USOO9253332B2
`
`(12) United States Patent
`Dye et al.
`
`(10) Patent No.:
`(45) Date of Patent:
`
`US 9.253,332 B2
`*Feb. 2, 2016
`
`(54)
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`(71)
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`(72)
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`VOICE CONFERENCE CALLUSING PSTN
`AND INTERNET NETWORKS
`
`Applicant: VPN Multicast Technologies LLC,
`Plano, TX (US)
`Inventors: Thomas Dye, Austin, TX (US); Thomas
`Dundon, San Marcos, TX (US)
`Assignee: VPN Multicast Technologies LLC,
`Plano, TX (US)
`Subject to any disclaimer, the term of this
`patent is extended or adjusted under 35
`U.S.C. 154(b) by 166 days.
`This patent is Subject to a terminal dis
`claimer.
`
`Notice:
`
`Appl. No.: 13/674,227
`
`Filed:
`
`Nov. 12, 2012
`
`Prior Publication Data
`US 2013/O163480 A1
`Jun. 27, 2013
`
`Related U.S. Application Data
`Continuation of application No. 12/646,892, filed on
`Dec. 23, 2009, now Pat. No. 8,339,997, which is a
`continuation of application No. 10/796.560, filed on
`Mar. 9, 2004, now Pat. No. 7,664,056.
`Provisional application No. 60/453,307, filed on Mar.
`10, 2003.
`
`Int. C.
`H04L 2/66
`H04M 3/56
`
`(2006.01)
`(2006.01)
`(Continued)
`
`U.S. C.
`CPC ........... H04M 3/568 (2013.01); H04L 12/1818
`(2013.01); H04L 12/4641 (2013.01);
`(Continued)
`
`(58) Field of Classification Search
`CPC ....... H04M 3/56; H04M 3/561; H04M 7/006;
`H04L 29/06027; H04L 49/25; H04Q
`2213/13204
`USPC .......................... 370/260 263,265, 352–356
`See application file for complete search history.
`References Cited
`
`(56)
`
`U.S. PATENT DOCUMENTS
`
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`
`9, 1998 Civanlar et al.
`3/2000 Hart et al.
`(Continued)
`FOREIGN PATENT DOCUMENTS
`
`WO
`WO
`WO
`
`11, 2001
`WOO1f91397 A2
`WO 02/098063 A1 12/2002
`WOO3OO3665 A1
`1, 2003
`
`OTHER PUBLICATIONS
`Higgins, L. et al., “Tunneling Multicast Traffic Through Non
`Multicast Aware Networks and Encryption Devices.” MILCOM
`2001. Proceedings. Communications for Network-Centric Opera
`tions: Creating the Information Force, McLean, VAOct. 28-30, 2001;
`IEEE Military Communications Conference, New York, NY: IEEE,
`US, vol. 1 of 2, Oct. 28, 2001, pp. 296-300.
`(Continued)
`Primary Examiner — Suhan Ni
`(74) Attorney, Agent, or Firm — Martin & Ferraro, LLP
`(57)
`ABSTRACT
`A system and method for Supporting a multi-participant Voice
`conference call using PSTN and Internet networks is
`described. The method for Supporting a multi-participant
`voice conference call includes receiving voice from a PSTN
`client. The method also includes receiving Voice data from a
`moderator and from at least one remote client connected to
`the Internet. The method then proceeds to mix the voice data
`from the PSTN client with the voice data from the moderator
`into a first mixed voice data that is transmitted to the remote
`client that is connected to the Internet. The method also mixes
`the Voice data from the moderator with the voice data from the
`remote client connected to the Internet into a second mixed
`voice data that is transmitted to the PSTN client.
`
`12 Claims, 6 Drawing Sheets
`
`
`
`PSTN Group.
`412
`
`San
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 001
`
`

`

`US 9.253,332 B2
`Page 2
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`(2006.01)
`(2006.01)
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`(51) Int. Cl.
`HO)4N 7/15
`H04L 2/8
`H04L 2/46
`H04L 29/06
`H04M 7/2
`HO4M 7/OO
`(52) U.S. Cl.
`CPC ...... H04L29/06027 (2013.01); H04L 65/1009
`(2013.01); H04L 65/4038 (2013.01); H04M
`3/567 (2013.01); H04M 7/1225 (2013.01);
`H04N 7/15 (2013.01); H04M 7/009 (2013.01);
`H04M 2207/35 (2013.01)
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`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 002
`
`

`

`US 9.253,332 B2
`Page 3
`
`(56)
`
`References Cited
`
`OTHER PUBLICATIONS
`
`Rosen et al., “Multicast in MPLS/BGP VPNs.” Network Working
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`
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`
`* cited by examiner
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 003
`
`

`

`U.S. Patent
`
`Feb. 2, 2016
`
`Sheet 1 of 6
`
`US 9.253,332 B2
`
`413
`
`412
`
`SR
`
`Wireless
`phone
`
`
`
`PBXI
`
`8
`Wireless
`phone
`
`H.323.Network
`i.e. Ethernet
`
`
`
`FIG. 1
`(Prior Art)
`
`
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 004
`
`

`

`U.S. Patent
`
`Feb. 2, 2016
`
`Sheet 2 of 6
`
`US 9.253,332 B2
`
`
`
`
`
`
`
`ŒŽ (d.18)
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 005
`
`

`

`U.S. Patent
`
`Feb. 2, 2016
`
`Sheet 3 of 6
`
`US 9.253,332 B2
`
`38||808000101000 „ , !
`
`
`
`
`
`
`
`
`
`
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 006
`
`

`

`U.S. Patent
`
`Feb. 2, 2016
`
`Sheet 4 of 6
`
`US 9.253,332 B2
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 007
`
`

`

`U.S. Patent
`
`Feb. 2, 2016
`
`Sheet 5 of 6
`
`US 9.253,332 B2
`
`
`
`J???T?
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 008
`
`

`

`U.S. Patent
`
`Feb. 2, 2016
`
`Sheet 6 of 6
`
`US 9.253,332 B2
`
`Remote Speaker
`401
`Remote
`DEE
`
`D
`
`
`
`VPN NetWork
`Transport 461
`Remote Transport input for AW Clients
`542
`
`VPN Network
`OEc ZO- Transport 461
`544
`Remote Transport Output
`for AW Clients
`
`4.18
`Remote AW Clients
`
`as as as as as as a sess up as as a
`
`as as
`
`a sm as
`
`ess TTTT
`
`Remote Client/Moderator boundary 510
`Local Mic
`452
`Oase
`
`s
`
`VPN Network
`Transport 461
`
`VPN Network
`Transport 461
`
`Local Transport ogy
`
`
`
`Local Transport input
`548
`
`
`
`VoIPEncoder
`522
`
`
`
`552
`Conference Audio Output
`Remote Clients
`Audio
`
`Decoden3
`
`V
`
`Audio D
`CIO LeCompressors
`N
`- 527Remote mixed Audio
`2 ?ec.co.... ca.
`Sos Sound VF bounday 564.
`
`--
`
`\
`WOP Decoder
`524
`
`Audio Mixer 534
`
`as as a - - - - - - - - - - - - - - - - - - -
`
`internet interfaces 435
`
`VPN Network Transport 461
`.
`.
`. essa sease seases as are s - - - - - - - - - - as as a a
`VolP Server (H.323) 409
`PSTN Boundary 515 2 p-y
`PSTNTransport 433
`
`ea
`4O1
`Local Moderator Client
`550
`Moderator Audio input
`Local Speaker Output 564
`Local Speaker S-D
`Conference Application
`Boundary 562
`-
`... Wop
`as w-
`was a
`VRecMix
`VoIP Application S
`Boundary 566
`Moderator Volp
`Mixer 568
`Local Client VoIP
`Boundary 520
`a
`K art
`. .
`
`.
`
`.
`
`.
`
`.
`
`.
`
`g es
`
`PSTN Client
`412
`
`F.G. 6
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 009
`
`

`

`1.
`VOICE CONFERENCE CALLUSING PSTN
`AND INTERNET NETWORKS
`
`PRIORITY CLAIM
`
`This application is a continuation of U.S. patent applica
`tion Ser. No. 12/646,892, filed Dec. 23, 2009, which is a
`continuation of U.S. patent application Ser. No. 10/796.560,
`now U.S. Pat. No. 7,664,056, filed Mar. 9, 2004, which claims
`priority of U.S. Provisional Application No. 60/453,307, filed
`Mar. 10, 2003; wherein all of the above referenced applica
`tions are incorporated by reference in their entirety.
`
`BACKGROUND OF THE INVENTION
`
`1. Field of the Invention
`The present invention relates to computer system architec
`tures and more particularly to audio and video telecommuni
`cations for collaboration over hybrid networks.
`2. Description of the Related Art
`Since their introduction in the early 1980s, audio/video
`conferencing systems ("video conferencing systems) have
`enabled users to communicate between remote sites using
`telephone lines based on dedicated or switched networks.
`Recently, technology and products to achieve the same over
`Internet Protocol have been attempted. Many such systems
`have emerged on the marketplace. Such systems produce
`low-frame-rate and low quality communications due to the
`unpredictable nature of the Internet. Such connections have
`been known to produce long latencies with limited band
`width, resulting in jerky video, dropped audio and loss of lip
`Sync.
`Therefore, most video conferencing solutions have relied
`on dedicated Switched networks such as T1/T3, ISDN or
`ATM. These systems have the disadvantage of higher cost and
`complexity and a lack of flexibility due largely to interoper
`ability issues and higher cost client equipment. High costs are
`typically related to expensive conferencing hardware and
`dedicated pay-per-minute communications usage. Most often
`these dedicated communications circuits are Switched cir
`cuits which use a fixed bandwidth allocation.
`In most prior art systems the public switched telephone
`network (PSTN) is used to transfer audio during conferencing
`and collaboration with remote parties. It is known that quality
`of audio reception is poor over typical prior art Internet pro
`tocol (IP) systems. Prior art audio/video conferencing sys
`tems which use IP networks for audio and video transport lack
`the ability to terminate audio to client end systems through
`both PSTN and IP networks. Thus, it is desirable to achieve a
`hybrid mix of audio and video data over PSTN and IP-based
`audio/video conferencing to achieve full duplex real-time
`operation for all conference participants.
`Modem voice over IP telephony systems have used the
`H.323 standard from the international telecommunications
`union (1 TU). The H.323 standard focuses on the transmis
`sion of audio and video information through the Internet or
`switched private networks. FIG. 1 illustrates a prior art H.323
`system. The block diagram of FIG. 1 includes a number of
`major components, including the general Internet 435. Inter
`net H.323 bridges or gateways 411, telecommunications
`PSTN 433 (Public Switched Telephone Network), wireless
`and land-line phone handsets 412/413, standard Internet
`router 453, an optional gatekeeper 205, a multipoint control
`unit 203, a standard local area network 457, a voice over IP
`server running the H.323 protocol 201, and multiple I/O and
`display terminals 455. FIG. 1 is an example of the prior art
`conferencing system used between hybrid networks connect
`
`2
`ing the PSTN and Internet. Hybrid networks are used to
`communicate audio on internal LAN and WAN networks as
`well as transfer of audio to the existing telephone or PSTN
`network. While the H.323 recommendation allows for video
`conferencing, the prior art systems use private Switched net
`works to establish transport that require expensive H.323
`bridges between dedicated networks and the PSTN. Each of
`the components in FIG. 1 serves this purpose to achieve audio
`telecommunications between multiple parties.
`Referring again to FIG. 1, the components of FIG. 1 are
`interconnected as follows. Prior art technology uses PC or
`client terminals 455 connected through a local area network
`457 to either a data server or a specialized audio/video server
`201. The network server 201 contains the application neces
`sary to generate the H.323 network protocol. The data server
`201 may be connected to a local gatekeeper 205 that is
`responsible for management control functions. As known the
`gatekeeper 205 is responsible for various duties such as
`admission control, status determination, and bandwidth man
`agement. Data server 201 functions are specified and handled
`through the ITU-H.225.0RAS recommendations. In addition,
`management control unit (MCU) 203 is connected to the data
`server 201. The multipoint control unit of a 203 is required by
`the eight-step ITU-5 H.323 recommendation for flexibility to
`negotiate endpoints and determine compatible setups for any
`conference media correspondents. The multipoint control
`unit 203 enables communication between three or more end
`points. Similar to a multipoint bridge, the gatekeeper 205 and
`the multipoint control unit 203 are optional components of
`the H.323 enabled network. Another useful job of the multi
`point control unit 203 is to determine whether to unicast or
`multicast the audio or video streams. As known by one skilled
`in the art, these decisions are dependent on the capability of
`the underlying network and the topology of the multipoint
`conference. The multipoint control unit 203 determines the
`capabilities of each client terminal 455 and status each of
`media stream.
`Again referring to FIG. 1, a standard network router 453 is
`connected between the local area network 457 and the Inter
`net 435. At the outer edges of the Internet, “points of pres
`ence' are located at multiple end points or call termination
`sites. Gateways 411 are used to the transcode the H.323
`network information onto the PSTN 433. Standard telephone
`handsets 413 or wireless phones 412 are connected to the
`PSTN telephony system.
`FIG. 2 illustrates the embodiment of the H.323 protocol
`stack 200, its components and their interfaces to the local area
`network computers at the network interface 300. The input
`and control devices 455 along with a local area network 457
`of FIG. 1 are shown in FIG. 2, consisting of the audio input
`output block 452, the video input and output block 451, the
`system control unit and data collaboration unit 459. These
`input devices are largely responsible for the delivery of media
`data to the H.323 protocol stack 200 shown in FIG. 2.
`Again referring to FIG. 2, the sub blocks of functionality
`that make up the H.323 protocol stack 200 is described. The
`H.323 protocol stack consists of an audio codec 214, and a
`video Codec 213 connected to the audio/video input and
`output blocks 452 and 451, respectively. The audio and video
`CoDecs are responsible for compression and decompression
`of the audio and video sources. The real-time network proto
`col component 215 is connected to the audio video Codecs
`and is also responsible for preparation of the media data for
`transport according to the RTP (real-time protocol) recom
`mendations.
`Again referring to the prior art system of FIG. 2, the H.323
`protocol stack has a system control unit 459 which connects
`
`US 9,253,332 B2
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`25
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`30
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`40
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`45
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`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 010
`
`

`

`US 9,253,332 B2
`
`3
`to multiple control blocks within the H.323 protocol stack
`200. The system control unit connects to the RTC Protocol
`block 217 for real time transport of the control information
`used to set-up and tear down the conference. The system
`control unit 459 also connects to the call-signaling units 221
`and 219 for call signaling protocols and media stream pack
`etization application used for packet-based multimedia com
`munications. The system control unit 459 also connects to the
`control signaling block 223 used for control of protocols for
`multimedia communications. Lastly, the H.323 recommen
`dation defines a data collaboration capability as known and
`outlined in the T. 120 data collaboration unit 225.
`All of the defined blocks make up the H.323 protocol
`network interface to the Transport protocol and network inter
`face unit 300 for transport of data through the modem or
`router 453 to the Internet 435.
`
`SUMMARY OF THE INVENTION
`
`10
`
`15
`
`25
`
`30
`
`35
`
`4
`FIG. 2 illustrates an H.323 protocol stack and its compo
`nents implemented in accordance with prior art;
`FIG.3 illustrates one embodiment of the present invention;
`FIG. 4 illustrates an embodiment using multicast Protocol;
`FIG. 5 illustrates the audio and video data flow over hybrid
`networks; and
`FIG. 6 illustrates the local client data mixing used in the
`preferred embodiment.
`
`DETAILED DESCRIPTION OF THE PREFERRED
`EMBODIMENT
`
`Incorporation by Reference
`
`The following applications and references are hereby
`incorporated by reference as though fully and completely set
`forth herein.
`U.S. application Ser. No. 10/446,407 titled “Transmission
`Of Independently Compressed Video Objects Over Internet
`Protocol, Dye et al. filed May 28, 2003
`U.S. application Ser. No. 10/620,684 titled “Assigning Pri
`oritization During Encode Of Independently Compressed
`Objects, Dye, et al. filed on Jul. 16, 2003.
`International Telecommunications Union Recommenda
`tion H.323, Titled “Packet Based Multimedia Communica
`tion System.” November, 2000
`International Telecommunications Union Recommenda
`tion H.261, Titled “Video Coding for Audio Visual Services at
`Px64 kbps.”
`International Telecommunications Union Recommenda
`tion H.263, Titled “Video Coding for Low Bit-Rate Commu
`nications' February, 1998
`One embodiment of the present invention uses a decentral
`ized model for multipoint conferencing. The multipoint con
`trol unit insures communication capability once the media
`stream is transcoded to the H.323 standard as known. How
`ever, this embodiment mixes media streams at each terminal
`prior to multicast.
`FIG. 3 illustrates one embodiment of the invention. This
`embodimentallows audio video and data collaboration infor
`mation to be securely transferred between a plurality of local
`and remote clients preferably within a virtual private network.
`This embodiment provides the ability for a moderator (single
`member of the conference) to dial out from a desktop com
`puter or terminal (using a novel hybrid network structure)
`connecting an external telephone user's audio into the audio/
`video conference. The embodiment integrates full duplex
`audio, video, and data connections between clients confer
`encing on the Internet and clients conferencing on standard
`telephone systems. The Internet/PSTN hybrid network is the
`medium used for transport. FIG. 3 depicts the necessary
`equipment and protocols to complete the dial out to PSTN
`network method and process.
`Now referring to FIG. 3, the voice over IP moderator 401
`(call initiator or caller) typically has a number of peripherals
`used for real input output devices at the desktop. These
`include a client computing devices Such as a PC or other
`computer 459, a client terminal 455 including a keyboard and
`mouse for input output control, a standard desktop telephone
`457, a video input device or camera 451 and the audio input
`device, microphone 452. In one embodiment each conference
`call connected to the Internet will have similar peripheral
`hardware devices. FIG.3 illustrates a multi-party virtual con
`ference connected over the Internet. Internet clients include
`audio video client 415, audio video client 418, and audio
`video client number and 417. In addition, FIG. 3 shows two
`possible telephony clients using standard wired 413 or wire
`
`A system and method for Supporting a multi-participant
`voice conference call using PSTN and Internet networks is
`described. The method for Supporting a multi-participant
`Voice conference call includes receiving Voice from a public
`switched telephone network (PSTN) client. The method also
`includes receiving Voice data from a moderator and from at
`least one remote client connected to the Internet. The method
`then proceeds to mix the voice data from the PSTN client with
`the voice data from the moderator into a first mixed voice data
`that is transmitted to the remote client that is connected to the
`Internet. The method also mixes the voice data from the
`moderator with the voice data from the remote client con
`nected to the Internet into a second mixed voice data that is
`transmitted to the PSTN client.
`A system for Supporting a multi-participant voice confer
`ence call is also described. The system includes an audio
`mixer, a first transport output, a VOIP mixer and a second
`transport output. The audio mixer mixes Voice data from a
`Public Switched Telephone Network (PSTN) client with
`voice data from a moderator into a first mixed voice data. The
`first transport output transmits the first mixed Voice data to at
`least one remote client. The VOIP mixer mixes voice data
`from the moderator with voice data from the remote client
`connected to the Internet into a second mixed Voice data. The
`second transport output transmits the second mixed Voice
`data to the PSTN client.
`Another system for Supporting a multi-participant Voice
`conference call that includes a PSTN client, a first participant
`client, at least one remote client, a first audio mixer and a
`second audio mixer is described. The PSTN client, the first
`participant, and each remote client device are each configured
`to receive audio. The first audio mixer mixes the audio from
`the PSTN client with the audio from the first participant into
`a first mixed audio. The first mixed audio is transmitted to
`each one remote client connected to the Internet. The second
`audio mixer mixes audio from the first participant with the
`audio from each remote client connected to the Internet into a
`second mixed audio. The second mixed audio is transmitted
`to the PSTN client.
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`BRIEF DESCRIPTION OF THE DRAWINGS
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`A better understanding of the present invention can be
`obtained when the following detailed description of the pre
`ferred embodiment is considered in conjunction with the
`following drawings, in which:
`FIG. 1 illustrates a typical H.323 audio and video confer
`encing system implemented in accordance with prior art;
`
`65
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1001 – 011
`
`

`

`5
`less telephone 412 systems. PSTN client #1412 is connected
`to a wireless cellphone that in turn is connected to the global
`dial network 450, as specified by the PSTN 433. Remote
`telephony user client #2 413 is connected to a standard tele
`phone handset 413 which is connected to the global dial
`network 450 based on the PSTN 433.
`Again referring to FIG. 3 the Internet-based clients 401,
`415, 418, and 417 are connected through routers or modems
`453 preferably in a virtual private network configuration 461.
`A virtual private network bridge 461 is used to connect local
`10
`and remote clients together within a secure private network. A
`local connection from the VPN bridge 407 to the voice over IP
`server 409 is used to transfer conference audio from any
`participant on the IP network to any participant in the PSTN.
`Thus, the voice over IP server 409 is responsible for transcod
`ing audio information from the virtual private network 461 to
`and from the PSTN gateway 411, thus bridging the PSTN and
`VPN together.
`FIG. 4 illustrates one embodiment of the present invention.
`The system of FIG. 4 performs audio transport between mul
`tiple client groups who all share the same multicast group
`address such that audio/video and data may be shared inter
`actively without the need of central servers. Multicast proto
`col and encapsulated media packets are implemented so that
`media data may be routed through public or private IP net
`works without the need for special hardware and software
`during the majority of the network transport. FIG. 4 shows a
`system of virtual networks that interconnect as a virtual pri
`vate network 423. Each VPN tunnel can be connected in a
`series or star topology between one or more multicasting
`30
`appliances 447-457. One or more central servers or VPN
`bridge(s) 407 are at the center of the network topology. Mul
`ticasting enabled appliances 447,449,451,453,455, and 457
`are used at the origination or termination points for audio,
`video, or data (media data) to and from the backbone of the
`35
`transport path. PSTN gateways are used to provide “points of
`presence' throughout and are responsible for origination or
`termination of audio data on and off of the PSTN from the IP
`network topology. Multicast enabled routing allows remote
`clients to be PCs or PSTN gateways which become “Listen
`ers' of media data. Thus, media data is presented or broadcast
`onto a network with one or more group addresses. This
`method uses less bandwidth and reduces latency during trans
`port.
`Again referring to FIG. 4. PSTN group if 1 412 has three
`analog telephones which are switched into a PSTN gateway
`and VoIP server 471 which is networked over public or private
`network connection to a multicast enabled VPN appliance
`447. Appliance 447 is connected to a VPN bridge server 407
`also by means of a virtual private network. The VPN Bridge
`407 is used to authenticate clients, assign multicast IP group
`addresses to various PC clients and VoIP gateway servers. In
`addition the VPN Bridge Server 407 may have additional
`meeting room or conferencing features necessary to carry out
`a multi-party conference. Connected to the VPN Bridge 407
`are various virtual private networks which form network tun
`nels to one or more other multicasting appliances 449, 451,
`453,455,457 which connect to one or more PSTN gateways
`typically located in geographically dispersed areas.
`For the purpose of the illustration of FIG. 4, PSTN group
`#1 412 is audio conferencing with PSTN client #3 414 and
`PSTN client #5416, each of which are audio conferencing
`with Audio/Video client group #4415. In the illustration of
`FIG. 4, each member of audio/video client group #4 share
`audio with all the clients and video with each other. One
`example may be illustrated again referring to FIG. 4. If tele
`phone client #5 416 is talking, the analog audio is converted
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`US 9,253,332 B2
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`6
`from switched network (PSTN) to IP in the VoIP/PSTN gate
`way 475. The digital IP is routed via Internet to an appliance
`455 at the edge of the network typically co-located with the
`VoIP/PSTN gateway 475. The appliance has been configured
`to have a virtual private network creating a tunnel through
`Internet to appliance 453 which also has Internet-based vir
`tual private tunnels to appliance 457 and appliance 447.
`Audio from PSTN client #5416 is broadcast from appliance
`457 whereby all the audio/video client PCs of group #4 are
`“listeners’ and receive the audio from PSTN client 416 at the
`same time. Additionally, PSTN client #5's 416 audio is routed
`over another virtual private network to one or more appli
`ances in this case appliances 447 and 449. PSTN Client group
`#1 412 are also “listeners' of the multicast group as well as
`PSTN Client #3 414. Thus, audio is broadcast to multiple
`audio devices in both IP networks and the PSTN using a
`unique group address and a virtual private network structure.
`Interactivity is gained by using the same process no matter
`who in the group is the broadcaster of audio or video.
`FIG.5 shows a more detailed block diagram of the embodi
`ment of the present invention. The moderator client #1 401
`initiates the call using the application code running on the
`voice over IP server 409. Call initiation and call transfer may
`be accomplished through a VPN tunnel 421 connected to the
`moderator client 401. Two connections to the Moderator cli
`ent #1 401 through the VPN tunnel 421 ar