System and method for simulating telephone use in a network...

Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching

Reexamination Certificate

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Details

C379S093090

Reexamination Certificate

active

06487196

ABSTRACT:

A. FIELD OF THE INVENTION
The present invention relates to telephony services using a wide-area network (WAN) as a transport medium, and more particularly to the simulation of telephone use in an Internet-based telephone system.
B. BACKGROUND OF THE INVENTION
The growth of the Internet has made it possible for users to obtain information from sources located virtually anywhere in the world. Users communicate over the Internet by connecting computers and computer networks to the Internet's data transport facilities.
In order to facilitate such communication, industry and international standards bodies have established sets of functional requirements, conventions or rules that govern the transmission of data over both telephone and packet switched computer networks. These functional requirements or rules are known in the art as “protocols.” The implementation of protocols is necessary in order to bring order, and standardization, to the communications field and allow equipment of diverse manufacturers to be interoperable.
Some protocols are considered low level transmission media related modulation protocols, such as modulation schemes implemented in a modem, for example V.34, V.22 bis, etc. Other protocols are considered higher level, and relate to such features as error control, transmission control protocols and network level routing and encapsulation of data. Examples of such protocols are the Point-to-Point Protocol (PPP), the Serial Line Interface Protocol (SLIP), and the Real-time Transport Protocol (RTP). The requirements of these latter protocols are typically prepared as a “Request For Comment” document, circulated among the industry, and eventually adopted by developers.
Developers have applied the various functions defined in protocols to develop devices and systems that improve the performance and capabilities of the Internet. One such device is a “network access server”. The network access server is a device that is capable of receiving a plurality of simultaneous incoming calls from the Public Switched Telephone Network (PSTN) and routing them to a packet switched computer network for transmission to a host computer system, or telephone or other device connected to the computer network. The network access server is also capable of handling multiple simultaneous calls from the computer network and directing them onto a communications link in the PSTN for transmission to the remote user.
The patent to Dale M. Walsh et al., U.S. Pat. No. 5,525,595, which is fully incorporated by reference herein, describes an integrated network access server suitable for use in the present invention. Such a device has been commercialized widely by 3Com Corporation (previously U.S. Robotics Corp.) under the trade designation Total Control™ Enterprise Network Hub. Network access servers similar in functionality, architecture and design are available from other companies, including Ascend Communications, Livingston Enterprises, Multitech, and others. The invention is suitable for implementation in network access servers from the above companies, and other similar devices.
Improvements in network access servers and in the development of protocols for a variety of functions have resulted in the development of other applications for the Internet. For example, Internet Telephony would use the Internet to connect two or more plain old telephones (POTS). Internet telephony would make long distance calls substantially less costly. Internet telephony would potentially add functions to phone service that may currently not be easy to provide. For example, impromptu conferences (like CHAT ROOMS), anonymous meeting services, etc. may be needed.
Examples of Internet telephony systems are disclosed in U.S. patent application Ser. No. 08/970,834, “DISTRIBUTED PROCESSING OF HIGH LEVEL PROTOCOLS, SUCH AS REAL TIME TRANSPORT PROTOCOLS, IN A NETWORK ACCESS SERVER” to Daniel Schoo et al., which is hereby incorporated by reference. Schoo et al. discloses a network access server that uses the Real-time Transport Protocol (RTP) to connect a H.323 client PC to a telephone. In the system in Schoo et al., telecommunications devices may connect via the PSTN to a network access server. The network access server converts audio and video signals to separate audio and video streams in H.323 format. The streams are transported over the Internet to H.323 computers connected to the Internet.
The systems disclosed in Schoo et al. are particularly suited for use in real-time video conferencing. It would be desirable to have an Internet telephony system that would provide the functions and services of a plain-old telephone service. One way to connect two telephones over a wide-area network (WAN), such as the Internet, is to connect them to separate, called and calling, network access servers by using methods and systems disclosed in Schoo et al. The first telephone would communicate with the second telephone by starting Internet sessions to carry audio signals from the called network access server to the calling network access server closest to the second telephone. The calling network access server may dial a call to the second telephone according to call-setup instructions from the called network access server. The advantage of using the Internet for telephone service is that long distance calls may be made for the cost of local calls and new features may be added to phone service. When the first telephone initiates the connection by connecting to the first network access server, it makes a local call.
In order to communicate audio signals in an Internet-based telephone system, the network access server uses the audio signals received from the first telephone over the PSTN. These audio signals are typically pulse code modulated (PCM) signals according to the international G.711 standard. As described in Schoo et al., G.711 audio signals may need to be transcoded to G.723 or G.729 compressed audio signals to conserve bandwidth. The compressed audio signals are packetized and communicated in streams of packets over the Internet as discussed in Schoo et al.
The sound made on a telephone in an Internet-based telephone system is transported to the destination telephone number in a digitized form. Consequently, many of the sounds associated with plain old telephone service (POTS) over analog lines are not transported. For example, periods of silence in a POTS telephone create a sound while the telephone connection is available. The sound may be a soft noise, however, it is perceptible and different from the complete silence that a user would hear when the connection is no longer available. Many of the sounds in a POTS telephone may arise from limitations of the telephone system, however, the sounds provide feedback to the user that informs the user as to the status of the connection. Many of these sounds are so familiar that they are capable of communicating the status of a call more efficiently than a voice message.
The digitized form in which sound is transported in an Internet-based telephone system may not allow a user to determine the status of a call. The digitized form may sound completely silent during periods of silence. The sound heard by the user is different from the sound heard on a POTS telephone. In an Internet-based telephone system, a user picks up a telephone and hears complete silence before beginning to dial. There would be no indication that the telephone system is even working. Tones may be generated during dialing. However, until voices are generated at the telephone, the user hears nothing. If the other party in the telephone call hangs up first, only silence is heard. The user may let several seconds elapse before wondering if perhaps the other party hung up. Users may find it difficult to accept such differences in Internet-telephone service.
It would be desirable for the network-based telephone system to sound more like a plain old telephone to provide a user of the network-based telephone the “feel” of plain old analog telephone service.
SUMMARY OF THE INVENTION
In view of the above, a network

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