servers. The system includes a plurality of software modules within each of the gateway servers for performing digital signal processing (DSP), and an account manager placed at an arbitrary location on the Internet for monitoring transactions between client stations to produce billing information. The system is characterized by low latency, full duplex voice communication, and permits telephone to telephone or PC to telephone connections.

Abstract: A system and method for managing roaming billing records for a telephone caller's home subscriber network. The invention includes a method for roaming networks to push roaming billing records in real time back to a subscriber's home network.

Abstract: A system and method for managing roaming billing records for a telephone caller's home subscriber network. The invention includes a method for roaming networks to push roaming billing records in real time back to a subscriber's home network.

Abstract: The invention is concerned with improvements in full duplex Internet telephone systems with a system architecture having low latency and permitting voice communication with telephone to telephone or PC to telephone connections. The architecture permits dynamic packet-to-packet change in codec to adjust for Internet conditions. The voice port creates self-describing packet conditions so that the higher level software of the system is independent of codec selection. In addition to adjusting the codec, the voice port has the capability of dynamically and concurrently selecting other factors such as the level of error correction redundancy, the packet size and packet bundling on a packet-to-packet basis. The invention further includes a technique to eliminate dead air spaces in the voice data transmission stream by speeding up or slowing down the data rate in the buffer while maintaining a constant pitch of speech.

Abstract: An Internet telephone system architecture having low latency and permitting voice communication between telephones and computers is disclosed. The architecture permits dynamic packet-to-packet change in various factors to adjust for Internet conditions. A forward error correction algorithm provides a variable level of redundancy from zero to three in the transmission of data packets. The level of error correction redundancy, the codec selection and other factors, are dynamically changeable by a voice port to replace lost packets without interpolation and to thereby maintain the highest voice quality and data compression ratio that is consistent with the quantity of packet loss.