KR100819328B1 - A call processing apparaus and method for internet phone using H.323 and SIP protocol - Google Patents

Abstract

The call processing apparatus and method for Internet phone using the H.323 protocol and the SIP protocol to process the call using the SIP protocol and the call procedure using the H.323 protocol,

When an incoming / outgoing call occurs for Internet phone that supports H.323 protocol and SIP protocol, SIP proxy server that performs call setup procedure using SIP protocol collects called party information stored in database through SIP location server. Requesting a call by number; When the call request procedure is completed, a second step of establishing a call path using the H.323 protocol and then conducts a voice call through the call path.

The present invention has the advantage of simplifying the call processing procedure by performing the call processing using the SIP protocol and making a voice call using the gateway and the gatekeeper using the existing H.323 protocol. The service can be provided, and the internet phone can provide a lifetime telephone number service.

Description

A call processing apparaus and method for internet phone using H.323 and SIP protocol

1 is a block diagram of a call processing apparatus of an Internet phone using the H.323 protocol and the SIP protocol according to an embodiment of the present invention.

2 is a flowchart for explaining a log-in procedure in an Internet phone using the H.323 protocol and the SIP protocol according to an embodiment of the present invention.

3 is a configuration example of a directory screen to which the present invention is applied.

4 is a flowchart illustrating a call signal processing in the call processing method of the Internet phone using the H.323 protocol and the SIP protocol according to an embodiment of the present invention.

5 to 6 is a signal flow diagram illustrating a call processing method of an Internet phone using the H.323 protocol and the SIP protocol according to an embodiment of the present invention.

<Description of Symbols for Major Parts of Drawings>

100,200: Internet phone 101: H.323 stack

102: SIP stack 103: user interface

104: scenario processing unit 105: directory service provider

106: client DB 107: gateway                 

108: Gatekeeper 109: SIP Proxy Server

110: directory server 111: SIP location server

112: Invitation Handler (IH)

113: VP-UA (Virtual Presence-User Agent)

114: VP-CC (Virtual Presence-Call Controller)

115: database

The present invention relates to a call processing apparatus and method of an Internet phone using the H.323 protocol and the SIP protocol, and more particularly, the call processing procedure in the Internet phone using the H.323 protocol and SIP (session initiation protocol) The call procedure is related to the call processing apparatus and method of the Internet phone using the H.323 protocol and the SIP protocol to perform call processing using the H.323 protocol.

Along with the general spread of the Internet, communication between computers and computers using the Internet, general telephone periods of computers and public telephone networks, and mobile communication terminals of computers and mobile communication networks is gradually increasing and foresee changes in the communication market in the future.

In order to make a call between a computer, a computer, a general telephone terminal, and a mobile communication terminal, an Internet phone terminal using the H.323 protocol or an Internet phone terminal using the SIP protocol is installed on the client side (computer).                         

H.323 manages subscribers and sets up call through gatekeeper, and voice call is connected to PSTN through gateway. However, H.323 has a disadvantage in that the call setup procedure is complicated because the call setup is a binary type and the protocols such as H.225, H.245, and Q.931 must be observed.

In order to overcome these disadvantages, the internet phone using SIP has the advantage of simplifying the call setup procedure compared to H.323 because the call setup is required to be text type and SIP protocol only needs to be complied with. In addition, since the HTTP protocol used on the web is a text type, when a web developer makes a call request, a call request can be simplified by using a SIP of the same type.

However, since SIP cannot use a gateway and a gatekeeper using an existing H.323 protocol for a voice call, a separate gateway needs to be established.

Accordingly, the present invention has been made to solve the above-mentioned conventional problems. When setting up a call for an Internet phone, the call is processed using the SIP protocol, and the call procedure is processed by using the H.323 protocol. The purpose of the present invention is to provide a call processing apparatus and method for Internet phones using the H.323 protocol and the SIP protocol to make voice calls using a gateway and a gatekeeper using the existing H.323 protocol.

In order to achieve the above object, the call processing apparatus of the Internet phone using the H.323 protocol and the SIP protocol according to the preferred embodiment of the present invention supports the H.323 protocol and the SIP protocol, and the call setup procedure uses the SIP protocol. An internet phone to be carried out through the H.323 protocol and a call procedure;

A database including information including subscriber information, terminal information about a subscriber, and group / member information for a messenger service;

A gateway and a gatekeeper for providing a voice call path using the H.323 protocol for the Internet phone;

A SIP proxy server performing a call setup procedure using the SIP protocol of the Internet phone;

A SIP location server configured to register information of an Internet phone (terminal) on the database while performing a call setup procedure in association with the SIP proxy server and to provide information of an Internet phone corresponding to a destination to the SIP proxy server. It features.

In order to achieve the above object, the call processing method of the Internet phone using the H.323 protocol and the SIP protocol according to a preferred embodiment of the present invention includes an incoming / outgoing signal for an Internet phone supporting the H.323 protocol and the SIP protocol. Generating a called number information stored in a database through a SIP location server, and requesting a call to the called number by a SIP proxy server performing a call setup procedure using a SIP protocol;

When the call request procedure is completed, a second step of establishing a call path using the H.323 protocol and then conducting a voice call through the call path is characterized.

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

Figure 1 is a block diagram of a system to which the call setup and call method of the Internet phone according to an embodiment of the present invention is applied. Internet phone 100, gateway 107, gatekeeper 108, SIP proxy server 109, directory server 110, SIP location server 111, database 115.

The Internet phone 100 supports the SIP protocol and the H.323 protocol to perform a call setup procedure through the SIP protocol and to perform a call procedure through the H.323 protocol.

The Internet phone 100 includes system control devices (H.245, H.225) and provides signaling for normal operation of the H.323 terminal, and opens call control, capability exchange, command and indication signaling, and logical channels. An H.323 stack 101 for providing messages used to describe the contents of the channel; A SIP stack 102 providing methods (REGIST, INVITE, ACK, ...) in accordance with RFC 2543 to make authentication and call requests; A user interface 103 that enables a user to log in and call requests, and performs an integrated interface in conjunction with a client address book and a directory server; A scenario processing unit 104 for allowing scenarios to be generated using H.323 and SIP in a call from an Internet phone to an Internet phone or an Internet phone to a PSTN, a PSTN to an Internet phone, and a PSTN to a PSTN; A directory service provider 105 that provides services such as group registration / deletion, group member registration / deletion, connection status of registered friends, registration information, etc. in association with the directory server 110; And a client DB 106 in which a non-subscriber address book and a directory structure of non-subscriber groups and members on the client side are stored.

The gateway 107 is an H.323 gateway, which functions as a voice data path between the PSTN and the Internet phone in conjunction with the gatekeeper 108.

The gatekeeper 108 provides signaling for normal operation of H.323.

When the user requests a call with a destination ID, the SIP proxy server 109 finds the actual location in association with the SIP location server 111 and is responsible for setting up a call between the sender and the receiver.

The directory server 110 performs connection management, group registration / deletion, and member registration / deletion information change of the Internet phone terminal in conjunction with the database 115.

The SIP location server 111 works with the SIP proxy server 109 to find out the actual location corresponding to the destination ID. In particular, the existing Universal Personal Telecommunications (UPT) service over the Internet SIP which finds the current destination in the extended Virtual Presence (VP) service and connects the call, that is, the VP-CC 114 (Virtual Presence-Call Controller) and the VP-UA 113 (Virtual Presence-User Agent) It utilizes as a location server (111).

IH (Invitation Handler) constituting the SIP location server 111 is a module that connects the SIP proxy server 109 and the VP-CC 114 as a module for requesting a call from the PSTN to the SIP proxy server or originating from an Internet phone. The call is connected to the PSTN through the VP-CC (114).

In addition, the VP-UA 113 configured in the SIP location server 111 performs an algorithm and directory service for finding a subscriber's destination (primary secondary) for performing a VP service, which is an extension of a lifetime telephone service. Perform the DB operation for the VP-CC 114, the PSTN network EH performs a function to control the call using the Internet network.

The database 115 receives a terminal registration from the SIP proxy server 109 as a server-side DB, stores terminal information, and stores a directory user table, a directory group, and a member table.

With reference to the accompanying Figure 2 will be described in detail the log-in process applied to the call setup and call method of the Internet phone according to an embodiment of the present invention.

Here, the user is a subscriber registered in the lifetime number service, and the log-in process stores the location of the user in the gatekeeper and a procedure for storing the user's location in the database 115 in which the information of the lifetime number service subscriber is stored. It is an apparatus for doing so.

When an ID and password are input from a user (200), the user interface 103 transmits the ID, password and IP address to the SIP stack 102, and the SIP stack 102 returns the ID, password, and IP address. A register (REGIST) method is generated and transmitted to the SIP proxy server 109. Accordingly, the SIP proxy server 109 performs an authentication procedure (202).                     

That is, the SIP proxy server 109 requests the VP-UA 113 to inquire the ID and password, and the VP-UA 113 sends the user information stored in the database 115 according to the inquiry request. After searching and performing an inquiry about the ID and password, and transmitting the inquiry result to the SIP proxy server 109, the SIP proxy server 109 determines whether authentication is successful based on the inquiry result (204). .

 As a result of the determination, if the authentication procedure is successfully performed because the ID and password are registered in the database 115 (Yes at 204), the SIP proxy server 109 uses the VP-UA 113 to perform the authentication. The user's IP address is registered by storing it in the database 115 (206).

Then, the SIP proxy server 109, upon successful registration of the terminal with the SIP proxy server 109 (Yes at 208), establishes the SIP stack 102 and the user interface 103 and the H.323 stack 101. When the terminal is requested to the gatekeeper 108 through the terminal registration and the terminal is successfully registered in the gatekeeper 108 (Yes at 212), the user interface 103 is requested. ) Notifies the user of the registration success, then transitions to the waiting state and waiting for the user input, and simultaneously calls the group information stored in the database 115 and transmits the group information to the user interface 103 to display the directory window through the user's screen. (Window) is output (214).

At this time, the terminal registration in the SIP proxy server 109 is stored in the form of matching ID and IP address, and the terminal registration in the gatekeeper 108 is matching for Internet phone number (wired number) and IP address. It is stored in the form, and the registration of the Internet phone number to the gatekeeper 108 is for writing when a call is requested from the PSTN to the Internet phone.

However, if the terminal is not registered in the gatekeeper 108, the user interface 103 transitions the call mode to the H.323 call mode to enable computer-to-computer Internet phone communication (216).

On the other hand, if authentication is not successful at step 204 (No at 204) or registration of the terminal to the SIP proxy server 109 fails (No at 208), the process proceeds to step 216 described above and transitions to the H.323 call mode.

Meanwhile, the configuration example of the directory window output on the user screen in step 214 is as shown in FIG. 3.

According to the directory window, the members include both subscribers and non-subscribers in the group 1 (800) and the group 2 (805), the information on the subscriber member collected in the database 115 by the directory service provider 105 And information about the non-subscriber member collected from the client DB 106 is based on the information.

In particular, when a user registers a non-subscriber member with a group, inputs a PSTN telephone number or IP address for the non-subscriber, and enters an ID for the non-subscriber, information about the non-subscriber member is sent to the directory service provider 105. It is stored in the client DB 106 so that the user can send a call to the non-subscriber even if the user clicks in the directory window without inputting the non-subscriber information.

Application of the above-described functions to the messenger service enables the use of the messenger between the subscriber and the non-subscriber.                     

When one or more of the user's ID or IP address is entered through the keyboard by the user logged in through the above-described process of FIG. 2 or one of the members designated as a group for the current user is designated through the on-screen directory window. 302, the user interface 103 analyzes the destination type (304) to determine if the destination is an IP address (306) (see Figure 4).

If the determination result indicates that the destination information is an IP address (Yes at 306), the user interface 103 transmits the input IP address to the H.323 stack and uses another Internet phone (B; called Internet phone) using the H.323 protocol. Call 308 to the H.323 stack.

However, otherwise, if the destination information is not an IP address, that is, an ID or PSTN number, the user interface 103 requests a call setup from the SIP stack 102 to the SIP proxy server 109 (310).

Accordingly, the SIP proxy server 109 inquires the called number for the corresponding ID stored in the database 115 to the VP-UA 113 through the IH 112 of the SIP location server 111 (312), and When the inquiry result is delivered to the IH 112, the IH 112 analyzes the type of the called number and if the called number is a PSTN number, requests for call setup to the PSTN network through the VP-CC (316). If the IP number is not the PSTN number, IP address information is transmitted to the SIP proxy server 109 (318).

The SIP proxy server 109 transmits the IP address to the user interface 103 via the SIP stack 102 (320).

On the other hand, although not shown in Figure 4, when the user enters the IP address PSTN phone number for the non-subscriber on the directory screen, the call setup request to the corresponding IP address via the H.323 stack 101 by the user interface 103 Or a call setup request to the PSTN via the SIP stack 102 and the SIP proxy server 109, the IH 112, and the VP-CC 114.

Therefore, according to the above operation, the subscriber can make a call to the non-subscriber as well as the member corresponding to the subscriber subscribed to the service.

Hereinafter, the operation of the present invention will be described in more detail with reference to the scenario shown in FIGS. 5 to 8.

4 is a scenario in which the Internet phone A proposed in the present invention makes a call request to another Internet phone B. Referring to FIG.

When the IP address for the called number is transmitted from the user interface 103, the scenario processor 104 transmits the corresponding IP address to the SIP stack 102, and the called number information is transmitted through the SIP stack 102. In step 401, a text-type INVITE method including its own ID is generated and a call request is made to the SIP proxy server 109 using the INVITE method.

For reference, the INVITE method according to the text-based SIP protocol has the following format.

INVITE

FORM: Caller Information

TO: called party information                     

Contact: Your ID

Meanwhile, the SIP proxy server 109 receiving the Invite method makes a call to the IH 112 of the SIP location server 111 to confirm the entity location of the called party (402), and the IH 112 is a VP-CC. In operation 403, the VP-CC 114 confirms the information of the called party stored in the database 115 through the VP-UA 113 and informs the actual location information of the called party 404. ).

The IH 112, which has secured the actual location information of the called party, makes a call request to the SIP proxy server 109 (405), and the SIP proxy server 109 transmits the INVITE method to the Internet phone B corresponding to the destination. (406).

Upon receiving the INVITE method, the Internet phone B sends 180 / Ring to the SIP proxy server 109 (407), and the SIP proxy server 109 forwards it to the Internet phone A (100) (408).

At this time, when the Internet phone B responds (409), the Internet phone B transmits 200 / OK to the SIP proxy server 109 (410), and the SIP proxy server 109 sends it to the Internet phone A ( In step 411, the Internet phone A confirms the response and sends an ACK signal to the Internet phone B through the SIP stack 102 to complete the call setup (412).

The Internet phone B, which has received the ACK signal, switches to the H.323 automatic acceptance mode (413).

On the other hand, the H.323 stack 101 of the Internet phone (A) delivers the setup (SETUP) message to the Internet phone (B), the Internet phone (B) receiving the setup message is automatically accepted while the Internet Delivers the CONNECT message to the phone A.

Through the series of operations described above, a call channel is formed to enable voice calls between the Internet phone A and the Internet phone B. FIG.

6 is a scenario when the Internet phone A proposed in the present invention makes a call request to a PSTN terminal.

When the PSTN number corresponding to the called number is transmitted from the user interface 103, the scenario processor 104 transfers the corresponding phone number to the SIP stack 102, and the call number information is transmitted through the SIP stack 102. And generate a text-based INVITE method including its ID and the like and make a call request to the SIP proxy server 109 using the INVITE method (501).

The SIP proxy server 109 requests a call to the IH 112 of the SIP location server 111 (503), the IH 112 informs the call request to the VP-CC 114 (504), and the VP If the call is identified by the CC 114 to be connected to the PSTN, the VP-CC 114 makes a ring request to the SIP proxy server 109 (506), and receives the ring request. 109 transmits a 180 ring signal (507).

When the Internet phone A, which has received the 180 ring signal, transmits a setup message from the gatekeeper 108 (509), the internet phone A is switched to the H.323 auto accept mode (510). In operation 511, the controller transmits a connect message to the processor 108.

Subsequently, the gatekeeper 108 receiving the connection message connects the gateway 107 to the H.323 stack 101 to enable voice communication between the Internet phone A and the PSTN terminal through the gateway 107. (512).

7 is a scenario in which the Internet phone A proposed in the present invention is received by the PSTN terminal.

When a call request from a PSTN terminal is delivered to the VP-CC 114, the VP-CC 114 forwards a Notify request message to the IH 112 (602), and the IH 112 sends the SIP proxy server. The Notify request message is forwarded to 109 at 603.

Subsequently, the SIP proxy server 109 receiving the Notify request message sends an INVITE message to the Internet phone A, and when the Internet phone A responds with a 200 / OK message (605), the SIP proxy server ( 109 forwards the Notify response message to the IH 112 (606), and the IH 112 forwards the corresponding Notify response message to the VP-CC 114 (607).

In this state, when the setup message is transmitted from the gatekeeper 108 (610), the Internet phone 100 is switched to the H.323 automatic acceptance mode (609), the connection message to the gatekeeper 108 If the gateway is connected to the H.323 stack 101 by the gatekeeper 108, voice communication between the calling PSTN terminal and the destination Internet phone A is enabled (612).

8 is a scenario of connecting a first PSTN terminal and a second PSTN terminal by an Internet phone (A) request proposed in the present invention.

In the Internet phone (A), using the INVITE method, set the first PSTN terminal as the source in the From item of the SIP message, the second PSTN terminal as the destination in the To item, and put its ID in the Contact item. When transmitting to the SIP proxy server 109 (701), the SIP proxy server 109 transmits the 180 / Ring signal to the Internet phone (A) (702).

Then, when the SIP proxy server 109 requests connection between the first and second PSTN terminals to the IH 112 (703), the IH 112 sends a first and second PSTN terminals to the VP-CC 114. To request the connection between the two (704), the call between the first and second PSTN terminal is connected by the VP-CC (114).

On the other hand, if the Internet phone (A) receives the 180 ring signal from the SIP proxy server 109 terminates the call request session (705).

By the operation according to the above-described scenario, the present invention enables the call to the Internet phone or PSTN terminal through the Internet phone, can also connect the call from the PSTN terminal to the Internet phone, between the PSTN terminal at the request of the Internet phone You can also link them together.

In addition, the call setup procedure uses text-based SIP, and the call procedure uses the H.323 protocol, so that the call processing of the Internet phone can be performed while efficiently utilizing the gateway and gatekeeper according to the currently secured H.323 protocol. It becomes possible.

The present invention enables the SIP server to receive the call request in conjunction with the SIP server, so that the lifetime number service can be used in the Internet phone, and the non-subscriber can request the call from the messenger service.

 According to the present invention as described above, by simplifying the call processing procedure by performing the call processing using the text-based SIP used on the web, it is possible to interwork with the currently secured H.323 gatekeepers, gateways The advantage is that it can be effectively used, and there is a good effect that it can provide various services to Internet phone users.

On the other hand, the present invention is not limited to the above-described embodiment, but can be modified and modified within the scope not departing from the gist of the present invention, such modifications and changes should be regarded as belonging to the following claims. will be.

Claims (22)

An Internet phone which supports the H.323 protocol and the SIP protocol while making a call setup procedure through the SIP protocol and making a call procedure through the H.323 protocol; A database including information including subscriber information, terminal information about a subscriber, and group / member information for a messenger service; A gateway and a gatekeeper for providing a voice call path using the H.323 protocol for the Internet phone; A SIP proxy server performing a call setup procedure using the SIP protocol of the Internet phone; A SIP location server configured to register information of an Internet phone (terminal) on the database while performing a call setup procedure in association with the SIP proxy server and to provide information of an Internet phone corresponding to a destination to the SIP proxy server. Call processing apparatus of Internet phone using H.323 protocol and SIP protocol. The method of claim 1, In the Internet phone,  H.323 and SIP protocols, characterized by a client database that stores information about nonsubscribers and a directory structure of groups and members on the client side, allowing users to edit groups of nonsubscribers and subscribers. Call processing device of the Internet phone using. The method of claim 1, The Internet phone using the H.323 protocol and the SIP protocol, characterized in that it further comprises a directory server for changing the information according to the group registration / deletion, member registration / deletion for the subscriber of the Internet phone in conjunction with the database. Call processing device. When an incoming / outgoing call occurs for Internet phone that supports H.323 protocol and SIP protocol, SIP proxy server that performs call setup procedure using SIP protocol collects called party information stored in database through SIP location server. Requesting a call by number; Call completion of the Internet phone using the H.323 protocol and the SIP protocol, characterized in that the second step of establishing a call path using the H.323 protocol and then making a voice call through the call path when the call request procedure is completed Treatment method. The method of claim 4, wherein A first step of transmitting a call request message to a SIP proxy server using a SIP protocol when an outgoing call to the Internet phone occurs in the first step; A second step of the SIP proxy server searching for the actual IP address of the Internet phone corresponding to the destination in the database through a SIP location server and transmitting a call request message to the destination Internet phone of the retrieved IP address; And a third step of terminating the call setup procedure by transmitting an ACK signal to the called Internet phone when the called Internet phone responds. The method of claim 4, wherein The first process, A first step of transmitting a call request message to a SIP proxy server using a SIP protocol when an outgoing call to the PSTN terminal occurs in the Internet phone; A second step of, when the SIP proxy server transmits a call request message to a SIP location server, the SIP location server requests a call to a PSTN terminal and simultaneously transmits a ring request message to the SIP proxy server; The SIP proxy server transmits a 180 ring signal to the Internet phone, and then completes a call setup procedure. The method of claim 1, wherein the SIP proxy server comprises the H.323 protocol and the SIP protocol. The method of claim 6, In the second step, The SIP location server determines whether the called number corresponds to the calling lifetime telephone number service, and if the called number corresponds to the calling lifetime telephone number service, collecting the actual called number for the called number through a database. A call processing method for an Internet phone using the H.323 protocol and the SIP protocol, wherein a call request is made using the actual called number. The method of claim 7, wherein If the collected destination number is a PSTN number, the SIP location server attempts a call request to the corresponding PSTN terminal. If the collected destination number is a mobile communication number, the SIP location server attempts a call request to the corresponding mobile communication terminal. Call processing method of internet phone using SIP protocol. The method of claim 4, wherein The first process, A first step of forwarding the call request message to the Internet phone when the Notify request message is transmitted to the SIP proxy server through the SIP location server due to an incoming call from the PSTN terminal to the Internet phone; When the Internet transmits a response message, the SIP proxy server transfers a Notify response message to the SIP location server to complete a call setup procedure. The Internet uses the H.323 protocol and the SIP protocol. How to call a pawn. The method of claim 4, wherein The first process, If the Internet phone transmits the call request message including the called number of the first PSTN terminal, the called number of the second PSTN terminal and the information of the Internet phone, the SIP location server establishes a call between the first PSTN terminal and the second PSTN. Call processing method of the Internet phone using the H.323 protocol and the SIP protocol, characterized in that performed. The method of claim 4, wherein The method of claim 1, wherein the database stores information including subscriber information, terminal information on subscribers, group / member information for messenger services, and the H.323 protocol and the SIP protocol. SIP proxy server that performs authentication procedure for the user when ID and password are input by the user of Internet phone supporting H.323 protocol and SIP protocol. Then, if authentication is successful, SIP proxy server performs call setup procedure using SIP protocol. And a first step of registering the Internet phone with a gatekeeper for establishing a call path using the H.323 protocol; A second process in which a SIP proxy server performing a call setup procedure using a SIP protocol collects called number information stored in a database through a SIP location server and requests a call to the called number when an incoming / outgoing signal is generated for the Internet phone; and; When the call request procedure is completed, the call of the Internet phone using the H.323 protocol and the SIP protocol comprising a third step of establishing a call path using the H.323 protocol and then making a voice call through the call path. Treatment method. The method of claim 12, If the authentication fails or the registration of the Internet phone to the SIP proxy server and the GatorKeeper fails, the H.323 protocol and the SIP protocol enable the Internet phone call through the computer by transitioning to the H.323 call mode. Call processing method of Internet phone using The method of claim 12, If the authentication is successful, the SIP proxy server registers the Internet phone with the SIP proxy server by storing the actual location information of the Internet phone in the database through the SIP location server. Call processing method of Internet phone. The method of claim 12, The SIP proxy server registers the Internet phone using the H.323 protocol and the SIP protocol. The method of claim 12, The gatekeeper registers the Internet phone number and IP address for the Internet phone by registering the H.323 protocol and the SIP protocol. The method of claim 12, When the first process is completed, the Internet phone using the H.323 protocol and the SIP protocol, characterized in that further comprising the step of displaying a directory window showing the directory of the member divided into groups on the screen of the Internet phone Call processing method. The method of claim 17, The member of the call processing method of the Internet phone using the H.323 protocol and the SIP protocol, characterized in that both the subscriber and the non-subscriber of the service. The method of claim 12, Transmitting a call request message to a SIP proxy server using a SIP protocol when an outgoing call from the Internet phone is generated for a member's PSTN number; A second step of, when the SIP proxy server transmits a call request message to a SIP location server, the SIP location server requests a call to a PSTN terminal and simultaneously transmits a ring request message to the SIP proxy server; The SIP proxy server transmits a 180 ring signal to the Internet phone, and then completes a call setup procedure. The method of claim 1, wherein the SIP proxy server comprises the H.323 protocol and the SIP protocol. The method of claim 19, In the second step, The SIP location server determines whether the called number corresponds to the calling lifetime telephone number service, and if the called number corresponds to the calling lifetime telephone number service, collecting the actual called number for the called number through a database. A call processing method for an Internet phone using the H.323 protocol and the SIP protocol, wherein a call request is made using the actual called number. The method of claim 12, When the outgoing call from the Internet phone is generated for the member's IP address, a call request is made to the Internet phone of the corresponding IP address. The method of claim 12, A first step of transmitting a call request message to a SIP proxy server using a SIP protocol when an outgoing call from the Internet phone is generated for a member ID; Wherein the SIP proxy server retrieves the actual location of the member ID from the database through the SIP location server and makes a call request to the location of the retrieved member. How to call a pawn.

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