Conferencing system

CONFERENCE SYSTEM

Field of the invention

The present invention relates to a method and apparatus for adjusting conference calls in communication systems, and in particular but not exclusively to wireless communication systems.

Background of the invention The concept of conference calls in public switched telephone networks (PSTN) is well known. PSTN conferences are typically set by a first participant who calls a specific customer support number and is provided with a conference bridge number and a PIN code. The first participant can then provide this information to any other potential participants. Participants wishing to join the call could each dial the conference bridge number, provide the PIN code upon request, and could subsequently be admitted to the conference call.

As an alternative, the Internet could conceivably be used to dial conference calls. A specific website could be accessed by a first participant, and a bridge number and PIN code could be obtained. The first participant could then be able to provide the details to other participants.

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Both of these procedures allow a mobile terminal to be involved in the conference call. However, both procedures have two main disadvantages. First, a conference call must be planned in advance. Multiple participants should contact each other to let each other know when the call is about to take place and can dial the conference bridge number at that time. Second, participants themselves need to arrange for the bridge number and PIN code to be distributed to all participants.

There have been several models proposed to provide conferencing services on Third Generation Internet Protocol Multimedia Subsystem (IMS) wireless communication systems, for example in the IETF draft SIP multiparty conferencing models, J. Rosenberg and H. Schulzrinne . Each of the templates in this draft uses Session Initiation Protocol (SIP) messages. The SIP protocol is discussed in Internet Standards RFC 3261 and RFC 2543. Some of the models are briefly outlined.

The first model, known as end-system mixing, requires that a terminal involved in a conference call mix (fuse) signals and media streams sent to and from other terminals in the call. Figure 1a is a representation of a three-way call using this model. In this example, users A and B are involved in a two-way call. At some point during the call, user A decides to involve user C in the call. To do this, user A calls user C using a completely separate SIP call. There is no call setting between B and

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C. In contrast, A receives mean currents from B and C and mixes them. Terminal A sends a current containing currents from A and B to terminal C, and a current containing currents from A and C to terminal B. In this model, terminal B and C do not realize from a SIP perspective that the call involves more than two parts.

In the case of a call involving more than three terminals, more than one terminal may mix and signal to support the call. For example, as an extension of the above example, user C may decide to invite a fourth user D to the conference call. 0 user C could then call user D and terminal C could mix the currents he receives from terminal A with his own current, and send the combined current to D, and mix his own current with that of D and send this to A. This setting is shown in Figure 1b.

Serious disadvantages of this model are that when a mix terminal leaves the call, the conference must terminate, and there is no way for a mix terminal to determine whether a signaling message sent to it was intended for that single terminal or for all terminals in the call. conference.

Another model, which uses dial-in conferencing servers, accurately reflects the above-described PSTN system. A participant defines a uniform resource identifier (URI) to identify the conference call, and sends it to other participants. Participants then each call the server using the conference URI while maintaining peer-to-peer SIP relationships.

ΕΡ1540907Β1 point with participant who calls. The server receives average from each participant, mixes it, and sends the appropriate mixed current to each participant separately. This model is shown in Figure 2, which shows 4 AD users participating in a conference call.

Dial-in conferencing servers are versatile in that they can be used for pre-arranged conferences or ad hoc conferences. However, this model suffers from the fact that it is possible for the same URI to be used for more than one conference. This could cause conference sessions to be mixed.

It is an object of the present invention to provide a solution to one or more of the above problems.

US 2002/0093948 discloses a packet-based multimedia communication system that extends IP host functionality to wireless terminals served by wireless connections. A communication system service controller manages communications services such as voice calls, video calls, web browsing, video conferencing, and / or Internet communications over a wireless packet network between source and destination host devices. A communication system multimedia content server provides access to one or more multimedia communication services. A bandwidth manager of the

<td>system</td><td>in</td><td>Communication</td><td>determines a</td><td>availability</td><td>in</td>

<td>width</td><td>in</td><td>band to</td><td>the requisitions</td><td>of service and,</td><td>if</td>

<td>width</td><td>in</td><td colspan="2">band is available,</td><td>reserve width</td><td>in</td>

enough bandwidth to support service requests. Wireless Communication Manager (s)

ΕΡ1540907Β1 manages wireless communication capabilities required to support service requests. Methods are disclosed that include the service controller that generates a call request for audio and video calling; the multimedia content server that accommodates a request for multimedia information (e.g., video replay request or web browsing); the bandwidth manager that accommodates a request for a bandwidth buffer to support a video call and audio call; two-way video call execution, video playback calls, and web browsing requests.

WO 02/21816 discloses a method of adjusting conference calling in a communications system. In the method, at least one subscriber group having two or more subscribers is established on a server connected to the communication system; Subscribers registered in the setting group talk to each other by messages in such a way that the server in the communications system receives a message from a subscriber participating in the message conversation and transmits it to other subscribers belonging to the group in question. In the method, it is also possible to go from message conversation to conference call by one of the subscribers in the subscriber group sending a predefined message to the server, which message acts as a signal to the server to go to the conference call; The server adjusts a conference call between subscribers registered as active in the subscriber group in response to receive said predefined message.

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Summary of the Invention

The present invention is as set forth in the independent claims.

According to a first aspect of the present disclosure there is provided a method for administering conference resources in a communication system comprising a plurality of terminals and a conference server, the method comprising: transmitting a first message from a first terminal to a server comprising a request for a feature capable of supporting the conference call; assigning through the server a network address that identifies a resource capable of supporting the conference call; and transmitting from the server to the first terminal a second message comprising the network address.

Advantageously, the method allows conferences to be adjusted on an ad hoc basis so conferences need not be pre-scheduled. Additionally, the method avoids the problem of overlapping conference sessions. This problem is overcome by providing a server that assigns a resource for a conference and a corresponding address for that resource. Thus, an address may be unique to a particular conference at any given time.

Another advantage of the method is that it allows the use of standard SIP messages in establishing a conference call. In addition, no significant user configuration is required in conferencing resource allocation.

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Preferably, a user transmits the network address of the first terminal to terminals of other users who will participate in the conference call. Preferably connections are initiated between the terminals and network address to establish the conference call.

According to a second aspect of the disclosure there is provided a conference server for administering conference resources in a communication system comprising a plurality of terminals, the conference server comprising: a receiving unit for receiving from the first terminal a first message which comprises a request for a feature capable of supporting a conference call; an assignment unit for assigning a network address that identifies a resource capable of supporting the conference call; and a transmission unit for transmitting to the first terminal a second message comprising the network address.

Preferably the receiving unit is arranged to receive from the first terminal a first message comprising a request for a resource capable of supporting a conference call.

Preferably the assignment unit is arranged to assign a network address that identifies a resource capable of supporting the conference call. Preferably the transmission unit is arranged to transmit to the first terminal a second message comprising the network address, server could be provided at a location.

401540907Β1 or by functionality that is distributed between two or more locations.

According to a third aspect of the disclosure there is provided a communication system comprising: a conference server for administering conference resources in a communication system comprising a plurality of terminals, the conference server comprising: a receiving unit for receiving from the first terminal a first message comprising a request for a resource capable of supporting a conference call; an assignment unit for assigning a network address that identifies a resource capable of supporting the conference call; and a transmission unit for transmitting to the first terminal a second message comprising the network address; and a plurality of terminals including the first terminal.

Brief Description of the Drawings

The present invention will now be described by way of example with reference to the accompanying drawings, in which:

Figure 1 shows an earlier technology model for a conferencing system;

Figure 2 shows a prior art dial-in conferencing system;

Figure 3 shows a conference system according to the invention.

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Detailed Description of the Invention

The invention is not described herein by reference to a non-limiting embodiment. In particular, the invention is described with respect to SIP signaling in an IMS 3G mobile communications network. However, the invention is not limited to such signaling or such a network.

Referring to Figure 3, two user agents 10 and are shown. A first user using user agent 10 wishes to start a conference call involving user agent 11 and sends a SIP INVITE 21 message to a well known URI at an operator to initiate the conference process. This URI could be stored by user agent 10. The INVITE message 21 indicates that that user agent 10 wishes to start a conference, and the URI Request could appropriately take the form:

sip: conferences@sonera.fi

The INVITE message could include details of the type of conference requested, for example a preferred data rate. These details could be contained in the media components listed in the INVITE 21 Session Description Protocol (SDP) payload.

Message 21 is received at a conferencing server which is preferably a generic user agent server. At this stage, server 12 could authenticate the conference request by transmitting a SIP message containing an authentication challenge to user agent 10 requesting details such as a username and password. In this case, the user could then provide

Tais1540907Β1 such details, ie valid authentication credentials, so that the conference request is authorized.

Either in response to message reception 21 or in response to receipt of valid user agent authentication information 10, server 12 assigns a dynamic SIP URI to be used for the requested conference. The dynamic URI identifies a feature 13 that is available to be used to support the requested conference according to the specifications listed by the first user in INVITE 21. The network is willing to route to the resource communications, or the resource providing unit, directed to that address. To facilitate this the server is preferably arranged to assign conference addresses according to a predefined standard so that they will also refer to a suitable conference resource. Server 12 may reserve this feature to remain available until the requested conference begins. Alternatively, no resources can be reserved by server 12 but instead an available resource could be located at the time when the requested conference is started. The feature is capable of establishing peer-to-peer connections to a terminal of each participant in a conference call. It can fuse the traffic signals it receives from each terminal and transmit the fused signals to the other terminals participating in the conference call. These traffic signals could carry voice data or other data such as video or graphic data (for example, whiteboarding). The feature could have data handling capability, bandwidth, or any other feature needed to support a conference call. The feature could be provided by physical equipment as a part of a server.

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Dynamic URI is transmitted to user agent 10 by conference server 12 in a SIP message 22. Message 22 is preferably a redirect message with a code in extension 3xx, and the URI is preferably contained in the message contact field. An example of how the contact field could suitably take is:

Contact: &lt;sip: DKLSKX87KKJ989SHFKJH@conference.sonera.fi&gt;

Upon receipt of the redirection message including the assigned URI, user agent 10 then transmits an INVITE 23 message to the URI. The URI identifies reserved conference resource 13, and responsive to receiving INVITE message 23, resource 13 sends an acknowledgment, such as a 200 OK message 24, to user agent 10.

When user agent 10 receives the message OK 200, the first user can then transfer the URI assigned to the second user as user agent 11. Another message, such as a SIP REFER 25 message, which includes the URI is transmitted from user agent 10 to user agent 11. REFER could adequately take the form:

REFER sip: user_b@pp.radiolinja.fi SIP / 2.0

With the following heading:

Transfer to: sip:

DKLSKX87KKJ989SHFKJH@conference.sonera.fi; Method = INVITE

Alternatively, the URI could be sent from the first user to the second user in another way.

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Through the above mechanism, a user can reserve a hurried conference resource. Without any significant input on the part of the user, other participants may be linked together to form a conference call.

In response to receiving user agent REFER message 10, user agent 11 transmits an acknowledgment, such as an accepted message 202 26, back to user agent 10.

User Agent 11 now transmits a request message, such as an INVITE 27 message, to reserved resource 13, in response to which resource 13 sends an acknowledgment, such as a message 200 OK 28, to user agent 11. Server 12 and resource 13 are able to communicate with each other. Thus, the server can obtain authentication information obtained by a user's resource so that each user can be authenticated before being allowed to join the conference. For example, it may be desirable to verify that a user has a subscription that allows them to attend conferences. Subscription details could be contained in authentication information. Alternatively, or additionally, a user may be asked to enter a PIN for transmission to resource 13 to confirm their identity for security reasons.

Following message 28, an acknowledgment, such as a NOTIFY 29 message with response code 200 OK, is sent from user agent 11 to user agent 10, and the conference can begin.

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It will be apparent that user agent 10 may also send or DOWNLOAD the dynamic URI to a number of other users to participate in the conference. Alternatively, the REFER 25 message could be directed to the conference URI. In other words, instead of transferring user B to the conference, the conference could be transmitted to user Β. The same set of messages could be used as above, but in this case external link semantics are used.

A summary of the messages required to set up a conference according to a preferred embodiment of the invention is as follows.

INVITE sip: conferences@sonera.fi

3xx redirection that includes dynamic URI

INVITE URI

200 OK

REFER TO AU 11

202 accepted

INVITE URI

200 OK

NOTIFY with 200 OK Response Code

The above mechanism can then facilitate dial-in conferencing. In the case of dial-up, the mechanism could function in essentially the same way as above except that the conference URI could be delivered to prospective participants in a different way, for example via an instant message or email, than using a REFER message. to invite them.

Applicant draws attention to the fact that the present invention may include some feature or combination of features disclosed herein either implicitly or explicitly or some generalization thereof, without limitation to the scope of any of the above definitions. In view of the foregoing description it will be apparent to a person skilled in the art that various modifications may be made within the scope of the invention as defined by the appended claims.