DESC:INTERNET PROTOCOL (IP) BASED AUDIO CONFERENCING SYSTEM WITH SERVER-LESS OPERATION AND METHOD THEREOF

Field of the invention:

The present invention relates to an audio conferencing system and more particularly to an Internet Protocol (IP) based multicast communication system with open line audio conferencing and paging features which employs multiple fallback servers’ architecture.

Background of the invention:

Present telecommunications systems provide one-to-one, one-to-many and many-to-many communications in a form of intercom, talkback or conferencing. The present conferencing communication systems are available with central media conference servers or distributed media mixing.

The systems with central media conference servers are normally, an audio conference system in an IP environment and are established using a conference unit/server. This server does the media mixing as well as setting up and arbitration of the conference for all participants and sends out separate unicast streams to each participant. Such conferences need to be set up specifically and are not available 24 x 7.

Systems as per some proposed patents use multicast transmission by contributing subscribers and receiving subscriber units undertake media mixing locally. However, usually the number of contributors in a given conference is limited and secondly the arbitration needed to mix the streams from various contributors is not uniform and deterministic. This method may not ensure uniform conferencing experience for all subscribers.
There is a need for 24 x 7 conferencing for large groups for critical communication, where low latency and operation in absence of server is essential.

The US patent application US20120307688A1 by Trilogy discloses an audio conference system and a need of an audio mixing and routing hub / server. In turn the hub sends unicast audio streams to every participant in a conference group resulting in lot of traffic in the network. It may also result in large latency delay.

Another US patent US9736312B2 of Avaya discloses a method and system for controlling audio signals in multiple concurrent conference calls. Further, a complete media server and mixer are needed in the system.

Still another patent CN103095939B of ZTE Corp discloses a meeting voice control system to solve the limited processing resources of the media server in multiparty conference systems which are not applicable for the proposed invention.

In the systems with central media conference server, the IP conferencing feature generally requires a dedicated conference server for media mixing and arbitration and routing of requests. The conference server is a complex dedicated unit that needs to be installed separately in any IP communication system such as IP PBX (Internet Protocol private branch exchange). Also, the number of participants is often limited to a maximum of ten. Further, the conference server needs to feed a separate uni-cast RTP (Real-time Transport Protocol) stream to each of the participants resulting in a huge requirement of bandwidth in the local network. Normally, the conference needs to be setup every time the participants wish to talk to each other. Hence, the system cannot provide instant on-the-fly communication in a group. Further, in these systems dropping in and out of multiple conferences is generally not allowed or cannot be done on the fly. Most importantly, in a critical communication environment, the participants rely on the conference server to communicate with each other. If the conference server fails, the participants cannot communicate with each other.

In the systems with distributed media mixing, the media streams are mixed in the end points. Additionally, the arbitration of speech requests is done by a central server or by the end-point itself.

The US patent US6466550B1 of Cisco discloses a system with distributed media mixing. The system comprises a local transceiver responsible for relaying the speaker's voice over a packet network by multicast transmission to transceivers local to every other conference participant. The total number of simultaneous speakers, however, is limited by an arbitration function resident in each transceiver (IP/Internet Protocol Phone). In particular, the arbitrator contained in each transceiver, enables distributed conferencing by providing control over the number of simultaneous talk Streams, present in the system. The arbitrator monitors source attributes of conference data appearing on each data path. Whenever the number of sources of conference data; exceeds a preset maximum number; the arbitrator opens the switch to prevent transmission of the local signal and the local signal loses arbitration against the remote Sources. The arbitrator may preferably also provide guidance to an input selector inserted in the second data path, allowing it to discard packets from remote sources that have lost arbitration. There is an arbitration time needed for selecting/rejecting incoming multicast streams based on SSRC (Synchronization Source).

In the systems with distributed media mixing, the arbitration mechanism in each transceiver changes dynamically depending on the number of incoming streams resulting in not-so-predictable conferencing experience. There is an arbitration time needed for selecting /rejecting incoming multicast streams based on SSRC. This will produce a different conferencing experience for different participants depending on how the arbitration logic in each end point operates. This may result in different participants listening to different combinations of participants at a given point of time and possibly loss of speech for some participants. In this arrangement there is no synchronization between various participants and is not a very deterministic model; resulting into a sub-optimal conferencing experience.

Accordingly, there exists a need to provide an internet protocol (IP) based audio conferencing system with server-less operation for a proper synchronization between various participants which overcomes drawbacks of the prior arts.

Objects of the invention:

An object of the present invention is to provide an audio conferencing system for large groups and for un-interrupted server-less operation and method thereof.

Another object of the present invention is to provide an IP multicast communication system for an optimum use of bandwidth in the network and method thereof.

Still another object of the present invention is to provide a proper synchronization between various participants and ensure that all participants get identical conferencing experience.

Summary of the invention:

Accordingly, in one aspect the present invention provides an internet protocol (IP) based audio conferencing system with server-less operation. The system comprises at least one main server, a standby / redundant server, a plurality of field communications units, a programming unit and an internet protocol (IP) network provided for communication between the each of the servers, each of the plurality of field communications units and the programming unit. Each of the plurality of field communications units includes a master internet protocol (IP) station, at least two or more internet protocol (IP) substations/end point devices, and at least two or more fallback servers. Each of the plurality of field communications units includes input-output modules and control modules for communicating with each of the servers and communicating with each other. The programming unit has an application to be configured on each of the servers and each of the plurality of field communications units. The internet protocol (IP) network is provided for communication between each of the servers, each of the plurality of field communications units and the programming unit. The main and the standby / redundant servers are responsible only for call routing / arbitration function of the conference calls. The fallback servers are standard field communication units having an additional capability of making call routing/arbitration of conference calls for monitoring and ensuring an un-interrupted large 24 x 7 open-line communications using multicast RTP (Real-time Transport Protocol) streams, in absence of the main server and the standby server. The system provides a very deterministic and predictable conferencing experience with at least one arbitrator/router present in the system at any point of time.

In another aspect, the present invention provides a method of an internet protocol (IP) based audio conferencing with server-less operation. The system configuration is downloaded from the programming unit in each of the plurality of field communications units and saved in a persistent file locally. Every conference group channel of each of the end poine device is assigned a unique multicast IP address. The members send their speech on this multicast address with a specific source id that identifies the contributing source. Further, the receiving station selects the streams on the same multicast address based on the source id of the contributing source and mixes these streams. The number of streams is maximum Two /three or four as configured in the system. Next, the requests from the participants (for contribution) are channelized through the Active server (main or standby or the fallback server) depending on the state of each of those servers. The active server maintains a scroll of the active contributors, which is updated dynamically. In case the scroll is full as per the configured capacity (number of contributors allowed in the system), then the next subscriber is intimated that capacity is full and he will have to wait till a contributor stops contributing. This results in ensuring conflict-free contribution by two / three or four contributors to the conference, which is uniformly received by all the members in the conference.

Brief description of the drawings:

The objects and advantages of the present invention will become apparent when the disclosure is read in conjunction with the following figures, wherein

Figure 1 shows a block schematic diagram of an internet protocol (IP) based audio conferencing system with server-less operation, in accordance with the present invention.

Detailed description of the invention:

The foregoing objects of the present invention are accomplished and the problems and shortcomings associated with the prior art, techniques and approaches are overcome by the present invention as described below in the preferred embodiments.

The present invention is illustrated with reference to the accompanying drawings, throughout which reference numbers indicate corresponding parts in the various figures. These reference numbers are shown in bracket in the following description.

Referring to figure 1, an internet protocol (IP) based audio conferencing system (100) with server-less operation (hereinafter referred as, “the system (100)”), in accordance with the present invention is shown. Specifically, the figure 2 shows a schematic block diagram of an architecture of the system (100) with open line audio conferencing features (anybody in the group can talk and everybody else listens), employing multiple fallback server’s architecture for ensuring un-interrupted server-less operation.

The system (100) comprises of at least one main server (10), a standby / redundant server (20), a plurality of field communications units, a programming unit (60) and an internet protocol (IP) network (70). Each of the plurality of field communications units includes a master internet protocol (IP) station (50), at least two or more internet protocol (IP) substations/end point devices (40), and at least two or more fallback servers (41, 42, 43). Each of the plurality of field communications units includes input-output modules (not shown) and control modules (not shown) for communicating with each of the servers and communicating with each other. The input-output modules of each of the plurality of field communications units include a keypad, a microphone, a display and a speaker, and like, but not limited thereto.

The programming unit (60) has an application to be configured on each of the servers and each of the plurality of field communications units. The internet protocol (IP) network (70) is provided for communication between the each of the servers (10, 20), each of the plurality of field communications units and the programming unit (60).

The system (100) provides a very deterministic and predictable conferencing experience with at least one arbitrator/router present in the system at any point of time. The main server (10) and the standby server (20) are responsible for only call routing / arbitration function of the conference calls and are not responsible for any media processing or mixing.

The fallback servers (41, 42, 43) are standard field communication units having an additional capability of making call routing/arbitration of conference calls for monitoring and ensuring an un-interrupted large 24 x 7 open-line communications using multicast RTP (Real-time Transport Protocol) streams, in absence of the main server (10) and the standby server (20).

The system (100) handles large 24 x 7, open-line conferences using multicast RTP streams. Here, the communication points themselves handle the transmission and reception of multicast audio signals; hence the operation can be independent of the servers. The participants follow a particular discipline/protocol in which two, three or four participants can talk simultaneously into the conference or open-line group, while all others can listen into this conversation. Such multiple audio conferences or open-line groups can be pre-setup, while configuring the system (using programming unit 60) and participants can drop-in or drop-out of these conferences depending on the permissions programmed in the system. The end point device of the participant should be capable of accepting multicast RTP streams and locally mixing the RTP packets from different sources from two / three or four such sources to be reproduced on the speaker. The number of participants that can talk simultaneously in a given conference, is dependent on the capability of end point device to mix number of streams.

The system (100) provides a very deterministic and predictable conferencing experience with at least one arbitrator/router present in the system at any point of time. The conferencing or media mixing is done at the end points (40); hence there is no requirement of a central media mixing server. The speech stream for each conference channel is sent over a multicast address resulting in efficient bandwidth utilization of the network. The between the main server (10), the standby server (20) and the fallback servers (41, 42, 43) is minimal as each server has full knowledge of all conference calls at any point of time. As the participants themselves are transmitting and receiving media packets, they can easily hop in to different pre-configured conference or open-line groups at any point of time. The open-line calls continue to operate even during changeover of servers in the system.

Normally, the system (100) operates under the control of the main server (10) or the standby server (20). Here, every conference group channel, is assigned a unique multicast IP address. The members send their speech on this multicast address with a specific source id that identifies the contributing source. Further, the receiving station (40) selects the streams on the same multicast address based on the source id of the contributing source and mixes these streams. The number of streams is maximum Two /three or four as configured in the system. Next, the requests from the participants (for contribution) are channelized through an active server (main or standby or the fallback server) depending on the state of each of those servers. The active server maintains a scroll of the active contributors, which is updated dynamically. In case the scroll is full as per the configured capacity (number of contributors allowed in the system), then the next subscriber is intimated that capacity is full and he will have to wait till a contributor stops contributing. This results in ensuring conflict-free contribution by two / three or four contributors to the conference, which is uniformly received by all the members in the conference.

Advantages of the invention:
1. The system (100) provides 24 x 7 conferencing for large groups for critical communication, where low latency and operation in absence of server is essential.
2. The conferencing or media mixing is done at the end points (40); hence there is no requirement of a central media mixing server.
3. The switching time between the main server (10), the standby server (20) and the fallback servers (41, 42, 43) is minimal as each server has full knowledge of all conference calls at any point of time.
4. As the participants themselves are transmitting and receiving media packets, they can easily hop in to different pre-configured conference or open-line groups at any point of time.
5. The open-line calls continue to operate even during changeover of servers in the system.

The foregoing objects of the invention are accomplished and the problems and shortcomings associated with prior art techniques and approaches are overcome by the present invention described in the present embodiment. Detailed descriptions of the preferred embodiment are provided herein; however, it is to be understood that the present invention may be embodied in various forms. Therefore, specific details disclosed herein are not to be interpreted as limiting, but rather as a basis for the claims and as a representative basis for teaching one skilled in the art to employ the present invention in virtually any appropriately detailed system, structure, or matter. The embodiments of the invention as described above and the methods disclosed herein will suggest further modification and alterations to those skilled in the art. Such further modifications and alterations may be made without departing from the scope of the invention. ,CLAIMS:We claim:

1. A internet protocol (IP) based audio conferencing system (100) with server-less operation comprising:
at least one main server (10),
a standby / redundant server (20),
a plurality of field communications units having,
a master internet protocol (IP) station (50),
at least two or more internet protocol (IP) substations/end point devices (40), and
at least two or more fallback servers (41 ,42, 43);
wherein, each of the plurality of field communications units includes input-output modules and control modules for communicating with each of the servers (10, 20) and communicating with each other;
a programming unit (60) having an application to be configured on each of the servers (10, 20) and each of the plurality of field communications units; and
an internet protocol (IP) network (70) provided for communication between the each of the servers (10, 20), each of the plurality of field communications units and programming unit (60);
wherein, each of the servers (10, 20) are responsible for only call routing / arbitration function of the conference calls and
wherein, the fallback servers (41 ,42, 43) are standard field communication units having an additional capability of making call routing/arbitration of conference calls for monitoring and ensuring an un-interrupted large 24 x 7 open-line communications using multicast RTP (Real-time Transport Protocol) streams, in absence of the main server (10) and the standby server (20). And,
wherein, the system (100) provides a very deterministic and predictable conferencing experience with at least one arbitrator/router present in the system (100) at any point of time.

2. The audio conferencing system (100) as claimed in claim 1, wherein the system configuration is downloaded from the programming unit (60) in each of the plurality of field communications units and saved in a persistent file locally.

3. The audio conferencing system (100) as claimed in claim 1, wherein the input-output modules of each of the plurality of field communications units include a keypad, a microphone, a display and a speaker.

4. The audio conferencing system (100) as claimed in claim 1, wherein each of the end point device (40) of a participant is capable of accepting multicast RTP streams and locally mixing the RTP packets from different sources to be reproduced on the speaker.

5. The audio conferencing system (100) as claimed in claim 1, wherein number of participants that can talk simultaneously in a given conference, is dependent on the capability of the end point device (40) to mix number of streams.

6. The audio conferencing system (100) as claimed in claim 1, wherein a switching time between the main server (10), the standby server (20) and the fallback servers (41, 42, 43) is minimal as each server has full knowledge of all conference calls at any point of time.

7. A method of an internet protocol (IP) based audio conferencing system (100) with server-less operation comprising steps of:
downloading system configuration from the programming unit (60) in each of the plurality of field communications units and saving in a persistent file locally;
assigning a unique multicast IP address to every conference group channel of each of the end poine device (40), wherein members send their speech on the multicast address with a specific source id that identifies a contributing source;
selecting the streams on the same multicast address based on the source id of the contributing source and mixeing these streams by a receiving station selcted from the each of the end poine device (40), wherein number of streams is maximum two /three or four as configured in the system;
channelizing requests from the participants (for contribution) through an active server (main or standby or the fallback server) depending on the state of each of the servers (10, 20, 41, 42, 43), wherein the active server maintains a scroll of the active contributors, which is updated dynamically and in case the scroll is full as per the configured capacity (number of contributors allowed in the system), then the next subscriber is intimated that capacity is full and have to wait till a contributor stops contributing, thereby ensuring conflict-free contribution by two / three or four contributors to the conference, which is uniformly received by all the members in the conference.

Date this 12th day of September 2020

Prafulla Wange
(Agent for the Applicant)
(IN/PA-2058)