KR100625123B1 - Method and System for Implementing of Push-to-Talk Service by Using SIP and SDB for Use in Mobile Data Communication Network - Google Patents

Abstract

The present invention provides a PTT service method using a SIP and SDB to efficiently provide a PTT service by reducing the call setup time required for PTT transmission and PTT reception using the SDB in the SIP-based PTT service system of the CDMA 2000 mobile communication data network; It's about the system.

In order to realize the present invention, the present invention calls a PTT terminal for incoming / outgoing calls, a SIP server for the conversion and processing of messages, a mobile communication network connecting the PTT terminal and the SIP, a PDSN, an MRC for copying the caller's voice, and a PTT member to speak. It includes a PTT server to manage and a presence server for displaying the status change information of the PTT service subscriber.

According to the present invention, in the SIP-based PTT service system of the CDMA 2000 mobile data network, the SDB is used to reduce the traffic generated in the network by reducing the call setup time required for PTT transmission and PTT reception, as well as PTT service. It is possible to provide a satisfactory service to PTT service subscribers by providing certainty of promptness and immediate connectivity.

PTT, SIP, SDP, group call, selective call, presence, MRC, call setup method, binary message

Description

Method and System for Implementing of Push-to-Talk Service by Using SIP and SDB for Use in Mobile Data Communication Network}

1 is a configuration diagram briefly showing a PTT service providing system using SIP and SDB according to an embodiment of the present invention;

2 is an exemplary view of a terminal screen receiving a PTT service using SIP and SDB according to an embodiment of the present invention;

3 is a structural diagram showing a structure of a binary message transmitted for a selective call in a PTT service using SIP and SDB according to an embodiment of the present invention;

4 is a structural diagram showing a structure of a binary message transmitted for a group call in a PTT service using SIP and SDB according to an embodiment of the present invention;

5 is a flowchart illustrating a call setup method between a calling terminal and a PTT server in a selective call in a PTT service using SIP and SDB according to a preferred embodiment of the present invention.

6 is a flowchart illustrating a call setup method between a PTT server and a receiving terminal in a selective call in a PTT service using SIP and SDB according to an embodiment of the present invention;

7 is a flowchart illustrating a call setup method between a calling terminal and a PTT server in a group call in a PTT service using SIP and SDB according to a preferred embodiment of the present invention.

8 is a flowchart illustrating a call setup method between a PTT server and a receiving terminal in a group call in a PTT service using SIP and SDB according to an exemplary embodiment of the present invention.

<Description of Symbols for Main Parts of Drawings>

110: calling terminal 120, 190: mobile communication network

122, 192: base station transmitter 124, 194: base station controller

130, 180: PDSN 140: SIP server

150: presence server 160: PTT server

170: MRC 200: receiving terminal

202: user status display unit 204: registrant status display unit

206: Outgoing button 208: Ending button

210: call type display unit 212: call participant list display unit

214: menu button 216: message window

218: message transmission button 302: message type display unit

304: direction display unit 306: method display unit

308: ID display unit 310: margin insertion unit

312: caller phone number display unit 314: receiver phone number display unit

316: From_Tag display unit 318: To_Tag display unit

320: Via_Branch display unit 322: Call-ID display unit

402: margin unit 404: group ID display unit

The present invention relates to a method and system for implementing a PTT service using SIP and SDB in a mobile communication data network. More specifically, in the SIP-based PTT service system of the CDMA 2000 mobile data network, PTT service using SIP and SDB that efficiently provides PTT service by reducing call setup time for PTT origination and PTT reception using SDB It relates to a method and a system.

BACKGROUND With the rapid development of computer, electronic and communication technologies, various wireless communication services using a wireless telecommunication network have been provided. The most basic wireless communication service is a wireless voice call service that provides voice calls wirelessly between users of communication terminals regardless of time and place.

Among the various types of wireless voice call services, the common carrier radio frequency (TRS: Trunked Radio System, hereinafter referred to as 'TRS') supports one-to-many group communication by eliminating the existing one-to-one wireless communication. It is effectively and conveniently used in military and government agencies. In addition, the TRS has been used in various ways because it is possible to talk immediately and quickly by pressing a button provided on the TRS terminal without going through a dial operation.

In addition, with the development of a wireless communication network, a wireless Internet service has emerged, in which a user of a communication terminal provides a communication service similar to a service provided by the Internet through a wireless communication network. Accordingly, the wireless communication subscriber can receive information necessary for life, such as news, weather, sports, securities, exchange rate, traffic information, etc. in various forms such as text, voice, image, and video through wireless Internet service. have. In particular, the instant messenger service, which has been recently implemented on a wired internet network, that is, a web application, is also provided through a wireless communication network. IM (Instant Messenger, hereinafter referred to as 'IM') is a service that provides convenience to users in that they can make instant communication with the status information of friends and organization members registered on the messenger. It also includes the features of group calls, where multiple members can talk simultaneously.

Recently, a push-to-talk (PTT) service, which combines TRS and IM service, which has characteristics of immediate connectivity, speed, and group call, has emerged. PTT service has the advantage of low installation cost because it can utilize mobile communication data networks such as CDMA 1x network and EV-DO network without establishing a new dedicated network.

In addition, the TRS is limited in the group member setting and the presence information can not be confirmed, but the PTT has an advantage that the presence information can be viewed visually because the IM technology is also incorporated. In addition, IM is inconvenient to input text-type characters for simultaneous conversations of several members, but PTT can be easily used in wireless messenger service compared to IM because PTT can communicate using voice in addition to text.

However, the current PTT service takes 8 to 15 seconds on average for a call and an average of 2 seconds or more when talking to each other. Many complaints from subscribers have been raised.

In order to solve the above problems, the present invention, SIP in the SIP-based PTT service system of the CDMA 2000 mobile data network using SDB to reduce the call setup time required for PTT transmission and PTT incoming SIP to efficiently provide a PTT service And to provide a PTT service method and system using the SDB.

In order to achieve the above object, the present invention is connected to a plurality of users in a mobile communication data network, a PTT service that provides a push-to-talk (PTT) call in which a conversation or a message is transmitted between the users immediately and quickly A system, comprising: a PTT server for calling a member for a PTT call and managing a voice in the PTT call member; A presence server that manages the presence information of the PTT service subscriber and a list of set groups, and displays status change information of the PTT service subscriber; MRC (Media Rdsource Controller) for copying the talker's voice and forwarding it to other members of the PTT call; SIP server for message processing and conversion delivered from the PTT service system using the SIP protocol; A mobile communication network which connects a SIP server and a PTT call member to deliver messages and data; And a packet data service node (PDSN) for processing packet data between the mobile communication network and the SIP server.

In addition, according to a second object of the present invention, a calling terminal, a mobile communication network, a Packet Data Service Node (PDSN), a Session Initiation Protocol (SIP) server, a presence server, a push-to-talk server, A call setup method for a selective call in a PTT service system using a SIP and SDB including a media resource controller (MRC) and a receiving terminal, the method comprising: (a) an initial call setup request signal transmitted from an originating terminal using an SDB; step; (b) requesting the MRC to occupy a media resource for the calling terminal; (c) transmitting subscriber information of the selected call member from the calling terminal; (d) checking status information of the selected call member using the presence server; (e) transmitting a call signal to a receiving terminal using the SDB; (f) requesting the MRC to occupy Media resources for the receiving terminal; And (g) notifying the calling terminal of the fact that the called terminal participates in the selective call.

In addition, according to a third object of the present invention, a calling terminal, a mobile communication network, a Packet Data Service Node (PDSN), a Session Initiation Protocol (SIP) server, a presence server, a push-to-talk server, A call setup method for a group call in a PTT service system using a SIP and SDB including a Media Resource Controller (MRC) and a receiving terminal, the method comprising: (a) a PTT group call request signal using an SDB is transmitted from an originating terminal; step; (b) checking a list and status information of the group call members using the presence server; (c) requesting the MRC to occupy a media resource for the calling terminal; (d) transmitting a call signal from the PTT server to the receiving terminal using the SDB; (e) requesting the MRC to occupy a media resource for a receiving terminal; And (f) notifying the calling terminal of the fact that the called terminal participates in the group call.

According to a fourth object of the present invention, a binary message is used as an indicator structure of a binary message used in a PTT service system in which a selective call and a group call are made using a Session Initiation Protocol (SIP) and a Short Data Burst (SDB). A message type display unit indicating that the message has a binary structure in the SIP protocol stack; A direction indicator which displays a transmission direction of a binary message; Method display unit for displaying the nature of the binary message; And an ID display unit for discriminating whether a binary message is used in a selective call or a group call.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings. First of all, in adding reference numerals to the components of each drawing, it should be noted that the same reference numerals are used as much as possible even if displayed on different drawings. In addition, in describing the present invention, when it is determined that the detailed description of the related well-known configuration or function may obscure the gist of the present invention, the detailed description thereof will be omitted.

1 is a configuration diagram briefly showing a PTT service providing system using SIP and SDB according to an embodiment of the present invention.

In the PTT service providing system using the SIP and the SDB according to the present invention, the originating terminal 110, the mobile communication network (120, 190), the packet data service node (130, 180), the SIP server 140, the presence server 150 , The PTT server 160, the MRC 170, and the receiving terminal 200 are configured.

The originating terminal 110 is a terminal capable of making a call through the mobile communication network 120 and automatically performing a session establishment procedure and transmitting packet data according to a session establishment request from the mobile communication network 120. The originating terminal 110 transmits a PTT request for call setup to the PTT server 160 using a short data burst (SDB) of an access channel. The access channel is a channel used to transmit an origination signal and a registration signal. The access channel has a low data rate for the stability of the radio channel. Here, SDB refers to an IP packet included in a DBM field of an access channel and transmitted to a data network.

In addition, the calling terminal 110 should be a PTT terminal for receiving a PTT service, and a SIP terminal for using a SIP method.

The mobile communication network 120, 190 includes a base station transmitter 122, 192 and a base station controller 124, 194. The mobile communication network 120, 190 transmits an outgoing request signal from the originating terminal 110 to the SIP server 140. It receives a message and data from the server 140 and transmits it to the called terminal 200.

A packet data service node (PDSN) (hereinafter referred to as PDSN) (130, 180) processes the packet data between the mobile communication network (120, 190) and the SIP server 140, the originating terminal 110 And the receiving terminal 200 enables the connection to the PTT server 160 via the SIP server 140. When the user receives data while moving, the PDSN (130, 180) provides a mobility management function so that the mobile terminal (110, 200) can be provided with the Internet data service without changing the IP address.

The Session Initiation Protocol (SIP) server 140 performs a function of providing a mobile communication service such as audio, video, etc. using the SIP protocol. Here, SIP is a simple text-based application layer control protocol, which is a signaling protocol for Internet video conferencing, Internet telephony, and the like. SIP is a client / server protocol, unlike H.323. In addition, the SIP server 140 processes a standard SIP message and a binary SIP message received from the terminal 110 or 200 or the PTT server 160 to convert each other. In addition, when an incoming or outgoing call is made using a binary SIP message, a 200 OK signal or an ACK is transmitted to the terminal 110 or 200 or the PTT server 160.

The presence server 150 is a server that manages the presence information of PTT service subscribers and a list of friends and groups. The presence information indicates a change in the status of the PTT service subscriber, and the PTT subscriber may display his / her status as offline, online, busy, away, in a meeting, or while eating. In addition, the PTT service subscriber may access the presence server 150 and check the status information of another subscriber using the calling terminal 110 or the called terminal 200.

The PTT server 160 manages PTT group sessions and manages voice in group members. In addition, it may be connected to a PTT database (not shown) for managing group member information or may include a PTT database therein. When a selective call or a group call request signal from the calling terminal 110 is transmitted to the PTT server 160, the PTT server 160 calls a corresponding call member and receives a response signal from each member to receive the calling terminal ( 110) to inform the result of participation of the member.

The media resource controller (MRC) 170 is responsible for copying the caller's voice and transmitting it to other members of the PTT call during the selective or group call. In order to transmit the voice data of the caller transmitted to the PTT server 160 to each member, that is, the caller, the voice data of the number of members participating in the PTT call must be duplicated. Therefore, the voice data of the caller transmitted to the PTT server 160 is replicated in the MRC 170 and transmitted to each called party through the PTT server 160 again.

Like the calling terminal 110, the receiving terminal 200 can make a call through the mobile communication network 190, and automatically performs a session setting procedure and transmits packet data according to a session setting request from the mobile communication network 190. Is a PTT terminal and a SIP terminal.

The PTT service system using the SIP and the SDB configured as described above provides a selective call in which a call is made with some members by a user's selection among the members, and a group call in which a call is made with the whole member.

2 is an exemplary view of a terminal screen receiving a PTT service using SIP and SDB according to an embodiment of the present invention.

2A is a terminal screen in a standby state, and includes a user status display unit 202, a registrant status display unit 204, a call button 206, and an end button 208.

The user status display unit 202 is a part for setting the user's own status to notify other users of his / her status. The user status display unit 202 may be set to offline, online, busy, away, meeting, meal, or the like.

The registrant status display unit 204 is a portion in which a list and other current statuses of registered users are displayed. Registered users can be displayed with various icons according to their status, and in front of each user, there is a rectangular box to designate a corresponding user when a user wants to make a call, send a message, or check information. In addition, it is possible to list registered users in various ways such as by group, by on / offline status, and the like.

The call button 206 is a button for calling with one or more users registered in the registrant status display unit 204. If one or more users want to call and inputs the call button 206, the call is requested to the designated user. The signal is sent.

The end button 208 is a button used when you do not want any more PTT service to terminate.

2B is a terminal screen in a call state, and is a screen used for group call and screening call. The terminal screen in the call state includes a call type display unit 210, a call participant list display unit 212, a menu button 214, a message window 216, and a message transmission button 218.

The call type display unit 210 automatically displays whether the corresponding call is a selected call or a group call. When the user designates a registered group on the screen of 2A, the call type display unit 210 is a group call. If a registered user is selected, the selected call is displayed. If the call type is a group call, the corresponding group is displayed.

The call participant list display unit 212 displays a user who is currently participating in a screening call or a group call. In addition, as shown in the registrant status display unit 204, it is also possible to display the status of the call participant, and the user who is displayed offline may not be displayed.

The menu button 214 is an invitation button for inviting other users who are not in the call, an expulsion button for expelling a user who is present in the call but does not need to participate anymore, and information on the user who is in the call. And a confirmation button for confirming information and an end button for ending a call.

In PTT service, text service is available as well as voice call. Accordingly, a message window 216 for inputting a character to be transmitted or confirming the received character is also required on the terminal screen. In addition, a message transmission button 218 for transmitting an input message is also provided.

2A and 2B are merely one exemplary screen for explaining a screen for providing a PTT service according to the present invention, and may be modified as much as possible in implementing the present invention.

3 is a structural diagram illustrating a structure of a binary message transmitted for a selective call in a PTT service using SIP and SDB according to a preferred embodiment of the present invention.

According to an embodiment of the present invention, a binary message used for a selective call among PTT services using SIP and SDB is a binary invite message and a binary 200 ok signal. The binary invite is classified into a message transmitted from the terminals 110 and 200 to the SIP server 140 and a message transmitted from the SIP server 140 to the terminals 110 and 200.

An indicator structure for distinguishing types of binary messages according to the present invention is shown in FIG. 3A. In this case, the indicator structure is a part which is located at the front of the message to explain the characteristics of the message. The indicator according to the present invention is composed of 1 byte, that is, 8 bits.

The binary message indicator (hereinafter, referred to as 'BMI') according to the present invention includes a message type display unit 302, a direction display unit 304, a method display unit 306, an ID display unit 308, and a margin insertion unit 310. It is composed of

The message type display unit 302 is the first 4 bits of the binary message, and is indicated by '1110' according to the present invention. This indicates that this message is a binary structured message in the SIP protocol stack.

The direction indicator 304 is denoted by 'D', which means a direction. The direction display unit 304 displays a signal transmission direction between the terminals 110 and 200 and the SIP server 140. When the signal is transmitted from the terminals 110 and 200 to the SIP server 140, the direction indicator 304 is displayed as '0'. When a signal is transmitted from the SIP server 140 to the terminals 110 and 200, it is displayed as '1'.

The method display unit 306 is indicated by 'M', which means Method. The method display unit 306 displays the nature of the message. If the message is an Invite message, it is displayed as '0', and if the response signal is 200 OK signal, it is displayed as '1'.

 ID display unit 308 is denoted by 'B', which means 'Buddy ID'. The ID display unit 308 indicates whether the message is a message used in a selected call or a group call, and is displayed as '0' when used in a group call, and when used in a selected call. It is displayed as '1'.

The margin inserter 310 is denoted by 'F', which is a 'Padding Bit'. Padding Bit is a bit that fills the empty space, and can be used as either '0' or '1'. Here, the margin inserter 310 may be filled with identifiers of other conditions for distinguishing binary messages according to a setting.

3B is a structure of Binary Invite, 1) shows the structure of Binary Invite message transmitted from the terminals 110 and 200 to the SIP server 140, and 2) is from the SIP server 140 to the terminals 110 and 200. Represents the structure of a binary Invite message.

Binary Invite is composed of BMI, caller's phone number display 312, recipient's phone number display 314, From_Tag display 316, To_Tag display 318, Via_Branch display 320 and Call-ID display 322. .

The BMI unit is set to '1110001F' in 1) by the method described in 3A above. This is an Invite message transmitted from the terminals 110 and 200 to the SIP server 140, which has a binary structure, and is used for a selective call.

The caller phone number display unit 312 is a part for displaying the caller's phone number, and is composed of 4.5 bytes, or 36 bits. Since 4 bits are required to display a single digit phone number, the Binary Invite message according to the present invention can display a 9 digit phone number. Currently, the number of digits of a communication terminal telephone number commonly used in Korea is 11 digits, and in the present invention, in consideration of being composed of '01x-xxxx-xxxx', the present invention is composed of 9 digits except the preceding two digits '01'. However, in implementing the present invention, it is possible to adjust the number of bits of the calling party telephone number display 312.

The receiver phone number display unit 314 is also made up of 36 bits in the same manner as the caller phone number display unit 312 and configured to display a 9-digit phone number. The receiver phone number display unit 314 may also adjust the number of bits in implementing the invention.

The From_Tag display unit 316 is composed of 4 bits, and is a part for displaying index information values transmitted from the terminals 110 and 200 through the SDB. That is, the tag transmits the index information in the terminals 110 and 200 to the SIP server 140, and when receiving the 200 OK signal from the SIP server 140, the tag receives the corresponding information and uses the tag information used in creating the session. Find and use Here, Tag is a key value having terminal information and is an identifier for distinguishing a request generated from the same user group.

The To_Tag display unit 318 is also composed of 4 bits and is a part for displaying index information values transmitted to the terminals 110 and 200.

Via_Branch display unit 320 is configured in the same way as the From_Tag display unit 316, and puts the branch information of the terminals (110, 200) in the SDB, and receives the corresponding information when receiving the 200 OK signal to find the branch information used when creating a session use. Here, Branch is an identifier for distinguishing the movement history of SIP.

The Call-ID display unit 322 is composed of 8 bits and is a part for displaying a call ID. Here, the call ID refers to an identifier for displaying the terminals 110 and 200 and generates call IDs by two methods. One is a method of continuously using the call ID generated by the SIP server 140 afterwards, and one is a method of arbitrarily creating and using ID based on the calling number.

The BMI part of 3B to 2) is set to '1110101F' by the method described above, and is an Invite message transmitted from the SIP server 140 to the terminals 110 and 200 in a binary structure, indicating that it is used for a selective call. do. Parts other than the BMI part are the same as the rest of 1) transmitted from the terminals 110 and 200 to the SIP server 140.

3C is a Binary 200 OK signal, and the BMI unit is set to '1110011F' to indicate that the 200 OK signal transmitted from the terminals 110 and 200 having the binary structure to the SIP server 140 is used for a selective call. .

4 is a structural diagram illustrating a structure of a binary message transmitted for a group call in a PTT service using SIP and SDB according to an embodiment of the present invention.

The binary message structure used for group calls is similar to the binary message structure used for select calls.

4A shows the Binary Invite message structure used in group currency. Here, 1) shows the structure of the message transmitted from the terminals 110 and 200 to the SIP server 140, and 2) shows the structure of the message transmitted from the SIP server 140 to the terminals 110 and 200.

In 1), the BMI unit is denoted by '1110000F', which indicates that the BMI unit is an Invite signal transmitted from the terminals 110 and 200 to the SIP server 140 having a binary structure and is used for a group call. Behind the BMI is a caller's telephone number display section 212, which occupies 36 bits.

After the caller's telephone number display section 312, a four-bit margin 402 consisting of 'FFFF' is located. The margin portion 402 is represented by 'F' in the same manner as the margin insertion portion 310 described above. This part may be used as either a '0' or a '1' as a margin bit for classifying the caller's telephone number display unit 312 and the group ID in the binary invite message. Here, the margin part 402 may be deleted or replaced with another part according to a setting.

The group ID display unit 404 is set to 16 bits and is a part where the ID of the reception group is set. Using the screening method described above to set all the members of the group as the destination, the long data of 36 bits must be transmitted as the number of members, which can cause traffic on the network at the time of call and consumes a lot of resources. The network is less efficient. However, according to the present invention, if the group ID is transmitted and the telephone number of the group member is transmitted from the PTT server 150, the efficiency of the network is increased. Therefore, the Binary Invite message uses the group ID display unit 404 during a group call.

The From_Tag display unit, the To_Tag display unit, the Via_Bra display unit, and the Call-ID display unit after the group ID display unit 404 are the same as those described above with reference to FIG. 2.

The BMI part in 2) of 4A is denoted by '1110100F', and indicates that the BMI part, which has a binary structure, is an Invite signal transmitted from the SIP server 140 to the terminals 110 and 200 and used for a group call.

4B indicates that the BMI portion of the binary 200 OK signal is set to '1110010F', and is a 200 OK signal transmitted from the terminals 110 and 200 having the binary structure to the SIP server 140 and used for a group call.

The calling terminal 110 uses a message having the above-described structure in order to transmit a call request signal to the receiving terminal 200 through a PTT service providing system using SIP and SDB. In the process of transmitting the call request signal from the calling terminal 110, there is a need for a method of minimizing call setup time to ensure promptness and immediate connectivity, which are advantages of the PTT service.

5 is a flowchart illustrating a call setup method between a calling terminal and a PTT server in a selective call in a PTT service using SIP and SDB according to an exemplary embodiment of the present invention.

The caller selects the conversation counterparts to call using the calling terminal 110. The calling terminal 110 constructs a binary INVITE message to call the first selected receiving terminal 200 and requests a PTT service using the SDB of the access channel. At this time, the calling terminal 110 establishes a traffic channel with the mobile communication network 120 (S502).

The binary INVITE message transmitted to the mobile communication network is transmitted to the SIP server 140 via the PDSN 130 (S504).

The SIP server 140 converts the transmitted binary INVITE message into a SIP INVITE message (S506), and transmits a SIP INVITE message to the PTT server 160. The PTT server 160 requests the MRC 170 to occupy the media resource for the calling terminal 110 through the SIP INVITE message (S508).

The MRC 170 occupies a resource for the calling terminal 110 and then transmits a 200 OK message to the PTT server 160. The PTT server 160 receiving the 200 OK message transmits the 200 OK message to the calling terminal 110 through the PDSN 130 and the mobile communication network 120 using the traffic channel (S510).

The calling terminal 110 transmits the SIP ACK signal for the 200 OK message to the PTT server 160, and the PTT server 160 transmits back to the MRC 170 (S512). In addition, the originating terminal 110 transmits the information on the remaining receiving terminals to the PTT call to the PTT server 160 using the SIP INFO message (S514).

The PTT server 160 calls subscribers who want to make a PTT call using the receiving terminal information transmitted from the calling terminal 110 (S516). The procedure of calling the receiver terminal 200 will be described in detail with reference to FIG. 6.

The PTT server 160 notifies the calling terminal 110 of the fact that the receiving terminals participate in the PTT call when the response from the receiving terminal 200 that wants the PTT call is transmitted (S518).

The calling terminal 110 notified of the participation of the PTT selective call member transmits a 200 OK signal to the PTT server 160, and accordingly, a call setup procedure for the PTT selective call is completed (S520).

In the above call processing, when the originating terminal 110 is in the dormant state (Dormant), the procedure using the SDB is performed, but when the originating terminal 110 is in the active state (Standard) is set to perform a standard SIP procedure .

6 is a flowchart illustrating a call setup method between a PTT server and a receiving terminal in a selective call in a PTT service using SIP and SDB according to an exemplary embodiment of the present invention.

In step S514 of FIG. 5, the PTT server 160 which has received the reception terminal information of the members who want the PTT selective call from the originating terminal 110 calls the receiving terminal 200 of each member.

First, the PTT server 160 transmits a SIP Subscribe message requesting the status information of the member to the presence server 150 (S602).

The presence server 150 receiving member status information from the PTT server 160 transmits a 200 OK signal, which is a response signal, to the PTT server 160 (S604), and simultaneously transmits status information of the requested member as SIP Notify. (S606).

When the status information of the member is transmitted, the PTT server 160 transmits a 200 OK signal to the presence server 150 (S608), and sends a SIP INVITE message for calling the receiving terminal 200 to the SIP server 150. It transmits (S610).

The SIP server 150 converts the delivered SIP INVITE message into Binary INVITE (S612), and transmits the converted Binary INVITE message to the mobile communication network 190 through the PDSN 180 (S614). The mobile communication network 190 transmits a binary INVITE message to the receiving terminal 200 using the SDB of the paging channel (S616), and the receiving terminal 200 transmits a binary 200 OK response signal using the SDB of the access channel. Transmit to 190. At the same time, the receiving terminal 200 establishes a traffic channel with the mobile communication network 190 (S618).

The mobile communication network 190 transmits a Binary 200 OK signal to the SIP server 140 via the PDSN 180 (S620), and the SIP server 140 converts it to a SIP 200 OK signal (S622), and the PTT server. And transmits to 160 (S624).

The PTT server 160 transmits a SIP INVITE message to the MRC 170 to occupy the media resource for the receiving terminal 200 (S626), and the MRC 170 occupies the media resource of the receiving terminal 200. In step S628, the PTT server 160 transmits 200 OK.

The PTT server 160 notified of the resource occupancy transmits the SIP ACK to the receiving terminal 200 through the PDSN 180 and the mobile communication network 190 using the traffic channel (S630), and also the SIP ACK to the MRC 170. It is transmitted (S632).

The call setup between the PTT server 160 and the receiving terminal 200 is made at the receiving terminal 200 of several subscribers at the same time, thereby reducing the call setup time for the receiving terminal call.

When the receiving terminal call of the selected call members is completed through the above process, the calling terminal 110 is notified of the participation of the receiving terminals in step S518 of FIG. 5.

7 is a flowchart illustrating a call setup method between a calling terminal and a PTT server in a group call in a PTT service using SIP and SDB according to a preferred embodiment of the present invention.

The caller selects a group to be called using the calling terminal 110. The calling terminal 110 constructs a binary INVITE message to call the receiving terminal 200 and requests a PTT service using the SDB of the access channel. At this time, the calling terminal 110 establishes a traffic channel with the mobile communication network 120 (S702).

The binary INVITE message transmitted to the mobile communication network is transmitted to the SIP server 140 via the PDSN 130 (S704).

The SIP server 140 converts the transmitted binary INVITE message into a SIP INVITE message (S706), and transmits a SIP INVITE message to the PTT server 160 (S708).

The PTT server 160 receiving the SIP INVITE message transmits the SIP Subscribe message requesting the member list and the status information of the reception group to the presence server 150 (S710).

The presence server 150 which receives the list and status information of the group members from the PTT server 160 transmits a 200 OK signal, which is a response signal, to the PTT server 160 and simultaneously sends the list and status information of the requested group members. It transmits to SIP Notify (S712).

When the list and status information of the group members are delivered, the PTT server 160 transmits a 200 OK signal to the presence server 150 (S714), and captures the media resource for the calling terminal 110 through the SIP INVITE message. Request to MRC 170 (S716).

The MRC 170 occupies a Media resource and then transmits a 200 OK message to the PTT server 160. The PTT server 160 receiving the 200 OK message transmits the 200 OK message to the calling terminal 110 through the PDSN 130 and the mobile communication network 120 using the traffic channel (S718).

The calling terminal 110 transmits the SIP ACK signal for the 200 OK message to the PTT server 160, and the PTT server 160 transmits back to the MRC (170) (S720).

The PTT server 160 calls the receiving terminal 200 of the group member using the group member information transmitted from the presence server 150 (S722). Here, the calling procedure of the receiving terminal 200 will be described in detail with reference to FIG. 8.

When a response from the receiving terminal 200 of the PTT group member is transmitted to the PTT server 160, the calling terminal 110 notifies the calling terminal 110 that the receiving terminal participates in the PTT call (S724).

The calling terminal 110 notified of the participation of the PTT group members transmits a 200 OK signal to the PTT server 160, and thus the call setup procedure for the PTT group call is completed (S726).

8 is a flowchart illustrating a call setup method between a PTT server and a receiving terminal in a group call in a PTT service using SIP and SDB according to a preferred embodiment of the present invention.

The method for establishing a call between the PTT server 160 and the receiving terminal 200 for a group call according to the present invention is one of the method for establishing a call between the PTT server 160 and the receiving terminal 200 for the selective call described with reference to FIG. 6. Since it is the same as the steps of S610 to S632, detailed description thereof will be omitted.

Here, the call setup between the PTT server 160 and the receiving terminal 200 is made simultaneously with the receiving terminal 200 of all the group members, thereby reducing the call setup time for calling the receiving terminal.

When the reception terminal call of the group members is completed through the above process, the originating terminal 110 is notified of the PTT call participation of the reception terminals 200 by step S724 of FIG. 7.

In the above description of FIGS. 5 to 8, the originating terminal 110 and the receiving terminal 200 use a Binary SIP message for using the SDB in the dormant state, and in the case of the active state, the SIP. Use standard SIP messages abbreviated by the method.

In addition, when the originating terminal 110 uses a Binary SIP message, the SIP stack of the originating terminal forms message information using only information received from 200 OK without information on the SIP INVITE.

On the other hand, when the receiver terminal 200 transmits a Binary SIP message, the SIP stack of the receiver terminal forms message information using only information received as an ACK without information about SIP INVITE and 200 OK.

In addition, in order to set the calling terminal 110 and the receiving terminal 200 to use the binary SIP message and the standard SIP message according to the state, the SIP server 140 processes both the standard SIP message and the binary SIP message. You should be able to.

The above description is merely illustrative of the technical idea of the present invention, and those skilled in the art to which the present invention pertains may make various modifications and changes without departing from the essential characteristics of the present invention. Therefore, the embodiments disclosed in the present invention are not intended to limit the technical idea of the present invention but to describe the present invention, and the scope of the technical idea of the present invention is not limited by these embodiments. The protection scope of the present invention should be interpreted by the following claims, and all technical ideas within the equivalent scope should be interpreted as being included in the scope of the present invention.

As described above, according to the present invention, in the SIP-based PTT service system of the CDMA 2000 mobile data network, the traffic generated in the network can be reduced by reducing the call setup time required for PTT transmission and PTT reception using SDB. In addition, it is possible to provide a satisfactory service to PTT service subscribers by providing a certainty of prompt and immediate connectivity, which is an advantage of the PTT service.

Claims (61)

A PTT service system connected to a plurality of users in a mobile communication data network and providing a push-to-talk (PTT) call in which a conversation or a message is immediately and quickly transmitted between the users. A PTT server for calling a member for the PTT call and managing a voice in the PTT call member; A presence server that manages the presence information of a PTT service subscriber and a list of set groups, and displays status change information of the PTT service subscriber; A Media Resource Controller (MRC) for copying a talker's voice and delivering it to other members of the PTT call; A SIP server in charge of message processing and conversion delivered from the PTT service system using a SIP protocol; A mobile communication network connecting the SIP server and the PTT call member to deliver a message and data; And Packet data service node (PDSN) for processing packet data between the mobile communication network and the SIP server Including; The message is a binary message, the binary message is an indicator structure, a message type display unit for indicating that the binary message is a binary message in a SIP protocol stack, a direction display unit for displaying a delivery direction of the binary message, and And an ID display unit for distinguishing whether or not the binary message is used in a selective call or a group call. According to claim 1, The PTT service system, An originating terminal for inputting a PTT call request signal to the PTT service system through the mobile communication network; And Receiving terminal for receiving messages and data from the PTT service system PTT service system using SIP and SDB, characterized in that it further comprises. The method of claim 2, wherein the calling terminal and the receiving terminal, In the dormant state (Dormant), the Binary SIP message for using the SDB, and in the active state (Active), the PTT service system using the SIP and SDB, characterized in using the standard SIP message abbreviated by the SIP method . delete delete The method of claim 2, wherein the calling terminal and the receiving terminal, PTT service system using SIP and SDB, characterized in that for transmitting a PTT request for call setup to the PTT server using a Short Data Burst (SDB) of an access channel. The method of claim 3 or 6, wherein the SDB, PTT service system using SIP and SDB, characterized in that the IP packet in the DBM (Data Burst Message) field of the access channel to transmit to the data network. delete delete delete delete delete delete delete delete The method of claim 1, wherein the PTT call, SIP and SDB, characterized in that the invitation of the PTT service subscriber who is not participating in the PTT call by the PTT call member or the expulsion of the PTT call member who participates in the PTT call but does not need any further participation. PTT service system using. Includes originating terminal, mobile communication network, Packet Data Service Node (PDSN), Session Initiation Protocol (SIP) server, presence server, push-to-talk server, media resource controller (MRC) and receiving terminal A call setup method for selective call in a PTT service system using SIP and SDB configured by (a) transmitting an initial call setup request signal using SDB from the calling terminal; (b) requesting the MRC to occupy Media resources for the calling terminal; (c) transmitting subscriber information of the selected call member from the calling terminal; (d) checking status information of the selected call member using the presence server; (e) transmitting a call signal to the receiving terminal using the SDB; (f) requesting the MRC to occupy Media resources for the receiving terminal; And (g) notifying the calling terminal of the fact that the called terminal participates in the selective call; Including; Binary INVITE message is used as the initial call setup request signal, The binary INVITE message has an indicator structure, and includes a message type display unit for indicating that the binary message is a message having a binary structure in a SIP protocol stack, a direction display unit for indicating a delivery direction of the binary message, and a characteristic of the binary message. And a display unit and an ID display unit for distinguishing whether the binary message is used in a selective call or a group call. delete delete 18. The method of claim 17, wherein step (a) comprises: Method for establishing a call in the PTT service system using a SIP and SDB, characterized in that using the SDB of the access channel (Access Channel) for the initial call setup signal transmission. delete delete delete delete delete delete 18. The method of claim 17, wherein step (e) comprises: (e1) generating a SIP INVITE message in the PTT server; (e2) converting the SIP INVITE message into a binary INVITE message at the SIP server; (e3) transmitting the binary INVITE message to the mobile communication network; (e4) transmitting the binary INVITE message to the receiving terminal by using the SDB of a paging channel in the mobile communication network; (e5) transmitting, by the receiving terminal, a binary 200 OK signal, which is a response signal, to the SIP server by using an SDB of an access channel; (e6) converting the binary 200 OK signal to a SIP 200 OK signal at the SIP server; And (e7) transmitting the SIP 200 OK signal to the PTT signal Call setup method for the selective call in the PTT service system using a SIP and SDB characterized in that it comprises a. delete delete delete The method of claim 17, wherein the call setup for the selective call, When the calling terminal is in the dormant state (Dormant), the procedure using the SDB is performed, and when the calling terminal is in the active state (Active) using a SIP and SDB, characterized in that performing a procedure using a standard SIP Call setup method for selective call in PTT service system. delete delete Includes originating terminal, mobile communication network, Packet Data Service Node (PDSN), Session Initiation Protocol (SIP) server, presence server, push-to-talk server, media resource controller (MRC) and receiving terminal Call setup method for a group call in a PTT service system using a SIP and SDB configured by (a) transmitting a PTT group call request signal using SDB from the calling terminal; (b) checking the list and status information of the group call members using the presence server; (c) requesting the MRC to occupy Media resources for the calling terminal; (d) transmitting a call signal from the PTT server to the receiving terminal using the SDB; (e) requesting the MRC to occupy Media resources for the receiving terminal; And (f) notifying the calling terminal of the group call participation of the called terminal; Including; In the step (a), a Binary INVITE message is used as a group call request signal, and the Binary INVITE message is an indicator structure, indicating that the binary message is a message having a binary structure in a SIP protocol stack, and the binary SIP including a direction display unit for displaying the delivery direction of the message, a method display unit for displaying the nature of the binary message, and an ID display unit for distinguishing whether the binary message is used in a selective call or a group call. Call setup method for group call in PTT service system using SDB and SDB. delete delete delete delete delete delete delete delete The method of claim 34, wherein step (d) (d1) generating a SIP INVITE message in the PTT server; (d2) converting the SIP INVITE message into a binary INVITE message at the SIP server; (d3) transmitting the binary INVITE message to the mobile communication network; (d4) transmitting the binary INVITE message to the receiving terminal using the SDB of a paging channel in the mobile communication network; (d5) transmitting, by the receiving terminal, a binary 200 OK signal, which is a response signal, to the SIP server by using an SDB of an access channel; (d6) converting the binary 200 OK signal into a SIP 200 OK signal at the SIP server; And (d7) transmitting the SIP 200 OK signal as the PTT signal Call setup method for a group call in a PTT service system using a SIP and SDB comprising a. delete delete delete delete delete delete As an indicator structure of binary messages used in Push-to-Talk (PTT) service system in which selective and group calls are made using Session Initiation Protocol (SIP) and Short Data Burst (SDB), A message type display unit indicating that the binary message is a message having a binary structure in a SIP protocol stack; A direction display unit which displays a delivery direction of the binary message; A method display unit displaying a property of the binary message; And ID display unit for distinguishing whether the binary message is used in a selective call or a group call Binary message indicator structure in the PTT service, comprising a. 51. The method of claim 50, wherein the binary message indicator structure is: Binary message indicator structure in the PTT service, characterized in that it further comprises a margin insert for filling the empty space. 51. The apparatus of claim 50, wherein the message type display unit The binary message indicator structure of the PTT service, which is located in the first 4 bits of the binary message and is indicated by '1110'. The method of claim 50, wherein the direction display unit, Binary message indicator structure in the PTT service characterized by indicating the signal transmission direction between the PTT terminal and the SIP server. The method of claim 53, wherein the signal transmission direction, When the signal is transmitted from the PTT terminal to the SIP server, '0' is displayed, and when the signal is transmitted from the SIP server to the PTT terminal, '1' is displayed. rescue. The method of claim 50, wherein the method display unit, The binary message indicator structure in the PTT service, characterized in that the binary message is displayed as '0' if the message is Invite, '1' if the binary message is 200 OK. 51. The method of claim 50, wherein the ID display unit, The binary message indicator structure in the PTT service characterized in that the binary message is displayed as '0' when used in a group call, and '1' when the binary message is used in a selective call. 51. The method of claim 50, In the PTT service, if the binary message is an Invite message used in a selective call, a caller's phone display unit, a receiver phone display unit, a From_Tag display unit, a To_Tag display unit, a Via_Branch display unit, and a Call-ID display unit are added after the binary message indicator. Binary message indicator structure. 59. The system of claim 57, wherein the caller's phone display unit and the receiver's phone display unit, Binary message indicator structure in the PTT service, characterized in that consisting of 36bit 9-digit phone number. 51. The method of claim 50, If the binary message is an Invite message used in a group call and transmitted from a PTT terminal to a SIP server, a caller's phone display unit, group ID display unit, From_Tag display unit, To_Tag display unit, Via_Branch display unit, and Call-ID display unit are added after the binary message indicator. Binary message indicator structure in the PTT service, characterized in that. 60. The method of claim 59, wherein the Invite message, The binary message indicator structure in the PTT service, characterized in that further comprising a margin consisting of 4 bits between the caller display and the group ID. 51. The method of claim 50, If the Binary message is used in a group call and is an Invite message transmitted from a SIP server to a PTT terminal, after the Binary message indicator, a caller's telephone display unit, a receiver telephone display unit, a From_Tag display unit, a To_Tag display unit, a Via_Branch display unit, and a Call-ID display unit are added. Binary message indicator structure in the PTT service, characterized in that.

Images