Multiplex communications – Special services – Conferencing
Reexamination Certificate
2000-06-28
2004-01-27
Patel, Ajit (Department: 2664)
Multiplex communications
Special services
Conferencing
C379S202010
Reexamination Certificate
active
06683858
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates generally to computer-based telephony networks and more particularly to servers that manage telephony conferencing.
2. Related Art
In today's technological environment, there exists many ways for several people who are in multiple geographic locations to communicate with one another simultaneously. One such way is audio conferencing. Audio conferencing applications serve both the needs of business users (e.g., national sales force meeting) and leisure users (e.g., audio chat room participants) who are geographically distributed.
Traditional audio conferencing involved a central conferencing server which hosted an audio conference. Participants would use their telephones and dial in to the conferencing server over the Public Service Telephone Network (PSTN) (also called the Plain Old Telephone System (POTS)).
The availability of low-cost personal computers, networking equipment, telecommunications, and related technology, however, has dramatically changed the way people communicate. One example of such change is the explosion of people connected to the global (sometimes referred to as the “public”) Internet.
The connectivity achieved by the Internet—connecting numerous, different types of networks—is based upon a common protocol suite utilized by those computers connecting to it. Part of the common protocol suite is the Internet Protocol (IP), defined in Internet Standard (STD) 5, Request for Comments (RFC) 791 (Internet Architecture Board). IP is a network-level, packet (i.e., a unit of transmitted data) switching protocol.
In recent years, the possibility of transmitting voice (i.e., audio) over the worldwide public Internet has been recognized. Voice over IP (VoIP) began with computer scientists experimenting with exchanging voice using personal computers (PCs) equipped with microphones, speakers, and sound cards.
VoIP further developed when, in March of 1996, the International Telecommunications Union-Telecommunications sector (ITU-T), a United Nations organization, adopted the H.323 Internet Telephony Standard. Among its specifications, H.323 specifies the minimum standards (e.g., call setup and control) that equipment must meet in order to send voice over the IP, and other packet-switched network protocols where quality of sound cannot be guaranteed. Thus, conferencing servers (also called multipoint control units (MCUs)) were developed to host audio conferences where participants connected to a central MCU using PC-based equipment and the Internet, rather than traditional phone equipment over the PSTN.
More recently, several alternatives to H.323 have been developed. One such alternative is the Session Initiation Protocol (SIP) developed within the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group. SIP, which is well-known in the relevant art(s), is a signaling protocol for Internet conferencing and telephony. SIP addresses users using an e-mail-like address and utilizes a portion of the infrastructure used for Internet e-mail delivery. SIP is more powerful than H.323 in providing call control and extended feature sets. It handles basic setup functions as well as enhanced services (e.g., call forwarding).
Given the rapid pace of development in the telephony industry—both in protocols and equipment—and the existence of legacy equipment and protocols (e.g., telephones and switching networks such as the PSTN), it is desirable for conferencing servers (or MCUs) to provide support for users of both new (i.e., packet-based) and legacy (i.e., switching-based) systems. Therefore, what is needed is a hybrid server architecture for mixing and non-mixing client conferencing. The hybrid server should realize the capabilities of the various participants' equipment (e.g., PC-based client versus phone-based clients) and provide the appropriate audio data to each participant.
SUMMARY OF THE INVENTION
The present invention is directed to a hybrid server architecture, that meets the above-identified needs, whereby mixing (e.g., PC-based clients) and non-mixing (e.g., phone) clients can simultaneously participate in a single audio conference application.
The system of the present invention includes a receiver capable of receiving audio packets from each client, means for determining and keeping a list of clients who are currently active speakers, and means for storing information (e.g., database, list, linked list, table, flag, or the like) indicative of whether each client has the capability to mix multiple audio streams.
The system also includes a multiplexor capable of multiplexing the packets of audio data received from each client on the list of active speakers into a multiplexed stream, and a mixer capable of mixing the packets of audio data received from each client on the list of active speakers into one combined packet.
The system further includes means for sending the multiplexed stream to each of the clients which have the capability to mix multiple audio streams, and the combined packet to each of the plurality of clients which do not have the capability to mix multiple audio streams.
The method and computer program product of the present invention include the steps of receiving audio packets from each client, determining which are active speakers and forming an active speakers list. Then, the clients are divided into two categories—those which have the capability to mix multiple audio streams and those which do not. For those clients which can mix, the server multiplexes the packets of audio data received from each client on the active speakers list into a multiplexed stream. For those clients which cannot mix, the server mixes the packets of audio data received from each client on the active speakers list into one combined packet.
The method and computer program product of the present invention then send the multiplexed stream to each of the clients that can mix, and send the combined packet to each of the clients that cannot mix. The method and computer program product of the present invention also perform an “echo suppression” during the sending of either the multiplexed stream or combined packet so that each client, if they are an active speaker, will not hear themselves speaking.
An advantage of the present invention is that a single server or multipoint control unit (MCU) can provide conferencing services to multiple clients that are using varying equipment and protocols.
Another advantage of the present invention is that servers or MCUs, by realizing the audio mixing capabilities of their clients, can distribute the computational burden of mixing audio streams of the active speakers.
Another advantage of the present invention is that by providing multiplexed packets to clients who are capable of mixing, better sound quality is achieved by reducing the effect of “transcoding artifacts.”
Yet another advantage of the present invention is that by providing multiplexed packets to clients who are capable of mixing, servers or MCUs can be scaled to support more simultaneous conferences due to the efficiency gained by not having to mix for every client.
Further features and advantages of the invention as well as the structure and operation of various embodiments of the present invention are described in detail below with reference to the accompanying drawings.
REFERENCES:
patent: 5914940 (1999-06-01), Fukuoka et al.
patent: 6418125 (2002-07-01), Oran
Chu Frank J.
Dobjanschi Virgil Patrick
Gates Corey
Kwan Katherine W.
Wright Daniel W.
Paltalk Holdings, Inc.
Patel Ajit
Pham Brenda
Skadden, Arps Slate Meagher & Flom LLP
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