CA2242426A1 - Method of and apparatus for communications conferencing - Google Patents
Method of and apparatus for communications conferencing Download PDFInfo
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- CA2242426A1 CA2242426A1 CA 2242426 CA2242426A CA2242426A1 CA 2242426 A1 CA2242426 A1 CA 2242426A1 CA 2242426 CA2242426 CA 2242426 CA 2242426 A CA2242426 A CA 2242426A CA 2242426 A1 CA2242426 A1 CA 2242426A1
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- 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
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- 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
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- 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
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- 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
- H04M3/563—User guidance or feature selection
- H04M3/564—User guidance or feature selection whereby the feature is a sub-conference
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/0024—Services and arrangements where telephone services are combined with data services
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- Engineering & Computer Science (AREA)
- Multimedia (AREA)
- Signal Processing (AREA)
- Telephonic Communication Services (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
- Two-Way Televisions, Distribution Of Moving Picture Or The Like (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 communications conferencing and is particularly concerned with audio conferencing.
Background 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 side 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 communications between remotely disposed video terminals.
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 lo network. They do not 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 MultipIe Output (MIMO) voice mixing process.
Stevens et al. (Volume control for digital communication 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 communication system, however it is limited by its interface as it has no data control network. They do not claim any other characteristics such as pitch/tone modification supported by the present invention.
Summary of the Invention In accordance with an embodiment of the present invention there is provided 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.
In a further embodiment of the present invention there is provided 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 users.
o In a further embodiment of the present invention there is provided 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.
In still a further embodiment of the present invention there is provided 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 users, allowing each user to establish side conferencing with selected others of the plurality by controlling signals associated therewith.
Brief Description of the Drawings The present invention will be further understood from the following description with reference to the accompanying drawings in which:
Fig. 1 illustrates command 1 control API between client application and server applicat1on;
Fig. 2 illustrates server command 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 0 of Fig. 4i Figs. 6-8 illustrate in flow charts steps to establish 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 "Voice Fonts" 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 received 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.
The enhanced mixing function is able to implement side conversations without establishing any new connections.
The mixing function 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 that 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 "locked" (i.e.
0 cannot be modified by the other participants) for the duration of the side conversation.
The enhanced mixing function is not restricted to volume modification, but also allows modification of other voice characteristics such as tone/pitch. For example, by issuing function calls speclfied in the client-server API, a participant may modify the perceived volume of another ("soft-spoken") participant.
The conference server can be accessed from the SCN. This allows POTS based users to participate in internet based audio conferences without the need of Gateways.
Referring to Fig. 4, there is illustrated an audio mixing architecture in accordance with an embodiment of the present invention The mixing architecture provides an input bus 10 and an output bus 12, plural input paths 14, 16, 18, and 20 coupled to the input bus 10, plural conference mixers 22 and 24 coupled between the input and output buses 10 an 12 and plural output paths 26 and 28.
Each input path 14-20 includes a decoder 30, a filter 32, and a queue 34. The decoder 30, in the case of switched circuit network (SCN) voice calls, provides tone detection for conference number and password validation. The filter 32 provides gain, equalization, and voice fonts. The queue 34 is a 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.
The conference mixers 22 and 24 each include a plurality of input filters 36a-d, a software mixer 38, and an output filter 40. The input filters 36a-d provide gain, equalization, and optional voice fonts. The software 0 mixer 38 is connected to the input filters 36a-d that provide individual filters and volume controls for each input. These controls are software selectable through a software interface. The output filter 40 provides gain, equalization, and optional voice fonts.
The input bus 10 includes a plurality of buses, input bus 1, 42; input bus 2, 44; input bus 3, 46; input bus 4, 48;
and input bus 5, 50. Similarly the output bus 12 includes an output 1, 52; an output bus 2, 54; an output bus 3, 56; and an output bus 4, 58. The input bus 10 is a software selectable input bus. The input bus allows multiple mixers to process the data from each input queue.
Each mixer acts independently of every other mixer, such that an input stream can be processed by zero, one, or many mixers, but the data streams are placed on the software selectable output bus.
The output bus 12 is software selectable. One output channel for example output bus 1, 52, is used to carry the stream of information of one mixer, in this instance conference mixer 22. The stream may be sent to one or more encoders or links to provide a channel of the input bus. An encoded stream may be sent to one or many destinations.
Each output path 24 and 26 includes a filter 62 and 64, respectively, that provides gain, equalization, and optional voice fonts and an encoder 66 and 68, respectively. The encoder includes, as required, packetization and formatting. The preferred embodiment of the present invention includes, an encoder and decoder but is not limited to ITU-T G.711, ITU-T G.723.1, and ITU-T
G.729-a CODECs.
A SCN telephony user 70 is illustrated connected to channel 4. There is no data control channel for the voice data stream unless a data terminal representing the SCN
user is connected to the mixer control algorithm.
An IP voice user 72 is illustrated connected to channel 3.
A block 74 represents a mixer control algorithm. An ITU-T
T.120 connectivity, represented by line 76, carries the ITU-T T.132 protocol information.
Referring to Fig. 5, there is illustrated a network implementing the embodiment of Fig. 4. The network 80 includes telephony components and data components. The telephony components are represented by a switch 82 and terminal 84 and the data components by a router 86, a data terminal 88 and a data terminal with video 90. Coupling the telephony and data components is a conferencing platform 92. The conferencing platform 92 optional is coupled to a firewall 94 to the internet 96 to which are connected a computer user 98 and a telephony user 100 via an interface 102. The conferencing platform includes an analog voice gateway 104, multipoint control 106, multipoint participation 108 and a digital voice mixer 110 , in accordance with an embodiment of the present invention as illustrated in Fig.4. Also included are Internet protocol (IP) gateways for data 112 and voice 114.
Referring to Fig. 6, there is illustrated in a flow chart steps to establish enhanced conferencing connection in accordance with an embodiment of the present invention.
The method of establishing enhanced conferencing connection includes a step of subscriber 0 calling conference center 0, as represented by a block 120. The conference center requests whether the conference is data or voice, as represented by block 122. If voice, a tone receiver is activated and an announcement is played, as o represented by block 124. Digits are collected, as represented by a block 126. Valid password and identification are verified, as represented by a block 128. A check for sufficient resources is made, as represented by a block 130. Whether there are sufficient resources is queried, as represented by a block 132. If yes, the telephone caller is connected to the conference, as represented by a block 134. If the password or identification is invalid or the resources are insufficient, the user is alerted, as represented by a block 136; the caller is rejected, represented by a block 138; and the connection sequence ends, as represented by a block 140.
If the caller is data, the user is provided with login and password prompts, as represented by a block 142. The password is checked for validity, as represented by a block 144. If valid, resources are checked, as represented by a block 146. And if determined sufficient, as represented by a block 148, a data connection to the conference is made, as represented by a block 150. If resources are insufficient, the user is alerted, as represented by a block 152. The call is rejected and the connection ends, as represented by blocks 154 and 156.
Referring to Fig. 7, there is illustrated in a flow chart steps to set up enhanced voice conferencing in accordance with an embodiment of the present invention. The method of setting up enhanced conferencing for phone includes connecting the caller as shown in Fig. 6 and represented here by a block 160. The conference center sets input source address = x, as represented by a block 162. A
default telephony filter is assigned, as represented by a block 164. An acknowledgement tone is played to the caller, as represented by a block 166. Caller x is attached audio is received to output bus and command Set_bus_output(,,,&dest) is issued, as represented by a 0 block 168. A check for participating user is made, as represented by a block 170. If yes, an input bus is assigned to input source x SET_INPUT_BUS( ), as represented by a block 172. The telephone caller (x) has audio attached to mixer: set_mixer (mixer_number, DN,(current_settings + vol_x=FULL_VOLUME)), as represented by a block 174. An announcement is played, as represented by a block 176. The conference is active, represented by a block 178. If the caller is not an active participant, an announcement is played, as represented by a block 180 and the conference is active as represented by a block 182.
Referring to Fig. 8, there is illustrated in a flow chart steps to set up enhanced data conferencing in accordance with an embodiment of the present invention. The method of setting up enhanced conferencing for data terminal includes making a data connection as shown in Fig. 6 as represented by a block 190. The conference center retrieves client IP address (x), as represented by a block 192. The client voice connection method is queried, as represented by a block 194. Whether by phone, as represented by a block 196. If yes, dial back?, as represented by a block 198. If yes, retrieve client dial back DN, as represented by a block 200. Determine if dial back allowed on this DN, as represented by a block 202.
If yes, dial back as represented by a block 204. If no to block 198, establish PSTN setup, as represented by a block 206. If no to block 202, alert user, as represented by a block 208 and go back to block 194.
If no to block 196 query is connection is via IP phone as represented by a block 210. If yes to this query, assign an IP telephony filter, as represented by a block 212.
Query mixer resources, as represented by a block 214.
Query availability of mixer resources, as represented by a block 216. If yes, proceed with call setup, as lo represented by a block 218, then proceed to D in Fig. 7.
If no to block 210, no voice connection is requested, represented by a block 222; and the conference is active, as represented by a block 224. If no to block 216, alert user, block 226 and end 228.
Referring to Fig. 9, there is illustrated in a flow chart steps to establish enhanced conferencing connection in accordance with an embodiment of the present invention.
During an active conference as represented by circle 250 a volume adjustment requested query is made as represented by a block 252. If YES the system obtains the user to adjust and the mixer number as represented by a block 254.
The system associates the user to a bus number as represented by a block 256 and sets a value using said mixer (mixer number, ..., new vol) as represented by a block 258. It returns to an active conference as represented by circle 260. If NO to the block 252 a side conference requested query is made as represented by a block 262. If YES the system obtains a user list X as represented by a block 264 queries resources as represented by a block 266 and determines if a new user is required as represented by a block 268. If YES the system queries whether there is sufficient mixing and channels available as represented by a block 270. If yes system invokes a new mixer Y as represented by block 272, selects a new output bus Z as represented by a block 274 and queries whether original conference audio is desired as represented by a block 276. If YES the method goes to B
of Figure 10 as represented by circle 296 if NO the method goes to A of Figure 10 as represented by circle 288. If NO to block 268 querying whether a new mixer is required the method passes to C of Figure 11 as represented by circle 308. If NO to the block 270 the system alerts the user as represented by a block 278 and returns to the active conference as represented by circle 280.
If no to block 262, a filter requested query is made as represented by a block 282, a yes leads to application of a filter as represented by a block 284, while a no leads back to an active conference as represented by a block 286. Note other requests are possible, but are not illustrated in Figure 9, and can be added as extensions from decision block 282.
Referring to Figure 10, A represented by circle 288, leads the system to select inputs for a mixer Y as represented by a block 290 using the command:
for (all (X) in user list) set_mixer (Y,ON,X=DEFAULT_VOLUME) and mute side conference participants on main mixer S as represented by a block 292 using the command:
for (all main mixers i) for (all (X) in user list) set_mixer (i,ON,X=VOLUME_OFF) leading back to an active conference as represented by circle 294. B represented by circle 296 leads to select inputs for mixer Y using the command:
for (all (X) in user list) set_mixer (Y,ON,X=DEFAULT_VOLUME) as represented by 298, mute side conference participants on main mixer S using the command:
for (all main mixers i) for (all (X) in user list) set_mixer (i,ON,X=VOLUME_OFF) select main mix loop back buses K using command for (all mixers j) set_bus_input(K[j], main mix bus (j)) as represented by a block 302 and select main mix inputs K
for side conference mixer Y using for (all input buses (K)) set_mixer 0 (Y,ON,K=DEFAULT_VOLUME) as represented by a block 304, then going back to the main active conferences represented by circle 306.
Referring to Figure 11, C represented by a circle 308, leads to a mute/gain adjustment of user channels as represented by a block 310 using the instructions:
for (all main mixers i) for (all participants in side conf(x)) set_mixer (i,x=0);
followed by a query whether main mix is desired as represented by a block 312, if YES select attenuation factors represented by a block 314 if NO set attenuation factor = 0 as represented by a block 316 then adjust overall conference volume on each side participant mixer as represented by a block 318 using the instructions:
for (all side_conf_participant mixers i) for (all non-participants x) set_mixer (i,x=x*attenuation _factor);
leading back to active conferences represented by circle 320.
Referring to Figure 12 filter as represented by 284 leads to the step of obtaining filter action type A as represented by a block 322, obtain action target list T as represented by a block 324 and obtain scope of action S as represented by a block 326 leads to a question is user U
permitted to perform action A on target T with scope of S
CA 02242426 l998-07-06 as represented by decision block 328. A YES leads to invoking filter of type A as represented by a block 330 and using S, T and U to connect filter to processing streams represented by block 332 leading back to an active conference as represented by 334. A NO to the block 328 leads to alerting the user as represented by a block 336 and then a return to active conference as represented by a circle 334.
o Where A, S, T and U are:
A = action, several actions possible {volume, filter, equalization, on, off, ...}
T = target {user, filter, mixer, bus, ...}
S = scope {user's channel only, entire conference, sub-conference, entire system, .. }
~ = user ID
Referring to Figure 13 PSTN setup represented by circle 206 leads to listening for PSTN connections represented by a block 340, querying the existence of that connection asrepresented by block 342, if YES obtaining user ID
information as represented by a block 344, querying user connection via data as represented by a block 346 if YES
associating data control session with PSTN connection as represented by a block 348 and returning to active conference as represented by a circle 350. If NO to the connection at block 342 returning to step 340. If NO to decision at block 346 the step of assigning ID to the PSTN
connection as represented by a block 352 and returning to active conference as represented by circle 354. Dial back as represented by circle 204 leads to steE: Of obtaining user dial back information as represented by a block 360 and allocating dial back resources as represented by a block 362. Whether sufficient resources exist is queried by a block 364, if YES the user is dialled as represented by a block 366 whether connection has been made is queried as represented by a block 368, if YES a data control session is associated with PSTN connection as represented by a block 370 leading to active conference as represented by 372. If NO to query at 364 alert user as represented by a block 374 and return to active conference as represented by circle 376. If NO to query concerning connection at 368 query whether retry limit is exceeded as represented by a block 382 if YES alert user as represented by a block 380 and return to active conference as represented by a circle 376. If NO then dial the user lo as represented by a block 380.
The present invention relates to communications conferencing and is particularly concerned with audio conferencing.
Background 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 side 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 communications between remotely disposed video terminals.
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 lo network. They do not 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 MultipIe Output (MIMO) voice mixing process.
Stevens et al. (Volume control for digital communication 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 communication system, however it is limited by its interface as it has no data control network. They do not claim any other characteristics such as pitch/tone modification supported by the present invention.
Summary of the Invention In accordance with an embodiment of the present invention there is provided 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.
In a further embodiment of the present invention there is provided 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 users.
o In a further embodiment of the present invention there is provided 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.
In still a further embodiment of the present invention there is provided 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 users, allowing each user to establish side conferencing with selected others of the plurality by controlling signals associated therewith.
Brief Description of the Drawings The present invention will be further understood from the following description with reference to the accompanying drawings in which:
Fig. 1 illustrates command 1 control API between client application and server applicat1on;
Fig. 2 illustrates server command 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 0 of Fig. 4i Figs. 6-8 illustrate in flow charts steps to establish 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 "Voice Fonts" 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 received 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.
The enhanced mixing function is able to implement side conversations without establishing any new connections.
The mixing function 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 that 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 "locked" (i.e.
0 cannot be modified by the other participants) for the duration of the side conversation.
The enhanced mixing function is not restricted to volume modification, but also allows modification of other voice characteristics such as tone/pitch. For example, by issuing function calls speclfied in the client-server API, a participant may modify the perceived volume of another ("soft-spoken") participant.
The conference server can be accessed from the SCN. This allows POTS based users to participate in internet based audio conferences without the need of Gateways.
Referring to Fig. 4, there is illustrated an audio mixing architecture in accordance with an embodiment of the present invention The mixing architecture provides an input bus 10 and an output bus 12, plural input paths 14, 16, 18, and 20 coupled to the input bus 10, plural conference mixers 22 and 24 coupled between the input and output buses 10 an 12 and plural output paths 26 and 28.
Each input path 14-20 includes a decoder 30, a filter 32, and a queue 34. The decoder 30, in the case of switched circuit network (SCN) voice calls, provides tone detection for conference number and password validation. The filter 32 provides gain, equalization, and voice fonts. The queue 34 is a 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.
The conference mixers 22 and 24 each include a plurality of input filters 36a-d, a software mixer 38, and an output filter 40. The input filters 36a-d provide gain, equalization, and optional voice fonts. The software 0 mixer 38 is connected to the input filters 36a-d that provide individual filters and volume controls for each input. These controls are software selectable through a software interface. The output filter 40 provides gain, equalization, and optional voice fonts.
The input bus 10 includes a plurality of buses, input bus 1, 42; input bus 2, 44; input bus 3, 46; input bus 4, 48;
and input bus 5, 50. Similarly the output bus 12 includes an output 1, 52; an output bus 2, 54; an output bus 3, 56; and an output bus 4, 58. The input bus 10 is a software selectable input bus. The input bus allows multiple mixers to process the data from each input queue.
Each mixer acts independently of every other mixer, such that an input stream can be processed by zero, one, or many mixers, but the data streams are placed on the software selectable output bus.
The output bus 12 is software selectable. One output channel for example output bus 1, 52, is used to carry the stream of information of one mixer, in this instance conference mixer 22. The stream may be sent to one or more encoders or links to provide a channel of the input bus. An encoded stream may be sent to one or many destinations.
Each output path 24 and 26 includes a filter 62 and 64, respectively, that provides gain, equalization, and optional voice fonts and an encoder 66 and 68, respectively. The encoder includes, as required, packetization and formatting. The preferred embodiment of the present invention includes, an encoder and decoder but is not limited to ITU-T G.711, ITU-T G.723.1, and ITU-T
G.729-a CODECs.
A SCN telephony user 70 is illustrated connected to channel 4. There is no data control channel for the voice data stream unless a data terminal representing the SCN
user is connected to the mixer control algorithm.
An IP voice user 72 is illustrated connected to channel 3.
A block 74 represents a mixer control algorithm. An ITU-T
T.120 connectivity, represented by line 76, carries the ITU-T T.132 protocol information.
Referring to Fig. 5, there is illustrated a network implementing the embodiment of Fig. 4. The network 80 includes telephony components and data components. The telephony components are represented by a switch 82 and terminal 84 and the data components by a router 86, a data terminal 88 and a data terminal with video 90. Coupling the telephony and data components is a conferencing platform 92. The conferencing platform 92 optional is coupled to a firewall 94 to the internet 96 to which are connected a computer user 98 and a telephony user 100 via an interface 102. The conferencing platform includes an analog voice gateway 104, multipoint control 106, multipoint participation 108 and a digital voice mixer 110 , in accordance with an embodiment of the present invention as illustrated in Fig.4. Also included are Internet protocol (IP) gateways for data 112 and voice 114.
Referring to Fig. 6, there is illustrated in a flow chart steps to establish enhanced conferencing connection in accordance with an embodiment of the present invention.
The method of establishing enhanced conferencing connection includes a step of subscriber 0 calling conference center 0, as represented by a block 120. The conference center requests whether the conference is data or voice, as represented by block 122. If voice, a tone receiver is activated and an announcement is played, as o represented by block 124. Digits are collected, as represented by a block 126. Valid password and identification are verified, as represented by a block 128. A check for sufficient resources is made, as represented by a block 130. Whether there are sufficient resources is queried, as represented by a block 132. If yes, the telephone caller is connected to the conference, as represented by a block 134. If the password or identification is invalid or the resources are insufficient, the user is alerted, as represented by a block 136; the caller is rejected, represented by a block 138; and the connection sequence ends, as represented by a block 140.
If the caller is data, the user is provided with login and password prompts, as represented by a block 142. The password is checked for validity, as represented by a block 144. If valid, resources are checked, as represented by a block 146. And if determined sufficient, as represented by a block 148, a data connection to the conference is made, as represented by a block 150. If resources are insufficient, the user is alerted, as represented by a block 152. The call is rejected and the connection ends, as represented by blocks 154 and 156.
Referring to Fig. 7, there is illustrated in a flow chart steps to set up enhanced voice conferencing in accordance with an embodiment of the present invention. The method of setting up enhanced conferencing for phone includes connecting the caller as shown in Fig. 6 and represented here by a block 160. The conference center sets input source address = x, as represented by a block 162. A
default telephony filter is assigned, as represented by a block 164. An acknowledgement tone is played to the caller, as represented by a block 166. Caller x is attached audio is received to output bus and command Set_bus_output(,,,&dest) is issued, as represented by a 0 block 168. A check for participating user is made, as represented by a block 170. If yes, an input bus is assigned to input source x SET_INPUT_BUS( ), as represented by a block 172. The telephone caller (x) has audio attached to mixer: set_mixer (mixer_number, DN,(current_settings + vol_x=FULL_VOLUME)), as represented by a block 174. An announcement is played, as represented by a block 176. The conference is active, represented by a block 178. If the caller is not an active participant, an announcement is played, as represented by a block 180 and the conference is active as represented by a block 182.
Referring to Fig. 8, there is illustrated in a flow chart steps to set up enhanced data conferencing in accordance with an embodiment of the present invention. The method of setting up enhanced conferencing for data terminal includes making a data connection as shown in Fig. 6 as represented by a block 190. The conference center retrieves client IP address (x), as represented by a block 192. The client voice connection method is queried, as represented by a block 194. Whether by phone, as represented by a block 196. If yes, dial back?, as represented by a block 198. If yes, retrieve client dial back DN, as represented by a block 200. Determine if dial back allowed on this DN, as represented by a block 202.
If yes, dial back as represented by a block 204. If no to block 198, establish PSTN setup, as represented by a block 206. If no to block 202, alert user, as represented by a block 208 and go back to block 194.
If no to block 196 query is connection is via IP phone as represented by a block 210. If yes to this query, assign an IP telephony filter, as represented by a block 212.
Query mixer resources, as represented by a block 214.
Query availability of mixer resources, as represented by a block 216. If yes, proceed with call setup, as lo represented by a block 218, then proceed to D in Fig. 7.
If no to block 210, no voice connection is requested, represented by a block 222; and the conference is active, as represented by a block 224. If no to block 216, alert user, block 226 and end 228.
Referring to Fig. 9, there is illustrated in a flow chart steps to establish enhanced conferencing connection in accordance with an embodiment of the present invention.
During an active conference as represented by circle 250 a volume adjustment requested query is made as represented by a block 252. If YES the system obtains the user to adjust and the mixer number as represented by a block 254.
The system associates the user to a bus number as represented by a block 256 and sets a value using said mixer (mixer number, ..., new vol) as represented by a block 258. It returns to an active conference as represented by circle 260. If NO to the block 252 a side conference requested query is made as represented by a block 262. If YES the system obtains a user list X as represented by a block 264 queries resources as represented by a block 266 and determines if a new user is required as represented by a block 268. If YES the system queries whether there is sufficient mixing and channels available as represented by a block 270. If yes system invokes a new mixer Y as represented by block 272, selects a new output bus Z as represented by a block 274 and queries whether original conference audio is desired as represented by a block 276. If YES the method goes to B
of Figure 10 as represented by circle 296 if NO the method goes to A of Figure 10 as represented by circle 288. If NO to block 268 querying whether a new mixer is required the method passes to C of Figure 11 as represented by circle 308. If NO to the block 270 the system alerts the user as represented by a block 278 and returns to the active conference as represented by circle 280.
If no to block 262, a filter requested query is made as represented by a block 282, a yes leads to application of a filter as represented by a block 284, while a no leads back to an active conference as represented by a block 286. Note other requests are possible, but are not illustrated in Figure 9, and can be added as extensions from decision block 282.
Referring to Figure 10, A represented by circle 288, leads the system to select inputs for a mixer Y as represented by a block 290 using the command:
for (all (X) in user list) set_mixer (Y,ON,X=DEFAULT_VOLUME) and mute side conference participants on main mixer S as represented by a block 292 using the command:
for (all main mixers i) for (all (X) in user list) set_mixer (i,ON,X=VOLUME_OFF) leading back to an active conference as represented by circle 294. B represented by circle 296 leads to select inputs for mixer Y using the command:
for (all (X) in user list) set_mixer (Y,ON,X=DEFAULT_VOLUME) as represented by 298, mute side conference participants on main mixer S using the command:
for (all main mixers i) for (all (X) in user list) set_mixer (i,ON,X=VOLUME_OFF) select main mix loop back buses K using command for (all mixers j) set_bus_input(K[j], main mix bus (j)) as represented by a block 302 and select main mix inputs K
for side conference mixer Y using for (all input buses (K)) set_mixer 0 (Y,ON,K=DEFAULT_VOLUME) as represented by a block 304, then going back to the main active conferences represented by circle 306.
Referring to Figure 11, C represented by a circle 308, leads to a mute/gain adjustment of user channels as represented by a block 310 using the instructions:
for (all main mixers i) for (all participants in side conf(x)) set_mixer (i,x=0);
followed by a query whether main mix is desired as represented by a block 312, if YES select attenuation factors represented by a block 314 if NO set attenuation factor = 0 as represented by a block 316 then adjust overall conference volume on each side participant mixer as represented by a block 318 using the instructions:
for (all side_conf_participant mixers i) for (all non-participants x) set_mixer (i,x=x*attenuation _factor);
leading back to active conferences represented by circle 320.
Referring to Figure 12 filter as represented by 284 leads to the step of obtaining filter action type A as represented by a block 322, obtain action target list T as represented by a block 324 and obtain scope of action S as represented by a block 326 leads to a question is user U
permitted to perform action A on target T with scope of S
CA 02242426 l998-07-06 as represented by decision block 328. A YES leads to invoking filter of type A as represented by a block 330 and using S, T and U to connect filter to processing streams represented by block 332 leading back to an active conference as represented by 334. A NO to the block 328 leads to alerting the user as represented by a block 336 and then a return to active conference as represented by a circle 334.
o Where A, S, T and U are:
A = action, several actions possible {volume, filter, equalization, on, off, ...}
T = target {user, filter, mixer, bus, ...}
S = scope {user's channel only, entire conference, sub-conference, entire system, .. }
~ = user ID
Referring to Figure 13 PSTN setup represented by circle 206 leads to listening for PSTN connections represented by a block 340, querying the existence of that connection asrepresented by block 342, if YES obtaining user ID
information as represented by a block 344, querying user connection via data as represented by a block 346 if YES
associating data control session with PSTN connection as represented by a block 348 and returning to active conference as represented by a circle 350. If NO to the connection at block 342 returning to step 340. If NO to decision at block 346 the step of assigning ID to the PSTN
connection as represented by a block 352 and returning to active conference as represented by circle 354. Dial back as represented by circle 204 leads to steE: Of obtaining user dial back information as represented by a block 360 and allocating dial back resources as represented by a block 362. Whether sufficient resources exist is queried by a block 364, if YES the user is dialled as represented by a block 366 whether connection has been made is queried as represented by a block 368, if YES a data control session is associated with PSTN connection as represented by a block 370 leading to active conference as represented by 372. If NO to query at 364 alert user as represented by a block 374 and return to active conference as represented by circle 376. If NO to query concerning connection at 368 query whether retry limit is exceeded as represented by a block 382 if YES alert user as represented by a block 380 and return to active conference as represented by a circle 376. If NO then dial the user lo as represented by a block 380.
Claims (16)
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 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 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 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 users;
allowing each user to establish side conferencing with selected others of the plurality by controlling signals associated therewith.
5. A method as claimed in claim 1 wherein each user is allowed to control volume of each other user.
6. A method as claimed in claim 1 wherein each user is allowed to control tone of each other user.
7. A method as claimed in claim 1 wherein each user is allowed to control pitch of each other user.
8. A method as claimed in claim 2 wherein each user is allowed to control volume of each other user.
9. A method as claimed in claim 2 wherein each user is allowed to control tone of each other user.
10. A method as claimed in claim 2 wherein each user is allowed to control pitch of each other user.
11. A method as claimed in claim 3 wherein each user is allowed to control volume of each other user.
12. A method as claimed in claim 3 wherein each user is allowed to control tone of each other user.
13. A method as claimed in claim 3 wherein each user is allowed to control pitch of each other user.
14. A method as claimed in claim 4 wherein each user is allowed to control volume of each other user.
15. A method as claimed in claim 4 wherein each user is allowed to control tone of each other user.
16. A method as claimed in claim 4 wherein each user is allowed to control pitch of each other user.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA 2242426 CA2242426A1 (en) | 1997-07-07 | 1998-07-06 | Method of and apparatus for communications conferencing |
Applications Claiming Priority (3)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA002209707A CA2209707A1 (en) | 1997-07-07 | 1997-07-07 | Method of and apparatus for communications conferencing |
| CA2,209,707 | 1997-07-07 | ||
| CA 2242426 CA2242426A1 (en) | 1997-07-07 | 1998-07-06 | Method of and apparatus for communications conferencing |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CA2242426A1 true CA2242426A1 (en) | 1999-01-07 |
Family
ID=25679463
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CA 2242426 Abandoned CA2242426A1 (en) | 1997-07-07 | 1998-07-06 | Method of and apparatus for communications conferencing |
Country Status (1)
| Country | Link |
|---|---|
| CA (1) | CA2242426A1 (en) |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP1366607A1 (en) * | 2001-02-07 | 2003-12-03 | Paltalk Holdings Inc. | System architecture for linking packet-switched and circuit switched clients |
| GB2492103A (en) * | 2011-06-21 | 2012-12-26 | Metaswitch Networks Ltd | Interrupting a Multi-party teleconference call in favour of an incoming call and combining teleconference call audio streams using a mixing mode |
-
1998
- 1998-07-06 CA CA 2242426 patent/CA2242426A1/en not_active Abandoned
Cited By (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP1366607A1 (en) * | 2001-02-07 | 2003-12-03 | Paltalk Holdings Inc. | System architecture for linking packet-switched and circuit switched clients |
| EP1366607A4 (en) * | 2001-02-07 | 2005-09-21 | Paltalk Holdings Inc | System architecture for linking packet-switched and circuit switched clients |
| GB2492103A (en) * | 2011-06-21 | 2012-12-26 | Metaswitch Networks Ltd | Interrupting a Multi-party teleconference call in favour of an incoming call and combining teleconference call audio streams using a mixing mode |
| GB2492103B (en) * | 2011-06-21 | 2018-05-23 | Metaswitch Networks Ltd | Multi party teleconference methods and systems |
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