KR100629889B1 - Method And System Of Redirecting Call/Session Based On SIP - Google Patents

Abstract

A method and system for providing a service for switching a call by calling party number or time zone in a call setup process based on SIP is disclosed. The SIP-based call / session switching method comprises the steps of: a) when receiving an INVITE message from an originating party, determining whether the receiving party is a subscriber of the call switching service; And b) in step a), when the called party determines that the call transfer service is a subscriber, the call is transferred based on call transfer service information of the called party. Therefore, based on SIP, it is possible to transfer a call requested by a subscriber who is previously designated by the subscriber to an address designated by the subscriber, or to transfer a call that is requested by the subscriber to the address specified by the subscriber in a predetermined time zone. .

Description

Method and System of Redirecting Call / Session Based On SIP}

1 is a configuration diagram schematically showing a SIP-based call transfer system according to an embodiment of the present invention.

FIG. 2 is a signal flowchart illustrating a call transfer process when a receiver to be transferred is a SIP terminal (Client) in the system shown in FIG. 1.

FIG. 3 is a signal flowchart illustrating a signal processing process when a terminal to be received in the system shown in FIG. 1 is a PSTN or PLMN terminal.

FIG. 4 is a signal flowchart illustrating a signal processing process when the system illustrated in FIG. 1 is in a redirect mode.

The present invention relates to a Session Initiation Protocol (SIP) based call / session switching method, and more particularly, to a method and system for providing a service for switching a call by calling party number or time zone in a call setup process based on SIP. will be.

SIP is being developed by the SIP Steering Group within the Internet Engineering Task Force (IETF). SIP has been published as IETF RFC 2543 and is currently proposed as a standard. SIP cooperates with the present invention.

Accordingly, the present invention has been made in view of the above circumstances, and an object of the present invention is to provide a method and a system capable of switching a call based on SIP.

It is another object of the present invention to provide a method and system that can divert a call requested by a subscriber who has been previously designated by the subscriber to an address (or destination) specified by the subscriber.

It is still another object of the present invention to provide a method and system capable of converting a call requested at a subscriber's predetermined time period to an address designated by the subscriber.

In order to achieve the above objects, an SIP-based call / session switching method according to the first aspect of the present invention comprises: a) when receiving an INVITE message from an originating party, whether the receiving party is a call switching service subscriber; Determining; And b) in step a), when the called party determines that the call switching service is a subscriber, the call is switched based on the call switching service information of the receiving side. Preferably, the call transfer service information includes information on whether the called party is a subscriber of a call transfer service by calling number or a subscriber of a call transfer service by time zone.

Also, preferably, in step b), when the receiving side determines that the call switching service subscriber is in step b-1), whether the receiving side is the call switching service subscriber by the calling number or the time transfer Determining whether a subscriber is a service; And b-2) in step b-1), if the called party determines that the call forwarding service subscriber is called by the calling number, determining whether the calling party is a target of the call forwarding service; b-3) if the calling party is a call forwarding service subject, executing the call forwarding; b-4) if the receiving party is a subscriber of the time zone transfer service, determining whether the reception time of the invite message belongs to a time zone of a call transfer service; And b-5) executing the call transfer when the reception time of the invite message belongs to a call transfer service time zone. Preferably, the call transfer service information includes a call transfer destination address of a terminal for which a receiver wants to transfer a call, and the call transfer is based on the call transfer destination address.

More preferably, the method further comprises c) analyzing the call forwarding destination address prior to step b), wherein step b) is performed according to the analysis result of step c). It features. Here, the call transfer destination address includes a PSTN or PLMN Uniform Resource Locator (tel-URL), a SIP URL, an HTTP URL, and an e-mail address. Preferably, the step b) is characterized in that when the call forwarding destination address is a SIP URL, generates an invite message based on the call forwarding destination address. Further, step b), if the call transfer destination address is a PSTN or PLMN Uniform Resource Locator (tel-URL), performs a circuit switched mobile telephone call request according to the call transfer destination address. In step b), when the call transfer destination address is an HTTP URL, the Internet browser drive command signal is generated based on the call transfer destination address, and the generated Internet browser drive command signal is provided to the calling party. If the call transfer destination address is an e-mail address, step b) generates an e-mail program driving command signal including the call transfer destination address, and sends the generated e-mail program driving command signal to the calling party. To provide.

When the system according to the second aspect of the present invention receives an invite message, the system determines whether the receiver is a call switching service subscriber based on the profile of the receiver, and when the receiver determines that the receiver is a call switching service subscriber, A SIP server for providing an invite message to the application server; And at least one subscriber service profile, and when receiving the invite message from the SIP server, generates an invite message based on a call transfer address included in a service profile of a receiver, and generates the invite message. An application server for providing a byte message to the SIP server, wherein the SIP server implements a call setup according to the invite message from the application server. Preferably, the subscriber service profile includes call transfer service information, and the call transfer service information includes information on whether the called party is a subscriber of a call transfer service by calling number or a subscriber of a call transfer service by time zone. .

Also preferably, when the receiving side determines that the receiving party is a call switching service subscriber, the application server determines whether the receiving side is a subscriber of the call switching service according to the calling number or a subscriber of the call switching service according to the time zone. If the receiving side determines that the call forwarding service subscriber by the calling number, the caller determines whether the calling party is the target of the call transfer service, if the calling party is the target of the call transfer service, generates the invite message, If the receiving side is a subscriber of the time zone call transfer service, it is determined whether the reception time of the invite message belongs to a call transfer service time zone, and if the reception time of the invite message belongs to a call transfer service time zone, Generate the invite message for call transfer. Here, the call transfer service information includes a call transfer destination address of a terminal for which a receiver wants to transfer a call, and the call transfer is based on the call transfer destination address. Preferably, the application server analyzes the call transfer destination address before generating the invite message for the call transfer, and generates the invite message based on the analysis result. The call transfer destination address may be tel-URL, SIP URL, HTTP URL, or e-mail address.

Preferably, the application server generates an invite message according to each of tel-URL, SIP URL, HTTP URL, and e-mail address. Preferably, when the call forwarding address is an HTTP URL, the application server generates an invite message including an Internet browser driving command based on the HTTP URL, wherein the call forwarding address is an e-mail address. In this case, an invite message including an e-mail program driving command including the call transfer destination address is generated.

According to the present invention, a SIP-based call transfers a call requested by a subscriber who is previously designated by the subscriber to an address (or destination) designated by the subscriber, or transfers a requested call to an address designated by the subscriber in a time zone designated by the subscriber. You can make a call transfer.

Hereinafter, the present invention will be described in more detail with reference to the accompanying drawings.

First, prior to describing a specific embodiment of the present invention, the following items are noted in the present embodiment to assist in understanding the present embodiment:

1. Definition of terms

a) Call Transfer Service by Calling Number: A service that transfers a call to a third address (or destination) predetermined by the subscriber according to the caller's call number to the service subscriber. In this context, the caller ID does not mean only a number or a number but an overall identifier that distinguishes the caller (for example, caller name, SIP URL, or image).

b) Time-dial call transfer service: A service that transfers a call that is received to a service subscriber by time-division to an address (or a third destination) previously designated by the subscriber.

2. Service Settings

a) Call transfer service subscriber (terminal B) registers (activates) the service so that the call can be switched to another place by call number or time zone through operation at the terminal, setting up service through the web, or through a customer center.

b) Call divert setting by calling number: A call is set to be switched to a third destination (terminal C) for a specific calling number (terminal A), and one or more numbers can be set.

c) Set call forwarding by time zone: Set the call to be forwarded to a third destination (terminal C) for all or specific outgoing calls attempted to reach the specified time zone (e.g. year, month, day, hour, minute, day of the week, etc.). In addition, this third destination may be more than one set also.

d) Terminal C to be transferred is not only SIP terminal but also non-SIP terminal. (For example, PSTN (tel-URL), Internet homepage (HTTP URL), E-mail, etc.)

3. Service profile change: It is a service profile change managed in the network subscriber DB (Location Server or Home Subscriber Server) according to service activation.

4. Apply service profile updates

The changed service profile is delivered to SIP Server B and Application Server, which is currently responsible for this subscriber's service, so that Subscriber B's profile is updated in real time. That is, the call transfer service logic by calling number or time zone is applied to the call / session to be received by subscriber B in the future. If the subscriber has not yet registered the network, the modified service profile and script can be downloaded to the SIP server B and the application server when the registration procedure is performed to provide the corresponding service.

4. In addition to the above definition, the names of all nodes in the present specification used general terms according to the SIP standard, and these names can be changed according to implementation. In addition, each node represents a logical functional element, and thus, multiple functional elements may be mounted in one system as needed in an implementation.

In addition, all service flows only indicate the delivery of the first INVITE message, and after "INVITE" the "183 Session Progress", "PRACK", "UPDATE", "180 Ringing", Messages related to session negotiation and setup, resource reservation, etc., such as "200 OK", "ACK", etc. should be additionally used, but this is to follow the delivery path of INVITE and is independent of the core of the present invention. Therefore, it is omitted in the present specification.

1 is a configuration diagram schematically showing a SIP-based call transfer system according to an embodiment of the present invention.

The SIP-based call / session switching system according to the present invention includes a SIP server 201, a location server 203, and a SIP application server (AS). Although only three servers 201, 203, and 205 are shown in FIG. 4, those skilled in the art will understand that each of these servers may be one or more according to embodiments. Therefore, it should be noted that the present invention is not limited by the number of the servers.

In FIG. 1, the SIP server 201 responds to an invite message entered into the IP network according to the SIP. The SIP server 201 sets up a call / session in response to the invite message. The SIP server 201 stores information of terminals registered in the SIP server 201. In addition, the SIP server 201 stores profiles of registered terminals. When an invite message for call / session setup is input to a terminal (or receiving side) managed by the SIP 201, the SIP server 201 receives "Initial Filter Criteria" information in a service profile of the receiving side. On the basis of this, it is determined whether the receiving party is subscribed to the call switching service by calling number and / or the time switching call service according to the present invention. If the terminal B is subscribed to call switching service by calling number or / and call switching service by time zone, the SIP server 201 sends an invite message to the SIP application server according to the present invention. Receive. It should be noted that the term "Initial Filter Criteria" refers to all mechanisms corresponding to the predefined service triggering criteria in connection with the provision of services by the subscriber.

The location server 203 stores location information of terminals registered in each SIP server 201 and responds to a location query from the SIP server 201.

The SIP AS 205 stores service profiles of all subscribers. Upon receiving the invite message, the SIP AS 205 recognizes that the terminal (or subscriber) has set up call switching service by calling number and / or time zone based on the subscriber service profile of the corresponding terminal. The service logic is driven according to the recognition result. At this time, the SIP AS 205 determines that if the receiving side of the invite message is a call switching service subscriber, the service to the call side should be switched by operating in the "Proxy mode" as a result of the service logic driving. The call number information of the call is included in the "From" header of the SIP invite message, and the time information may preferably use system time information. In addition, the service logic is implemented in the same language as the CPL, such as service subscriber information, subscription service type, the operation mode of the AS (in this case, "Proxy mode"), service application conditions, and application results (call transfer destination), etc. Note that it refers to the code included. In addition, the SIP AS 205 generates a new invite message with a new address for a call request to a new destination address (or destination) according to the present invention, and sends the invite message having the new address to a SIP server ( 201).

For example, when the receiving side determines that the receiving party is a call switching service subscriber based on the received invite message, the SIP AS 205 determines whether the receiving side is a subscriber of the call switching service by calling number or the time-based transfer service. Determine if you are a subscriber. When the called party determines that the call forwarding service subscriber by the calling number is determined, the SIP AS 205 determines whether the calling party is a target of a call forwarding service, and when the calling party is a call forwarding service target, Generate an invite message for call transfer. On the contrary, when the receiving side is a subscriber of the time zone transfer service, the SIP AS 205 determines whether the reception time of the invite message belongs to a call transfer service time zone, and the reception time of the invite message. If the call transfer service falls within the time zone, the invite message for the call transfer is generated.

The call transfer service information includes a call transfer destination address of a terminal for which a receiver wants to transfer a call, and the call transfer is based on the call transfer destination address. As the call forwarding destination address, the tel-URL (Uniform Resource Locator), SIP URL, HTTP URL, and e-mail address, the SIP AS 205 generates an incoming message according to the call forwarding destination address.

For example, the SIP AS 205 generates the invite message according to the tel-URL when the call transfer destination address is a uniform resource locator (tel-URL). If the call transfer destination address is an HTTP URL, the SIP AS 205 generates an invite message including an Internet browser driving command based on the HTTP URL. If the call transfer destination address is an e-mail address, the SIP AS 205 generates an invite message including an e-mail program driving command including the call transfer destination address.

Hereinafter, the operation of the present invention according to the state of the receiving side to be forwarded will be described in more detail with reference to the accompanying drawings.

1. Signaling process when terminal C (receiving side to be transferred) is SIP terminal (Client)

FIG. 2 is a signal flowchart illustrating a call transfer process when a receiver to be transferred is a SIP terminal (Client) in the system shown in FIG. 1.

Referring to FIG. 2, first, terminal A transmits an invite message to SIP server A for call setup to terminal B. FIG. The SIP server A is a server currently serving the terminal A (S101).

When receiving the invite message from the terminal A, the SIP server A analyzes the receiving party number included in the invite message, and registers the location and registration information of the terminal B for routing to the terminal B having the receiving party number. The location server B managed is inquired of the location of the terminal B. When receiving the location query of the terminal B from the SIP server A, the location server B provides the address of the SIP server B registered with the terminal B to the SIP server A (S102).

The SIP server A transmits the invite message to the SIP server B (Serving) according to the address of the SIP server B (S103).

The SIP server B analyzes the receiving party number included in the invite message from the SIP server A, and finds out that it is an incoming call to the terminal B managed by the analysis. Subsequently, the SIP server B determines whether the terminal B is subscribed to the call switching service by calling number and / or the time slot calling service based on the "Initial Filter Criteria" information in the service profile of the terminal B. If the terminal B is subscribed to call switching service by calling number and / or call switching service by time zone, the SIP server B exchanges an invite message with a SIP application server (AS). Here, the term "Initial Filter Criteria" refers to all mechanisms corresponding to a service triggering criterion defined in advance in connection with providing a service of a subscriber (S104).

Upon receiving the invite message, the SIP AS recognizes that the terminal B (or subscriber) has set up call switching service by calling number and / or time zone based on the subscriber service profile of the corresponding terminal B, and the corresponding service logic. As a result, it is determined that the call should be switched to the terminal C by operating in the "Proxy mode" as a result. The call number information of the call is included in the "From" header of the SIP invite message, and the time information may preferably use system time information. In addition, the service logic here is implemented in a language such as CPL, service subscriber information, subscription service type, the operation mode of the AS (in this case, "Proxy mode"), service application conditions, and application results (call transfer destination-terminal C). It should be noted that it refers to a code (Code) that contains the back (S105).

According to the above-mentioned call switching decision, the caller may be informed that the call is currently being transferred, which may be provided as an option according to the service condition, and indicates that the call is being transferred to the user according to the operator policy and the terminal implementation. It may not be (S106, S107, S108).

The SIP application server generates a new invite message of which the destination address is the terminal C for a call request to the terminal C which is the new destination address (or the destination) and transmits it to the SIP server B (S109).

In this embodiment (FIG. 2), steps S106 to S108 are described as being performed before the delivery of the invite message, but step S109 can be executed in parallel with steps S106 to S108. In other words, step S108 does not necessarily have to be executed afterward.

SIP server B analyzes the destination number (eg, SIP URL) of the invite message and performs a location query to location server C, which manages the registration and location information of terminal C for routing to terminal C, and performs a location query from location server C. Find out the address of SIP server C where C is registered. Subsequently, the SIP server B sends an invite message to the SIP server C according to the address of the SIP server C (S110 and S111).

The SIP server C analyzes the received invite message and confirms that the called party is the terminal C managed by the called party. The SIP server C receives the final call to the terminal C based on the contact information (eg, the IP address) of the terminal C. Routing so that this can occur (S112).

2. Signaling process when terminal C is PSTN or PLMN terminal (tel-URL)

FIG. 3 is a signal flowchart illustrating a signal processing process when a terminal to be received in the system shown in FIG. 1 is a PSTN or PLMN terminal (tel-URL).

In FIG. 3, the same reference numerals are attached to the same parts as in FIG. 1, and detailed description thereof will be omitted.

Referring to FIG. 3, the steps from 101 to 109 are the same as in FIG. 1.

The SIP server B analyzes the destination number (tel-URL) of the invite message, determines that the call should be routed to a circuit switched network such as a PSTN or PLMN, and then performs an MGCF function, which is an interworking function with the PSTN / PLMN in the network. The invite message is routed to a node (MGCF; Media Gateway Control Function) (S210).

Here, the MGCF is a system that performs interworking function between two networks by integrating the SIP signaling protocol of the packet network and the SS7 protocol such as the ISUP of the circuit network. The MGCF receives an invite message and analyzes the called number, and the receiving side analyzes the PSTN. Or confirm that the PLMN terminal and converts the invite message to an ISUP message (IAM) for the call setup with the PSTN or PLMN and exports it to the PSTN. Then, the PSTN exchange performs routing to terminal C of the last stage (S211, S212).

3. Signal Processing in Redirect Mode

FIG. 4 is a signal flowchart illustrating a signal processing process when the system illustrated in FIG. 1 is in a redirect mode.

In Fig. 4, the same parts as those in Fig. 2 or Fig. 3 are denoted by the same reference numerals, and detailed description thereof will be omitted.

This embodiment corresponds to the following cases:

1. When the terminal C exists in the PSTN / PLMN (tel URL) and there is no signaling conversion device such as MGCF in the network;

2. terminal C is an Internet home page (or HTTP URL) address;

3. Terminal C is an E-Mail address; And

4. In case of routing according to other service provider policy.

Referring to FIG. 3, steps 101 through 104 are the same as in FIG. 1.

In step 305, the SIP application server receiving the invite message confirms that the terminal B has set the call switching service by calling number or time zone based on the subscriber service profile of the terminal B, and drives the corresponding service logic. As a result (ie, due to the nature of the terminal C address or by the operator policy), it is determined that the call should be transferred to the terminal C by operating in the redirect mode. The caller number of the call is included in the "From" header of the SIP invite message, and the time information is preferably used by the system time information (S305).

Here, the service logic is implemented in a language such as CPL, and includes service subscriber information, subscription service type, operation mode of the application server (redirect mode in this case), service application condition, and application result (call switching address (destination destination); Refers to a code including C).

Subsequently, in steps S306 to S311, the SIP application server generates a driving command signal (eg, a "302 Moved Temporarily" message) including the address of the terminal C to be forwarded to the terminal A (S306 to S311). . Alternatively, according to an embodiment, a driving command signal may be provided that includes inquiring the terminal A of a call transfer intention to the terminal C.

If a driving command signal is provided to the terminal A, the corresponding application is automatically executed (not recognized by the user) according to the type of the address to be forwarded as follows to redial the call.

In this case, when the address of the terminal C is a SIP URL, a SIP client (Client) in the terminal A automatically generates a new byte that receives the address and sends it to the network. If the address of the terminal C is a general telephone (tel) URL, the circuit switched mobile telephone function of the terminal A is automatically driven to execute call setup for receiving this address. If the address of the terminal C is an HTTP URL (e.g., an Internet homepage, etc.), an Internet browser in the terminal A is automatically started to set up a session for receiving this address, and the terminal C has an e-mail address. ), The e-mail program in the terminal A that receives this address is automatically started.

On the contrary, when the terminal A is provided with a message inquiring about the intention to transfer the call to the terminal C, the call processing is terminated without the redialing process or the above-described process according to the response from the terminal A. For example, when the user of the terminal A selects "Yes" for the inquiry of the call forwarding intention, the terminal A activates a redialing program to redial the call. If NO is selected, the execution of the command is completed without driving the redialing program.

Accordingly, according to the above embodiments, the subscriber converts a call requested by a subscriber who has been previously designated by the subscriber to an address (or destination) designated by the subscriber based on SIP, or receives a call that is requested at a predetermined time zone. The call can be transferred to the address specified by.

As described above, according to the present invention, a method for switching a call based on SIP can be realized.

Although the present invention has been described in detail by the above embodiments, the present invention is not limited thereto, and variations and improvements can be made without departing from the ordinary knowledge of those skilled in the art.

Claims (20)

a) when receiving an INVITE message from the calling party, determining whether the called party is a call diverting service subscriber by calling number; b) in step a), if it is determined that the called party is a call diverting service subscriber by the calling number, determining whether the calling party is a target of call diverting service; And c) as a result of the determination of step b), when the calling party is the call switching service target, the SSI-based call / session switching method comprising the step of switching the call based on the call switching service information of the receiving side. . A) when receiving an INVITE message from the calling party, determining whether the receiving party is a time-based call transfer service subscriber; B) if it is determined in step A) that the receiving party is the time-dial call transfer service subscriber, determining whether the reception time of the invite message belongs to a time-division service time zone; And (C) if the reception time of the invite message falls within a call transfer service time zone as a result of the determination of step B), performing the call transfer based on the call transfer service information of the receiving side; SIP-based call / session switching method. delete The method according to claim 1 or 2, wherein the call transfer service information includes a call transfer destination address of a terminal for which a receiver wants to transfer a call. SIP-based call / session switching method characterized in that the call transfer is based on the call transfer destination address. delete The method of claim 4, wherein the call forwarding address is: SST-based call / session switching method comprising a PSTN or PLMN Uniform Resource Locator (tel-URL), a SIP URL, an HTTP URL, and an e-mail address. 7. The method of claim 6, wherein if the call forwarding destination address is a SIP URL, an invite message based on the call forwarding destination address is generated for the call forwarding. . 7. The method of claim 6, wherein the step b) is a circuit switched mobile telephone call request according to the call transfer destination address for the call transfer when the call transfer destination address is a PSTN or PLMN tel-URL. SIP-based call / session switching method characterized in that for implementing. The method according to claim 6, wherein when the call transfer destination address is an HTTP URL, an Internet browser drive command signal based on the call transfer destination address is generated and the generated Internet browser drive command signal is provided to the calling party. SPI-based call / session switching method characterized in that. 7. The method according to claim 6, wherein when the call transfer destination address is an e-mail address, an e-mail program driving command signal including the call transfer destination address is generated for the call transfer, and the generated e-mail program driving command is generated. SIP-based call / session switching method characterized in that providing a signal to the originating side. In a SIP-based network system, In the case of receiving the invite message, the receiver determines whether the called party is a call diverting service subscriber by calling number based on the profile of the called party. A SIP server for providing the application server; And Includes at least one subscriber service profile, and upon receipt of the invite message from the SIP server, determines whether the originating party is a subject of a call transfer service; An application server for generating an invite message based on a call divert address included in the service profile of the mobile station, and providing the generated invite message to the SIP server; And the SIP server implements call setup according to the invite message from the application server. In a SIP-based network system, In the case of receiving the invite message, it is determined whether the receiving party is a call switching service subscriber by time zone based on the profile of the receiving side. A SIP server for providing to an application server; And At least one subscriber service profile, and when receiving the invite message from the SIP server, it is determined whether the calling party is the target of the time zone call switching service, and as a result, the calling party is time zone call switching In the case of the service target person, it is determined whether the reception time of the invite message belongs to a call transfer service time zone, and when the reception time of the invite message belongs to the call transfer service time zone, In the case, it includes an application server for generating an invite message based on the call transfer address included in the service profile of the receiving side, and providing the generated invite message to the SIP server, And the SIP server implements call setup according to the invite message from the application server. delete The method according to claim 11 or 12, wherein the call transfer service information includes a call transfer destination address of a terminal for which a receiver wants to transfer a call, and the call transfer is based on the call transfer destination address. SIP-based call / session switching system. 13. The method of claim 11 or 12, wherein the application server Before generating the invite message for the call transfer, the call transfer destination address is analyzed, SPI-based call / session switching system characterized in that for generating the invite message based on the analysis result. The method of claim 15, wherein the call forwarding address, SIP-based call / session switching system comprising a tel-URL (Uniform Resource Locator), SIP URL, HTTP URL, and e-mail address. The system of claim 16, wherein the application server generates an invite message based on the SIP URL when the call transfer destination address is a SIP URL. The SIP-based call / session of claim 16, wherein the application server generates the invite message according to the tel-URL when the call transfer destination address is a tel-URL. Switching system. 17. The method of claim 16, wherein the application server, if the call forwarding destination address is an HTTP URL, generates an invite message including an Internet browser driving command based on the HTTP URL, and the generated invite message SIP-based call / session switching system characterized in that provided to the calling party through a SIP server. 17. The application server of claim 16, wherein when the call transfer destination address is an e-mail address, the application server generates an invite message including an e-mail program driving instruction including the call transfer destination address, and generates the invite message. SIP-based call / session switching system, characterized in that for providing a message to the calling party through the SIP server.

Images