JP2005252995A - B board connection system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To realize, in an SIP network, a B board connection method that has been realized in a line switching network. <P>SOLUTION: A B board server 40 is connected to an SIP network 2. The B board server 40 comprises a means for accepting and accumulating a plurality of calls which are incoming from a PSTN 1 via a GW 31; a means for managing the number of accumulated calls and incoming call information; a means for notifying one or more B board terminals 41-4n of the number of accumulated calls, the incoming call information and the like; and a means for connecting an incoming call to a corresponding B board terminal through a responding operations of the B board terminals. The B board server operates to look like a single SIP UA terminal from arbitrary B board terminals on the existent SIP network. Furthermore, the B board server operates internally to accumulate a plurality of incoming calls to a system and to handle these calls in an arbitrary order at the one or plurality of B board terminals. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a console connection system in a SIP (Session Initiation Protocol) network.

  The relay console connection system is a well-known connection method in a circuit-switched network, and an outline thereof is shown in FIG.

  In a circuit switched network as shown in FIG. 6, when a call is received from a PSTN (Public Switched Telephone Network) via a local line or a trunk line, one or a plurality of switches are accommodated by the switch. The simultaneous call of the attendant terminal is started and the increase in the number of overdue calls is stored. When any attendant terminal answers the call, the exchange connects the incoming call to the responding attendant terminal and stores the decrease in the number of overloaded calls. At this time, if the number of overdue calls becomes 0, the simultaneous call of the attendant terminal is stopped. By this method, a plurality of incoming calls can be stuck and handled by one or a plurality of attendant terminals in any order.

  A system and method for efficiently establishing a SIP session between a calling party and a called endpoint have been proposed (see, for example, Patent Document 1).

JP 2002-335267 (first page-second page, FIG. 2)

  However, since the relay console connection method realized in the above-described circuit switched network is not defined in the SIP basic specifications, the relay console connection method is not realized in the SIP network.

  The technology described in Patent Document 1 relates to intelligent call routing in an Internet telephony system compliant with SIP, and does not relate to a relay console connection system in a SIP network.

  Therefore, an object of the present invention is to realize a relay console connection system in a SIP network.

  The invention according to claim 1 is a relay that relays a terminal in the PSTN (1 in FIG. 1) and a relay terminal (41, 42... 4n in FIG. 1) connected to the SIP network (2 in FIG. 1). In the stand connection system, the relay stand connection system is characterized in that a relay stand server (40 in FIG. 1) having the following means is connected to the SIP network.

  The attendant server accepts an incoming call in the SIP network and receives a call overload (401 in FIG. 2), analyzes each SIP message received by the call acceptance means, and receives the call information and product of each incoming call. Call information management means (402 in FIG. 2) for managing the number of missed calls, and for each incoming call accepted by the call accepting means, the call information management means and the attendant terminal according to the call information managed by the call information management means and the number of stuck calls A relay terminal control means (403 in FIG. 2) for transmitting and receiving messages between them and controlling the relay terminal, and when any relay terminal responds to an incoming call, the call and the relay terminal And a call control means (404 in FIG. 2) for performing a call control operation for disconnecting the connection between the call and the attendant terminal when the call ends.

  The invention according to claim 2 is a relay connection method for relaying a terminal in the PSTN and a relay base terminal connected to the SIP network, wherein the relay base server is connected to the SIP network, and an arbitrary one in the PSTN When a call is made from the terminal to the attendant server, an INVITE message and various SIP responses defined in the SIP basic specifications are transmitted / received to / from the gateway via the SIP server, and a call arrives at the attendant server ( In S1) of FIG. 4, the attendant server stores an increase in the number of overdue calls and incoming call information by the arrival of a call, and notifies the attendant terminal of the number of overdue calls and the incoming call information (FIG. 4). S2) and a step of transmitting response information such as which call is to be answered to the relay console server from the relay console terminal that has performed the response operation (S3 in FIG. 4) When the attendant server transmits / receives the SIP message via the SIP server according to the response information (S4 in FIG. 4, S4 ′ in FIG. 5), the attendant server reduces the stored number of overdue calls. The call information of the incoming call that has been stored and answered is deleted, and notification to that effect is sent to each attendant terminal, and if necessary, the ringing of the attendant terminal is stopped and the number of overdue calls and the display of the incoming call information are updated. In step (S5 in FIG. 4) and in accordance with the description in the SIP message, a stage in which a call is made by transmitting / receiving media information such as voice and video between the responding console terminal and the gateway (S6 in FIG. 4). 5 (S6 ′ in FIG. 5).

  According to the present invention, an attendant server and one or more attendant terminals are newly added to an existing SIP network, and the plural terminals cooperate to provide a single SIP UA terminal as viewed from the existing SIP network. In addition, since it operates so as to realize the connection to the attendant console inside, it is possible to realize an attendant connection system that is not defined in the SIP basic specification without changing the existing SIP network. be able to.

  Furthermore, if a method of transmitting / receiving media information via the attendant server is adopted, it becomes possible to centrally manage calls handled by a plurality of attendant terminals, so that new services can be easily provided in the SIP network. The effect that it can do is also acquired.

  The relay console connection system of the present invention is a relay console connection system that relays a terminal in a PSTN and a relay console terminal connected to a SIP network. Call receiving means, call information managing means for analyzing each SIP message received by the call receiving means and managing the call information and the number of overdue calls for each incoming call, and call information for each incoming call received by the call receiving means An attendant terminal control means for controlling the attendant terminal by transmitting and receiving messages to and from the attendant terminal according to the call information and the number of overdue calls managed by the managing means, and an arbitrary attendant terminal for an incoming call A call control that performs a call control operation that connects the call and the attendant terminal when the answering operation is performed and disconnects the connection between the call and the attendant terminal when the call ends. It is obtained by connecting the relay base server and means.

  Referring to FIG. 1, the relay console connection system of the present invention includes PSTN 1, SIP network 2, SIP server 30, GW (SIP gateway device) 31, relay console server 40, and n relay console terminals 41, 42,. It is configured. .., 4n will be referred to as a relay terminal 4X.

  Note that the relay board server 40 and the relay board terminal 4X may be physically separate devices, or may be combined into one device. As an alternative to PSTN1, H. Other networks such as a H.323 network and an MGCP network may exist, or various SIP UA (user agent) terminals such as a SIP telephone terminal may exist instead of the GW 31.

  As shown in FIG. 2, the attendant server 40 includes call accepting means 401, call information managing means 402, attendant terminal control means 403, and call control means 404.

  The call reception unit 401 receives an SIP message transmitted from an existing SIP UA terminal such as the GW 31 or the SIP server 30 and receives an incoming call and is stuck.

  The call information management unit 402 analyzes each SIP message received by the call reception unit 401 and manages the call information and the number of overdue calls for each incoming call. This call information includes, for example, caller information such as SIP URI of the caller terminal, media negotiation information attached to the SIP message, and the like.

  The attendant terminal control means 403 transmits / receives a message to / from the attendant terminal 4X for each incoming call accepted by the call accepting means 401 in accordance with the call information managed by the call information managing means 402 and the number of overdue calls. Control each relay terminal 4X.

The call control unit 404 connects the call and the attendant terminal 4X when an arbitrary attendant terminal 4X responds to an incoming call, or connects the call and the attendant terminal 4X when the call ends.
The call control operation that disconnects the connection with is performed.

  As shown in FIG. 3, the attendant terminal 4 </ b> X includes a call information display unit 411, a calling unit 412, a call response unit 413, and a call unit 414. An input device 41 a such as a microphone or a camera and an output device 41 b such as a speaker or a monitor are connected to the call means 414.

  The call information display means 411 receives the message transmitted / received to / from the relay console server 40 by the relay console terminal control means 403 of the relay console server 40, and displays the call information regarding each call on the relay console terminal 4X. For example, a ringing tone at the time of an incoming call, display of caller information during a call, and the like are performed.

  The calling means 412 receives a call operation on the attendant terminal 4X and performs a call operation toward an arbitrary partner terminal.

  The call response means 413 receives a response operation on the attendant terminal 4X, notifies the attendant server 40 of a response to an arbitrary incoming call, requests call control by the attendant server 40, or prior to a call. Send and receive messages such as media negotiation information.

  The call means 414 transmits media information such as voice and video input from the input device 41a to the counterpart terminal in accordance with the media negotiation information exchanged by the call response means 413, or conversely, those media received from the counterpart terminal. Information is output to the output device 41b.

  Next, the relay stand connection method of the present embodiment configured as described above will be described with reference to FIG.

  First, when a call is made from an arbitrary terminal in the PSTN 1 to the relay console server 40, in the SIP network 2, between the GW 31 and the relay console server 40, either directly or via the SIP server 30, the SIP basic specifications are used. The specified INVITE message and various SIP responses are transmitted / received, and the call arrives at the relay console server 40 (step S1 in FIG. 4).

  When the call arrives, the attendant server 40 stores the increase in the number of overdue calls and the incoming call information, and notifies the attendant terminals 41, 42,... 4n of the number of overdue calls and the incoming call information (step S2). ). In response to the notification, each relay terminal 41, 42,..., 4n starts ringing the terminal as necessary, or displays the number of overdue calls and incoming call information.

  Subsequently, when a response operation is performed at an arbitrary relay console terminal 4X (referred to as the relay console terminal 41 in FIG. 4), response information such as which call the relay console terminal 41 responds to the relay console server 40. Is transmitted (step S3).

  The attendant server 40 transmits / receives an OK message, an ACK message, and the like directly or via the SIP server 30 according to the response information. In these SIP messages, it is described that SIP messages are transmitted and received between the GW 31 and the relay console server 40, and media information such as voice and video is transmitted and received between the GW 31 and the corresponding relay console terminal. (Step S4).

  At the same time, the attendant server 40 stores the stored decrease in the number of overdue calls and deletes the call information of the answering incoming call. Then, the attendant terminals 41, 42... 4n are notified to that effect, and if necessary, the ringing of the attendant terminals 41, 42... 4n is stopped and the number of overdue calls and the display of incoming call information are updated. (Step S5).

  Finally, according to the call control in step S4, a call is performed by directly transmitting and receiving media information such as voice and video between the responding console terminal 41 and the GW 31 (step S6).

  In the case of subsequent hold, transfer, end of call, etc., the call control is performed by the relay console server 40 transmitting / receiving the SIP message instead of the relay console terminal 41 in the same manner as the operation of step S4 at the time of response. The media information is directly transmitted and received between the relay terminal 41 and the communication partner. When a new call is received, the operations in steps S1 to S6 described above are repeated to handle a plurality of calls.

  In this attendant connection method, as described above, the attendant server 40 accumulates an incoming call instead of the attendant terminal 4X, and broadcasts the number of overloaded calls, call information, etc. to each attendant terminal 4X. Therefore, when viewed from an existing SIP UA terminal such as the GW 31, it is possible to behave so that call control is performed with a single device (the attendant server 40).

  Further, since the actual call (transmission / reception of media information) is performed with any relay terminal 41 that has performed a response operation among the plurality of relay terminal 4X, the SIP server 30 is not burdened. Relay board connection can be realized without affecting the SIP network 2.

  Next, a second relay stand connection method according to the present embodiment will be described with reference to FIG.

  First, when a call is made from an arbitrary terminal in the PSTN 1 toward the attendant server 40, the incoming call is overloaded and an arbitrary attendant terminal 4X (FIG. 5) is used as in steps S1 to S3 in the first method described above. In this case, a response is made by the relay terminal 41).

  The relay console server 40 transmits / receives an OK message, an ACK message, and the like directly or via the SIP server 30 according to the response information. In these SIP messages, unlike the first method, it is described that both the transmission and reception of SIP messages and the transmission and reception of media information such as audio and video are performed via the relay server 40 (steps). S4 ′).

  At the same time, the attendant server 40 stores the stored decrease in the number of overdue calls and deletes the call information of the answering incoming call. The relay terminal 41, 42,... 4n is notified to that effect, and the ringing of each relay terminal 41, 42,... 4n is stopped or the number of overdue calls and the display of incoming call information are updated as necessary. (Step S5).

Finally, according to the call control in step S4 ′, a call is performed by transmitting and receiving media information such as voice and video between the responding relay terminal 41 and the GW 31 via the relay server 40 (step S4 ′). S6 ')
In the case of subsequent hold, transfer, end of call, etc., the call control is performed by the relay console server 40 transmitting / receiving the SIP message instead of the relay console terminal 41 in the same manner as the operation of step S4 ′ at the time of response. The media information is also transmitted / received via the relay board server 40. When a new call is received, the operations in steps S1 to S3, S4 ′, S5, and S6 ′ described above are repeated to handle a plurality of calls.

  In the second attendant connection method, as described above, not only the SIP message but also the media information is configured to be transmitted / received via the attendant server 40. Therefore, the effect of the first attendant connection method described above is achieved. In addition, the relay server 40 centrally manages the media information transmitted and received by the relay console terminal 4X, so that it is possible to easily provide a new service. For example, it is possible to easily provide a conference call service by mixing audio and video, a recording / recording service by storing audio and video, and the like.

The block diagram which shows one Example of the console connection system of this invention The block diagram which shows the structure of the attendant server of FIG. The block diagram which shows the structure of the attendant terminal of FIG. The figure which shows the 1st attendant connection method of this invention The figure which shows the 2nd attendant connection method of this invention Block diagram showing a conventional attendant connection system

Explanation of symbols

1 PSTN (Public Switched Telephone Network)
2 SIP network 30 SIP server 31 GW (SIP gateway device)
40 attendant server 41 to 4n attendant terminal 401 call accepting means 402 call information managing means 403 attendant terminal control means 404 call controlling means 411 call information displaying means 412 calling means 413 call answering means 414 talking means 41a input device 41b output device

Claims (4)

In a relay console connection system that relays a terminal in the PSTN and a relay console terminal connected to the SIP network, the SIP network includes:
Call accepting means for accepting incoming calls in the SIP network and stagnating;
Call information management means for analyzing each SIP message received by the call acceptance means and managing the call information of each incoming call and the number of overdue calls;
For each incoming call accepted by the call acceptance means, a relay terminal that controls the relay terminal by transmitting and receiving messages to and from the relay terminal in accordance with the call information managed by the call information management means and the number of overdue calls Control means;
When any attendant terminal responds to an incoming call, a call control operation is performed to connect the call and the attendant terminal and disconnect the connection between the call and the attendant terminal when the call ends. An attendant connection system characterized by connecting an attendant server having call control means for performing connection. A relay base connection method for relaying a terminal in a PSTN and a relay base terminal connected to a SIP network, wherein a relay base server is connected to the SIP network,
When a call is made from any terminal in the PSTN to the attendant server, an INVITE message and various SIP responses defined by the basic SIP specifications are transmitted to and received from the gateway via the SIP server, and the attendant server is called. The stage of incoming calls,
The attendant server stores an increase in the number of overdue calls and incoming call information when a call is received, and notifies the attendant terminal of the number of overdue calls and the incoming call information, respectively,
A step of transmitting response information such as which call is answered from the attendant terminal that has performed the answering operation to the attendant server;
The attendant server transmits and receives a SIP message via the SIP server according to the response information;
The attendant server stores the stored decrease in the number of overdue calls, deletes the call information of the received incoming call, notifies the attendant terminal to that effect, and rings the attendant terminal as necessary. To stop or update the display of the number of overdue calls and incoming call information,
A relay console connection method comprising the step of performing a call by transmitting and receiving media information such as voice and video between the responding console terminal and the gateway according to the description in the SIP message. 3. The relay console connection method according to claim 2, wherein transmission / reception of the media information is directly performed between the gateway and the corresponding relay console terminal according to the description in the SIP message. 3. The attendant connection method according to claim 2, wherein transmission / reception of the media information is performed between the gateway and the corresponding attendant terminal via the attendant server according to the description in the SIP message.

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