KR20050008033A - Method And System Of Redirecting Call/Session Based On SIP When Called Party Is Busy - Google Patents

Abstract

PURPOSE: An SIP-based busy call/session redirecting method and a system are provided to perform a call redirecting process based on busy call redirecting service information of a receiver when the receiver is registered in a busy call redirecting service. CONSTITUTION: If the line is busy, an SIP server(201) confirms whether a receiver is registered in a busy call redirecting service based on the receiver's profile, and supplies an invite message to an SIP AS(Application Server)(205) if the receiver is registered in the service. The SIP AS(205) includes at least more than one subscriber service profile, generates an invite message based on busy call redirecting service information included in the receiver's profile if the invite message is received from the SIP server(201), and provides the generated invite message to the SIP server(201). The SIP server(201) carries out a call setup process according to the received invite message of the SIP AS(205).

Description

Method and System Of Redirecting Call / Session Based On SIP When Called Party Is Busy}

The present invention relates to a Session Initiation Protocol (SIP) based call / session switching method, and more particularly, to a method and system for switching a call or session to another address when a called party is busy during a call setup process based on SIP. It is about.

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 for converting a call / session to another address when a receiver is busy during SIP call / session setup.

Another object of the present invention is to provide a method and system for converting a call / session to an address previously designated by a subscriber when a receiver is busy during SIP call / session setup.

In order to achieve the above objects, a method of SIP-based busy call / session switching according to the first aspect of the present invention includes a) when the receiving side of the requested call or session is busy, the receiving side is busy calling switching service. Confirming whether the subscriber is a subscriber; And b) in step a), if the receiver determines that the call transfer service subscriber is in a call, performing call transfer based on the call transfer service information of the receiver. Advantageously, said busy call transfer service information comprises at least one call transfer address, wherein step b) is implemented in accordance with said at least one call transfer address, wherein said at least one call transfer address is at said receiving side. Is set by. In addition, the busy call transfer service information includes busy call transfer selection information indicating whether it is an immediate call transfer service or an optional call transfer service, and the busy call transfer selection information is set by the receiving side.

Also preferably, the step b) is a busy call of the called party based on the busy call transfer selection information when the receiving party determines in step b-1) that the receiving party is a busy call transfer service subscriber in step a). Determining whether a transfer service is the immediate call transfer service or the selective call transfer service; And b-2) in step b-1), if it is determined that the call transfer service on the receiving side is the instant call transfer service, transferring the call based on the call transfer service information on the receiving side; And b-3) in step b-1), transmitting the invite message to the receiving side when the busy call transfer service of the receiving side is the selective incoming service. In addition, the method further includes performing a call transfer based on the busy call transfer service information when a rejection message is generated from the receiving side in response to the invite message of step b-4) b-3). It further includes. The busy call transfer service information may include at least one call transfer address, and step b-4) may be performed based on the at least one call transfer address. Advantageously, the method further comprises c) analyzing said at least one call transfer address prior to performing said call transfer, and performing said call transfer in accordance with the analysis result of step c). It is characterized by. Here, the at least one call transfer address includes a PSTN or PLMN Uniform Resource Locator (tel-URL), a SIP URL, an HTTP URL, and an e-mail address.

Also preferably, the method generates an invite message based on the SIP URL for the call transfer when the at least one call transfer address is a SIP URL. If the at least one call transfer address is a PSTN or PLMN tel-URL, the call transfer is performed according to the PSTN or PLMN tel-URL. When the at least one call transfer address is an HTTP URL, an invite message including the HTTP URL and an internet browser driving command is generated, and the generated invite message is provided to the calling party. When the at least one call transfer address is an e-mail address, an invite message including the e-mail address and an e-mail program driving command is generated, and the generated invite message is provided to the calling party.

In the SIP-based busy call / session switching system according to the second aspect of the present invention, when the receiving party of the requested call or session is busy, the receiving party is a busy call switching service subscriber based on the profile of the receiving party. A SIP server for confirming whether the mobile station is configured to provide the invite message to an application server when the receiver is a busy call transfer service subscriber; And at least one subscriber service profile, and when receiving the invite message from the SIP server, generates an invite message based on the busy call transfer service information included in the service profile of the receiver; And the application server for providing the generated invite message to the SIP server, wherein the SIP server implements a call setup according to the invite message from the application server. Here, the busy call transfer service information includes at least one call transfer address, and the application server generates the invite message according to the at least one call transfer address. In addition, the at least one call transfer address is preferably set by the receiving side.

Also preferably, the busy call transfer service information includes busy call transfer selection information indicating whether it is an immediate call transfer service or an optional call transfer service, and the busy call transfer selection information is set by the receiving side. do.

Further, preferably, when the receiving server determines that the receiving party is a busy call transfer service subscriber, the busy call transfer service of the receiving side is based on the busy call transfer selection information. Whether it is the call transfer service or the selective call transfer service, and if the call transfer service on the receiving side determines that the call transfer service is the instant call transfer service, generates an invite message for an immediate call transfer; If the call transfer service is the selective incoming service, an invite message for selective incoming is generated. In addition, the SIP server, in response to the invite message, when a call rejection message is generated from the receiving side, provides the call reject message to the application server, and the application server rejects the call from the SIP server. When receiving a message, generate the invite message for call transfer during a call.

Also preferably, the SIP server analyzes the at least one call transfer address before performing the call transfer, and performs the call transfer according to the analysis result, wherein the at least one call transfer address is used. PSTN or PLMN tel-URL, SIP URL, HTTP URL, and e-mail address.

Also preferably, when the at least one call transfer address is a SIP URL, the application server generates an invite message based on the SIP URL for the call transfer. When the at least one call transfer address is a PSTN or PLMN tel-URL, the application server generates an invite message based on the PSTN or PLMN tel-URL. When the at least one call transfer address is an HTTP URL, the application server generates an invite message including the HTTP URL and an internet browser driving command. The at least one call transfer address, when the call transfer address is an e-mail address, the application server generates an invite message including the e-mail address and an e-mail program driving command.

According to the present invention, in a SIP-based network, when a subscriber is in a call, the requested call / session can be converted into a call / session to an address (or destination) designated by the subscriber.

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

FIG. 2 is a signal flow diagram illustrating a call / session switching process when the receiving side to be switched in the system shown in FIG. 1 is a SIP terminal (Client).

FIG. 3 is a signal flowchart illustrating a call / session switching 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 call / session switching process when the system illustrated in FIG. 1 is in a redirect mode.

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

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

1. Definition of terms

In-call transfer service: A third party designated in advance when the subscriber is busy (i.e. using another call or section) at the time a call or section is attempted (i.e., called) to a service subscriber. A service that diverts the attempted call or section to an address (or destination).

2. Service Settings

a) busy call transfer service subscriber (terminal B) sets up a service through a web site or operation on the terminal, or through a customer center, to another place (for example, terminal C) for calls received in the busy state. Register (activate) the service to be switched.

b) Terminal C to be transferred (other) can be a non-SIP terminal as well as a SIP terminal. (For example, PSTN (tel-URL), Internet homepage (HTTP URL), e-mail, etc.)

3. Service Profile Change: The change of the service profile managed by the network subscriber DB (Location Server or Home Subscriber Server) according to the 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 the subscriber B's profile is updated in real time. In other words, the call transfer service logic is applied to the call / session attempted to the subscriber B in the future.

-If the subscriber has not yet registered the network, the subscriber can download the changed service profile and script to SIP Server B and the application server when performing the registration procedure.

5. In addition to the above definitions, the names of all nodes in this specification used general terms according to the SIP specification, 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. Since it is omitted herein.

1 is a schematic diagram 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. 1, those skilled in the art will appreciate 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 received or input over an IP network according to SIP. The SIP server 201 sets up a call or 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 a call to a terminal (or receiving side) 101, 103, 105 managed by the SIP 201 or an invite message for session setup is inputted, the SIP server 201 may request a call or call according to the present invention. It is determined whether the receiving side (eg, terminal B 103) of the session is busy. As a result of the determination, when the receiving side (terminal B 103) is busy, the SIP server 201 receives the received information based on the "Initial Filter Criteria" information in the service profile of the receiving side (terminal B 103). It is determined whether the side (terminal B 103) is subscribed to the busy call transfer service according to the present invention. If the receiving side (terminal B 103) is subscribed to a call transfer service during a call, the SIP server 201 exchanges an invite message with the SIP application server 205 according to the present invention. (E.g., a "486 Busy Here" message). Here, it should be noted that 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.

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. When receiving the invite message (eg, a "486 Busy Here" message) from the SIP server 201, the SIP AS 205, according to the present invention, the subscriber service of the terminal (terminal B (103)) Based on the profile, the terminal (terminal B 103) recognizes that the call transfer service is set during the call, and drives the corresponding service logic according to the recognition result. At this time, the SIP AS 205 operates in "Proxy mode" as the service logic driving result if the receiving side (terminal B 103) of the invite message is a busy call switching service subscriber. An invite message is generated based on the busy call transfer service information included in the service profile of B 103), and the generated invite message is provided to the SIP server 201. The SIP server 201 then performs a call / session setup in accordance with the invite message from the SIP AS 205. Here, the service logic is implemented in a language such as CPL, and includes service subscriber information, subscription service type, operation mode of the AS (in this case, "Proxy mode"), service application condition, and application result (call transfer destination). It should be noted that it refers to a code.

Preferably, the busy call transfer service information includes a call transfer address, and the SIP AS 205 generates the invite message according to the call transfer address. It is also preferable that the call transfer address is set by the receiving side (terminal B 103). The busy call transfer service information also includes busy call transfer selection information indicating whether it is an immediate call transfer service or an optional call transfer service. The call transfer selection information during the call is set in advance by the receiving side (terminal B 103). The SIP AS 205 determines that the receiving side (terminal B 103) of the receiving side (terminal B 103) is determined based on the busy call transfer selection information when the receiving side (terminal B 103) determines that the subscriber is a busy call transfer service. It is determined whether the call transfer service during the call is the instant call transfer service or the selective call transfer service. As a result of the determination, when the call transfer service during the call is the immediate call transfer service, the SIP AS 205 generates an invite message for immediate call transfer and provides the SIP server 201. On the contrary, when the call transfer service on the receiving side is the selective incoming service, the SIP AS 205 generates an invite message for selective incoming and provides it to the SIP server 201. The SIP server 201 provides the call rejection message to the SIP AS 205 when a call rejection message is generated from the receiving side (terminal B 103) as a response to the invite message. When the SIP AS 205 receives the call rejection message from the SIP server 201, the SIP AS 205 generates the invite message for call transfer during a call.

In addition, according to the present invention, the SIP server 201 analyzes the at least one call transfer address before performing the call transfer, and performs the call transfer according to the analysis result. For example, the call transfer address may be a PSTN or PLMN tel-URL, a SIP URL, an HTTP URL, or an e-mail address. Accordingly, the SIP AS 205 generates an invite message based on the SIP URL for the call transfer when the call transfer address is a SIP URL. If the call transfer address is a PSTN or PLMN tel-URL, the SIP AS 205 generates an invite message based on the PSTN or PLMN tel-URL. If the call transfer address is an HTTP URL, the SIP AS 205 generates an invite message including the HTTP URL and an internet browser driving command. If the call transfer address is an e-mail address, the SIP AS 205 generates an invite message including the e-mail address and an e-mail program driving command.

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

1. Process when terminal C (receive 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 during a call 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 a call transfer service based on the "InitialFilter Criteria" information in the service profile of the terminal B. If the terminal B is subscribed to a call transfer service during a call, the SIP server B exchanges an invite message with a SIP application server (AS). At this time, the SIP server determines (or determines) whether the call transfer service of the terminal B is an immediate call transfer service or an optional call transfer service (S104).

If the terminal B is the immediate call transfer service, steps S105, S106, and S107 are omitted, and the SIP server executes step S108. On the contrary, when the terminal B is set as the selective incoming service, the SIP server B transmits an invite message to the terminal B (S105).

At this time, there may be two responses from the terminal B. One is a response to approve a call / session setup for a call / session attempt according to the invite message, and the other is a response to reject the call / session attempt. If the terminal B accepts the call / session attempt, the SIP server B establishes a call / session. Alternatively, when receiving a message (eg, "486 Busy Here") denying a call / session attempt, the SIP server B sends a busy message (eg, "486 Busy Here") to the SIP according to the present invention. Provided to the AS (S106, S107, S108, S109).

When receiving a "486 Busy Here" message from SIP server B, the SIP AS analyzes the "486 Busy Here" message and recognizes that it is necessary to apply a busy call transfer service according to the analysis result. In response to this, the corresponding service logic is driven, and as a result, it is determined that the call should be switched to the terminal C by operating in the "Proxy mode" (S110).

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 (S111, S112, S113).

The SIP AS generates a new invite message of which the destination address is a terminal C for a call request to a terminal C which is a new destination address (or a destination), and transmits the generated invite message to a SIP server B (S114). .

In the present embodiment (FIG. 2), steps S111 to S113 are described prior to delivery of the invite message, but it will be understood by those skilled in the art that step S114 can be executed in parallel with steps S111 to S113. In other words, step S114 is not necessarily executed after step S113.

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 (S115 and S116).

The SIP server C analyzes the received invite message and confirms that the called party is the terminal C managed by the called party. Based on the contact information (eg, the IP address) of the terminal C, the final call / session is performed to the terminal C. Routing so that it can be established (S117).

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 114 are the same as in FIG. 1.

The SIP server B analyzes the destination address (tel-URL) of the invite message and recognizes (or determines) that the call should be routed to a circuit switched network such as a PSTN or a PLMN, and then interworks with a PSTN / PLMN in the network. The invite message is routed to a node performing a MGCF function (Media Gateway Control Function) (S215).

Here, the MGCF is a system that performs interworking function between two networks by converting SIP signaling protocol of packet network and SS7 protocol such as ISUP of circuit network. The MGCF receives an invite message and analyzes a called number. 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 (S216, S217).

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 109 are the same as in FIG. 1.

In step 310, the SIP AS receiving the "486 Busy Here" message confirms that the terminal B has set up the busy call transfer service based on the subscriber service profile of the terminal B, and drives the service logic to execute the service logic. As a result (ie, due to the characteristics of the terminal C address or the operator policy), it is determined that the call should be transferred to the terminal C by operating in the redirect mode (S310).

Subsequently, 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 (S311 to S315). 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 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 driven.

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.

Therefore, according to the above embodiments, in a SIP-based network, when a subscriber is in a call, the requested call / session can be converted into a call / session to an address (or destination) designated by the subscriber.

As described above, according to the present invention, a method and system for switching a call / session to another address when the receiving party is busy during SIP call / session setup 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 (30)

a) if the receiving party of the requested call or session is busy, confirming whether the receiving party is a busy call transfer service subscriber; And b) in step a), if the called party determines that the call transfer service subscriber is in a call, performing call transfer based on the call transfer service information of the called party. How to transfer a call / session during a call. The SIP-based busy call of claim 1, wherein the busy call transfer service information includes at least one call transfer address, and step b) is performed according to the at least one call transfer address. How to switch sessions. 3. The method of claim 2, wherein the at least one call transfer address is set by the receiving party. The method of claim 1, wherein the busy call transfer service information includes busy call transfer selection information indicating whether the call transfer service is an immediate call transfer service or an optional call transfer service. . The method of claim 4, wherein the busy call transfer selection information is set by the receiver. The method of claim 4, wherein step b) b-1) in step a), if the called party determines that the call transfer service subscriber is busy, whether the busy call transfer service of the called party is the immediate call transfer service based on the busy call transfer selection information; Determining whether it is a selective call transfer service; And b-2) in step b-1), if it is determined that the call transfer service on the receiving side is the instant call transfer service, transferring the call based on the call transfer service information on the receiving side; b-3) in step b-1), when the busy call transfer service of the receiving side is the selective incoming service, transmitting an invite message to the receiving side. How to switch between calls / sessions. The method of claim 6, wherein the method b-4) in response to the invite message of step b-3), further comprising the step of: performing a call transfer based on the busy call transfer service information when a rejection message is generated from the receiving side. SIP-based busy call / session switching method characterized in that. The SIP-based method of claim 7, wherein the busy call transfer service information includes at least one call transfer address, and step b-4) is performed based on the at least one call transfer address. How to transfer a call / session during a call. 9. The method of claim 8, wherein the at least one call transfer address is set by the receiver. 10. The method according to any one of the preceding claims, wherein the method is c) analyzing the at least one call transfer address prior to performing the call transfer; SIP-based busy call / session switching method characterized in that for performing the call transfer according to the analysis result of step c). 11. The method of claim 10, wherein the at least one call transfer address is SST-based busy 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. 12. The SIP-based busy call / session according to claim 11, wherein when the at least one call transfer address is a SIP URL, an invite message based on the SIP URL is generated for the call transfer. How to switch. 12. The SIP-based busy call / session according to claim 11, wherein when the at least one call transfer address is a PSTN or PLMN tel-URL, the call transfer is performed according to the PSTN or PLMN tel-URL. How to switch. 12. The method of claim 11, wherein when the at least one call transfer address is an HTTP URL, generating an invite message including the HTTP URL and an internet browser driving command, and providing the generated invite message to the calling party. SIP-based busy call / session switching method characterized in that. 12. The method of claim 11, wherein when the at least one call transfer address is an e-mail address, an invite message including the e-mail address and an e-mail program driving command is generated, and the generated invite message is sent to the originating party. SIP-based busy call / session switching method characterized in that provided in the. In a SIP-based network system, When the called party of the requested call or session is busy, it is determined whether the called party is a busy call transfer service subscriber based on the profile of the called party, and if the receiving party is a busy call transfer service subscriber based on the profile of the called party. A SIP server for providing the invite message to an application server; And At least one subscriber service profile, when receiving the invite message from the SIP server, generating an invite message based on the busy call transfer service information included in the service profile of the receiving side; The application server for 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. 17. The SIP of claim 16, wherein the busy call transfer service information includes at least one call transfer address, and the application server generates the invite message according to the at least one call transfer address. In-call call / session switching system. 18. The SIP-based busy call / session switching system of claim 17, wherein the at least one call transfer address is set by the receiving side. 17. The system of claim 16, wherein the busy call transfer service information includes busy call transfer selection information indicating whether the call transfer service is an immediate call transfer service or an optional call transfer service. . 20. The method of claim 19, wherein the busy call transfer selection information is set by the receiver. 20. The system of claim 19, wherein the application server is When the receiver determines that the call transfer service subscriber is busy, it determines whether the busy call transfer service of the receiving party is the immediate call transfer service or the selective call transfer service based on the busy call transfer selection information. and, If it is determined that the call transfer service during the call on the receiving side is the instant call transfer service, an invite message for immediate call transfer is generated. SIP-based busy call / session switching system, characterized in that if the incoming call transfer service of the receiving side is the selective incoming service, generates an invite message for selective incoming. The method of claim 21, wherein the SIP server In response to the invite message, when a call rejection message is generated from the receiver, the call rejection message is provided to the application server. When the application server receives the call rejection message from the SIP server, SIP-based busy call / session switching system, characterized in that for generating the invite message for call transfer. The system of claim 22, wherein the busy call transfer service information includes at least one call transfer address. 24. The system of claim 23, wherein the at least one call transfer address is set by the receiving side. 25. The method of claim 16, wherein the SIP server is SIP based busy call / session switching system, characterized in that for analyzing the at least one call transfer address before performing the call transfer, and performing the call transfer according to the analysis result. The method of claim 25, wherein the at least one call transfer address is: An SIP-based busy call / session switching system comprising a PSTN or PLMN tel-URL, a SIP URL, an HTTP URL, and an e-mail address. 27. The system of claim 26, wherein the application server is If the at least one call transfer address is a SIP URL, SIP-based busy call / session switching system characterized in that for generating the invite message based on the SIP URL for the call transfer. 27. The system of claim 26, wherein the application server is SIP-based busy call / session switching system, if the at least one call transfer address is a PSTN or PLMN tel-URL, generates an invite message based on the PSTN or PLMN tel-URL. 27. The system of claim 26, wherein the application server is If the at least one call transfer address is an HTTP URL, SIP-based busy call / session switching system characterized in that for generating an invite message including the HTTP URL and the Internet browser drive command. 27. The system of claim 26, wherein the application server is The at least one call transfer address if the call transfer address is an e-mail address, an SIP-based busy call / session switch characterized in that it generates an invite message including the e-mail address and an e-mail program driving command. system.