JP4823876B2 - Call control system and message transmission method - Google Patents

Description

  The present invention relates to a Push-to-Talk over Cellular (PoC) communication in which a push-to-talk (hereinafter referred to as “PTT”) call that is a half-duplex one-to-many communication using a mobile terminal is performed on a cellular system. The present invention relates to a call control system and a message transmission method for displaying input message information of a caller on an incoming mobile terminal before responding to the callee.

  There is a technology called “Push-to-Talk over Cellular” (PoC) that realizes a call function in which a voice is alternately exchanged by a speaker pressing a button, such as a transceiver or a commercial radio. The PTT call service based on PoC communication can be realized by a packet-switched best-effort voice service, and data and voice are multiplexed on the same packet line to represent the user's voice acquisition state while talking. It is possible to confirm presence information.

  On the other hand, a service has been proposed in which character information, which is call notification information, is added to a call signal when a mobile phone is called, so that the character information is displayed before the incoming call is answered on the called mobile phone (for example, , See Patent Document 1). Since PTT may be called even for members whose participation is not required due to the characteristics of the service, if the same service as in Patent Document 1 can be provided in the PTT service, the member on the receiving side displays it before answering the incoming call. Participation / non-participation can be determined using the character information to be determined.

In addition, in the IP telephone system, a method has been proposed in which outgoing side character information or the like is added to a outgoing signal or a call signal and displayed on a receiving terminal (see, for example, Patent Document 2). In such an IP telephone system, the agent information on the calling side is added as a SIP extension parameter to the outgoing signal or calling signal sent by the outgoing side mobile phone and transmitted, and the agent information on the receiving side is displayed as the partner information. A program for displaying a transmission image on the side and outputting sound and a simple message are included.
JP2003-101615A JP 2003-188992 A

By the way, the PoC that is based on the standard specification of the OMA (open mobile alliance), the 3GPP (3 ^rd generation partnership project) is "IMS" (IP multimedia subsystem) and is referred to as one of the applications of mobile phone multimedia service platform The IMS performs call control by SIP on an IP network for mobile phones. Also in PoC communication, it is conceivable that character information input at the time of transmission by the calling terminal is added to the SIP command as a SIP extension parameter and transmitted.

  However, the IP telephone system described in Patent Document 2 is based on the premise that the callee can display character information added as a SIP extension parameter in one-to-one SIP communication. If it is applied as it is to the PoC communication that is likely to become a member of the mobile phone, there is a problem that it becomes impossible to accurately grasp whether or not the message is displayed on the callee terminal on the caller side.

  For example, if an INVITE signal with a message is transmitted to all called terminals without considering the function of the called terminal, an error may occur in the called terminal trying to interpret the message part of the INVITE signal with a message. There is sex.

  In addition, in the case of a destination terminal that supports PTT communication but does not support message display, an error occurs when trying to interpret the message, or a provisional response signal remains undisplayed in response to the INVITE signal. May reply. When the provisional response signal is returned, there is a possibility that the calling terminal may erroneously recognize that the message is displayed at the called terminal.

  In addition, how information (delivery OK / NG) indicating whether or not the message-displayed destination terminal has displayed the message is transmitted from the destination-side mobile unit to the source-side mobile unit before the destination side response. It becomes a problem. For example, when it is added to the 180 Ringing signal that is one of the provisional response signals, there is a possibility that the delivery OK / NG is not transmitted from other than the called terminal that has returned the 180 Ringing signal first. This is because only one 180 Ringing signal needs to be output to the calling terminal, and only the first 180 Ringing signal is notified to the calling terminal. PoC defines two answer modes, a manual answer mode and an automatic answer mode. In the manual answer mode, when the receiving terminal receives an INVITE signal for session establishment, it returns a 180 Ringing signal and rings to notify the user of the incoming call. On the other hand, in the automatic answer mode, when the user sets the automatic answer mode in the terminal in advance and receives an INVITE signal for session establishment, a 200 OK is directly returned as a final response and PoC communication is automatically started. . When manual answer mode and automatic answer mode are mixed in a plurality of called terminals, the called terminal having the automatic answer mode set returns 200 OK directly without returning a 180 Ringing signal. It is impossible to add delivery OK / NG to the 180 Ringing signal.

  The present invention can display the caller's input message information to the called terminal before responding in the PTT communication, and accurately grasp whether or not the message is displayed at the called terminal at the calling terminal. An object of the present invention is to provide a call control system and a message transmission method capable of performing the above.

  The call control system of the present invention receives an INVITE signal designating a plurality of destination mobile stations as receiving terminals from a source mobile station as a transmitting terminal by a calling side SIP server having the source mobile station under control, and In a call control system in which an INVITE signal is received by the destination mobile device from a destination SIP server having a mobile device under its control to perform call connection between the source mobile device and each destination mobile device, the destination SIP server includes: When an INVITE signal with message information added to the destination mobile device is received, the corresponding function of the destination mobile device is determined, and then the INVITE signal is received by the destination mobile device. The delivery result signal indicating whether or not the message information is displayed on the destination mobile device or the destination SIP server is notified to the source mobile device via the source SIP server. The features.

  According to this configuration, since the destination SIP server determines the corresponding function of the destination mobile device and causes the INVITE signal to arrive at the destination mobile device, the destination mobile device that cannot respond to the INVITE signal with message information is received. In this case, the destination SIP server can deal with it, and even if a mobile terminal having various functions is targeted, a delivery result signal indicating whether or not message information is displayed on each destination mobile device is accurately issued and moved. Can be sent to the machine.

  Further, the present invention provides the call control system, wherein the destination SIP server holds corresponding function information indicating whether or not the subordinate mobile terminal can display the message information added to the INVITE signal on the incoming call screen, When receiving an INVITE signal to which message information is added with the receiving mobile device being a subordinate mobile terminal as the destination, it is determined whether or not the destination mobile device can display the message information based on the corresponding function information, and cannot be displayed. If it is, a delivery result signal indicating whether or not the message information is displayed on the incoming call screen of the destination mobile station is transmitted to the originating SIP server.

  With this configuration, the destination SIP server holds the corresponding function information indicating whether or not the subordinate mobile terminal can display the message information added to the INVITE signal on the incoming call screen. Whether or not can be displayed can be quickly determined.

  Also, in the call control system according to the present invention, the destination SIP server determines whether the destination mobile station can display message information based on the corresponding function information, and if it can display the INVITE signal. The incoming mobile station is made to receive a call, and it is determined whether or not the incoming mobile station displays the message information added to the INVITE signal on the incoming call screen, indicating that the message information is displayed or not displayed on the incoming call screen. The delivery result signal is transmitted, and the originating SIP server receives the delivery result signal transmitted by the destination mobile device and notifies the delivery result to the originating mobile device.

  With this configuration, an INVITE signal with message information can be received only by a destination mobile device that can display message information, and in this case, a delivery result indicating whether or not the destination mobile device has displayed the message information on the incoming screen. Since the signal is transmitted, a correct delivery result signal is notified to the mobile station.

  Further, the present invention is characterized in that, in the call control system, the originating SIP server notifies the originating mobile station of the delivery result signal and displays the delivery result while calling each of the destination mobile stations. And

  With this configuration, since the delivery result signal is notified to the calling mobile device during the calling of the called mobile device, the delivery result from each called mobile device is displayed in real time on the terminal screen of the calling mobile device. can do.

  In the call control system according to the present invention, the originating SIP server receives a delivery result signal transmitted from the destination mobile device or a delivery result signal transmitted from the destination SIP server, and receives the received delivery result signal. Is charged in the case of a delivery result indicating that the message information is displayed.

  With this configuration, additional charge is performed when the delivery result signal is a delivery result indicating that message information has been displayed, so that it is possible to accurately charge the message information displayed.

  In addition, the message transmission method of the present invention displays the incoming call screen of the called mobile terminal serving as the receiving terminal at the time of call connection of PoC (Push-to-Talk over Cellular) communication, and the INVITE signal when the calling mobile station serving as the calling terminal transmits. A message transmission method for displaying the added message information, wherein the destination SIP server having the destination mobile station receives the INVITE signal with the message information added to the destination mobile station as a destination, and A delivery result signal indicating whether or not the message information is displayed on the incoming call screen of the destination mobile device after the arrival side SIP server determines the corresponding function of the destination mobile device and causes the destination mobile device to receive the INVITE signal Is transmitted from the destination mobile device or the destination SIP server to the source mobile device via the source SIP server having the source mobile device under its control.

  Further, the present invention is characterized in that, in the message transmission method, the destination SIP server collects a corresponding function indicating whether or not the message information added to the INVITE signal from the subordinate mobile terminal can be displayed on the incoming call screen. And

  With this configuration, the destination SIP server collects the corresponding function indicating whether or not the message information added to the INVITE signal from the subordinate mobile terminal can be displayed on the incoming call screen, so that the subordinate mobile terminal changes. However, you can always grasp the correct function.

  According to the present invention, in the PoC communication, the caller's input message information can be displayed before the response to the called terminal, and the calling terminal can accurately grasp whether or not the message is displayed at the called terminal. can do.

Hereinafter, embodiments of the present invention will be described in detail with reference to the accompanying drawings.
FIG. 1 is a network configuration diagram of a mobile communication system that provides a PTT call service. The PoC terminal 1 is equipped with a module (PoC engine) for processing such as SIP call control, presence information control, floor control, voice encoding and decoder, and can perform PoC communication with other mobile terminals. It is a mobile terminal. In the present embodiment, any message information to be displayed on the incoming call screen of the called mobile terminal serving as the called terminal can be input when the PTT member is selected by the calling mobile machine serving as the calling terminal.

  The radio access network 2 is composed of a plurality of base stations (BTS), and a PoC terminal 1 and a mobile terminal that does not support PoC communication are connected to the SIP server 4 from the radio access network 2 via an RNC (Radio Network Controller) 3. It is configured to be able to.

  The SIP server 4 is a part that performs SIP call control and packet transfer control between the PoC terminals 1. In this embodiment, the SIP server 4 is GGSN (Gateway General packet radio service Support Node) / SGSN (Serving General packet radio service Support Node) 5. , CSCF (Call Session Control Function) 6, MRFC (Media Resource Function Controller) 7, MRFP (Media Resource Function Processor) 8, and HSS (Home Subscriber Server) 9 and the like. If control etc. are realizable, it is not limited to the said component.

  The GGSN / SGSN 5 is a gateway function and a wireless packet integration processing unit. The CSCF 6 is a part that executes processing such as SIP registration, SIP transfer (PoC call control), SIP application distribution, and PoC incoming request. The MRFC 7 is a part that performs processing such as SIP termination and floor-right charging control. In this embodiment, it is assumed that charging is also performed for displaying message information. The MRFP 8 is a part that performs processing such as floor management control and packet duplication control in PoC communication. The HSS 9 is a part that manages subscriber management information.

  The PoC server 11 is a server that performs authentication processing and provides a predetermined application such as provision of a network phone book, and provides members, group list management, and a network phone book as functions related to PoC communication. In this embodiment, the PoC server 11 is composed of an AS (Application Server) 12, a GLMS 13, a Web 14, and the like, but is not limited to such components. The AS 12 is an application related to PoC communication, and performs group ID management, network phone book, presence inquiry / notification processing. The GLMS 13 is a part that manages member and group lists. The Web 14 is a part that functions as a Web customer control I / F, and communicates with the SIP server 4 via protocol conversion by the WPCG 15.

  In the SIP server 4, the MRFC 7 / MRFP 8 receives a SIP request from the originating mobile device via the CSCF 6 as a user agent server, and transmits a SIP request as a user agent client to the destination mobile device. The CSCF 6 is responsible for SIP call control in the network. When receiving the SIP message sent from the PoC terminal 1, the CSCF 6 performs distribution processing to each application server based on information downloaded from the HSS 9 that manages subscriber information. . For example, an application such as authentication processing or provision of a network phone book is distributed to the AS 12.

  FIG. 2 is a network configuration diagram showing a correspondence relationship between the radio access network 2 covering a wide area and the plurality of SIP servers 4-1 to 4-n. A plurality of SIP servers 4-1 to 4-n are installed in various places, and are configured to be connected to each other via the mobile phone communication network 20. For example, when the terminal A is a calling terminal and the terminals B and C are receiving terminals, the SIP server 4-1 becomes a calling side SIP server having the calling terminal (A) under its control, and the receiving terminals (B) and (C) are under control. SIP servers 4-2 and 4-n possessed by the server are called SIP servers.

Next, the operation of the present embodiment configured as described above will be described.
FIG. 3 shows a case where a PTT call is sent from a source mobile device (UE1) serving as a PoC terminal 1 to a member, and a delivery result indicating whether or not a message is displayed on the incoming call screen at the destination mobile device (UE2) is a source mobile device ( The sequence until it is notified to UE1) is shown.

  Note that the source mobile device (UE1) is under the SIP server 4-1, the destination mobile device (UE2) is under the SIP server 4-2, and the destination mobile device (UE3) is under the SIP server 4-3. To do. For convenience of explanation, the CSCF 6 in the SIP server 4-1 is described as CSCF (1), and MRFC7 / MRFP8 is described as MRF. Similarly, the CSCF 6 in the SIP server 4-2 is described as CSCF (2), and the CSCF 6 in the SIP server 4-3 is described as CSCF (3).

  After displaying the member list or group list registered in the phone book on the originating mobile device (UE1), selecting multiple call partners from the list, and then entering the message information you want to display on the call partner's incoming screen The transmission process is performed by pressing the PTT button. Here, the message information to be input is not limited to text information, but is information that conveys the purpose of the telephone, urgency, importance, and the like.

  The calling mobile device (UE1) sets the message information as an extension parameter in the INVITE signal in the calling process (step S1), and the destinations of the plurality of selected call partners are converted into user addresses SIP URL for PTT. An INVITE signal with message information in which the user address SIP URL of the destination mobile device selected as the call partner is set as the final destination is transmitted to the base station 2.

  The INVITE signal with message information transmitted from the originating mobile device (UE1) is sent to the originating SIP server 4-1 via the radio access network 2 and the RNC 3. In the calling side SIP server 4-1, the calling side CSCF (1) receives the INVITE signal with message information, and starts transmitting a 100 Trying signal to the calling mobile device (UE1).

  On the other hand, the originating CSCF (1) transfers an INVITE signal with message information to the AS 12 of the PoC server 11 to request an authentication process. Further, when the authentication process is completed, an INVITE signal with message information is transferred to the MRF.

  In the MRF of the calling-side SIP server 4-1, the MRFC 7 receives the INVITE signal with message information, and generates an INVITE signal with message information for the called mobile station that is the call destination. At this time, in PTT which is one-to-many communication, there is a possibility that a plurality of communication partners are set as destinations as described above. In such a case, MRFC 7 generates a plurality of INVITE signals with message information. For this purpose, the MRFC 7 extracts message information from the INVITE signal with message information received from the originating CSCF (1) and stores it in the memory (step T1). One of a plurality of call destinations selected in the originating mobile device (UE1) is selected and set as the destination of the INVITE signal, and the message information stored in step T1 is read out and set as the extension parameter of the INVITE signal. (Step T2). For example, if the destination of the INVITE signal is the destination mobile device (UE2), the INVITE signal with message information for the destination mobile device (UE2) is generated. Similarly, an INVITE signal with message information is generated for all destinations (duplication). In FIG. 3, only the INVITE signal with message information for the destination mobile station (UE2) is illustrated. The INVITE signal with message information whose final destination has been specified in this way is transferred to the originating CSCF (1).

  When the originating CSCF (1) receives an INVITE signal with message information from the MRF, it transmits a 100 Trying signal to the MRF, while having a destination mobile station as the destination of the INVITE signal with message information. The INVITE signal with message information is transferred to the SIP server 4. In FIG. 3, the INVITE signal with message information is transferred to the destination SIP server 4-2 under the control of the destination mobile device (UE2).

  In the incoming side SIP server 4-2, when the incoming side CSCF (2) receives the INVITE signal with message information, the incoming side CSCF (2) transmits a 100 Trying signal to the outgoing side CSCF (1). Furthermore, the called-side CSCF (2) determines the corresponding function of the called mobile device (UE2) designated as the call destination (step R1).

Here, the corresponding function determination method of the destination mobile station at the destination CSCF 6 will be described.
Each mobile terminal 1 periodically accesses the destination CSCF 6 to update the mobile device function information. The destination CSCF 6 manages the corresponding functions of the mobile terminal with the bit string information shown in FIG. 5 (hereinafter referred to as “corresponding function table”). For example, “push talk support availability”, “message display support availability at the time of incoming PushTalk”, and “OO service support availability” are managed by 1 bit in order from the upper bit. In the figure, the case of “1” can be handled, and the case of “0” is not supported. The callee CSCF 6 prepares the corresponding function table shown in FIG. 5 for the subordinate mobile terminals, and quickly determines the corresponding function of the called mobile device by referring to the corresponding function table. Note that the corresponding function of the destination mobile device may be determined by a method other than the method using the corresponding function table shown in FIG.

  In the destination-side SIP server 4-2 in which the destination mobile device is subordinate, the destination-side CSCF (2) determines the corresponding function of the destination mobile device from the corresponding function table of the destination mobile device (UE2) (step R1). When the destination mobile device (UE2) is “capable of displaying a message when a PushTalk call is received”, an INVITE signal with a message is transmitted to the destination mobile device (UE2).

  Upon receiving the INVITE signal with a message, the terminating mobile device (UE2) interprets the INVITE signal and returns a 100 Tring signal to the terminating CSCF (2). Furthermore, the destination mobile device (UE2) confirms the setting information preset in the terminal (step Q1). Message information added to the INVITE signal with a message if the setting content “Allow message display on incoming call” is set to permit display of the caller's message information on the incoming call screen displayed on the terminal screen when an incoming call is received Is taken out and displayed on the incoming call screen (step Q2). Then, a delivery result signal “delivery result OK” indicating that the message information is displayed on the incoming call screen is transmitted to the originating SIP server 4-1 using the NOTIFY signal (step Q3).

  The MRF of the originating SIP server 4-1 is notified of a delivery result indicating whether or not message information is displayed on the incoming call screen at the destination mobile station by a NOTIFY signal from the destination mobile station (UE2). If the notified delivery result is “delivery result OK”, the MRF generates a charge for displaying the message information (step T3). In PTT, the delivery result is returned from the landing mobile devices of a large number of members, but billing occurs only once.

  Further, the MRF of the originating SIP server 4-1 transmits a NOTIFY signal with the delivery result added to the originating mobile device (UE 1).

  When the originating mobile device (UE1) receives the NOTIFY signal for notifying the delivery result from the MRF of the originating SIP server 4-1, an image indicating that it is being called is displayed on the terminal screen. The calling mobile device (UE1) recognizes the notified delivery result and displays the delivery result related to the message information on the calling terminal screen. The structure of the delivery result screen regarding many members will be described later.

  The destination mobile station (UE2) that has received the INVITE signal rings the destination mobile station after transmitting the delivery result, and returns a 180 Ringing signal to the destination CSCF (2). The terminating CSCF (2) transfers the 180 Ringing signal to the originating CSCF (1) of the originating SIP server 4-1 to notify that it is calling. Further, after being transferred to the MRF, a 180 Ringing signal is transmitted from the originating CSCF (1) to the originating mobile device (UE1). Thereby, it is displayed on the terminal screen of the calling mobile device (UE1) that the called mobile device (UE2) is being called.

  If a delivery result is added to the 180 Ringing signal transmitted from the originating CSCF (1) to the originating mobile device (UE1), only the first received delivery result may be transmitted to the originating mobile device. . When the system configuration is such that only the first one 180 Ringing signal is transmitted to the originating mobile station for a plurality of 180 Ringing signals returned from a large number of destination mobile stations, the delivery results for the remaining destination mobile stations are displayed. Cannot notify the calling mobile device.

  Therefore, in the present embodiment, a system is provided so that the delivery result from each destination mobile station or destination CSCF (2) can be individually transmitted to the source mobile station by using the NOTIFY signal without using the 180 Ringing signal. It is composed. Thereby, the delivery result of all the arrival mobile devices can be individually notified to the source mobile device.

  On the other hand, in the destination side CSCF (2) of the destination side SIP server 4-2, it is determined that the destination mobile device is “not compatible with message display at the time of incoming PushTalk” or “not compatible with PushTalk” in the corresponding function determination in Step R1. If found, the destination CSCF (2) instead transmits the delivery result to the mobile station (UE1) in the sequence indicated by the dotted line in FIG.

  In the case of a mobile terminal that does not support message display when receiving a PushTalk call and a mobile terminal that does not support PushTalk, that the receiving CSCF (2) does not display message information on the incoming call screen at the destination mobile device. The delivery result NG shown is notified to the CSCF (1) of the originating SIP server 4-1 using the NOTIFY signal. The CSCF (1) of the originating SIP server 4-1 transmits a NOTIFY signal with the delivery result NG added to the originating mobile device (UE1). Even when the destination mobile station is out of service, a NOTIFY signal of the delivery result NG is transmitted from the destination CSCF (2) of the destination SIP server to the source CSCF (1).

  As in the case of the delivery result OK, the calling mobile device (UE1) recognizes the notified delivery result and displays the delivery result for the corresponding mobile device on the terminal screen that is being called.

  By the way, if it is determined in step R1 that the destination mobile device is “push talk compatible” and “push talk incoming message cannot be displayed” in the destination CSCF (2), the INVITE signal with message information is included. Will be transferred to the destination mobile device. In this case, since the message cannot be displayed when the incoming PushTalk message is received, the delivery result is not returned from the destination mobile device. However, as described above, since the destination CSCF (2) is transmitting instead, the source mobile device ( In UE1), the delivery result can be confirmed. Further, even when the destination CSCF (2) acts as a proxy for the delivery result, since the INVITE signal is notified to the destination mobile device, the PTT call is normally performed.

  In this way, in the destination CSCF (2), the corresponding function determination of step R1 is performed, and if the destination mobile station is “not compatible with message display when receiving a push talk” or “not compatible with push talk”, the destination CSCF Since (2) returns the delivery result NG to the calling SIP server 4-1 instead of the destination mobile device, an error occurs when the message information is interpreted in the destination mobile device, or an erroneous delivery result is displayed. It is possible to prevent problems such as replying.

  FIG. 4 is a sequence diagram in the case where the destination mobile device (UE2) is set to “can display message when receiving a PushTalk call” but cannot display the message. A part of the same portion as the sequence diagram shown in FIG. 3 is omitted.

  The CSCF (2) of the incoming SIP server 4-2 determines the corresponding function in step R1, and if “message display when incoming PushTalk is available” is received, the INVITE signal with message information is received by the mobile station (UE2). Forward to.

  Here, it is assumed that the destination mobile device (UE2) is “available to display messages when a PushTalk call is received”, but has a setting that does not display message information on the incoming call screen according to the user's terminal settings. The terminal setting is stored in a memory together with other setting information, and can be changed by a terminal setting operation by the user.

The destination mobile device (UE2) receives and interprets the INVITE signal with message information, and recognizes that the message information is added. If the terminal setting is confirmed and the message information is recognized not to be displayed (step Q1), step Q2 is skipped and the message information is not displayed on the terminal screen, and the NOTIFY signal is used for the delivery result NG. the notifies the sending SIP server 4-1 the MRF (step Q3).

In the sending SIP server 4-1 the MRF, to determine the delivery result notified by NOTIFY signal, wherein no issuance message information display if delivery result NG (step T3). And it notifies to a transmission origination apparatus (UE1) that it is a delivery result NG using a NOTIFY signal.

  As described above, even when an INVITE signal with message information is transmitted to the destination mobile device that is set to not display the message information at the time of incoming call according to the terminal setting of the user, the delivery result NG is returned to the source mobile device (UE1). The delivery result can be displayed on the terminal screen.

  FIG. 6 is a diagram showing a display example of the terminal screen of the originating mobile device during PoC communication. As shown in FIG. 6, in the outgoing call terminal screen M1 immediately after the outgoing call, the list information 31 of members requesting to participate in the PTT and the status information 32 of each member's destination mobile device included in the list are associated with each other. The message information 33 (important) input by the caller is displayed at the bottom of the screen. After that, when the transition is made to the calling terminal screen M2 and the delivery result is received from the destination mobile device or the destination CSCF (2), the delivery result indicating whether or not the message information transmitted by the caller has arrived at each destination mobile device. Information 34 is displayed. In the terminal screen M2, a cross indicates a delivery result NG, and a pencil mark indicates a delivery result OK.

  FIG. 7 is a diagram showing a display example of the terminal screen of the destination mobile device during PTT communication. As shown in FIG. 7, character information “important!” 41, which is message information transmitted by the caller, is displayed on the incoming call screen. Further, the member name 43 and the response status 44 selected this time are displayed on the incoming call screen. The caller may be displayed.

1 is a network configuration diagram of a mobile communication system that provides a PTT service according to an embodiment of the present invention. Network configuration diagram showing a correspondence relationship between a radio access network and a plurality of SIP servers in the above embodiment Sequence diagram until the delivery result OK is notified after the originating mobile station sends a PTT in the above embodiment Sequence diagram until the delivery result NG is notified after the originating mobile station sends a PTT in the above embodiment Explanatory drawing of the function determination table of the destination mobile device in the above embodiment The figure which shows the example of a display of the terminal screen of the origination mobile station at the time of PTT communication The figure which shows the example of a display of the incoming call screen of the arrival mobile unit at the time of PTT communication

Explanation of symbols

1 ... PoC terminal 2 ... Radio access network 3 ... RNC
4 ... SIP server 6 ... CSCF
7 ... MRFC
8 ... MRFP
9 ... HSS
11 ... PoC server 12 ... AS (PoC)
13 ... GLMS

Claims (7)

An INVITE signal designating a plurality of destination mobile stations as receiving terminals is received by a calling SIP server having the source mobile station under control of the source mobile station, and the destination side having the destination mobile station under control. In a call control system for making a call connection between the calling mobile device and each called mobile device by receiving an INVITE signal from the SIP server to the called mobile device,
When the incoming SIP server receives the INVITE signal with message information added to the destination mobile device as a destination, the corresponding incoming mobile device is judged and then the INVITE signal is received by the destination mobile device. Notification of a delivery result signal indicating whether or not the message information is displayed on the incoming screen of the destination mobile device from the destination mobile device or the destination SIP server to the source mobile device via the source SIP server. Feature call control system.   The destination SIP server holds corresponding function information indicating whether or not the subordinate mobile terminal can display the message information added to the INVITE signal on the incoming call screen, and receives the destination mobile station that is the subordinate mobile terminal. When receiving the INVITE signal to which the message information is added, it is determined whether or not the destination mobile device can display the message information based on the corresponding function information. If the destination mobile device cannot display the message information, the incoming call screen of the destination mobile device is determined. The call control system according to claim 1, wherein a delivery result signal indicating whether or not the message information is displayed is transmitted to the originating SIP server. The destination SIP server determines whether or not the destination mobile device can display the message information based on the corresponding function information. If the destination mobile device can display the message information, the destination mobile device receives an INVITE signal.
Determining whether or not the message information added to the INVITE signal is displayed on the incoming call screen by the destination mobile device and transmitting or not displaying the message information on the incoming call screen;
The call control system according to claim 2, wherein the originating SIP server receives a delivery result signal transmitted from the destination mobile device and notifies the delivery result to the originating mobile device.   4. The originating SIP server displays the delivery result by notifying the originating mobile station of the delivery result signal while calling each of the destination mobile stations. A call control system according to claim 1.   The originating SIP server receives a delivery result signal transmitted by the destination mobile device or a delivery result signal transmitted by the destination SIP server, and a delivery result indicating that the received delivery result signal indicates message information. The call control system according to any one of claims 1 to 4, wherein additional charge is performed in the case of (1). When a call is connected for PoC (Push-to-Talk over Cellular) communication, a message is sent to display the message information added to the INVITE signal by the calling mobile device serving as the calling terminal on the incoming call screen of the called mobile device serving as the receiving terminal. A method,
The destination SIP server having the destination mobile device receives an INVITE signal with message information added to the destination mobile device as the destination, and the destination SIP server determines the corresponding function of the destination mobile device. The INVITE signal is received at the destination mobile device and a delivery result signal indicating whether or not the message information is displayed on the incoming screen of the destination mobile device is sent from the destination mobile device or the destination SIP server. A message transmission method comprising: notifying the originating mobile device via an originating SIP server having the originating mobile device under its control.   7. The message transmission method according to claim 6, wherein the destination SIP server collects a corresponding function indicating whether or not message information added to an INVITE signal from a subordinate mobile terminal can be displayed on an incoming call screen.

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