U.S. patent application number 10/697810 was filed with the patent office on 2004-05-06 for large-scale, fault-tolerant audio conferencing in a purely packet-switched network.
Invention is credited to Baxley, Warren E., Clemson, Gregory A., Ernstrom, William P., Nylander, Eric J., Stark, Thomas W., Yackay, Thomas E. JR..
Application Number | 20040085914 10/697810 |
Document ID | / |
Family ID | 29401665 |
Filed Date | 2004-05-06 |
United States Patent
Application |
20040085914 |
Kind Code |
A1 |
Baxley, Warren E. ; et
al. |
May 6, 2004 |
Large-scale, fault-tolerant audio conferencing in a purely
packet-switched network
Abstract
A method of large-scale fault-tolerant audio conferencing in an
audio conferencing system using a purely packet-switched network.
An endpoint places a call to a conference gatekeeper indicating an
audio conference. The conference gatekeeper determines whether the
call contains sufficient information to establish the audio
conference. If there is insufficient information, the endpoint is
connected to an IVR server that obtains sufficient information from
the endpoint. Either way, a CACS selects an MCU hosting or that
will host the audio conference. The CACS then responds to the
endpoint with routing instructions indicating the selected MCU and
the endpoint connects or transfers to the selected MCU. The MCU
mixes input from all endpoints in the audio conference to form a
voice stream, which is then returned to each endpoint in the audio
conference. Audio conference participants can dial-out from the MCU
to bring additional participants into the audio conference. Once
established, the audio conference supports full service audio
conferencing. In addition, dynamic routing permits an operator to
service multiple MCUs, and an audio conference participant and/or
an entire audio conference to be moved between MCUs. The audio
conference can also be broadcast from a streaming protocol server
to passive participants.
Inventors: |
Baxley, Warren E.; (Arvada,
CO) ; Clemson, Gregory A.; (Superior, CO) ;
Ernstrom, William P.; (Golden, CO) ; Nylander, Eric
J.; (Morrison, CO) ; Stark, Thomas W.;
(Englewood, CO) ; Yackay, Thomas E. JR.;
(Broomfield, CO) |
Correspondence
Address: |
Robert C. Dorr, Esq.
Dorr, Carson, Sloan & Birney, P.C.
3010 East 6th Avenue
Denver
CO
80206
US
|
Family ID: |
29401665 |
Appl. No.: |
10/697810 |
Filed: |
October 30, 2003 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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10697810 |
Oct 30, 2003 |
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09426684 |
Oct 25, 1999 |
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6646997 |
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Current U.S.
Class: |
370/260 |
Current CPC
Class: |
H04M 3/562 20130101;
H04L 65/80 20130101; H04L 12/1822 20130101; H04L 29/06027 20130101;
H04L 65/4038 20130101; H04Q 3/0045 20130101 |
Class at
Publication: |
370/260 |
International
Class: |
H04L 012/16 |
Claims
We claim:
1. A method of large-scale fault-tolerant audio conferencing in a
purely packet-switched audio conferencing system, said method
comprising the steps of: placing a call from an endpoint to a
conference gatekeeper, said call indicating an audio conference;
querying a CACS from said conference gatekeeper for routing
instructions for said audio conference; determining in said CACS
whether said audio conference is active; selecting in said CACS an
MCU to host said audio conference when said audio conference is
inactive; selecting in said CACS an MCU hosting said audio
conference when said audio conference is active; responding from
said CACS to said endpoint with said queried routing instructions,
said queried routing instructions indicating said selected MCU;
connecting said endpoint to said selected MCU; mixing input from
all endpoints in said audio conference to form a voice stream; and
returning said voice stream to each endpoint in said audio
conference.
2. The method of claim 1 wherein the step of placing a call links
said endpoint to said conference gatekeeper through a gatekeeper
cloud.
3. The method of claim 1 wherein the step of placing a call links
said endpoint to said conference gatekeeper through said
packet-switched network.
4. The method of claim 1 wherein the routing instructions include
at least an LCF signal indicating the selected MCU.
5. The method of claim 1 wherein the call includes at least an LRQ
signal.
6. The method of claim 1 further comprising the steps of:
determining in said CACS whether the call from said endpoint
contains adequate information to establish said audio conference;
responding from said CACS to said endpoint with routing
instructions to an IVR server when there is inadequate information
to establish said audio conference; connecting said endpoint to
said IVR server when there is inadequate information to route said
call; gathering in said IVR server, after connecting said endpoint
to said IVR server, adequate information to establish said audio
conference; and transferring said endpoint from said IVR server to
said selected MCU after said IVR server gathers said adequate
information.
7. The method of claim 1 further having a dial-out method
comprising the steps of: initiating a call request from said
selected MCU to said conference gatekeeper, said call request
indicating an additional endpoint; transmitting an LRQ from the
conference gatekeeper to the gatekeeper cloud; returning a
destination address to said conference gatekeeper from said
gatekeeper cloud, said destination address corresponding to said
additional endpoint; forwarding said destination address from said
conference gatekeeper to said selected MCU; establishing a
point-to-point call from said MCU to said additional endpoint based
on said destination address, thereby bringing said additional
endpoint into said audio conference.
8. The method of claim 1 further supporting full service audio
conferencing using a reservation system and a call agent.
9. The method of claim 8 wherein the reservation system and the
call agent are tightly integrated.
10. The method of claim 8 wherein the reservation system and the
call agent are loosely integrated.
11. The method of claim 1 wherein said selected MCU is selected
from an MCU pool.
12. The method of claim 1 further including the step of dynamically
routing an operator voice path to service multiple MCUs.
13. The method of claim 1 further including the step of
renegotiating the destination of a voice path to move an audio
conference participant from said selected MCU to a second MCU.
14. The method of claim 1 further including the step of moving said
audio conference from said selected MCU to a second MCU.
15. The method of claim 1 further including the steps of: providing
said audio conference to a streaming protocol server from said
selected MCU; connecting a passive participant to said streaming
protocol server; and broadcasting said audio conference from said
streaming protocol server to a passive participant.
16. A large-scale fault-tolerant purely packet-switched audio
conferencing method, said method comprising the steps of:
establishing an audio conference by: connecting an endpoint to a
conference gatekeeper, said endpoint indicating an audio
conference; querying a CACS from said conference gatekeeper for
routing instructions for said audio conference; determining in said
CACS the status of said audio conference; selecting in said CACS an
MCU to host said audio conference when said status of said audio
conference is inactive; selecting in said CACS an MCU hosting said
audio conference when said status of said audio conference is
active; responding from said CACS to said endpoint with said
queried routing instructions, said queried routing instructions
indicating said selected MCU; connecting said endpoint to said
audio conference through said selected MCU; mixing input from all
endpoints in said audio conference to form a voice stream;
returning said voice stream to each endpoint in said audio
conference; and dialing out from the audio conference when said
status of said audio conference is active to connect additional
endpoints to the audio conference by: initiating in the endpoint
connected to said audio conference a call request from said
selected MCU to said conference gatekeeper, said call request
indicating said additional endpoint; transmitting an LRQ from the
conference gatekeeper to the gatekeeper cloud; returning a
destination address to said conference gatekeeper from said
gatekeeper cloud, said destination address corresponding to said
additional endpoint; forwarding said destination address from said
conference gatekeeper to said selected MCU; establishing a
point-to-point call from said MCU to said additional endpoint based
on said destination address, thereby bringing said additional
endpoint into said audio conference.
17. The method of claim 16 further comprising the steps of:
determining in said CACS whether the call from said endpoint
contains adequate information to establish said audio conference;
responding from said CACS to said endpoint with routing
instructions to an IVR server when there is inadequate information
to establish said audio conference; connecting said endpoint to
said IVR server when there is inadequate information to route said
call; gathering in said IVR server, after connecting said endpoint
to said IVR server, adequate information to establish said audio
conference; and transferring said endpoint from said IVR server to
said selected MCU after said IVR server gathers said adequate
information.
18. The method of claim 16 further supporting full service audio
conferencing using a reservation system and a call agent.
19. The method of claim 16 wherein said selected MCU is selected
from an MCU pool.
20. The method of claim 16 further including the step of
dynamically routing an operator voice path to service multiple
MCUs.
21. The method of claim 16 further including the step of
renegotiating the destination of a voice path to move an audio
conference participant from said selected MCU to a second MCU.
22. The method of claim 16 further including the step of moving
said audio conference from said selected MCU to a second MCU.
23. The method of claim 16 further including the steps of:
providing said audio conference to a streaming protocol server from
said selected MCU; connecting a passive participant to said
streaming protocol server; and broadcasting said audio conference
from said streaming protocol server to a passive participant.
24. A large-scale fault-tolerant audio conferencing method over a
purely packet-switched network, said method comprising the steps
of: initiating a call from an endpoint to a conference gatekeeper;
querying a CACS from said conference gatekeeper for routing
instructions for an audio conference; determining in said CACS
whether the call from said endpoint contains adequate information
to establish said audio conference; responding from said CACS to
said endpoint with routing instructions to an IVR server when there
is inadequate information to establish said audio conference;
connecting said endpoint to said IVR server when there is
inadequate information to route said call and: gathering in said
IVR server adequate information to establish said audio conference;
and transferring said adequate information to the CACS; determining
in said CACS the status of said audio conference; selecting in said
CACS an MCU to host said audio conference when said status of said
audio conference is inactive; selecting in said CACS an MCU hosting
said audio conference when said status of said audio conference is
active; responding from said CACS to said endpoint with said
queried routing instructions, said queried routing instructions
indicating said selected MCU; transferring said endpoint from said
IVR server to said audio conference at said selected MCU when there
is inadequate information to route the call; and connecting said
endpoint to said audio conference at said MCU when there is
adequate information to route the call.
25. The method of claim 24 further supporting full service audio
conferencing using a reservation system and a call agent.
26. The method of claim 24 wherein said selected MCU is selected
from an MCU pool.
27. The method of claim 24 further including the step of
dynamically routing an operator voice path to service multiple
MCUs.
28. The method of claim 24 further including the step of
renegotiating the destination of a voice path to move an audio
conference participant from said selected MCU to a second MCU.
29. The method of claim 24 further including the step of moving
said audio conference from said selected MCU to a second MCU.
30. The method of claim 24 further including the steps of:
providing said audio conference to a streaming protocol server from
said selected MCU; connecting a passive participant to said
streaming protocol server; and broadcasting said audio conference
from said streaming protocol server to a passive participant.
31. A large-scale fault-tolerant audio conferencing method in a
purely packet-switched network, said method comprising the steps
of: initiating a call from an endpoint to a conference gatekeeper
in a gatekeeper cloud; querying a CACS from said conference
gatekeeper for routing instructions for an audio conference;
determining in said CACS whether the call from said endpoint
contains adequate information to establish said audio conference;
responding from said CACS to said endpoint with routing
instructions to an IVR server when there is inadequate information
to establish said audio conference; connecting said endpoint to
said IVR server when there is inadequate information to route said
call and: gathering in said IVR server adequate information to
establish said audio conference; and transferring said adequate
information to the CACS; determining in said CACS the status of
said audio conference; selecting in said CACS a conference MCU from
an MCU pool, said conference MCU hosting said audio conference when
said status of said audio conference is inactive; selecting in said
CACS a conference MCU from an MCU pool, said conference MCU hosting
said audio conference when said status of said audio conference is
active; responding from said CACS to said endpoint with said
queried routing instructions, said queried routing instructions
indicating said selected MCU; transferring said endpoint from said
IVR server to said audio conference at said selected conference MCU
when there is inadequate information to route the call; connecting
said endpoint to said audio conference at said conference MCU when
there is adequate information to route the call, once said endpoint
is connected to said audio conference, said audio conference:
supporting full service conferencing in said audio conference to
said endpoint with a reservation system and a call agent;
supporting dynamically routed audio signals within said
packet-switched network; supporting passive participants in said
packet-switched network supporting dial out from said audio
conference to an additional endpoint.
Description
BACKGROUND OF THE INVENTION
[0001] 1. Related Application
[0002] This application is related to co-owned U.S. Patent
Application entitled LARGE-SCALE, FAULT-TOLERANT AUDIO CONFERENCING
OVER A HYBRID NETWORK, filed on the same date as this
application.
[0003] 2. Field of the Invention
[0004] The present invention relates generally to the field of
packet-switched network audio conferencing. More specifically, the
present invention discloses a method for large-scale,
fault-tolerant audio conferencing in a purely packet-switched
network.
[0005] 3. Statement of the Problem
[0006] The most common method to route calls for an audio
conference is to control a local switch in a GSTN (globally
switched telephony network). That is, a physical point-to-point
connection is made between each piece of equipment in the network
to create an overall point-to-point audio connection for the call.
However, such a switch-controlled application can only route calls
to devices connected to the switch, limiting the overall size of
the system and limiting the geographic distribution of multipoint
control units (MCUs) within the system. In addition, call transfer
(e.g., from one MCU to another) requires that the connection from
the switch to the new endpoint be established and the path to the
transferring endpoint be torn down, thus limiting its use in a
large-scale audio conferencing system.
[0007] Another conventional method to route calls for an audio
conference is to interface with the network signaling layer
(SS7/C7) directly.
[0008] Packet-switched call routing, on the other hand, facilitates
dynamic call routing and call transfer during a call. That is, no
dedicated point-to-point connection is required in a
packet-switched network. Each packet, including the call data and
associated control, is sent individually to a destination address
and the physical route taken from one endpoint to another can vary
from packet to packet, eliminating the need for a dedicated circuit
for each call. Thus, a call can be routed or even transferred
within the packet-switched network simply by renegotiating the end
point address. A need exists to provide audio conferencing using
packet-switched call routing.
[0009] There is a need for audio conferencing implemented on a
purely packet-switched network that provides both scalability and
fault tolerance. Specifically, a need exists to monitor a pool of
MCUs to determine which MCU can best handle the conference, and to
dynamically route calls within the purely packet-switched network
so that a conference participant in one conference call can be
transferred to another conference call and further, entire
conferences can be transferred to other MCUs in the MCU pool
without interrupting the audio conference (i.e., without tearing
down connections and reestablishing the connections within the
packet-based network). A need also exists for audio conferencing
for both receive-only or passive broadcast participants.
Specifically, a need exists to provide a voice stream to the
endpoints connected to the conference but that do not actively
participate in the conference itself (i.e., do not contribute to
the conference voice stream). Yet another need exists for full
service audio conferencing using both high-touch (operator
assisted) or reservation based audio conferencing and automated or
"ad hoc" audio conferencing using the same platform. Specifically,
a need exists to provide conferencing on a reservation basis and on
an impromptu basis by monitoring a pool of MCUs to efficiently
establish conferences in the packet-based network.
SUMMARY OF THE INVENTION
[0010] 1. Solution to the Problem. None of the prior art references
discussed above disclose large-scale, fault-tolerant audio
conferencing implemented in a purely packet-switched network.
[0011] This invention provides an audio conferencing method
implemented on a purely packet-switched network that provides
scalability and fault tolerance.
[0012] A primary object of the present invention is to provide
large-scale, fault tolerant audio conferencing using dynamically
routed, call transfer in a purely packet-switched network. That is,
the present invention monitors a pool of MCUs so that conferences
can be efficiently established and routed to different MCUs when an
MCU approaches capacity or when an MCU has to be taken out of
service. As the audio conferencing method is implemented in a
purely packet-switched network, the destination of each audio
packet can be rerouted seamlessly without interrupting the audio
conference.
[0013] Another object of the present invention is to provide an
audio conferencing method for receive-only or passive participants.
That is, participants that do not actively contribute to the
conference can be accommodated (i.e., receive the conference output
or voice stream).
[0014] Yet another object of the present invention is to provide
full service audio conferencing using both high-touch or
reservation-based audio conferencing and automated or "ad hoc"
audio conferencing on the same platform. That is, a conference need
not be reserved against a dedicated MCU and instead, the method of
the present invention allows a pool of MCUs to be monitored, thus
allowing for both advance conference reservations and ad-hoc
conferences.
[0015] 2. Summary. The present invention discloses a method of
large-scale fault-tolerant audio conferencing in an audio
conferencing system using a purely packet-switched network.
According to the method of the present invention, an endpoint
places a call to a conference gatekeeper indicating an audio
conference (i.e., containing a location-request or LRQ signal). The
conference gatekeeper determines whether the call contains
sufficient information to establish the audio conference. If there
is insufficient information, the endpoint is connected to an
interactive voice response (IVR) server that obtains sufficient
information (i.e., an account number) from the endpoint. Either
way, a conference allocation and control system (CACS) linked to
the conference gatekeeper selects an available multipoint control
unit (MCU) to either host the audio conference if the audio
conference has not been established yet, or the MCU that is already
hosting the audio conference. The CACS then responds to the
endpoint with routing instructions (i.e., a location-found or LCF
signal) indicating the selected MCU. The endpoint then uses the
routing instructions to connect to the selected MCU, or where the
endpoint was initially connected to the IVR server to gather
additional information, the endpoint is transferred from the IVR
server to the selected MCU. Once connected, the MCU mixes input
from all of the endpoints in the audio conference and forms a voice
stream, which the MCU then returns to each endpoint in the audio
conference.
[0016] Once an audio conference is established according to the
method of the present invention, the audio conference participants
(i.e., endpoints connected to the MCU in the audio conference) can
dial-out from the MCU to bring additional participants (i.e.,
another endpoint) into the audio conference. In addition, the
established audio conference supports full service audio
conferencing (i.e., both reservation-based, and ad hoc).
Furthermore, the established audio conference supports dynamic
routing which permits an operator to service multiple MCUs, for the
MCUs to be geographically dispersed, and for an audio conference
participant and/or an entire audio conference to be moved between
MCUs.. The audio conference can also be broadcast from a streaming
protocol server to passive participants. As such, the audio
conference established according to the method of the present
invention using a purely packet-switched network can be both large
scale, and is fault-tolerant.
[0017] These and other advantages, features, and objects of the
present invention will be more readily understood in view of the
following detailed description and the drawings.
BRIEF DESCRIPTION OF THE DRAWINGS
[0018] FIG. 1 is a high-level diagram of a packet-switched network
for use with the method of the present invention.
[0019] FIG. 2 is a flow chart illustrating an audio conferencing
method of the present invention.
[0020] FIG. 3 shows an example of the audio conferencing method of
FIG. 2 in which an IVR server is not used.
[0021] FIG. 4 shows an example of the audio conferencing method of
FIG. 2 in which an IVR server is used.
[0022] FIG. 5 is a flow chart illustrating a dial-out method of the
present invention.
[0023] FIG. 6 shows an example of the dial-out method of FIG.
5.
DETAILED DESCRIPTION OF THE INVENTION
[0024] 1. Overview. FIG. 1 shows a high-level diagram of a purely
packet-switched audio conferencing system 100 using a packet
network 110 (e.g., Internet Protocol or IP, ATM, Frame Relay or any
other packet-switched protocol) in which the method of the present
invention can be implemented. The hardware is conventionally linked
through packetized signals, as shown in FIG. 1. For purposes of
illustration, control or routing signals are shown by dashed lines
and audio (or voice stream) signals are shown by solid lines. An
endpoint 120 (E1, E2, . . . Ei) accesses 112 the conventional
packet network 110 via a gateway 130 and is conventionally linked
therein through a series of routers/hubs 115 to a conference
gatekeeper 150 (e.g., via 118).
[0025] Optionally, each endpoint 120 is also registered with a
gatekeeper 140 through which routing signals are sent and received
such as over links 147. Registration is conventionally used under
the H.323 protocol, however, registration is not required for the
audio conferencing method of the present invention. The endpoint
120 can be connected to the same gatekeeper 140 or different
gatekeepers within a gatekeeper cloud 145 having one or more
gatekeepers 140. The gatekeeper 140 is then linked 142 to the
conference gatekeeper 150. The conference gatekeeper 150 controls
an MCU pool 165 having one or more conferencing MCUs 160 (MCU 1,
MCU 2, . . . MCU n). The conference gatekeeper 150 is also linked
156 to a conference allocation and control system (CACS) 170.
Optionally, the conference gatekeeper 150 is also linked 158 to a
conventionally available interactive voice response (IVR) server
180 that is capable of gathering additional routing information
from the endpoint 120 via links 182, 112
[0026] In one embodiment, the conference system 100 of the present
invention also includes a conventional streaming protocol server
185 (e.g., a real-time standard broadcast server or RTSP) linked
152 to the CACS 170 and the packet network 110, a reservation
system 155 linked 157 to the CACS 170, and a call agent 175 linked
177 to the conference gatekeeper. The streaming protocol server 185
is conventionally available and uses the conference sum (i.e., the
mixed voice stream from all endpoints 120 actively participating in
the audio conference) as input for a broadcast signal to passive
participants (i.e., endpoints 120 not actively participating in the
audio conference). The reservation system 155 is also
conventionally available and used to reserve planned audio
conferences against an available MCU 160 (i.e., an MCU having
available ports 190). Likewise, the call agent 175 is
conventionally available and manages available ports 190 in the MCU
pool 165 and assigns calls on an "ad hoc" basis to available MCUs
160.
[0027] The endpoint 120 is a conventionally available client
terminal that provides real-time, two-way communications using
packetized audio signals. Packetized audio signals contain
digitized and compressed speech or touch tones. Any protocol can be
used under the teachings of the present invention and the specific
protocol will be based on design considerations. That is, different
ITU recommendations for digitizing and compressing signals reflect
different tradeoffs between speech quality, bit rate, computer
power, and signal delay (e.g., G.711, G.723, etc.). It is to be
expressly understood that the endpoint 120 can be either
packet-based or circuit-switched, as the gateway 130 hides the
physical transport to the endpoint 120.
[0028] The gateway 130 is optional under the teachings of the
present invention, and when used can be a part of the packet
network 110 itself. The purpose of the gateway 130 is to provide,
among other things, a translation function between conventional
transmission formats (e.g., H.323, H.225.0, H.221, etc.). It is to
be expressly understood that the gateway 130 can support endpoints
120 that comply with other protocols and the gateway 130 need only
be equipped with the appropriate transcoders. However, the gateway
130 is not required where connections to other networks are not
needed, and the endpoint 120 then communicates directly with
another endpoint 120 on the same network and a single translation
function is used.
[0029] Gatekeepers 140 (and hence the gatekeeper cloud 145) are
also optional. Where the gatekeepers 140 are used under the
teachings of the present invention, the purpose of gatekeepers 140
is to perform two call control functions. Specifically, the
gatekeeper 140 performs address translation and manages bandwidth.
Address translation is done conventionally (e.g., domain name to IP
address or touch tones to IP address) within the packet network 110
itself. Bandwidth is also conventionally managed within the packet
network 110 itself (e.g., as IP trunks reach capacity, the network
moves audio, data, etc. signals to other lower volume IP trunks).
When the gatekeeper 140 is not used, endpoints are connected
through the gateway 130 (i.e., for H.323) or directly through the
packet network 110.
[0030] The conference gatekeeper 150 in conjunction with the CACS
170 controls the creation and execution of audio conferences. The
CACS 170 determines an available MCU 160 (i.e., having sufficient
available ports 190) to host the audio conference and provides
routing instructions to the conference gatekeeper 150 to direct the
call from the endpoint 120 to the appropriate MCU 160. For
instance, if a network administrator has specified a threshold
(i.e., in the CACS) for the number of simultaneous audio
conferences (i.e., number of active conferences, number of
available ports, etc.), the CACS 170 can refuse to make any more
connections once the specified threshold is reached. In addition,
the CACS 170 also provides information concerning the audio
conference parameters to the MCU 160 and collects billing
information.
[0031] The MCU 160 supports audio conferences between three or more
endpoints 120. The MCU 160 is conventionally available and consists
of a multipoint controller (not shown) and optionally one or more
multipoint processors (not shown). For purposes of illustration,
and not intended to limit the scope of the present invention, four
ports 190, 195 are shown on each MCU 160, although a typical MCU
160 can handle approximately 1,500 active conference participants.
Available ports 190 are shown "open" while unavailable ports 195
are shown "closed". The MCU 160 handles negotiations between all
endpoints 120 to determine common capabilities for audio
processing. The MCU 160 also controls audio conference resources by
determining which, if any of the audio streams will be
multicast.
[0032] With respect to the audio conferencing system 100 shown in
FIG. 1, an audio conference is initiated when a call identifying a
particular audio conference is placed by an endpoint 120, as
explained in more detail below. Routing signals are transmitted 112
or 147 either through the packet network 110 (i.e., if the
gatekeeper 140 is not used) or through the gatekeeper 140,
respectively, to the conference gatekeeper 150. An MCU 160 is
selected by the conference gatekeeper 150 and the CACS 170 and the
audio conference is established by connecting the endpoint 120
through the packet network 110 over links 112, 114 to the MCU 160.
Additional endpoints 120 can place a call identifying the audio
conference and are similarly connected via links 112, 114 to the
identified audio conference over link 112 through the packet
network 110 to the MCU 160 by the conference gatekeeper 150 and the
CACS 170, as described in more detail below.
[0033] It is to be expressly understood that each of the hardware
components of the purely packet-switched conferencing system 100
described above are conventionally available, and it is the
arrangement and/or configuration of each component in the manner
described above, and the method of using each component in this
configuration as explained below that is new. Likewise,
communication using packetized signals and various protocols is
conventionally known. It is the combination of each of the
above-identified hardware components to form the conferencing
system 100 for use with the method of the present invention that is
new. It is also to be expressly understood that alternative
hardware configurations are possible under the teachings of the
present invention and that the method of the present invention is
not to be limited by the configuration shown in FIG. 1 nor by any
particular network protocol.
[0034] 2. Establishing a Conference. An embodiment of the audio
conferencing method of the present invention is illustrated in FIG.
2 and explained with reference to FIG. 1. At step 200, an endpoint
120 initiates a call to the audio conferencing system 100, for
example, by entering a destination, account number, URL, or IP
address. Optionally the call is routed 147 through the gateway 130
to an address serviced by the gatekeeper 140 in the gatekeeper
cloud 145. If the gatekeeper 140 is not used, the call is then
routed 112 directly to the packet network 110. Either way, the call
is routed to the conference gatekeeper 150 in step 210. The
initiating call contains conventional packetized control signals
for routing the call including any audio conference identification
information required to initiate the audio conference (i.e., a
conventional location request or LRQ). For example, see co-owned
U.S. patent application Ser. No. 09/366,355 and Ser. No. 08/825,477
(hereinafter, the on-demand teleconferencing methods), incorporated
herein by reference. The LRQ is received via 147, 142 (or 112, 118
when the gatekeeper 140 is not used) by the conference gatekeeper
150 which in turn queries 156 the CACS 170 for audio conference
routing instructions in step 220. The CACS 170 determines whether
the call (i.e., the LRQ) contains sufficient information to set up
and route the audio conference in step 230. If the call contains
sufficient information 232 (i.e., enough information to uniquely
identify a subscriber, such as a subscriber identification, pass
code, etc.), the CACS 170 determines whether the indicated audio
conference is active (i.e., whether other endpoints 120 are
currently connected to the indicated audio conference) in step 240.
That is, the CACS 170 starts all conferences with the MCU 160 and
thus stores all activity in memory. If a CACS 170 is disconnected
from an MCU 160, a conventional process is used to resync the CACS
170 and the MCU 160, and thus the CACS 170 is continuously updated
with respect to activity in the MCU pool 165. If the CACS 170
determines that the indicated audio conference is not active 242,
the CACS 170 selects an available MCU 160 from the MCU pool 165 to
host the audio conference in step 245. In step 260, the CACS 170
then returns (e.g., via 156) routing information to the conference
gatekeeper 150 and the conference gatekeeper 150 responds 142, 147
(or 118, 112 when gatekeeper cloud 145 is not used) to the endpoint
120 with a conventional location found signal (LCF) indicating the
selected MCU 160 to host the audio conference. The endpoint 120
then establishes 112, 114 a point-to-point call via the packet
network 110 with the selected MCU 160 in step 270, and an audio
conference is established with one participant (i.e., the
initiating endpoint). In step 280, the MCU 160 mixes the input from
all endpoints 120 participating in the audio conference, and the
MCU 160 returns (e.g., via 114, 112) a voice stream to the endpoint
120 in step 290. The term "voice stream" as used herein, means the
mixed sum of input from all actively participating endpoints in the
conference. Further, the voice stream returned to an actively
participating endpoint does not include input from the same
endpoint 120.
[0035] Additional endpoints 120 can join an active audio conference
in a manner similar to that outlined above. That is, an additional
endpoint 120 initiates over link 147 (or 112 when gatekeeper cloud
145 is not used) a call to an address identifying the audio
conference in step 200. A conventional LRQ is sent 147, 142 (or
112, 118 when the gatekeeper cloud 145 is not used) to the
conference gatekeeper 150 as discussed above. The conference
gatekeeper 150 queries 156 the CACS 170 for routing instructions in
step 220. If there is sufficient information to set up and route
the audio conference in step 230, the CACS 170 proceeds to
determine whether the audio conference is active in step 240,
selects the active MCU 160 in step 250 if the audio conference is
active 247, and responds 156 with appropriate routing instructions
to the conference gatekeeper 150 in step 260. The conference
gatekeeper 150 responds 142, 147 (or 118, 112 where the gatekeeper
140 is not used) to the endpoint 120 with a conventional LCF signal
indicating the selected MCU 160 hosting the active audio conference
and the endpoint 120 establishes a point-to-point call via links
112, 114 with the selected MCU 160 in step 270, as discussed above.
The MCU 160 mixes the input from each endpoint 120 participating in
the audio conference in step 280 and returns an appropriate voice
stream over links 114, 112 to each endpoint 120 in step 290.
Additional endpoints can continue to join the audio conference in a
similar manner to that just described.
[0036] An example of the audio conferencing method of the present
invention in which there is sufficient information associated with
the call (i.e., an IVR server 180 is not required to gather
additional information such as an account number) is shown in FIG.
3. A new call identifying the audio conference (e.g., containing a
URL, conference access number, etc.) is placed 300 from the
endpoint 120 (e.g., E1, in this example and H.323 compliant
endpoint) via link 147 to a gatekeeper 140 in the gatekeeper cloud
145 (step 200). An LRQ is transmitted 310 from the gatekeeper cloud
145 to the conference gatekeeper 150 via 142 (step 210), which in
turn requests 320 routing instructions (i.e., the details for the
LCF) from the CACS 170 via 156 (step 220). The CACS 170 selects an
available MCU 160 from the MCU pool 165 (steps 230, 240, 245) and
returns 325 routing instructions to the conference gatekeeper 150
via link 156, which in turn forwards 330 an LCF signal through the
gatekeeper cloud 145 and back 335 to the endpoint 120 (E1) via
links 142, 147 (steps 270, 280, and 290). The endpoint 120 (E1)
uses the LCF to setup 340, 345 a point-to-point connection with the
MCU 160 identified by the LCF signal and establish 347 the
requested audio conference (i.e., an audio conference having only
the initiating endpoint E1) via links 112, 114. For example, see
the on-demand teleconferencing methods, incorporated herein by
reference. Additional endpoints 120 (i.e., E2) join the established
audio conference 347 as follows. A new call identifying the
established audio conference 347 is placed 350 to the gatekeeper
cloud 150 via link 147 (step 200). An LRQ is transmitted 360 from
the gatekeeper cloud 145 to the conference gatekeeper 150 via link
142 (step 210), which in turn requests 370 routing instructions to
the established audio conference from the CACS 170 via link 156
(step 220). The CACS 170 selects the active MCU 160 identified as
hosting the requested audio conference (steps 230, 240, and 250)
and returns 375 routing instructions identifying the MCU 160
hosting the audio conference 347 to the conference gatekeeper 150
via link 156, which in turn forwards 380, 385 an LCF signal through
the gatekeeper cloud 145 to the endpoint 120 (E2) via links 142,
147 (step 260). The endpoint 120 (E2) uses the routing information
from the LCF to establish a connection 390, 395 to the appropriate
MCU 160 (via 112, 114), and an active audio conference 397 is
established (i.e., between E1 and E2) (steps 270, 280, and 290).
Additional endpoints 120 (E3, E4, . . . Ei) can participate in the
active audio conference 397 by accessing the appropriate MCU 160 as
just described with respect to the endpoint 120 (E2) or through a
dial out request, as described below. It is to be expressly
understood that the above example is presented to be illustrative
of the audio conferencing method of the present invention, and in
no way should be interpreted to limit the scope of the present
invention.
[0037] In another embodiment, also shown in FIG. 2, where the call
does not contain sufficient information 237 (i.e., additional
information such as an account number is required), the endpoint
120 must first connect (via links 112, 182) to an IVR server 180
capable of gathering the required information in step 235 (e.g., by
querying the endpoint 120 for an account number). Routing proceeds
as described above with respect to steps 240 through 260 and in
step 270, the endpoint 120 is then transferred from the IVR server
180 to the MCU 160 selected in step 245 or 250 before mixing the
input and returning a voice stream in steps 280 and 290,
respectively. Thus, there is no requirement to collocate the device
gathering the information and the MCU 160 which will be the final
destination.
[0038] An example of the audio conferencing method of the present
invention in which an IVR server 180 is used is shown in FIG. 4.
Steps 300, 310 and 320 in FIG. 4 correspond to those shown in FIG.
3. However, in FIG. 4, the CACS determines that the routing request
contains insufficient information to establish an audio conference.
Hence, a signal is returned 325, 330, and 335 via links 118, 112 to
the endpoint 120 (E1) to route the endpoint 120 (E1) to an IVR
server 180. The endpoint 120 (E1) establishes a connection with the
IVR server 180 (400 and 405), and the IVR server 180 gathers 410
additional information (e.g., an account number) from the endpoint
120 (E1) to establish an audio conference (step 235). Once the IVR
server 180 has gathered this information, the IVR server 180 sends
420 a routing request to the CACS 170 via links 158, 156, which in
turn returns 425 routing information to the IVR server 180 (steps
240, 245, and 260). Based on the routing information, the call is
then transferred 430 from the IVR server 180 and a point-to-point
connection is established 340, 345 between the endpoint 120 (E1)
and the MCU 160 and an audio conference 347 is established via
links 112, 114 (steps 270, 280, and 290). Additional endpoints 120
(e.g., E2) join the audio conference again by placing 350 a call
through 360 the gatekeeper cloud 145 to the conference gatekeeper
150 (step 200). Again, the conference gatekeeper 150 requests 370
routing information from the CACS 170 and is provided 375 with
routing information to an IVR server for obtaining additional
information from the endpoint 120 (E2) (steps 210, 220, 230, 237).
The LCF is transmitted 380, 385 to the endpoint 120 (E2) and a call
is established 440, 445 between the endpoint 120 (E2) and the IVR
server 180. The IVR server 180 gathers 450 the additional
information (i.e., an account number, access code, etc.) from the
endpoint 120 (E2) and transmits 460 a routing request to the CACS
170 (step 235). The CACS 170 responds 465 with routing information
identifying the MCU 160 hosting the audio conference, and the call
is then transferred from the IVR server 180 to the identified MCU
160, a point-to-point connection 470 is established between the
endpoint 120 (E2) (steps 240 to 290 discussed above). It is to be
expressly understood that the above example is presented to be
illustrative of the audio conferencing method of the present
invention, and in no way should be interpreted to limit the scope
of the present invention.
[0039] Communication with the gateway 130 and the gatekeeper 140,
and address resolution is conventional. Furthermore, it is to be
expressly understood that the use of the gateway 130 and the
gatekeeper 140 is optional and need not be used under the teachings
of the present invention. In an embodiment where the gateway 130
and the gatekeeper 140 are not used, the call is routed directly
through the packet network 110 (e.g., between routers/hubs 115). 3.
Dial-out Method. An embodiment of the dial-out method of the
present invention is illustrated in FIG. 5. The dial-out method is
used to connect to an endpoint 120 not currently connected to an
active audio conference. For example, the dial-out method can be
used when an active audio conference exists 500 between conference
participants (e.g., E1, E2, and E3) and the conference participants
wish to bring in an additional participant (e.g., E4).
[0040] In step 510, a conference participant conventionally
initiates the dial-out from an originating endpoint 120 (e.g., E1,
via touch tone or a web interface) and the CACS 170 requests a
dial-out from the MCU 160 and supplies the MCU 160 with the address
of the endpoint 120 to connect to (e.g., E4). The MCU 160 initiates
(via 154) a new call request to the conference gatekeeper 150 in
step 520. In step 530, the gatekeeper 140 (or packet network 110
when cloud 145 is not used) receives an LRQ from the conference
gatekeeper 150 and in step 540 the gatekeeper 140 (or packet
network 110) returns the destination address (i.e., via an LCF
message) which is forwarded 154 to the MCU 160 from the conference
gatekeeper 150. The MCU 160 then establishes a point-to-point call
to the endpoint 120 (E4) and mixes the input to form a voice stream
for all conference participants (E1-E4) in step 550, similar to
that described above with respect to the audio conferencing method.
Thus, the additional participant (E4) is brought into the active
audio conference. If the additional participant (E4) does not
answer the dial-out request, the line is disconnected by the
originating endpoint 120 (E1) and the originating endpoint 120 (E1)
is placed back in the audio conference.
[0041] An example of the dial-out method of the present invention
is shown in FIG. 6. In this example, an active audio conference 397
has already been established (e.g., according to the method of
establishing an audio conference discussed above), and the existing
participants (e.g., E1-E3) wish to bring in an additional endpoint
120 (E4) to participate in the active audio conference 397 (step
500). An initiating endpoint (i.e., E1) places a call identifying
the additional endpoint (E4) and the CACS 170 requests 600 a
dial-out from the MCU 160 (step 510). The MCU 160 transmits 610 the
new call to the conference gatekeeper 150 via link 154 (step 520),
which in turn requests 620 the location of the desired endpoint 120
(E4) from the gatekeeper cloud (via 142). The gatekeeper cloud 145
responds 630 via link 142 (step 530) to the conference gatekeeper
150 with an LCF signal which is in turn transmitted 635 to the MCU
160 via link 154 (step 540). The MCU 160 then uses the information
from the LCF to establish 640, 645 a point-to-point call between
the MCU 160 and the endpoint 120 (E4) via links 114, 112. Hence,
the endpoint 120 (E4) is brought into the active audio conference
650 as an additional participant (step 550). It is to be expressly
understood that the above example is presented to be illustrative
of the audio conferencing method of the present invention, and in
no way should be interpreted to limit the scope of the present
invention.
[0042] 4. Full Service Audio Conferencing. Planned audio
conferencing conventionally requires an advance reservation against
a specific MCU 160 or MCU pool 165 and operator assistance (i.e.,
high-touch) to facilitate the audio conference. Ad-hoc audio
conferencing conventionally is able to support an audio conference
without a reservation and without operator assistance by creating a
conference against a single MCU 160. On the other hand, once an
audio conference is established according to the method of the
present invention, the audio conferencing system 100 offers full
service audio conferencing that supports both planned and ad-hoc
audio conferencing.
[0043] The method of the present invention implements full service
audio conferencing by integrating the reservation system 155 of the
planned audio conferencing system and the call agent 175 of the
ad-hoc system. Ports 190, 195 utilized for each audio conferencing
type can be dynamically driven by current loads to achieve maximum
port utilization.
[0044] In one embodiment, the reservation system 155 and the call
agent 175 are loosely integrated. That is, the master reservation
system 155 conventionally used to reserve planned audio conferences
on specific MCUs 160 in pool 165 keeps the ad-hoc call agent 175
informed as to the number of available ports 190 on each MCU 160
and the ad hoc call agent 175 conventionally manages the available
ports 190. The number of available ports 190 on a given MCU 160 are
conventionally monitored to ensure that all reservations can be
serviced. For example, an available port 190 that will be required
to support a reservation in the next five minutes is not considered
available (i.e., 195). Likewise, statistically expected ad-hoc
usage is also monitored and accounted for.
[0045] In another embodiment, the reservation system 155 and the
call agent 175 are tightly integrated. That is, the reservation
system 155 is used to reserve planned audio conferences against MCU
pool 165 but the reservation is not bound to a specific MCU 160.
Instead, the audio conference is assigned to an MCU 160 by the call
agent 175 when it is created and the call agent 175 continuously
monitors the port 190, 195 usage and anticipated near term usage
(i.e., reserved ports) of each MCU 160 in the pool 165 to determine
the number and location of available ports 190. When an audio
conference needs to be created (either a planned audio conference
or an ad-hoc audio conference), the call agent 175 selects an
appropriate MCU 160 to host the audio conference and ensures that
all calls for a given audio conference are routed to the
appropriate MCU 160. Thus, the call agent 175 determines the
location of all audio conferences allowing for greater port 190,
195 utilization as well as better fault tolerance (i.e., audio
conference requests will seldom be denied because available ports
190 are closely monitored).
[0046] 5. Network Centric Call Transfer and Dynamic Call Routing.
Once an audio conference is established according to the method of
the present invention, the packet-switched audio conference system
100 also facilitates dynamic call routing. A point-to-point
connection is made using logical links (i.e., within the packet
network 110) and a dedicated physical connection is not required
(i.e., as in a GSTN). That is, the call data and associated control
are sent via packets through the packet network 110 and each packet
is sent individually to a destination address so that the physical
route taken from end-to-end may vary from packet to packet (i.e., a
call can be routed or transferred by simply renegotiating the
destination address).
[0047] Thus, under the teachings of the present invention, calls
can be routed to any MCU 160 within the MCU pool 165 allowing MCUs
160 to be geographically distributed and the audio conference
network 100 to be large-scale. In addition, the ability to transfer
a call from one MCU 160 to another allows the operator voice path
to be routed to any MCU 160 in the conference system 100. This in
turn allows an operator to service a large number of MCUs 160 and
to quickly switch which MCU 160 their voice path terminates on.
[0048] In addition, an audio conference established according to
the method of the present invention allows an audio conference
participant to be moved from one audio conference to another, even
where the audio conferences are on separate MCUs 160. The
destination address of the packets are simply renegotiated to
another MCU 160 instead of establishing a connection between the
two MCUs 160.
[0049] An audio conference established according to the method of
the present invention also allows a new audio conference to be
created on a different MCU 160 where an MCU 160 is taken out of
service or otherwise unavailable to take additional participants
(e.g., due to overflow, etc.). By transferring calls, the audio
conference can be serviced by any MCU 160 in the system 100. All
calls destined for a "moved" audio conference are still statically
routed to the original MCU 160, but immediately transferred to the
correct MCU 160, thus service to the audio conference is not
interrupted.
[0050] 6. Receive Only Support. Audio conference participants can
be either active or passive. Participants that can both contribute
to and receive audio input from an audio conference are active
participants. Those that can only receive a voice stream from an
audio conference are passive participants. Once an audio conference
is established according to the method of the present invention,
the audio conference supports both active and passive
participants.
[0051] Support for passive participants can still be provided where
there are only a limited number of participants by the MCU 160 the
same as it is in a conventional circuit-switched network. That is,
a full duplex connection can be established and the receive path
simply ignored. However, the method of the present invention can
also use broadcasting to support passive participants. That is, the
audio conference output is directed to a streaming protocol server
185 (e.g., a real-time standard broadcast server RTSP). The
streaming protocol server 185 uses the audio conference sum as its
input, and passive participants can connect to the streaming
protocol server 185 using conventional standards of service. As
such, a large number of broadcast protocols can be supported, and a
virtually unlimited number of passive participants can be supported
with little or no impact on the conferencing MCU 160.
[0052] The foregoing discussion of the invention has been presented
for purposes of illustration and description. Further, the
description is not intended to limit the invention to the form
disclosed herein. Consequently, variation and modification
commensurate with the above teachings, within the skill and
knowledge of the relevant art, are within the scope of the present
invention. The embodiment described herein and above is further
intended to explain the best mode presently known of practicing the
invention and to enable others skilled in the art to utilize the
invention as such, or in other embodiments, and with the various
modifications required by their particular application or uses of
the invention. It is intended that the appended claims be construed
to include alternate embodiments to the extent permitted by the
prior art.
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