CA2209707A1 - Method of and apparatus for communications conferencing - Google Patents
Method of and apparatus for communications conferencingInfo
- Publication number
- CA2209707A1 CA2209707A1 CA002209707A CA2209707A CA2209707A1 CA 2209707 A1 CA2209707 A1 CA 2209707A1 CA 002209707 A CA002209707 A CA 002209707A CA 2209707 A CA2209707 A CA 2209707A CA 2209707 A1 CA2209707 A1 CA 2209707A1
- Authority
- CA
- Canada
- Prior art keywords
- conference
- conferencing
- signals associated
- users
- controlling
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Abandoned
Links
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
- H04M3/568—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities audio processing specific to telephonic conferencing, e.g. spatial distribution, mixing of participants
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M2207/00—Type of exchange or network, i.e. telephonic medium, in which the telephonic communication takes place
- H04M2207/20—Type of exchange or network, i.e. telephonic medium, in which the telephonic communication takes place hybrid systems
- H04M2207/203—Type of exchange or network, i.e. telephonic medium, in which the telephonic communication takes place hybrid systems composed of PSTN and data network, e.g. the Internet
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/40—Applications of speech amplifiers
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
- H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
- H04M7/129—Details of providing call progress tones or announcements
Landscapes
- Engineering & Computer Science (AREA)
- Multimedia (AREA)
- Signal Processing (AREA)
- Telephonic Communication Services (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
Abstract
This invention is based on a client-server conferencing architecture wherein enhanced audio conferencing is provided by utilizing a unique voice mixing method. It supports conference participants on both packet switched networks (such as Internet/Intranets) and Switched Circuit Networks (SCN) (such as the telephone network).
The enhanced voice mixing is performed within the conference server. The conference server can support multiple conference instances.
The enhanced voice mixing is performed within the conference server. The conference server can support multiple conference instances.
Description
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.
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.
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;
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 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 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.
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 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.
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.
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.
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;
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 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 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.
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 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.
Claims (4)
1. A method of controlling a communications 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.
conferencing a plurality of users together and allowing each user to independently control signals associated with others of the plurality of users.
2. A method of controlling a communications 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.
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.
3. A method of controlling a communications 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.
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.
4. A method of controlling a communications 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.
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.
Priority Applications (2)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA002209707A CA2209707A1 (en) | 1997-07-07 | 1997-07-07 | Method of and apparatus for communications conferencing |
| CA 2242426 CA2242426A1 (en) | 1997-07-07 | 1998-07-06 | Method of and apparatus for communications conferencing |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA002209707A CA2209707A1 (en) | 1997-07-07 | 1997-07-07 | Method of and apparatus for communications conferencing |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CA2209707A1 true CA2209707A1 (en) | 1999-01-07 |
Family
ID=4161012
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CA002209707A Abandoned CA2209707A1 (en) | 1997-07-07 | 1997-07-07 | Method of and apparatus for communications conferencing |
Country Status (1)
| Country | Link |
|---|---|
| CA (1) | CA2209707A1 (en) |
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP1112646A1 (en) * | 1998-09-09 | 2001-07-04 | Motorola, Inc. | Voice over internet protocol telephone system and method |
| EP1146722A2 (en) * | 2000-04-14 | 2001-10-17 | Lucent Technologies Inc. | Method and apparatus for providing telephony services switch-based processing of media streams |
| EP1786190A1 (en) * | 2005-11-15 | 2007-05-16 | Alcatel Lucent | Whisper feature for conference systems |
-
1997
- 1997-07-07 CA CA002209707A patent/CA2209707A1/en not_active Abandoned
Cited By (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP1112646A1 (en) * | 1998-09-09 | 2001-07-04 | Motorola, Inc. | Voice over internet protocol telephone system and method |
| EP1112646A4 (en) * | 1998-09-09 | 2004-10-13 | Motorola Inc | Voice over internet protocol telephone system and method |
| EP1146722A2 (en) * | 2000-04-14 | 2001-10-17 | Lucent Technologies Inc. | Method and apparatus for providing telephony services switch-based processing of media streams |
| EP1146722A3 (en) * | 2000-04-14 | 2005-04-13 | Lucent Technologies Inc. | Method and apparatus for providing telephony services switch-based processing of media streams |
| EP1786190A1 (en) * | 2005-11-15 | 2007-05-16 | Alcatel Lucent | Whisper feature for conference systems |
| US7974399B2 (en) | 2005-11-15 | 2011-07-05 | Alcatel Lucent | Enhanced whisper feature |
| CN1968316B (en) * | 2005-11-15 | 2013-08-28 | 阿尔卡特公司 | Telephony system and method for providing enhanced whisper feature |
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| FZDE | Dead |