Note: Descriptions are shown in the official language in which they were submitted.
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METHOD OF AND APPARATUS FOR COMMUNICATIONS CONFERENCING
The present invention relates to c~mmllnications
conferencing and is particularly concerned with audio
conferencing.
Back~round of the Invention
A major shortcoming of audio conferencing today is the lack
of mechanisms enabling participants to break off and hold
side conversations during a conference. Currently, the
only way to do this would be to establish a new connection
dynamically between the parties wishing a slide
conversation. This method is extremely resource intensive
and is implemented only by expensive conferencing systems
of ISDN users.
Another problem with audio conferencing systems today is
that participants do not have any control over the voice
characteristics, especially volume, of other participants
in the audio signal they receive.
Prior Art
Barraclough et al. (audio Conferencing Systems, U.S. Patent
No.: 5,539,741) describe a mixing architecture for LAN-
based audio conferencing which allows participant volume
customization. This patent does not cover modification of
other characteristics such as pitch/tone which are included
in our invention. This patent is based on a single
hardware implemented audio mixer whereas in the present
invention, mixing can be both hardware or software based,
and an arbitrary number of mixers can be invoked.
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Tompkins et al. (Video Conferencing Network, U.S. Patent
No.: 4,710,917) describe a video conferencing network for
providing video, audio and data comm~ln-cations between
remotely disposed video term;n~ls.
Baxter et al. (Distributed Digital Conferencing System,
U.S. patent No.: 4,389,720) describe a digital conferencing
system for TDM (Time Division Multiplexing) based networks
such as the public switched Telephony network. They do not
lo address the internet/intranets, which are Packet Switched
Networks.
Gunner et al. (Volume control in digital teleconferencing,
U.S. Patent No.: 5533112) claim participant volume
customization through a Multiple Input Multiple Output
(MIMO) voice mixing process.
Stevens et al. (Volume control for digital comml]nication
systems U.S. Patent No.: 5420860) claim hardware
implementation of a volume control system which maybe used
for an audio conference. The application of their invention
is not limited to audio conferencing, and can be applied to
any digital commlmlcation system. They do not claim any
other characteristics such as pitch/tone modification
supported by the present invention.
Summarv of the Invention
In accordance with an embodiment of the present invention
there is provided a method of controlling a comm-ln;cations
server comprising the steps of conferencing a plurality of
users together and allowing each user to independently
control signals associated with others of the plurality of
users.
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In a further embodiment of the present invention there is
provided a method of controlling a comm~]n;cations
conference comprising the steps of for each member of the
conference, providing a mixer for controlling signals
associated with other members of the conference and
allowing each user to independently control signals
associated with others of the plurality of of users.
In a further embodiment of the present invention there is
0 provided a method of controlling a com~ln;cations server
comprising the steps of conferencing a plurality of users
together and allowing each user to independently control
signals associated with others of the plurality of users.
allowing each user to establish side conferencing with
selected others of the plurality by controlling signals
associated therewith.
In still a further embodiment of the present invnetion
there is provided a method of controlling a comml~n;cations
conference comprising the steps of for each member of the
conference, providing a mixer for controlling signals
associated with other members of the conference and
allowing each user to independently control signals
associated with others of the plurality of of users,
allowing each user to establish side conferencing with
selected others of the plurality by controlling signals
associated therewith.
Brief Descri~tion of the Drawinas
The present invention will be further understood from the
following description with reference to the accompanying
drawings in which: -
Fig. 1 illustrates c~mm~n~ 1 control API between clientapplication and server application;
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-- 4 --
Fig. 2 illustrates server comm~n~ 1 control API;
Fig. 3 schematically illustrates symbols used in Fig. 4;
Fig. 4 illustrates an audio mixing architecture in
accordance with an embodiment of the present invention;
Fig. 5 illustrates a network implementing the embodiment of
Fig. 4;
Figs. 6-8 illustrate in flow charts steps to éstablish
conferencing in accordance with an embodiment of the
present invention;
Figs. 9-13 illustrate in flow charts steps to establish
enhanced conferencing in accordance with an embodiment of
the present invention.
Detailed Description
In accordance with an embodiment of the present invention
an enhanced mixing method is provided.
In accordance with an embodiment of the present invention a
command/control API (Application Programming Interface)
between the client application and the conference server to
performs this mixing included by reference in figure 1.
In accordance with an embodiment of the present invention a
command/control API (Application Programming Interface)
between the server application and the enhanced software-
based mixing function included by reference in figure 2.
A conference can support an arbitrary number of mixers.
New mixers can be instantiated arbitrarily. This allows
every conference instance to have a unique, optimized
mixing architecture, unlike hardware based mixing methods
CA 02209707 1997-07-07
today. This mixing architecture can be dynamically changed
during the progress of a conference.
Mixers are software instantiated hardware/software mixers.
The choice and number of mixers would depend on the optimal
mixing architecture for use of conferencing resources, and
which meets real-time requirements. Mixing options range
from one mixer per conference participant, to one mixer per
' all conference participants.
Side conversation support through voice characteristics
customization and arbitrary software mixer invocation.
Customization of participants voice characteristics such as
volume, pitch/tone etc. This involves the application of
the nVoice Fontsn concept to audio conferencing.
The embodiments of the invention described have the
following advantages:
customize voice input streams: Participants can
customize the audio signal received form the conference
server. This allows for modification of voice
characteristics of each individual participant in the
conference. This includes modifying volume and tone.
customized voice output streams: Participants can
customize their voice characteristics being heard by other
participants such as tone/pitch.
Mixing technology provides simultaneous support for POTS
users on the SCN and Computer users on the internet
Our enhanced mixing function is able to implement side
conversations without establishing any new connections. It
does this by modifying the voice characteristics of the
conference. The conferencing server invokes a new mixer
CA 02209707 1997-07-07
instance, and may also reduce the volume of the
participants not in the side conversation. In event where
each participant is allocated a mixer, a side conversation
is created by various muting/volume control configurations.
An indication is sent to each participants conferencing
interface of the side conversation. The voice
characteristics of the participants in the side
conversation are Ulocked~ (i.e. cannot be modified by the
other participants) for the duration of the side
0 conversation.
It is not restricted only to volume modification, but also
allows modification of other voice characteristics such as
tone/pitch. For example, by suing function calls specified
in the client-server API, a participant may modify the
perceived volume of another Usoft-spoken~ participant.
The conference server can be accessed form the SCN. This
allows POTS based users to participate in internet based
audio conferences without the need of Gateways. Gateways,
as specified in International Telecommlln;cations Union
recommendations, are expensive to implement. They convert
audio signals into internet protocol packets and vice
versa.
The overview structure is shown in Figure 3, with the
architectural components shown in Figure 4. These
components are described below.
Component 1 is decode. In the case of SCN (Switched
Circuit Network) voice calls, the decode component provides
tone detection for conference number and password
validation.
Component 2 is the filter. The filter will provide gain,
equalization, and voice fonts.
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Component 3 is the software queue for each input stream.
The data stream information is queued for a software-
selectable period of time to reduce loss and jitter from
packet inter-arrival time variation. From the software
queue, the data stream is associated with a software
selectable input bus.
Component 4 is a software selectable input bus. The input
bus allows multiple mixers to process the data from each
0 input queue. Each mixer would be acting independently from
every other mixer, such that an input stream may be
processed by zero, one, or many mixers, but the data stream
on the input bus is never affected by the mixers. All
processed streams are placed on the software selectable
output bus.
Component 5 is the filter. The filter will provide gain,
equalization, and optional voice fonts.
Component 6 is the software mixer which has individual
filters and volume controls for each input. These controls
are software selectable through a software interface.
Component 7 is the filter. The filter will provide gain,
equalization, and optional voice fonts.
Component 8 is the software selectable output bus. One
output channel is used to carry the stream of information
of one mixer. The stream may be sent to one or more
encoders, or link to provide a channel of the input bus.
An encoded stream may be sent to one or many destinations.
Component 9 is the filter. The filter will provide gain,
equalization, and optional voice fonts.
Component 10 is encoding. The encoding component includes,
as required, packetization, and formatting. The preferred
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embodiment of the present invention includes, as encoder
and decoder means, but is not limited to ITU-T G.711, ITU-T
G.723.1, and ITU-T G.729-a CODECs.
Component 11 is an SCN telephony user. There is no data
control channel for the voice data stream unless a data
term; n~ 1 representing the SCN user is connected to the
mixer control algorithm.
Component 12 is an IP voice user.
Component 13 is the mixer control algorithm. ITU-T T.120
connectivity carries the ITU-T T.132 protocol information.
