Abstract: A system and method is disclosed for real-time data and voice transmission over an internet network. A PSTN voice packet is received and digitized at a network gateway. A destination gateway and destination transmux is identified and a destination gateway address and a destination transmux address are appended to the digitized voice packet. The voice packet is received at an originating transmux and broken into gateway subpackets. The gateway subpackets are aggregated and the destination transmux address is removed from the gateway subpackets. The transmux voice packets are then transmitted over the network to a destination transmux, where they are broken into transmux subpackets. The subpackets are sorted and aggregated by a destination gateway address, and the destination gateway addresses are then removed. Voice packets are received from the destination transmux by a destination gateway and converted to analog voice packets and transmitted to a destination PSTN.

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.