At the present time a number of companies commercially offer computer hardware and software products which permit computer users to talk to each other using voice communication, and their computers, over the Internet. The Internet is a network using digital data and a large number of computers which communicate with each other over various types of communication channels, including data telephone lines, such as ISDN (Integrated Services Digital Network) T1, T2, T3 channels.
The digital data is organized into a bitstream consisting of packets. A packet is a series of bits comprising communicated data and a header, which gives control and/or address information, such as the source and destination of the packet.
The user's voice, an analog signal, is converted into digital data. For example, the analog voice signal is sampled at 8,000-100,000 times a second, and the voltage level at each sample is represented by a digital number. The voice Internet telephony software, which is commercially available, converts the voice signal to digital data, compresses it and transmits it as packets over a telephone line to an Internet server (provider or host server) which is generally a high speed and high capacity computer connected to a high bandwidth communication channel as part of the Internet.
For example, a telephone full duplex (two parties may talk simultaneously) real time voice conversation may be initiated over the Internet from one computer to another. Both computers generally require the same Internet telephony software, a microphone, a soundcard and a minimum of processor speed (Intel 486 or Pentium (.TM.)) modem (14.4 bps--bits per second), software (Microsoft 3.1 Windows or higher) and connection to an Internet provider. Both computers must be on line at the same time; which requires a pre-arrangement or a separate telephone toll phone call. Software for that type of system is available from IDT, Hackensack, N.J. ("Net 2 Phone" .TM.), Vocal Tec, Inc. ("Internet Phone" .TM.), or Digiphone .TM. and others.
It has also been announced, in February 1996, by Dialogic Inc.,.Parsippany, N.J., and Vocal Tec Inc., Northvale, N.J., that, in a product available in about the third quarter 1996, users may place voice calls from their PCs (Personal Computers) or a telephone connected to a gateway server, to an Internet "hopoff" server close to the person being called. The hopoff server would place a local call to the person being called. The telephone-to-telephone system would require "back-to-back" servers, i.e., each telephone is connected to a server.
The use of the Internet, in the United States, for voice communication, is presently being opposed by petition to the FCC (Federal Communications Commission) filed by America's Carriers Telecommunication Association. Data packets traverse the Internet by being routed from one node to the next. Each of these hops takes the packet closer to its destination. Each node along the route is designated by a globally unique IP address. Each node in the route looks at the destination address contained in the header of an IP packet and sends the packet in the direction towards its destination. At any time, a node along a particular route can stop accepting, or block one or more packets. This may be due to any number of reasons: congestion, maintenance, node crash, etc. Each routing node constantly monitors its adjacent nodes and adjusts its routing table when such problems occur. As a result, sequentially numbered packets may take different routes as they traverse the Internet.
The audio quality of duplex phone conversation over the Internet is often poor because of delays of transmission of packets, lost packets and lost connections. The delays are unpredictable and are usually caused by the dynamically changing data loads on the network and the changing and often long routes through which the data must pass. Existing methods for reducing this delay problem have included the use of (1) dedicated transmission lines, (2) permanent virtual circuits in which a route is reserved for the duration of the real-time data transmission, and (3) redundantly sending all of the critical data so that the delay experienced by the user will be only the delay of the shortest path.
Methods 1 and 2 above are undesirable for two-way voice communications due to the high cost of the dedicated path (channel) which must be present during the entire conversation. Additionally, these methods (1) and (2) are not universally available to most Internet users. Method (3) is undesirable because it wastes network resources by sending multiple copies of the data, although long delays along a given path are generally only occasional.
The recent increase in consumer interest in the Internet, for example the downloading of graphics using the World-Wide Web, has placed an increased demand for transmission and processing time. It is believed, by some, that such increased demand will result in even poorer audio duplex phone quality.
USPTO Backgrounds
USPTO Backgrounds