KR101560641B1 - Call set up method of internet phone - Google Patents

Abstract

The present invention relates to a call setup server installed between a sender client system having an Internet phone, an exchange, a receiver-side telephone, a transmitter-side client and an exchange and setting up a call between the Internet phone and the receiver- The present invention relates to an Internet phone call connection method for connecting an Internet phone and a recipient phone on an Internet phone system including a first step of displaying a call connection indication on a transmission side client system, And a call setup server for setting a transmission side client identifier, and a call setup server for receiving the reception side telephone identification information from the transmission side client system and transmitting the reception side telephone identification information to the exchange, Perform call setup including step 4 of setting up And an Internet phone call connection method is provided.

Description

{CALL SET UP METHOD OF INTERNET PHONE}

The present invention relates to an Internet phone call connection method, and more particularly, to an Internet phone call connection method in which a user clicks a telephone connection indication to which a user wishes to make a telephone call displayed by e-mail or a web browser without joining a specific call service or logging into a specific call system And a method of connecting a telephone of an Internet phone capable of connecting to a telephone of the side.

Currently, call connections over Internet phones are already widely used on a global scale, and the frequency and volume of calls are increasing exponentially.

A method of making a call using an Internet phone is known as a so-called 070 service, and a method of communicating with a receiver using an Internet phone phone (hereinafter referred to as 'VoIP phone') connected to the Internet, (Hereinafter, referred to as a " VoIP softphone " to distinguish it from a " VoIP phone ") equipped with a microphone. In the former case, the form and method of use are similar to those of the conventional phone, while in the latter case, the call is carried out using a headset attached to the computer terminal.

A typical Internet phone service provider providing Internet phone service using VoIP softphone is Skype (USA). In order to make a call using the service provided by Skype, the sender must (1) become a member of Skype, (2) access the corresponding communication server provided by Skype while inputting the login ID and password , And (3) by inputting the telephone number of the receiving side to be put on the receiving side.

In order to use the Internet phone service provided by Skype, the sender who wants to make a call must join the Skype company and log in. That is, when joining as a member, an identifier that can be distinguished from others is given using the login ID, and the identifier is recognized as an identification number that can identify the VoIP softphone of the sender with other VoIP softphone . However, there are disadvantages that the use is restricted due to the inconvenient use due to the registration procedure and the login procedure.

It is an object of the present invention to provide an Internet phone call connection method capable of establishing a call connection with a reception-side telephone without going through a separate registration procedure or a login procedure when making a call to an Internet phone .

The above object of the present invention can be also achieved by a call set up communication server which sets up a call between the transmitting client system and the receiving-side telephone using the exchange, a transmitting side client system having an Internet phone, And a call set up web server for providing the sender client system with the URL information of the call setup communication server and the receiver telephone identification information, A method for connecting an Internet phone call,

Wherein the call setup web server displays a call connection indication that includes the call setup communication server URL information and the receiver telephone identification information and indicates a receiver to which the call is to be made, After providing the call setup communication server URL information and the receiver telephone identification information to the transmitting client system,

The call setup communication server includes a first step of setting a transmitting client identifier for identifying the Internet phone, which is recognizable by the exchange and is distinguishable from other Internet phones, ; And

The call setup communication server includes a second step of receiving the receiver-side telephone identification information from the transmission-side client system and transmitting the received telephone identification information to the exchange to set up a call line,

And the transmitting client identifier is maintained while the Internet connection between the call setup server and the transmitting client system is maintained.

According to the Internet phone call connection method according to the present invention, it is possible to click on the display of the recipient's phone number displayed on the e-mail or web page, and to make a call using the VoIP softphone without registering a separate member or logging in. In other words, the sender can enter any subscription procedure, ID, password, and make phone calls without paying the fee.

In particular, a user providing an e-mail or web page can easily provide his / her phone number to an e-mail or a web page so that it can be easily exposed to consumers.

The Internet intermediary service provider temporarily distinguishes the sender's phone from the other sender's phone and temporarily issues a sender's phone identifier (DN) that can be recognized by the exchange to only the sender requesting the phone call, Can be reused as soon as the connection ends.

1 is a block diagram of an Internet phone call connection system according to an embodiment of the present invention;
BACKGROUND OF THE INVENTION 1. Field of the Invention [0001]
3 is a block diagram of an Internet phone call connection system according to another embodiment of the present invention;
4 is a block diagram of an Internet phone call connection system according to another embodiment of the present invention.
5 is a flowchart illustrating a call connection using an Internet phone call connection system according to an embodiment of the present invention;
FIG. 6 is an example of a telephone connection indication displayed to a sending client through a web browser and a telephone connection indication displayed to a sending client via email.
FIG. 7 shows an example of a web page that is provided to the transmitting client system from the call set-up server, and which inputs LN information after visiting a web site.
FIG. 8 is a flowchart of call setup and call connection of an Internet phone call connection system according to the present invention; FIG.
9 is a detailed configuration diagram and DN assignment flowchart of a transmitting client system and a call setup server;

First, terms used in the present invention will be defined. The transmitting client identifier is an identifier that can be recognized by the exchange and identifies the Internet phone provided in the transmitting client system in the exchange from the other Internet phones. In the present invention, this is referred to as a DN (Dial Number). The receiving-side telephone identifier is an identifier for identifying the receiving-side telephone, which receives a telephone call recognizable by the exchange, differently from other telephone sets. In the present invention, this is referred to as an LN (Location Number).

Preferred embodiments of the present invention will be described in detail with reference to the accompanying drawings.

1 is a block diagram of an Internet phone call connection system according to an embodiment of the present invention. The Internet phone call connection system is composed of a transmission side client system 100, a call setup server 250, an exchange 350 and a reception side telephone 600.

Side client system 100 receives a call setup server URL (Uniform Resource Locator) information and a reception-side telephone identification number from the call setup server and receives a call setup telephone number to be. The transmission-side client system 100 has a hardware configuration such as a personal computer or a notebook computer having a general input / output device, or an electronic device connected to a mobile phone or the Internet. The transmitting client system 100 establishes a call by connecting to a web browser and a call setup server 250 that interprets and displays HTML (Hyper Text Markup Language), connects to the packet network, and performs voice or video call of the VoIP softphone Lt; / RTI >

 A user agent is a software module that is temporarily or permanently installed in the memory of a sending client system during call setup or processing, and is a module that can be implemented in software called a flash. In other words, the service can be implemented using flash software, a general purpose general-purpose program. In this case, the call setup communication server 210 is provided with a flash application, and performs a call setup operation such as receiving a DN number usable in the exchange in place of the user agent. Therefore, the transmitting client system 100 can be temporarily installed even when the user does not perform a specific operation only at the time of a telephone call by a simple user agent interlocked with the flash agent and taking charge of only the flash phone function.

The user agent receives the URL information of the call setup server 250 at the moment the sender clicks a telephone icon or a button (telephone connection display) displayed on the web browser, sets up a call by connecting to the call setup server 250, And proceeds with a series of processes such as conducting a call with the receiving-side telephone 600 connected to the exchange 350 and stopping the call when the call between the sender and the receiver is completed.

The call setup server 250 includes a call setup web server 200 and a call setup communication server 210. The call setup web server 200 displays a telephone connection indication in which the URL of the call setup communication server and the LN are stored in the transmission side client system 100 and when the transmission side client system 100 clicks the telephone connection indication, And provides the client system 100 with the URL of the call setup communication server and the LN. The call setup communication server 210 is a server system that sets a temporary DN number to be assigned to the transmitting client system 100 and sets up a call between the exchanger 250 and the transmitting client system 100 and proceeds with the conversation . The call setup server 250 is composed of a memory and a normal server system having a plurality of operation control devices, and is connected to the transmission side client system 100 through an Internet network.

In the embodiment of FIG. 1, the call setup server 250 is configured to include the call setup web server 200 and the call setup communication server 210. However, the call setup server 250 may be configured as one server according to various hardware configurations, It goes without saying that the above-described server apparatus can be constructed.

The call setup communication server 210 communicates with the call setup web server 200 and the IP based telephone exchange 300 with each other. The call setup communication server 210 communicates with the user agent and relays it to connect the IP-based telephone switching network with the communication. Generally, the call setup communication server 210 and the user agent communicate based on the public IP, and the call setup communication server 210 and the IP-based telephone exchange 300 communicate with each other based on private IP or public IP. According to the present invention, the call setup communication server 210, which automatically allocates the DN when connected, eliminates the need for the transmission client system 100 to go through a login process in advance to use the service.

The user agent can not communicate with the receiving-side telephone 600 directly or indirectly connected to the IP-based telephone exchange 300 by itself. The reason is that the IP-based telephone exchange 300, which has to connect the calls to each other, can recognize the IP-based telephone exchange 300 in order to connect the receiver telephone 600 with the LN to the transmitting client system, This is because the connection information can not be understood unless the user agent uses a possible DN. Thus, the call setup communication server 210 selects the temporary telephone identifier (DN) from which the IP-based telephone exchange 300 can recognize the sending client system 100 and sends a call to the IP-based telephone exchange 300 Upon request, the IP-based telephone exchange 300 can perform a function of connecting a telephone call between the telephone identifier (DN) used by the transmitting client system 100 and the receiving telephone identifier (LN). That is, the user agent assigned the temporary telephone identifier (DN) is operated with the same function as the virtual telephone service provider's internal telephone.

The switching center 350 is a device for switching the transmitting-side client system 100 and the receiving-side telephone 600. The exchange 350 includes an IP based telephone exchange 300 operating on a packet network basis, a general telephone exchange 500 operating in a switched circuit network and an IP based telephone exchange 300, And a gateway 400 that plays a role of connecting a signal between the public telephone exchange 500 and the public telephone exchange 500.

The receiving-side telephone 600 is used as a generic term for a general phone connected to the ordinary telephone exchange 500, and a VoIP phone and a VoIP softphone connected to the IP-based telephone exchange.

The IP-based telephone exchange 300 is a telephone exchange based on the Internet protocol, and is a device that performs a function known as an existing telephone exchange on the basis of IP (Internet Protocol). It mainly manages call processing, signaling and dial plans, and manages the functions provided by the telephone. IP-based telephone exchanges are commonly referred to as IP PBXs.

The telephone 600 is a term collectively referred to as a telephone which is attached to the IP-based telephone exchange 300 and is capable of making a call or a general telephone connected to the telephone exchange 500. In the example of the telephone 600, (Also called "VoIP softphone") for installing a software module on a client computer and carrying out a call using a headset with a microphone, an Internet phone (or 'VoIP Phone ") and a public telephone. The telephone 600 may be any type of telephone connected to the exchange 350.

The configuration of the exchanger 350 has various configurations according to the type of the receiving-side telephone 600. [ For example, as shown in FIG. 1, when the receiving-side telephone 600 is composed of a general phone and an IP phone (used as a term collectively referred to as a VoIP phone and a soft VoIP phone), the exchange 350 exchanges information between the IP- And a general telephone exchange 500, and a gateway 400 is provided between both exchanges. 2 and 4, when the receiving-side telephone 600 is composed only of a general phone, the exchanger 350 is constituted by the gateway 400 and the normal telephone exchange 500 as shown in FIG. 4, the IP-based telephone exchange 300, the gateway 400, and the public telephone exchange 500 may be configured.

The call setup server 250 and the exchange 350 may be formed as a communication line as shown in FIGS. 1 and 2, or may be formed of an Internet network as shown in FIGS. The exchanger 350 and the receiving-side telephone 600 may be connected to a communication / communication line of wired / wireless communication as shown in Figs. 1, 2 and 4, or may be connected to the Internet network as shown in Fig. Of course.

In the present invention, it is preferable to use the SIP protocol for voice communication or signaling of a video call between the user agent and the receiving-side telephone, but the present invention is not limited thereto. Although the SIP protocol is an international protocol, 'Skype' is known to use its own protocol instead of using the SIP protocol.

When the telephone connection indication is provided to the transmitting client system 100 included in the body of the e-mail or the like, the transmitting client system 100 does not receive any information from the call setup web server 200 and directly transmits the call setup communication server 210 ), And the call setup process is performed.

2 is a block diagram of an Internet phone call connection system according to another embodiment of the present invention. The difference between the system configuration of FIG. 2 and the system configuration of FIG. 1 is that the call setup server 210 is connected to the public switched telephone network 500 via the gateway 400, And is connected to the exchange 500. In the case where only the ordinary telephone exchange 500 is used in the system of the service provider without using the IP-based telephone exchange 300 depending on the service provider, the connection is used as shown in FIG.

The connection between the call setup communication server 210 and the gateway 400 generally uses the SIP protocol and the connection between the gateway 400 and the public switched telephone exchange 500 is a telephone set used in a switched circuit network Use the connection method. The method of connecting the call setup communication server 210 directly to the normal telephone exchange 500 without using the gateway 400 is not generally used but the gateway 400 is connected to the call setup communication server 210. [ If the function is included, it can be applied.

3 is a block diagram of an Internet phone call connection system according to another embodiment of the present invention. The system configuration of FIG. 3 is similar to the system configuration of FIG. 3 except that the receiving-side telephone is subscribed to the VoIP phone service of the communication service provider and the service provider system does not have the IP-based exchange 300 or the general telephone exchange 400 ≪ / RTI >

The IP-based switching center 300 refers to an IP-based switching center 300 operated by a communication provider, and the receiving-side telephone 600 is a VoIP telephone 600 used by a service provider. Therefore, most of the IP-based telephone exchange 300 and the receiving-side telephone 600 of the communication provider are connected to the Internet and may be connected by a private line.

4 is a block diagram of an Internet phone call connection system according to another embodiment of the present invention. 3 is different from FIG. 3 in that a receiving-side telephone is subscribed to a general telephone service of a communication provider, and if the IP-based telephone exchange 300 or the general telephone exchange 400 is not provided in the service provider system, It is about how to do it.

The IP-based telephone exchange 300 means an IP-based exchange 300 operated by a communication carrier and includes a gateway 400 connecting the IP-based telephone exchange 300 and the public telephone exchange 500, and a public telephone exchange 500 The receiving-side telephone 600 is a general telephone. It is also technically possible to directly connect the call setup communication server 210 with the gateway 400 or the public telephone exchange 500. [ However, in most cases, when the communication company is used for data communication connection with the outside, it is common that the communication company is linked with the IP-based telephone exchange 300.

The call setup server 250 shown in FIGS. 1 to 4 may be configured to configure the call setup web server 200 and the call set up communication server 210 as a service provider internal system or an external third party system Of course. The sending client system 100 and the call setup communication server 210 may be connected to each other by a public IP address, but may be connected to a private IP network if the user is connected by a private line or is inside the service provider system. The call setup communication server 210 and the IP-based telephone exchange 300 or the public telephone exchange 400 can be implemented in any manner based on a public IP-based or a private IP-based.

It is needless to say that additional modifications regarding the DN issuance, which is a temporary number used in the user system 100 in the embodiment shown in FIGS. 1 to 4, are also possible. Although the DNs are allocated in the call setup communication server 210 in the drawings and corresponding descriptions shown in FIGS. 1 to 4, it is needless to say that the DN number issuance logic can be incorporated into other parts or separately configured. For example, a call setup Web server 200, an IP-based telephone exchange 300, a public telephone exchange 500, or a third system interworking with a service provider system, to the transmitting client system 100 Logic can be implemented.

5 is a flowchart illustrating a call connection using an Internet phone call connection system according to an embodiment of the present invention. 5, the call setup server 250 is divided into a call setup web server 200 and a call setup communication server 210, and the call setup server 250 includes a single integrated server Only the flow with the difference will be described.

Step 501 is a step in which the transmitting client system 100 accesses the call setup web server 200 through a web browser. That is, the call setup web server 200 displays the telephone connection indication including the URL information of the call setup communication server 210 and the LN information through the web browser of the client system 100. In addition to the web browser, the display of the telephone connection can be displayed through e-mail. The telephone connection indication may be provided in various ways such as a text form, an image form, or a link statement. FIG. 6 is an example of a telephone connection indication displayed on a transmitting side client through a web browser and an example of a telephone connection indication displayed on a transmitting side client via e-mail. 6A shows telephone connection marks 601 and 603 provided in the form of images and text, respectively, and FIG. 6B shows an example of a telephone connection mark 605 provided in an image form. Although the telephone connection display shown in FIG. 6 apparently does not include the URL and LN information of the call setup communication server 210, the corresponding telephone connection display includes an execution statement executed when the telephone connection indication is clicked as the meta information The URL of the call setup communication server 210 and the LN information.

Step 503 is a step in which the client clicks a telephone connection indication that the client wants to make a call in the connected transmitting client system 100. As soon as the client clicks the telephone connection display, the call setup web server 200 transmits the URL information and LN information of the call setup communication server 210 to the transmitting client system 100.

In step 505, after the web browser of the transmitting client system accesses the web server function existing in the call setup communication server 210 using the URL information of the call setup communication server 210, the transmitting client system 100 Means that the index.html storing the order of jobs to be connected to the communication server 210 is downloaded. In general, one server system includes various functional logic, and the call setup communication server 210 also includes logic to provide a web server function (providing information when a web browser is connected).

In step 507, a web browser of the transmitting client system recognizes the tag, and a port of a web server function using the 8080 port in the call setup communication server 210 To download a user agent that performs a call function. In the embodiment of the present invention, since the phone call is performed using the flash function, the client does not need to download a separate program in the case of the transmitting client system having the flash player. That is, in the call setup communication server 210, the call setup communication server 210 temporarily uses a user agent (hereinafter referred to as a 'flash phone') in the transmitting client system by using the web server function using the port 8080 ) Port to perform a call function.

Step 509 is a step in which the flash phone automatically downloaded to the transmitting client system is automatically connected to the flash server and the flash application existing in the call setup communication server 210.

Step 511 is a procedure in which the DN is assigned while the flash phone performs the automatic login function. That is, when the flash phone attempts to log in (meaning access to the flash application), the flash application inside the call setup communication server 210 receives a request from the SIP server having the logic function of assigning the DN number used by the user system It attempts to log in with a DN number assigned to it. In the case of a Skype or a Nate phone, a user logs in by inputting an ID and a password before dialing. In the present invention, the user of the client system does not need to input a separate login and password. Therefore, the client system can make a call even if it is not registered as a member of the Internet phone call connection system of the present invention.

However, the assignment logic of the DN number assigned by the SIP server can be easily configured to be performed in the call setup web server 200, the other logic part in the call set up communication server, or the third independent server according to the usage needs.

Step 513 is a procedure for checking whether the connected flash phone has an LN number to which a connection is desired. Step 515 is to check the LN number and make a telephone connection. Step 517 is a procedure for checking whether or not the telephone is connected. Step 519 is a step in which the call is terminated first in the DN after the telephone conversation between the LN and the DN. In step 521, the call is terminated first in the LN after a telephone call between LN and DN. Step 523 is a procedure for displaying a termination screen to the client when the flash phone call is terminated, and Step 525 is a procedure for providing a standby screen for requesting the client system to input the LN number.

In the case of Step 525 in the above-described step, after the proceeding of Step 523, the sender client system 100 tries to input new LN information by pressing a Reset button or the like. In this case, if the call setup web server 200 inputs counterpart LN information to be called and clicks the call button, the procedure proceeds from Step 513 shown in FIG. Since the client system 100 is connected to the call setup communication server 210 and has already been assigned a DN number, it is not necessary to perform steps 505, 507, 509, and 511 again.

In yet another example, a more modified embodiment will be described. The Internet phone call connection system according to the present invention does not provide a telephone connection indication, and the call setup server 250 provides a web site to the transmission side client system and inputs counterpart LN information to be called. FIG. 7 shows an example of a Web page that is provided to a transmitting client system from a call setup server and that inputs LN information after visiting a Web site. When the client inputs the LN information (e.g., vicwave-0126) and clicks the call button, the procedure proceeds from Step 505 shown in FIG.

If the call setup server 250 is provided as one call setup server 250 that is not separated into the call setup web server 200 and the call setup communication server 210 but has two functions integrated, Step 507 without going through it. This is because when the transmitting client system 100 accesses the call setup server 250 through the web browser, it is not necessary to transmit the URL information of the server. Also, the telephone connection display includes only the LN information and does not need to include the URL information of the call setup server 250. [ In step 503, the client proceeds to step 507 without going through step 505 as soon as the client clicks the telephone connection display.

Even if the call setup server 250 is provided as one call setup server 250 that is not separated into the call setup web server 200 and the call setup communication server 210 but has two functions integrated therein, If the client clicks the telephone connection display when it is displayed through e-mail, it goes through Step 501 shown in FIG. 5.

FIG. 8 is a flowchart of call set-up and call connection in the Internet phone call connection system according to the present invention. 8 is a flowchart illustrating a call setup and call connection flow based on the Internet phone call connection system shown in FIG. The call setup step (a), which is the most important feature of the present invention, will now be described. The call setup step (a) of FIG. 8 is a flowchart for explaining steps 509, 511, and 515 in FIG. 5 in detail. When the user agent of the sending client system 100 is connected to the call setup server 250, the flash application of the call setup server 250 sends the SIP server a temporary DN to be assigned to the sending client system 100 (Get User ID), the SIP server gives the flash application a temporary DN (User Id). Then, the sender client system is logged into the SIP server (REGISTER), and a confirmation of the register request is sent (200 OK). The Register step and the 200 OK step are commonly performed in all SIP communications.

Next, the sending client 100 user agent requests a call corresponding to the LN to the flash application, and the flash application sends the receiving telephone 600 through the SIP server and the IP- the call setup step (a) is completed through invite and ringing steps. The invite is then sent to the SIP server in the form of an invite LN number (invite to LN).

The user agent and the flash application of the transmitting client system 100 proceed in a net stream manner and the flash application and the receiving terminal 600 perform an rtp stream, . In the call progress step (c), the flash application and the receiving-side telephone 600 are directly connected to the Internet to perform packet communication, but the packet communication may be performed through the IP-based telephone exchange 300 . In the call termination step (d), all the devices are notified that the call has ended.

9 is a detailed configuration diagram and DN assignment flowchart of the transmitting client system and the call setup server. The call setup server shown in FIG. 8 has a flash application and a SIP server. However, the call setup server shown in FIG. 9 and FIG. Similarly, a flash server is provided. In FIG. 8, the flash server has been omitted for convenience of explanation.

The call client system 100 is provided with a user agent (referred to as a "flash phone" in the case of flash implementation), and the call setup server 250 is provided with a flash server, a flash application and a SIP server. More preferably, the flash server, the flash application, and the SIP server are preferably provided in the call setup communication server 210 among the call setup servers. When the user agent accesses the flash server (ST1), the flash server delivers it to the flash application. The flash application requests the DN from the SIP server (ST2), and the SIP server transmits the DN to the flash application (ST3). At this time, the SIP server must store the DN available in the exchange 350 in advance in the memory. Alternatively, when the SIP server receives the ST2 step request, the SIP server may request a usable DN to the exchange 350 or the third server, and may proceed to transmit the DN transmitted from the exchange or the third server to the SIP server have. The SIP server then completes the automatic assignment of the temporary DN by logging in the client. Next, the transmitting client system 100 transmits an LN while requesting a call (ST4). The flash server transmits the call request and the LN information to the flash application. The flash application requests the SIP server to invite the corresponding LN (ST5), and invites the SIP server to the exchange 350 to invite the corresponding LN (ST6).

As described with reference to FIG. 9, the call setup server 250 automatically grants the DN assigned to the flash application and used by the user agent even if there is no request from the client, and the granted DN is not delivered to the user agent have. The granted DN is stored in the flash application, and the flash application and the user agent maintain the connection through the flash server. Such a connection configuration is a method commonly used in the implementation of a game program implemented by a flash program, and thus a detailed description thereof will be omitted.

While specific embodiments of the invention have been illustrated and described, it will be obvious that the same may be varied in many ways by those skilled in the art without departing from the spirit of the invention. Such modified embodiments are not to be understood individually from the spirit and scope of the present invention, but are to be regarded as falling within the scope of the appended claims. In the present invention, description has been made mainly on voice call, but it can be easily applied to a video call. In this case, the receiving-side telephone 600 will refer to the video phone and use a protocol suitable for the video instead of the SIP protocol.

100: transmitting side client system 200: call setup web server
210: call setup communication server 250: call setup server
300: IP-based telephone exchanger 350:
400: gateway 500: public telephone exchanger
600: receiving-side telephone

Claims (1)

A call setup communication server that sets up a call between the transmitting client system and the receiving-side telephone using the exchange, and a call set-up communication server that sets up a call between the transmitting client system and the receiving- And a call setup web server for providing the sending side client system with the URL information of the calling terminal and the receiving side telephone system identification information, ,
Wherein the call setup web server displays a call connection indication that includes the call setup communication server URL information and the receiver telephone identification information and indicates a receiver to which the call is to be made, After providing the call setup communication server URL information and the receiver telephone identification information to the transmitting client system,
The call setup communication server includes a first step of setting a transmitting client identifier for identifying the Internet phone, which is recognizable by the exchange and is distinguishable from other Internet phones, ; And
The call setup communication server includes a second step of receiving the receiver-side telephone identification information from the transmission-side client system and transmitting the received receiver-telephone identification information to the exchange so as to set up a call line,
Wherein the first step is performed without using the information input from the user of the transmitting client system to identify the transmitting client system to the other Internet phone for connection to the Internet phone call,
Wherein the transmitting client identifier is maintained while an Internet connection between the call setup communication server and the sending client system is maintained.

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