KR100588480B1 - Internet based telephone communication system and communication method thereof - Google Patents

Abstract

The present invention relates to an internet telephony communication system and an internet telephony communication method. The Internet telephony communication system of the present invention can transmit and receive a voice signal with a counterpart through an internet network, send and receive a voice signal with a general telephone through a public telephone network, and at the request of a general telephone connected through a public network. A plurality of internet telephones for establishing a call path to a general telephone and a counterpart internet telephone, and at least one call server capable of connecting to the plurality of internet telephones through an internet network and managing the plurality of internet telephones; It is composed. Accordingly, the system of the present invention provides a high quality telephone-to-phone Internet phone system in which any Internet phone can talk to a general telephone through a public telephone network, and can also talk to other Internet phones through the Internet network. It can be implemented, and it is possible to talk with a general telephone connected through a public telephone network and another internet telephone connected through an internet network without installing a separate internet phone gateway between the Internet network and a public payphone network.

Description

Internet telephony system and communication method {INTERNET BASED TELEPHONE COMMUNICATION SYSTEM AND COMMUNICATION METHOD THEREOF}

1 is a view showing the overall concept of the Internet telephony communication system of the present invention.

FIG. 2 is a block diagram of the main circuit configuration of the Internet telephone shown in FIG.

FIG. 3 is a diagram showing a signal flow in a general telephone call mode of the Internet telephone shown in FIG.

FIG. 4 is a diagram showing the signal flow of the Internet telephone call mode of the Internet telephone shown in FIG.

FIG. 5 is a diagram schematically showing the registration procedure of the Internet telephone shown in FIG.

6A to 6C are diagrams schematically illustrating the procedure of the call mode from the calling Internet phone to the called Internet phone as the Internet phone call mode of the Internet phone shown in FIG.

7A to 7C show an Internet telephone call mode of the Internet telephone shown in FIG. 1, wherein the calling Internet telephone is used as a gateway to a general analog telephone connected to the called Internet telephone through a public telephone network. Is a diagram schematically illustrating a procedure of a call mode.

FIG. 8 is a signal flow diagram of a gateway mode for implementing a communication connection between a general analog telephone connected to the Internet telephone shown in FIG. 1 through a public telephone network and another Internet telephone.

9A to 9C are gateway call modes of the Internet telephone shown in FIG. 1, and a call from a general analog telephone connected to a calling Internet telephone to a called Internet telephone using the calling Internet telephone as a gateway; A diagram schematically illustrating the procedure of the mode.

10A to 10C show a gateway call mode of the Internet telephone shown in FIG. 1, in which a general analog telephone connected to a calling Internet telephone is used as a gateway and the calling Internet telephone is used as a gateway. 2 is a diagram schematically illustrating a procedure of a call mode to a general analog telephone connected to a called Internet telephone through a public telephone network.

FIG. 11 shows a dual channel for enabling communication connection between a general analog phone and a third Internet phone connected to the first Internet phone through a public telephone network simultaneously with a call connection between the first Internet phone and a second Internet phone. channel) A diagram illustrating a signal flow in a call mode.

<Explanation of symbols for the main parts of the drawings>

NP11 to NP1n, NP21 to NP2n: Analog phone IP, IP1 to IPn: Internet phone

CS: call server 10: handset

12: speakerphone 14: external voice input and output connection

16: multiplexer 18: first codec

20: digital signal processor 22: second codec

24: data connection arrangement (DAA) 26: telephone line connection

28: central processing unit (MCU) 30: I / O buffer

32: ROM 34: call intention detection unit

36: DRAM 38: keypad

40: display 42: digital network connection

The present invention relates to an internet telephony communication system for implementing a telephone-to-phone internet phone and a communication method thereof.

One of the most important means of communication in everyday life is the telephone. In particular, telephones have the highest status in the communication of business or business activities with family or friends in the distance. However, in recent years, the so-called Internet phone, which is a combination of the existing general phone and the Internet in a telephone, especially an international phone, has replaced the ordinary phone.

The types of internet phones that are currently put into practical use are classified into PC-to-PC and PC-to-Phone. The biggest advantage of these phones is that they can be used at low cost. have. Such low rates can be said to be at the heart of the low cost implementation of packet transmission.

In addition, since the Internet phone uses a public Internet network, it is possible to provide a service by connecting an Internet dedicated line to a domestic Internet Service Provider (ISP) and installing Internet phone equipment. The service can be provided at a significantly lower investment cost than the existing public telephone network.

The first Internet phone was developed by Vocaltec, a PC that connects PCs through the Internet and sends and receives voices through microphones and speakers (headsets). Two people who want to talk on the phone are connected to the Internet at the same time, and they are connected to Bocaltech's Internet phone server through the same program. And compress it (compress 64K into 5.3K, 6.3K, 8K; G723, G729 method) and use the transmission protocols such as H323, RSVP, RTP / RTCP, When the call is sent to the other party's PC, the call is made by decompressing programmatically and converting the digitized voice signal into an analog voice signal and restoring the voice through the speaker.

However, the PC-to-PC method was not as good as expected because of the user's PC performance, the state of the Internet line, and the performance of the program. In addition, there are many inconveniences in opening the era of full-fledged Internet phones because there are limitations such as whether two people who want to use the phone must use the same program or have to set a time to use the Internet phone in advance.

Therefore, the developed method is to connect a PC and a regular telephone. This is installed on a PC and connected to the Internet network, and then connected to a public telephone network (PSTN) through a device called an Internet Phone Gateway installed in a specific area. It is a method of talking with a regular telephone by connecting. The person who calls through a PC is the same as the PC-to-PC method mentioned above, but the receiver is called by a general telephone, which is an advanced method compared to the PC-to-PC method. While the PC-to-PC method described above is all done using the Internet method, the PC-to-phone method uses the Internet method for making a phone call and the caller makes the call using an existing phone method. In other words, the role of the Internet phone gateway is to connect the Internet with the existing public telephone network. However, when connected to the place where the call is received, the existing public telephone network is used, so the caller or the person who brokers the call has no choice but to pay for the use of the telephone network. In addition, the PC-to-telephone method is inconvenient to connect to the Internet phone from the general telephone side.

Therefore, such a PC-to-phone method is not enough to replace the existing phone. After all, in order to replace the existing telephone method, it must be a telephone-to-phone method, and the method must install a gateway that connects the telephone network with the Internet network so that the PC side of the aforementioned PC-to-phone system can be used as a telephone. will be. In other words, this method can be implemented by a gateway that can be connected by a public telephone network-Internet network-public telephone network, and can make telephone calls, and can connect and detect signals between networks. Therefore, the development of equipment such as Internet phone gateway is no longer a program on a PC anymore, and existing world-class exchange development companies are jumping into equipment development. High-performance equipment, large-capacity Internet backbone networks, operational technologies, and more advanced protocols still need to be addressed. However, the telephone-to-phone system that implements the Internet phone using such a regular phone requires installation of a very expensive device called an Internet phone gateway, so it is questionable whether it is possible to make an international call at the current local call rate.

Accordingly, the present invention has been made in view of the above circumstances, and an object of the present invention is to provide an internet telephony communication system and a communication method capable of implementing a complete telephone-to-telephone Internet phone.

In addition, another object of the present invention is to provide an Internet telephone communication system and a method of communicating that an Internet telephone can make a call with a general telephone through a public telephone network and also with another Internet telephone through an internet network. will be.

In addition, another object of the present invention is a call between a general telephone connected to the Internet telephone through a public telephone network and another Internet telephone connected through the Internet network without installing a separate Internet phone gateway between the Internet network and the public telephone network. It is to provide an Internet telephony communication system and a method of communication thereof.

Still another object of the present invention is to provide a general telephone connected to a first Internet telephone through a public telephone network, when a telephone connection is made with a second Internet telephone connected through an Internet network using the first Internet telephone as a gateway. It is an object of the present invention to provide an internet telephony communication system and a communication method capable of conducting a two-channel call in which the first internet telephone is connected to a third internet telephone.

The present invention for achieving the above object is, in the Internet telephony communication system, the analog voice input and output means and the call path by the call from the telephone with the other Internet phone through the Internet network and the public telephone network A call path is established between a general telephone requesting a call and a call path between the general telephone requesting the call and the other party's Internet telephone through the Internet network by a call request from a general telephone through a public telephone network. A plurality of Internet telephones that send and receive signals,
Connect to the plurality of Internet telephones through an internet network, manage the plurality of Internet telephones, receive a call connection grant message from the caller's Internet telephone, determine whether the call is connected, and send a connection grant message to the caller's Internet telephone. This may be accomplished by having at least one call server transmitting to.
On the other hand, according to another aspect of the present invention, the object is a plurality of Internet telephones that can be connected via the Internet network to exchange voice signals with the other party, and one call to manage the plurality of Internet telephones through the Internet network In a communication method in a system provided with a server,
When the user connects to the Internet telephone using a regular telephone, the central processing unit of the Internet telephone recognizes this;
Requesting a specific key input by the central processing unit to select one of a normal telephone call mode and an internet telephone call mode;
A first gateway call mode in which the connected ordinary telephone is in communication with the called Internet telephone via the calling Internet telephone in the connected ordinary telephone; Requesting a specific key input for selecting one of the second gateway call modes for talking with another general telephone through the called Internet telephone;
In the case of the key input of the first gateway call mode, the central processing unit requests input of an identification number of the other party's Internet telephone.
In the case of a key input of the second gateway call mode, the central processing unit requesting input of an identification number of the counterpart Internet telephone and a telephone number of the counterpart general telephone;
The central processing unit may also be achieved by detecting a tone signal by key input in the ordinary telephone and setting the first gateway call mode or the second gateway call mode according to the detection result.
Hereinafter, with reference to the accompanying drawings will be described in detail a preferred embodiment of the Internet telephone communication system and a communication method according to the present invention.

1 is a view showing the overall concept of the Internet telephony communication system of the present invention. In the figure, a plurality of Internet telephones IP1 to IPn are connected via an internet network, and a call server CS which mediates the calls of these Internet telephones IP1 to IPn is connected to the Internet network. . Each of the Internet phones IP1 to IPn is connected to the general phones NP11 to NP1n, NP21 to NP2n through the public telephone network PSTN.

FIG. 2 is a block diagram of the main circuit configuration of one example of the Internet telephone shown in FIG. This Internet telephone includes analog voice input / output means, first signal conversion means, code / decoding means, central processing means (MCU), digital network connection means, data storage means, key input means, display means, and call intention detection means. It is configured by.

The analog voice input / output means includes an external voice input / output connection unit 14 to which a handset 10, a speakerphone 12, and an external voice input / output device such as a headset can be connected, and the voice input / output devices 10, 12, 14 are connected. It is configured to include a multiplexer (MUX) 16 to selectively switch the call of.

The first signal converting means converts an analog audio signal input through the voice input / output means into a digital signal, and simultaneously converts a digital signal from the code / decoding means into an analog signal (CODEC1) 18 It is configured to include).

The code / decoding means compresses and encodes the digital voice signal of the sender input through the first codec 18, and simultaneously decompresses and decompresses the compressed coded signal of the other party, which is input through a central processing means described later; A digital signal processor (DSP) for generating a digital tone signal corresponding to a DTMF tone signal under the control of a central processing means described later, for determining an input tone signal, and for performing an echo cancellation function. 20 is configured to include. Here, the digital signal processor 20 performs signal compression encoding and signal decompression decoding using, for example, G.711, G723, G.726, G.727, G.728, and G729 methods.

The central processing unit (MCU) as the central processing means uses the H.323 protocol, for example, call signaling such as registration processing of its own Internet telephone and call request with other Internet telephones to the call server CS. A packetizer processing for packetizing the compressed speech coded signal by the digital signal processor 20, and at the same time, decomposing the packetized speech coded signal inputted through the digital network connection means. It is in charge of packetizer processing and overall control of the telephone set of the present invention.

The digital network connection means serially outputs a transmission signal transmitted from the central processing unit 28 through a digital network connection unit 42 for connecting to a digital network such as a LAN or a WAN connected to the Internet network. At the same time, the input / output buffer 30 which transmits serial data input from the digital network connection unit 42 via the LAN or WAN from the Internet network to the central processing unit 28 in units of packets, and LAN or As a registered jack for connecting to a digital network, such as a WAN, it is comprised including the digital network connection part 42 which consists of RJ-45, for example.

The data storage means stores a control program of the central processing unit 28, and includes a nonvolatile memory 32 capable of recording and reading, a DRAM 36 and a flash memory (not shown) for storing various data. It is composed. In addition, the key input means comprises a keypad 38 including a numeric key for inputting a telephone number and the like, and various function keys installed on the exterior of the telephone as shown in FIG. The input signal generated accordingly is transmitted to the central processing unit 28. In addition, the display means includes a liquid crystal display panel for displaying a setting state, an operating state, and the like of the telephone of the present invention under the control of the central processing unit 28. In addition, the call intention detecting means detects the hook on / off state of the handset 10 of the telephone of the present invention, the operating state of the speakerphone 12 and the operating state of the external voice input / output connection unit 14, and the like. 28 is a call intent detection unit 34 for transmitting to.

Here, the H.323 protocol basically uses Transmission Control Protocol / Internet Protocol (TCP / IP), and TCP / IP supports TCP and User Datagram Protocol (UDP) that can be used by an application. In the central processing unit 28, for example, UDP (User Datagram Protocol) in call signaling processing with a call server CS, TCP in call signaling processing with other Internet telephones, and RTP (Real-) for transmission of voice data. time transport protocol). The RTP transmits voice data to a network using UDP.

First, UDP (User Datagram Protocol) processing of the central processing unit 28 at the time of transmitting the call signaling message to the call server CS will be described. In UDP, the call signaling message is attached to the IP layer by adding a UDP header including the port number of the other party's application (i.e., the call server) and the port number of its own application to the IP layer in the form of a packet. It adds IP header including address and IP address of counterpart (ie call server) and drops it to Ethernet device driver in packet form.In Ethernet device driver, Ethernet protocol (for example, IEEE 802.2 / 802.3 protocol) Header is converted into a packet having a protocol format for communication with a local gateway connected through a LAN, and outputted to the input / output buffer 30.
On the other hand, the UDP processing of the central processing unit 28 at the time of receiving the call signaling message from the call server CS decomposes the packet in the reverse order of the above-described transmission processing, and determines whether the data is correct using its own header for each layer. Then removes its header and puts the packet to the upper layer.

The TCP processing of the central processing unit 28 at the time of transmitting the call signaling message to the other Internet telephone will be described. In TCP, the call signaling message is sent to the IP layer in the form of a packet by attaching a TCP header including the port number of the application of the other party (that is, the Internet phone of the other party), the port number of its application, and the transmission confirmation field. It adds the IP header including the IP address of the other party and the IP address of the other party (that is, the other party's Internet phone) to the Ethernet device driver in the form of a packet.In the Ethernet device driver, the Ethernet protocol (for example, IEEE 802.2) header is attached to The data is converted into a form and output to the input / output buffer 30.
On the other hand, the UDP processing of the central processing unit 28 at the time of receiving the call signaling message from the other Internet telephone is processed in the reverse order of the above-described transmission processing.

The following describes the Real-time Transport Protocol (RTP) processing of the central processing unit 28 during voice data transmission with the other Internet telephone. In RTP, the compressed and encoded voice data is attached by attaching an RTP header including a voice data format and a sequence number, and in UDP, the port number of an application of the other party (ie, the other party's Internet phone) and the port number of its own application are included. Attach the UDP header to the IP layer in the form of a packet, and at the IP layer, add an IP header including its own IP address and the IP address of the other party (ie, the other party's Internet phone), and drop it to the Ethernet device driver in the form of a packet. In the Ethernet device driver, an Ethernet protocol (e.g., IEEE 802.2) header is attached, converted into a packet form, and output to the input / output buffer 30.
On the other hand, the RTP processing of the central processing unit 28 at the time of reception of voice data from the other party's Internet telephone is processed in the reverse order of the above-described transmission processing.

By constructing the present system using the Internet phone as described above, it is possible to implement a phone-to-phone Internet phone between the Internet phones through the Internet network.

On the other hand, the Internet phone of Figure 1 can be configured to further include a public telephone network connection means and the second signal conversion means in the above configuration.

The public telephone network connection means is a registered jack (RJ) for telephone line connection, and a telephone line connection portion 26 made of, for example, RJ-11. Data access arrangement (DAA) for detecting a ring signal from an external telephone line, hooking on / off with an external telephone line under the control of the central control unit 28, separating and synthesizing an analog handset voice signal, and the like. Arrangement 24).

The second signal converting means converts an analog audio signal from a general telephone separated from the data connection array 24 into a digital signal, and simultaneously converts a digital signal from the digital signal processing unit 20 into an analog signal. It is configured to include a second codec (CODEC2) (22).

Here, the signal transmission of the digital signal processing unit 20 and the first codec 18 / second codec 22 is performed in a time slot manner, and the digital signal processing unit 20 has a first codec 18 in two channels. Audio signal from the second codec 22 and the audio signal from the second codec 22 are compressed and transmitted to the central processing unit 28. Here, the time slot method is a method of simultaneously transmitting and receiving multiple signals simultaneously on a signal line in a time-division manner, and transmitting / receiving signals of the first codec 18 and the digital signal processor 20 and the second codec 22 and the digital signal processor. The signal transmission / reception at 20 is performed in different time slots, and the central processing unit 28 divides the signal transmitted from each channel of the digital signal processor 20 into the requested counterpart IP address. In addition, the digital signal processing unit 20 further includes a function of monitoring a call of a general telephone connected through a public telephone network (PSTN) and a function of detecting a DTMF tone signal input from the general telephone.

When the system of the present invention is constructed using the above-described Internet telephone, not only calls between Internet telephones, but also calls between Internet telephones and ordinary telephones via public telephone networks, and general telephones connected to public telephone telephones through a public telephone network Calls to other Internet phones connected through the Internet network, and other Internet phones connected through the Internet network as well as general telephones connected to the Internet phone through a public telephone network simultaneously with calls from other Internet phones to other Internet phones. You can implement a call with

Next, the operation of the Internet telephony communication system constructed using the above-described Internet telephone is explained in detail.

First, the registration processing operation of the Internet telephone in the Internet telephony communication system of the present invention will be described with reference to Figs.

When the user operates the Internet phone (for example, IP1 or IP2) in an operable state by an operation such as power input, the central processing unit 28 undergoes an initialization operation, and then the call server request message ( Call-Server Request message (S1 or S1 'of FIG. 5) is output to the Internet through the input / output buffer 30 through the UDP protocol and transmitted to the call server CS.

Here, the call server request message (S1 or S1 ') is a message sent by the Internet phone (IP1, IP2) through the Internet network to find its call server (CS), for example, the information shown in Table 1 below Elements, and so on.

Information element name function Request sequence number Indicates the sequence number of the message. Protocol identifier Indicates protocol information of the endpoint. RAS address Indicates the transport address that the endpoint will use for RAS. Endpoint type Indicate the type of endpoint to register with the call server. Nonstandard Data Indicates information that is not defined in the specification.

As shown in Table 1, the Request Sequence Number information element indicates the sequence number of this message, and the response message associated with this message should have the same sequence number. The Protocol Identifier information element indicates information of a protocol (for example, ITU-T Recommendation H.225.0) for endpoints of Internet telephones IP1 and IP2. In addition, the RAS (Remote Access Services) address information element is a transport address used by an endpoint of the Internet telephones IP1 and IP2 to receive a RAS message, that is, a call server acknowledgment message that must be received from the call server CS. This indicates the address to receive a Call-Server Confirmation message. The transport address, which is the RAS address, may be configured to include its own IP address and application port number. Here, the own IP address may be set by an input by a user when the Internet telephone is installed or by an input method using a Dynamic Host Configuration Protocol (DHCP) protocol.

In addition, the endpoint type information element indicates a type of an endpoint, which is set to a terminal.

Here, the endpoint of the Internet telephone in the present invention can be defined as two, one is a terminal and the other is a gateway. Therefore, the registration procedure requires registration with the call server CS for each of the two endpoint types of the Internet phone. In the following description, only the registration procedure for the case where the endpoint type of the Internet phone is a terminal will be described for simplicity. In reality, the same registration procedure is performed even when the endpoint type of the Internet phone is a gateway. Endpoint Type Information elements can be distinguished according to whether they are terminals or gateways.

Then, when the call server CS receives the call server request message S1 or S1 ', the call server CS sends a Call Server Confirmation message (S2 or S2' in FIG. 5). It transmits to internet phone (IP1 or IP2) through internet network using UDP protocol.

Here, the call server confirmation message is a message sent to the Internet telephone (IP1 or IP2) corresponding to the case where the call server CS sends the call server request message to the Internet telephone (IP1 or IP2). For example, it includes information elements as shown in Table 2 below.

Information element name function Request sequence number Indicates the sequence number of the message. Protocol identifier Indicates the protocol identifier of the call server. RAS address Indicates the transport address that the call server will use for RAS.

As shown in Table 2 above, the Request Sequence Number information element indicates the sequence number of this message and must have the same sequence number as the call server request message (S1 or S1 ') related to this message. . The Protocol Identifier information element indicates information of a protocol (eg, ITU-T Recommendation H.225.0) of the call server CS. In addition, the RAS address indicates a transport address used by the call server CS to receive the RAS message, that is, an address to receive the RAS message from the corresponding Internet telephone IP1 or IP2. The transport address, which is the RAS address, may be configured to include an IP address and an application port number of a call server.

Here, the RAS message may include a call server request message, a call server confirmation message, a registration request message, a registration confirmation message, a registration request message, an access request message, and an access confirmation message. message), a disconnect request message, a disconnect confirmation message, and the like. In this specification, a message transmission flow for the other RAS messages is provided only for simplification of description. Omit the description. Such RAS messages are preferably followed by the RAS messages mentioned in, for example, ITU-T Recommendation H.225.0.

Subsequently, when the Internet phone IP1 or IP2 receives the call server confirmation message S2, S2 ', the central processing unit 28 of the Internet phone IP1 or IP2 confirms this and then requests registration through the Internet network. A message (Registration Request message) (S3 or S3 'in FIG. 5) is output to the Internet through the input / output buffer 30 using the UDP protocol, and transmitted to the call server CS.

Here, the registration request message (S3 or S3 ') is a message sent by the Internet phone (IP1 or IP2) through the Internet network to register itself with the call server (CS), for example, the information as shown in Table 3 below Elements, and so on.

Information element name function Request sequence number Indicates the sequence number of the message. Protocol identifier Indicates protocol information of the endpoint. Call signaling address Indicates the call signaling transport address of the endpoint. RAS address Indicates the transport address that the endpoint will use for RAS. Terminal type Indicate the type of terminal to register with the call server. Terminal alias Indicate an alias for an endpoint.

As shown in Table 3, the Request Sequence Number information element indicates the sequence number of the present message, and the response message related to this message should have the same sequence number. The Protocol Identifier information element indicates information of a protocol (for example, ITU-T Recommendation H.225.0) for an endpoint of an Internet telephone (IP1 or IP2).

In addition, the call signaling address information element may be a call signaling transport address (for example, an IP address of an Internet telephone) for receiving a call signaling message at an endpoint of the corresponding Internet telephone (IP1 or IP2). And an application port number), for example, a call signaling message specified in ITU-T Recommendation H.225.0 (specifically, a call specified in ITU-T Recommendation Q.931). Signaling messages, etc.).

In addition, the RAS address information element may include a transport address (including an IP address of the Internet phone and an application port number) that will be used by the endpoints of the Internet phones IP1 and IP2 to receive a RAS message, that is, a call server ( It indicates an address to receive a Registration Confirmation message from CS.

In addition, the Terminal Type information element indicates the type of endpoint of the Internet telephone to be registered with the call server, which is a terminal. On the other hand, if the endpoint of the Internet telephone to be registered in the call server is a gateway, this information element is a gateway.

In addition, the terminal alias information element indicates an alias of the Internet telephone. The terminal alias information element may comprise, for example, an H.323 identifier (ID) and an E.164 according to ITU-T Recommendation H.225.0. In the H.323 identifier, an identification code (for example, a serial number) of an Internet telephone is recorded, and E.164 is set to null so that the terminal alias information element is configured, the call server allocates the E.164. You can return it.

Next, upon receiving the registration request message (S3 or S3 '), the call server CS sends a registration confirmation message (S4 or S4' of FIG. 5) to the Internet through the Internet network using the UDP protocol. Send to the phone (IP1 or IP2).

Here, the registration confirmation message (S4 or S4 ') is a message that means to allow registration to the Internet phone (IP1 or IP2) that sent the registration request message, for example, including the information elements shown in Table 4 below It is composed.

Information element name function Request sequence number Indicates the sequence number of the message. Protocol identifier Indicates protocol information of the call server. Call signaling address Indicates the call signaling transmission address of the call server. Terminal alias Represents a list of aliases. Endpoint identifier Indicates the identifier of the terminal.

As shown in Table 4 above, the request sequence number information element indicates the sequence number of this message and should have the same sequence number as the registration request message S3 or S3 'associated with this message. The protocol identifier information element indicates information of a protocol of the call server CS (for example, ITU-T Recommendation H.225.0).

The call signaling address information element indicates an address for receiving a call signaling message in the call server CS (including, for example, an IP address of the call server and an application port number). Here, the call signaling message may be delivered between the Internet telephones via the call server CS or directly between the Internet telephones without the call server. The call signaling address information element may be a call. It can be used when a signaling message is delivered via a call server.

In addition, the terminal alias information element indicates an alias for the Internet telephone (IP1 or IP2) that sent the registration request message, which is a terminal. In this terminal alias information element, an identification code (e.g., a serial number) of an Internet telephone is described in an H.323 identifier, and an identification code (e.g., a phone number) assigned by a call server is described in an E.164. .

The endpoint identifier information element is a locally unique one assigned by the call server CS, and indicates an identifier to be used later by the Internet telephone (IP1 or IP2) that sent the registration request message. Here, two endpoint identifiers are assigned to the Internet telephone as the endpoint identifier information element. That is, they are allocated for the case where the endpoint is a gateway and the terminal, respectively. In this case, since the endpoint is a terminal, the endpoint identifier information element is assigned an endpoint identifier for the terminal.

Then, upon receiving the registration confirmation message S4 or S4 ', the central processing unit 28 of the Internet phone IP1 or IP2 checks this and recognizes that its Internet phone has been registered with the call server CS. . Therefore, the internet telephone after the above-mentioned registration processing is completed is in a standby state in which a normal internet telephone call can be made.

On the other hand, the above-described registration procedure of the Internet phone is described for the case of a terminal as an endpoint, the same procedure is also performed for the case of a gateway as an endpoint, the description thereof will be omitted here for simplicity of explanation. .

Next, an internet phone call mode according to the present invention will be described.

The call mode of the Internet phone according to the present system is divided into a general phone call mode, an internet phone call mode, and a gateway call mode. The general telephone call mode is a mode in which a telephone conversation with a general telephone is performed through a public telephone network (PSTN) like a normal telephone. The Internet telephone call mode may be divided into a first Internet call mode for talking between Internet telephones and a second Internet call mode for calling a general telephone through a called Internet telephone in the calling Internet telephone. In addition, the gateway call mode includes a first gateway call mode in which a general telephone is in communication with a called Internet telephone via a calling Internet telephone, and in a general telephone, a call is made from another ordinary telephone through a calling Internet telephone and a called Internet telephone. It may be divided into a second gateway call mode.

As a method of setting the general telephone call mode and the Internet telephone call mode, a method of using a voice guidance message pre-stored in the ROM 23 may be used.

First, a description will be given of how a user sets a normal telephone call mode and an internet telephone call mode using his Internet phone. When the user operates the voice input / output device 10 or 12 or 14 in a call-enabled state, the operating state is detected by the call intention detecting unit 34 and applied to the central processing unit 28, and the central processing unit 28 is A voice message for guiding a mode setting and a dialing method previously stored in the ROM 23 is output. Here, the voice guidance message requires a specific key input through the keypad 38 for the normal phone call mode and the Internet phone call mode.

In the case of a key input in a normal phone call mode, a voice message requesting the input of the other party's phone number is output. When the user inputs the desired counterpart phone number, the central processing unit 28 performs the dialing operation of the general phone call mode according to the input phone number.

On the other hand, in the case of the key input of the Internet phone call mode, a voice message for requesting a specific key input through the keypad 38 is output for the first Internet call mode and the second Internet call mode. In the case of the key input of the first internet call mode, the central processing unit 28 requests input of an identification number of the other party's Internet telephone through a voice message, and in the case of the key input of the second internet call mode, the central processing unit 28 28 requests input of an identification number of the other party's Internet telephone and an identification number (for example, a telephone number) of the other party's general telephone via a voice message. Thereafter, the central processing unit 28 sets the first Internet call mode or the second Internet call mode according to the key input signal from the keypad 38.

Next, when a user connects to an Internet telephone through a public telephone network (PSTN) using a regular telephone, a description will be given of a method of setting a general telephone call mode and a gateway telephone mode.

First, when a user connects to an Internet telephone using a regular telephone, the central processing unit 28 of the Internet telephone recognizes this and outputs a voice message for guiding mode settings previously stored in the ROM 32. Here, the voice announcement message requires a specific key input through the keypad for the normal phone call mode and the gateway call mode. Then, in accordance with the key input signal from the general telephone, the central processing unit 28 determines whether it is a general telephone call mode or a gateway telephone mode with its Internet telephone.

When the determination result of the key input signal is the general telephone call mode, the central processing unit 28 sets the general telephone call mode and operates a bell signal generator (not shown) to notify that there is a call connection request on its Internet telephone. do.

On the other hand, when the determination result of the key input signal is the gateway call mode, a voice message requesting a specific key input for the first gateway call mode and the second gateway call mode is output to the connected general telephone. Then, according to the key input signal from the general telephone, the central processing unit 28 determines whether it is a gateway 1 gateway mode or a second gateway mode.

In the case of a key input of the first gateway call mode, the central processing unit 28 requests input of an identification number of the other party's Internet telephone through a voice message. In the case of a key input of the second gateway call mode, the central processing unit 28 28 requests input of the identification number of the other party's Internet telephone and the telephone number of the other party's general telephone through a voice message. Then, the digital signal processing unit 20 detects the tone signal by the key input in the ordinary telephone and transmits the detection result to the central processing unit 28, so that the central processing unit 28 enters the first gateway call mode or the like. Set the second gateway call mode.

Next, the operation of the system according to the operation mode described above will be described in detail.

1. Normal Phone Call Mode

First, a general telephone call mode through a public switched telephone network (PSTN) will be described with reference to FIGS. 1 and 3.

1.1. Call from Internet Phone to Desk Phone

First, in FIG. 1, when a user wants to make a call from an Internet telephone IP1 to a general telephone (for example, NP12), the user operates the voice input / output device 10 or 12 or 14 in a call enabled state. It is detected by the intention detection unit 34 and applied to the central processing unit 28. Thereafter, it is assumed that the call mode to the general telephone is set by the user by the method such as voice guidance described above.

Then, the central processing unit 28 controls the digital connection arrangement 24 to be in a state of being able to connect with the PSTN network (i.e., sets the digital connection arrangement 24 in the hook-off mode). Then, when the telephone number of the telephone (for example NP12) to be called through the keypad 38 is inputted, the central processing unit 28 generates a digital tone signal corresponding to the DTMF tone of the input telephone number. 20). This digital tone signal is converted into an analog signal at the second codec 22 and then transmitted to the switch of the PSTN network through the data connection arrangement 24 and the telephone line connection 26. Thereafter, a communication path with the general telephone NP12 is formed by the process of forming the communication path between the exchange and the general telephone (for example, NP12). In the above process of establishing a call path with a general telephone, the central processing unit 28 transmits a ringing tone output on the call path and a signal returned from the exchange to the user. (22), the digital signal processing unit 20, the central processing unit 28, the digital signal processing unit 20, the first codec 18, and the call path to the voice input and output device is controlled to be formed.

Subsequently, the user's voice signal is converted into a digital signal by the first codec 18 and encoded by the digital signal processing unit 20 and then input to the central processing unit 28. At this time, since the central processing unit 28 knows that the call is through a public telephone network (PSTN), the coded signal is returned to the digital signal processing unit 20 to be decoded and output to the second codec 22. Thereafter, the user's decoded voice signal is converted into an analog signal in the second codec 22 to the general telephone unit NP12 via the data connection array unit 24, the telephone line connection unit 26, and the public telephone network PSTN. Delivered.

At the same time, the other party's voice signal from the general telephone unit NP12 is input through the telephone connection unit 26 and the data connection arrangement unit 24 to be converted into a digital signal by the second codec 22 and the digital signal processing unit 20. Is encoded and passed to the central processing unit 28. In this case, since the central processing unit 28 knows that the call is through a public telephone network (PSTN) (general telephone call mode), the coded signal is returned to the digital signal processing unit 20 and decoded to the first codec 18. To print. Thereafter, the decoded voice signal of the other party is converted into an analog signal in the first codec 18 and output to the voice input / output device 10 or 12 or 14 through the multiplexer 16. By this process, a call is made with a general telephone through the PSTN.

Then, when the user operates the voice input / output device 10 or 12 or 14 in the call termination state, the operation state is detected by the call intention detecting unit 34 and applied to the central processing unit 28. Accordingly, the central processing unit 28 controls the digital connection arrangement 24 to terminate the connection with the PSTN network (i.e., by setting the digital connection arrangement 24 to the hook-on mode), and then through the PSTN. Exit call mode.

1.2. Call from Desk Phone to Internet Phone

Next, a call connection from a general telephone (for example, NP12 in FIG. 1) connected via a PSTN to an Internet telephone (IP1 in FIG. 1) will be described.

First, a ring signal input through the telephone line connection section 26 is detected by the data connection arrangement section 24, where the data connection arrangement section 24 notifies the central processing unit 28 of the detection of the ring signal. The central processing unit 28 controls the data connection arrangement unit 24 to be in the hook-off mode, and also provides a voice guidance message (call request with oneself or call request with another Internet phone) stored in the memory means. Acknowledgment) is output to the digital signal processor 20, the second codec 22, the data connection array 24 and the telephone connection 26. According to the voice prompt message, when the other party operates the keypad of the telephone NP12, a DTMF tone signal is generated, and the tone signal is converted into a digital signal at the second codec 22 through the data connection arrangement 24 to convert the digital signal. It is input to the processing unit 20. The digital signal processing unit 20 detects this digitally converted tone signal, determines which key signal corresponds to which key input, and notifies the central processing unit 28. The central processing unit 28 determines the type of connection request of the counterpart phone NP12 according to the determination result of the digital signal processing unit 20 (call request with its own internet phone (ie, general telephone call mode) or other internet phone). Call request (ie, gateway call mode)]. At this time, assume that the connection request of the other party's telephone (NP12) is a call request (general phone call mode) with itself.

Then, the central processing unit 28 operates the bell signal generator (not shown). Accordingly, when the user recognizes that there is a call request with the user (normal phone call mode) and operates the voice input / output device 10 or 12 or 14 in a call enabled state, the operation state is determined by the call intention detecting unit 34. It is detected and applied to the central processing unit 28.

Thereafter, the central processing unit 28 includes the audio input / output means, the first codec 18, the digital signal processing unit 20, the central processing unit 28, the digital signal processing unit 20, as described in (1.1) above. The order of the second codec 22, the data connection arrangement 24, the telephone line connection 26 and the reverse of the call path are controlled.
Subsequent operations are the same as described in the above section (1.1), and thus description thereof is omitted.

2. Internet phone call mode

1, 4, 6a to 6c and 7a to 7c, the Internet telephone call mode through the internet network will be described.

2.1. Call between internet phones (first internet call mode)

First, the first Internet call mode according to the system of the present invention will be described with reference to FIGS. 1, 4 and 6A to 6C. As described with reference to FIG. 5, a description will be given of a case where a call from the Internet phone IP1, which has been registered in the call server CS, to each other Internet phone (for example, IP2).

If the user of the calling Internet phone IP1 operates the voice input / output device 10 or 12 or 14 in a call enabled state, the operating state (i.e., the hook-off state) is detected by the call intention detecting unit 34 to perform central processing. Is applied to the unit 28 (S10 in Fig. 6A). Thereafter, the central processing unit 28 outputs the voice guidance message to the calling user as described above (S11), and the Internet telephone IP1 is set to the first Internet call mode by the user's key input. Dialing input is made (S12).

Then, the central processing unit 28 of the Internet telephone IP1 sends an access request message (S13 in Fig. 6) through the Internet network through the input / output buffer 30 using the UDP protocol. Output to call server (CS).

In this case, the access permission request message is a RAS (Registration) sent by the Internet telephone IP1 to allow the call server CS to be authorized to use a packet-based network for a call with the called Internet telephone IP2. , Admission and Status) message and includes, for example, information elements as shown in Table 5 below.

Information element name function Request sequence number Indicates the sequence number of the message. Arc type Indicates the type of call. Endpoint identifier Indicates the identifier of the endpoint assigned by the call server. Call mode Indicate call mode. Destination information Indicate the nickname (phone number) of the destination. Destination call signaling address Indicates the call signaling transport address of the destination. Source Information Indicates a source nickname. Source call signaling address Indicates the call signaling transport address of the source. Bandwidth Represents the bandwidth requested for a two-way call. Call reference value Indicates a value to distinguish the call. Conference identifier Indicates a conference identifier. Call response Indicates a call response. Call identifier Unique call identifier set by the calling endpoint.

As shown in Table 5 above, the request sequence number information element indicates the sequence number of the present message, and the response message associated with this message should have the same sequence number. And, the Call Type information element indicates the type of call (ie, point-to-point, one-to-N, etc.), and the call server can use this value to measure the actual bandwidth used. The default value for all calls is point-to-point.

The endpoint identifier information element is an identifier assigned by the call server CS to a terminal which is an endpoint of the Internet telephone IP1 via the registration confirmation message S4 of FIG.

The Call Mode information element describes whether the Internet phone IP1 transmits the call directly to another Internet phone IP2 or the call via the call server CS. For the sake of simplicity, only the case where the information element directly transmits a call to another Internet telephone will be described.

The destination information information element is an alias of the destination, and is the telephone number of the called party dialed by the calling user (S12).

The destination call signaling address information element is not used when the originating Internet telephone IP1 sends the access permission request message S13 to the call server CS. This information element may be used when the called party's Internet telephone IP2 sends an access permission request message S17 to the call server CS.

In addition, the source information information element is an alias, that is, a telephone number, of the source originating Internet telephone IP1. Here, the alias is the telephone number of the endpoint for the Internet telephone IP1. These endpoints, terminals and gateways, each have different aliases. This means, for example, that if the telephone number assigned to the telephone book for an Internet telephone is "11100", this number "11100" is the telephone number of the terminal, and a number prefixed or preceded by the number (e.g. 9). The internet phone and the call server need to be promised to each other so as to use (eg, "911100" or "111009") as the gateway number.

The source call signal address information element includes an IP address and an application port number of the source originating Internet telephone IP1.

The bandwidth information element indicates the amount of bandwidth requested for a bidirectional call. For example, a call requiring 128 kbps may request 256 kbps.

In addition, the Call Reference Value information element is a locally unique value generated by the Internet telephone IP1 to distinguish a call, and represents a call reference value defined in ITU-T Recommendation Q.931. The Conference Identifier information element is a globally unique identifier generated by the originating Internet phone IP1.

In addition, the Answer Call information element is used to notify the call server CS whether it is the calling party or the called party with respect to the current call. The call server CS is connected to the calling Internet phone IP1. In the access permission request message S13 delivered to the user, this information element is set to FALSE, and in the access permission request message S17 transmitted from the called Internet telephone IP1 to the call server CS described later, The information element is set to TRUE.

In addition, the Call Identifier information element indicates a unique call identifier set by the Internet telephone IP1, which is the originating endpoint.

After that, when the call server CS receives the access permission request message S13, the call server CS sends an access confirmation confirmation message (S14 in Fig. 6A) using the UDP protocol. To the Internet phone (IP1).

In this case, the access permission confirmation message is used to allow the call server CS to use the packet network for a call with the called Internet phone IP2 to the Internet phone IP1 to which the access permission confirmation message S13 is sent. The outgoing message includes, for example, an information element as shown in Table 6 below.

Information element name function Request sequence number Indicates the sequence number of the message. Bandwidth Indicates the maximum bandwidth allowed for the call. Call mode Indicate call mode. Destination call signal address Indicates the call signal transmission address of the destination. Destination type Indicates the type of destination endpoint.

As shown in Table 6 above, the Request Sequence Number information element indicates the sequence number of the present message and should have the same sequence number as the connection permission request message S13 related to this message. The bandwidth information element indicates the maximum bandwidth allowed by the call server CS to the calling Internet telephone IP1, which may be smaller than the size requested in the access permission request message S13. .

In addition, the call mode information element indicates whether or not the call passes through the call server similarly to the call mode information element described in Table 5 above, and the call mode information element included in the access permission request message (S13). Is determined by the call server.

In addition, the destination call signal address information element may indicate that the called party's Internet telephone (IP2) is a call signaling message, for example, a call signaling message of ITU-T Recommendation H.225.0 (specifically, ITU-T). It indicates the address (including the IP address of the called Internet phone and the application port number) for receiving the call signaling message of Recommendation Q.931.

Then, upon receiving the connection permission confirmation message S14 from the call server CS, the central processing unit 28 of the calling Internet phone IP1 does not feel bored of the calling user, so that the digital signal processing unit 20 ) To output the virtual ringing tone to the first codec 18 (S15), so that the user attempts to make a call connection with the other party's Internet telephone through the voice input / output device 10 or 12 or 14. Make sure you know

In addition, the central processing unit 28 of the calling Internet telephone IP1 analyzes the connection permission confirmation message S14, and then inputs a setup message S16 to the input / output buffer 30 using the TCP protocol. ) Is output to the internet network and transmitted to the called Internet phone (IP2).

Here, the setup message S16 is a call signaling message sent to inform the calling Internet phone IP1 that it wants to set up a connection to the called Internet phone. For example, the setup message S16 is defined in ITU-T Recommendation H.225.0. The format of the setup message (specifically, the setup message of ITU-T Recommendation Q.931, etc.) may be used. An example of the format may include an information element as shown in Table 7 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call. Message type Indicates the type of message. Delivery ability Indicates the forwarding capability of the calling party. User-to-user Information on each of the called party and calling party is indicated.

As shown in Table 7, the Protocol Discriminator information element indicates a version supported by H.225.O as a protocol. The Call Reference information element is a locally unique value generated by the Internet phone IP1 so as to distinguish the call. In addition, the message type information element indicates a function of a message, which is set as a setup message.

In addition, the user-to-user information element is for indicating various types of information about the called party and the calling party, and may include a field as shown in Table 8 below. More specifically, UUIE (User-to-User Information Element) in the setup message specified in ITU-T Recommendation Q.931 can be used.

Field name Explanation Protocol discriminator Indicates the version supported by the protocol. H245 address Indicates the address of an H.245 call signaling channel. Source address Indicates an alias address for the source. Source Information Indicates the type of originating endpoint. Destination address Indicate the alias address of the destination. Destination call signaling address Indicates a transport address for call signaling at the destination. Conference identifier Indicates a unique identifier for the conference. Arc type Indicates the type of call connection. Source call signal address Indicate the source transfer address. Call identifier Indicates a globally unique call identifier. Overlap transmission Indicates whether or not to use overlap transmission on the calling side.

As shown in Table 8, the Protocol Discriminator field indicates a version supported by H.225.O as a protocol. The H245 address field is for indicating an address of a channel for call signaling by, for example, TIU-T Recommendation H.245 from the calling Internet phone IP1 to the called Internet phone. This field is optional. .

The source address indicates an alias address (e.g., a phone number) of a source IP phone number IP1 that is a source, and the source information field indicates that the type of endpoint for the source IP phone IP1 is a gateway. Specifies whether this is a terminal or terminal, where this field is set to terminal.

The Destination Address field indicates the alias address (e.g. telephone number) of the endpoint for the destination, i.e., the destination Internet phone IP2. In this case, since the endpoint is a terminal, it is the telephone number of the terminal for the Internet telephone (IP2).

The Destination Call Signaling Address field is an address (e.g., call signaling defined in ITU-T Recommendation H.225.0) of a destination, i.e., a destination Internet phone (IP2). For example, including an IP address and an application port number).

The conference identifier field has the same value as the conference identifier of the access permission request message S13. The Call Type field indicates the type of call (ie, point-to-point, one-to-N, etc.), and the default value for all calls is point-to-point. The Source Call Signal Address field indicates a transmission address (including an IP address and an application port number) of the source originating Internet telephone IP1.

The Call Identifier field indicates a globally unique call identifier as described in the access permission request message S13.

The overlap send field indicates whether the sender uses overlap transmission. Here, when the calling party uses overlap transmission, the overlap transmission field is set to TRUE, and when the calling party does not use overlap transmission, the overlap transmission field is set to FALSE. In this setup message (S16 in Fig. 6A), this field is set to "FALSE". Therefore, the Setup Acknowledge message does not follow this Setup message S16.

Thereafter, the central processing unit of the called Internet phone IP2 receiving the setup message S16 analyzes the setup message S16 and recognizes that there is an Internet phone call request with itself at the calling Internet phone IP1. Done. Subsequently, the central processing unit of the called Internet phone IP2 outputs an access request message (S17 in FIG. 6) to the Internet network through the input / output buffer using the UDP protocol through the Internet network. Send to call server (CS).

Here, the access permission request message is a message sent by the Internet telephone IP2 to allow the call server CS to permit the use of the packet network for the call with the calling Internet telephone IP1. For example, Information elements such as 9 and the like.

Information element name function Request sequence number Indicates the sequence number of the message. Arc type Indicates the type of call. Endpoint identifier Indicates the identifier of the endpoint assigned by the call server. Call mode Indicate call mode. Destination information Indicate the nickname (phone number) of the destination. Destination call signaling address Indicates the call signaling transport address of the destination. Source Information Indicates a source nickname. Source call signaling address Indicates the call signaling transport address of the source. Bandwidth Represents the bandwidth requested for a two-way call. Call reference value Indicates a value to distinguish the call. Conference identifier Indicates a conference identifier. Call response Indicates a call response. Call identifier Unique call identifier set by the calling endpoint.

As shown in Table 9 above, the request sequence number information element indicates the sequence number of the present message, and the response message associated with this message should have the same sequence number. The call type information element and the call mode information element are the same as those described in the access permission request message S13.

The endpoint identifier information element is an identifier assigned by the call server CS to the endpoint for the Internet phone IP2 via the registration confirmation message S4 'of FIG. The endpoint here is a terminal.

The destination information information element is an alias of the destination and is a telephone number of the destination Internet telephone (IP2), and the destination call signal address information element, source information information element, and source call signaling address information element. The bandwidth information element, the call reference value information element, the conference identifier information element, and the call identifier information element are the same as those described in the access permission request message S13. The call response information element is set to TRUE to receive an incoming call.

Thereafter, when the call server CS receives the access permission request message S17, the call server CS sends an access confirmation confirmation message (S18 in Fig. 6A) to the Internet network using the UDP protocol. To the Internet phone (IP2).

Here, the access permission confirmation message S17 is sent to the Internet phone IP2 to which the call server CS has sent the access permission confirmation message to permit the use of the packet network for the call with the calling Internet phone IP1. The message includes, for example, an information element as shown in Table 10 below.

Information element name function Request sequence number Indicates the sequence number of the message. Bandwidth Indicates the maximum bandwidth allowed for the call. Call mode Indicate call mode. Destination call signal address Indicates the call signal transmission address of the destination. Destination type Indicates the type of destination endpoint.

As shown in Table 10, the Request Sequence Number information element indicates the sequence number of the message, and should have the same sequence number as the connection permission request message S17 related to this message.

 The bandwidth information element indicates the maximum bandwidth allowed by the call server CS to the called Internet phone IP2, and the bandwidth may be smaller than the size requested in the access permission request message S17. .

In addition, the destination call signal address information element may include a call signaling message (eg, ITU-T) of a call signaling message, for example, ITU-T Recommendation H.225.0. It indicates the address (including the IP address of the called Internet phone and the application port number) for receiving the call signaling message of Recommendation Q.931. In addition, the destination type information element indicates the type of the called party's endpoint. In this case, since the called party's endpoint is the Internet phone IP1, the destination type information element is set to the terminal.

Thereafter, upon receiving the connection permission confirmation message S18 from the call server CS, the central processing unit of the called Internet phone IP2 analyzes the connection permission confirmation message S18, and then calls a call progress message. (S19) is output to the internet network through the input / output buffer using the TCP protocol and transmitted to the calling Internet phone (IP1).

Here, the call progress message S19 is sent to notify the calling Internet phone IP1 that the call request is received at the called Internet phone IP2 and that no further information for call establishment is needed. As a call signaling message to be used, for example, the format of a call progress message (specifically, a call progress message such as ITU-T Recommendation Q.931) specified in ITU-T Recommendation H.225.0 may be used. An example of the format may include an information element as shown in Table 11 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call Message type Indicates the type of message. User-to-user Information on each of the called party and calling party is indicated.

As shown in Table 11, the Protocol Discriminator information element and the Call Reference information element are the same as described in Table 7. The message type information element indicates a function of a message and is set as a call progress message.

In addition, the user-to-user information element is for indicating various types of information about the called party and the calling party, and may include fields such as the following Table 12. More specifically, UUIE (User-to-User Information Element) in the call progress message specified in ITU-T Recommendation Q.931 can be used.

Field name Explanation Protocol discriminator Indicates the version supported by the protocol. H245 address Indicates the address of an H.245 call signaling channel. Destination information Indicate the destination information. Call identifier Indicates a globally unique call identifier.

As shown in Table 12, the Protocol Discriminator field indicates a version supported by H.225.O as a protocol. The H245 address field is for indicating the address of a channel for call signaling by, for example, ITU-T Recommendation H.245 from the called Internet phone IP2 to the calling Internet phone IP1, and this field is selected. It is.

The destination information field specifies whether the type of the destination endpoint, i.e., the Internet telephone IP2, is a gateway or a terminal, and this field is set to a terminal.

The Call Identifier field is set to the same value as the call identifier received in the setup message S16.

Thereafter, the central processing unit 28 of the called party's Internet telephone IP2 operates the bell signal generator (not shown) to notify the called user of the telephone call (S20).

At the same time, the central processing unit 28 of the called Internet phone IP2 outputs an Alerting message (S21 in Fig. 6B) to the Internet network through the input / output buffer using the TCP protocol, and then to the calling Internet. Send to the phone IP1.

Here, the alert message S21 is a call signaling message transmitted to notify the calling Internet phone IP2 that an alert (for example, a telephone ringing) for the called user has been started in the called Internet phone IP2. For example, the format of the alert message specified in ITU-T Recommendation H.225.0 (specifically, an alert message such as ITU-T Recommendation Q.931) may be used. An example of the format may include an information element as shown in Table 13 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call Message type Indicates the type of message. User-to-user Information on each of the called party and calling party is indicated.

As shown in Table 13, the Protocol Discriminator information element and the Call Reference information element are the same as described in Table 7. The message type information element indicates a function of a message and is set as an alarm message.

In addition, the user-to-user information element is for indicating various types of information about the called party and the calling party, and may include a field as shown in Table 14 below. More specifically, UUIE (User-to-User Information Element) in the alert message specified in ITU-T Recommendation Q.931 can be used.

Field name Explanation Protocol identifier Indicates the version supported by the protocol. H245 address Indicates the address of an H.245 call signaling channel. Destination information Indicate the destination information. Call identifier Indicates a globally unique call identifier.

Each field in Table 14 is the same as that described in Table 12, and thus description thereof is omitted.

On the other hand, as the bell rings through the bell signal generator, the called user recognizes that there is a call request with the user and operates the voice input / output device 10 or 12 or 14 in a call enabled state, and the operation state is called. It is detected by the intention detection unit 34. As a result, the destination Internet telephone IP2 is in the hooked-off state S22.

Thereafter, the central processing unit 28 of the called Internet phone IP2 sends a connection message (S23 in FIG. 6B) through the Internet network according to the hook-off detection by the call intention detecting unit 34. Using the TCP protocol, it is output to the internet network through the input / output buffer and transmitted to the calling Internet phone (IP1).

Here, the access message S23 is a call signaling message transmitted from the called Internet phone IP2 to notify the calling Internet phone IP1 that the call has been accepted. The format of the connection message (specifically, a connection message of ITU-T Recommendation Q.931, etc.). An example of the format may include an information element as shown in Table 15 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call Message type Indicates the type of message. User-to-user Information on each of the called party and calling party is indicated.

As shown in Table 15, the Protocol Discriminator information element and the Call Reference information element are the same as described in Table 7. The message type information element indicates a function of a message, and is set here as a connection message.

In addition, the user-to-user information element is for indicating various types of information about the called party and the calling party, and may include fields such as the following Table 16. More specifically, UUIE (User-to-User Information Element) in the connection message specified in ITU-T Recommendation Q.931 can be used.

Field name Explanation Protocol identifier Indicates the version supported by the protocol. H245 address Indicates the address of an H.245 call signaling channel. Destination information Indicate the destination information. Conference identifier It indicates a unique consultation identifier.

In Table 16, the protocol identifier field, the H245 address field, and the destination information field are the same as those described in Table 12, and thus description thereof is omitted. The conference identifier is a conference identifier received in the setup message S16. If the H.245 address is not sent in the previous message, the H.245 address must be sent in the access message.

Thereafter, the originating Internet telephone IP1 and the called Internet telephone IP2 which have received the access message S23 mutually perform a signaling operation S24 specified in, for example, ITU-T Recommendation H.245. . In H.245 signaling, an endpoint-to-endpoint (i.e., between Internet telephones) control messages governing the operation of an H.323 entity using an H.245 control channel. Send them. These messages are used for capability exchange, opening and closing of logical channels, mode preference requests, flow control, and general commands and instructions.

H.245 signaling is established between two endpoints (ie the calling Internet phone and the called Internet phone). Each endpoint (Internet phone) establishes only one H.245 control channel for each call it participates in. For H.245 control channels, logical channel 0 shall be used. Logical channel 0 is considered permanently open from the establishment of the H.245 control channel until this channel is terminated. The general procedures used for opening and closing logical channels do not apply to H.245 control channels.

H.245 specifies many independent protocol entities that support endpoint-to-end signaling. H.323 endpoints must support the following protocol entities:

Master / slave Determination

Capability Exchange

Logical Channel Signaling

Bi-directional Logical Channel Signaling

Close Logical Channel Signaling

Mode Request

Round-trip Delay Determination

Maintenance Loop Signaling

General commands and instructions shall be selected from the message set contained in H.245. In addition, other commands and instructions defined to be included in the video, audio, or data stream may be transmitted.

H.245 messages fall into four categories: Request, Response, Command, and Indication. Requests and responses are used by protocol entities. The request message requires a specific action by the receiver, such as an immediate response, and the response message responds to the request. Command messages require specific actions but do not require a response. The indication message does not require any action or response and is merely for informational purposes. H.323 terminals must respond to all H.245 commands and requests and send instructions that reflect the state of the terminal.

The capability exchange shall be in accordance with the procedures of H.245 providing separate receiving and transmitting capabilities and the means by which terminals may describe their capabilities. Receive capability represents the receiving and processing capability of a terminal for an incoming information stream. The calling party shall limit the content of the information it transmits to the extent that the receiver has known it can accept it. If there is no reception capability, the terminal indicates that only transmission is possible. Transmission capability refers to the terminal's ability to transmit an information stream. The transmit capability provides the operable modes to the called party so that the receiver can request the mode it wishes to receive. The lack of transmission capability means that the terminal does not provide the operating modes that the receiver can select. However, the calling party can still transmit in any way that is within the capabilities of the called party.

In the logical channel signaling, each logical channel carries information from a transmitter to one or more receivers, identified by a unique logical channel number in each transmission direction. Logical channels are opened and closed using Open Logical Channel and Close Logical Channel messages and procedures in H.245. When a logical channel is opened, an open logical channel message describes the contents of the logical channel in detail, including the media type, the algorithm used, the options, and all other information required by the called party to interpret the contents of the logical channel.

Since most H.323 logical channels are unidirectional, asymmetric behaviors are allowed where the number of channels and the type of information streams vary from one transmission direction to another. However, if the called party allows only symmetric mode of operation, it can transmit a reception capability that reflects its limit. Terminals can also use a particular mode for only one direction of transmission. Certain media types, including data protocols such as T.120, basically require a bidirectional channel. In such a case, a pair of unidirectional logical channels may be opened and associated with each other in accordance with the H.245 bidirectional channel opening procedure. Channel pairs associated in this way do not need to share the same logical channel number.

In this mode preference, the called party may request the senders to transmit in a specific mode using an H.245 Request Mode message. The calling party shall follow this if possible.

In the master / slave determination, the H.245 master / slave determination procedure is used to resolve collisions between two terminals, which may be MCs in one conference, or between two terminals attempting to open a bidirectional channel. In this procedure, the two terminals decide the master / slave by exchanging random numbers in the H.245 master-slave decision message. The H.323 terminal must support both modes. The endpoint (i.e. the Internet phone) must set the Terminal Type and set the Status Determination Number to a random number between 0 and 224. The endpoint must choose only one random number for each call.

The call S25 is made between the calling party and the called Internet phone IP1 IP2 while the call channel is established by the H.245 signaling S24 as described above. At this time, the central processing unit 28 of the calling Internet telephone IP1 controls the digital signal processing unit 20 to terminate the generation of the virtual ring tone. Thereafter, the user's voice signal is converted into a digital signal at the first codec 18, compressed and encoded by the digital signal processor 20, and then input to the central processing unit 28. At this time, since the central processing unit 28 knows that the call is made through the Internet (first Internet call mode), the packetized voice data obtained by packetizing the compressed coded voice signal using the RTP protocol is loaded on the TCP / IP. The data is transmitted to the fur 30, and the input / output buffer 30 outputs the packet voice data to the data network connection 42 in serial. Accordingly, the output packet voice data is transmitted to the called party's Internet telephone IP2 through the Internet network.

On the other hand, the voice data signal from the destination Internet phone (IP2) is carried in TCP / IP in the form of a packet to reach the input / output buffer 30 in the form of serial data through the Internet network, the input / output buffer 30 is input serial The data is transmitted to the central processing unit 28 in packet units. The central processing unit 28 decomposes the RTP protocol packet data, checks the calling identification code (IP address) and the called party identification code (IP address), and the packet voice data is transmitted from the called partner's Internet telephone. After confirming that the application port number is confirmed, the decoded compressed coded voice signal is transmitted to the digital signal processor 20. Accordingly, the digital signal processing unit 20 decodes the compressed coded signal and outputs the decoded signal to the first codec 18. Thereafter, the decoded voice signal of the other party is converted into an analog signal in the first codec 18 and output to the voice input / output device 10 or 12 or 14 through the multiplexer 16. By this process, a call is made with an Internet telephone through the Internet network (S25 of FIG. 6C).

Thereafter, when the called user operates the voice input / output device 10 or 12 or 14 in the call termination state, the operating state (hook on state) (S26 in Fig. 6C) is detected by the call intention detecting unit 34 to perform central processing. Is applied to the unit 28.

Accordingly, the central processing unit 28 of the called Internet phone IP2 uses the TCP protocol to terminate the call for the call signal channel to which it is currently connected. As a call signaling message to be sent to the mobile station, the format of, for example, a release complete message (specifically, a release complete message of ITU-T Recommendation Q.931) specified in ITU-T Recommendation H.225.0 may be specified. It is available. An example of the format may include an information element as shown in Table 17 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call Message type Indicates the type of message. User-to-user Information on each of the called party and calling party is indicated.

As shown in Table 17, the Protocol Discriminator information element and the Call Reference information element are the same as described in Table 7. The message type information element indicates a function of a message, and is set here as a release complete message.

In addition, the user-to-user information element is for indicating various types of information about the called party and the calling party, and may include fields such as the following Table 18. More specifically, UUIE (User-to-User Information Element) in the release completion message specified in ITU-T Recommendation Q.931 can be used.

Field name Explanation Protocol identifier Indicates the version supported by the protocol. Reason Indicates why the connection was closed. Call identifier Indicates a globally unique call identifier.

In Table 18, the protocol identifier field is the same as that described in Table 12, and thus description thereof is omitted. The reason field indicates various reasons for the call being released, and the call identifier field is the same as the content described with reference to Table 5 above.

Thereafter, the central processing unit 28 of the called party's Internet telephone IP2 transmits the connection release complete message S27, and then a connection request message through the Internet network (S28 in Fig. 6C). By using the UDP protocol, and outputs to the Internet through the input and output buffer to the call server (CS).

Here, the connection release request message is a RAS (Registration, Admission and Status) message for notifying the call server CS that the call of the called Internet phone IP2 has been disconnected and receiving or releasing the resource. For example, the format of the Disengage Request message specified in ITU-T Recommendation H.225.0 may be used. An example of the format may include an information element as shown in Table 19 below.

Information element name function Request sequence number Indicates the sequence number of the message. Endpoint identifier Indicates the identifier of the endpoint assigned by the call server. Conference identifier Unique identifier. Call reference value Indicates a locally unique value for the call Reason for disconnection notification Indicate the reason for the request. Call identifier Unique call identifier set by the calling endpoint.

As shown in Table 19 above, the request sequence number information element indicates the sequence number of the present message, and the response message associated with this message should have the same sequence number. The endpoint identifier information element is an identifier assigned by the call server CS to the Internet telephone IP2, which is an endpoint, through the registration confirmation message S4 'of FIG. In addition, the conference identifier information element, the call reference value information element, and the call identifier information element are the same as those described with reference to Table 5 above. The disengagement reason information element indicates the reason for the disassociation notification request.

Thereafter, when the call server CS receives the disconnection request message S28, the call server CS sends a disconnection confirmation message (S29 in FIG. 6C) to the Internet network using UDP. To the Internet phone (IP2).

Here, the connection release confirmation message (S29) is a RAS (Registration, Admission and Status) message for the call server CS to allow a request for the release or return of resources by the connection release request message (S28), for example For example, the format of the Dis- surement Confirmation message specified in ITU-T Recommendation H.225.0 may be used. An example of the format may include an information element as shown in Table 20 below.

Information element name function Request sequence number Indicates the sequence number of the message.

As shown in Table 20 above, the Request Sequence Number information element indicates the sequence number of this message and should have the same sequence number as the connection permission request message (S17) related to this message.

Accordingly, the central processing unit 28 of the called party's Internet telephone IP2 analyzes the format of the Disconnect Confirmation message, confirms that a request for resource release or return has been made, and sets the call mode. Quit.

On the other hand, the central processing unit 28 of the originating Internet phone IP1 receiving the disconnection completion message S27 analyzes the format of the disconnection complete message S27 and calls the call by the called party Internet phone IP2. After confirming that the connection has been terminated, a connection request message (S30 in FIG. 6C) is output through the I / O buffer through the I / O buffer through the Internet network, and transmitted to the call server CS.

Thereafter, when the call server CS receives the connection release request message S30, the call server CS transmits a disconnection confirmation message S31 to the Internet telephone IP1 through the Internet network. do.

Here, since the connection release request message S30 and the connection release confirmation message S31 are the same as the connection release request message S28 and the connection release notification confirmation message S29, description thereof will be omitted.

Accordingly, the central processing unit 28 of the calling Internet phone IP1 analyzes the format of the Connection Confirmation message and confirms that a request for resource release or return has been made. 20 to output the virtual end tone to the first codec 18 (S32), so that the user recognizes that the Internet telephone connection is terminated through the voice input / output device 10 or 12 or 14. Do it. Thereafter, when the calling user operates the voice input / output device 10 or 12 or 14 in a call termination state, the operating state (hook on state) (S33 in Fig. 6C) is detected by the call intention detecting unit 34 to perform central processing. Is applied to the unit 28. Therefore, the central processing unit 28 of the calling Internet telephone IP1 ends the call mode.

On the other hand, in the above example, the case where the called party terminates the call has been described. However, when the calling party terminates the call, the disconnection completion message is transmitted from the calling Internet phone IP1 to the called Internet phone IP2. The same procedure as in the above example is performed except that the virtual end tone output occurs from the called Internet phone IP2 to the called user.

In addition, in the above embodiment, only the case where the call signaling messages are transmitted between the Internet telephones without passing through the call server CS has been described, but these messages are transferred between the Internet telephones via the call server CS. It may be easily understood from the above description that it may be configured as possible.

2.2. Call to Desk Phone Through Another Internet Phone (Second Internet Call Mode)

First, the second Internet call mode according to the system of the present invention will be described with reference to FIGS. 1, 4 and 7A to 7C. Here, as described with reference to FIG. 5, when a call is made from the Internet phone IP1, which has been registered in the call server CS, to the general phone NP21 via another Internet phone (for example, IP2). This will be described. Here, the endpoint for the calling Internet phone IP1 is a terminal and the endpoint for the called Internet phone IP2 is a gateway.

If the user of the calling Internet phone IP1 operates the voice input / output device 10 or 12 or 14 in a call enabled state, the operating state (i.e., the hook-off state) is detected by the call intention detecting unit 34 to perform central processing. Is applied to the unit 28 (S50 in Fig. 7A). Thereafter, the central processing unit 28 outputs the voice guidance message to the calling user as described above (S51), and the Internet telephone IP1 is set to the second Internet call mode by the user's key input, and Dialing input is made (S52).

Then, the central processing unit 28 of the Internet telephone IP1 sends an access request message (S53 in Fig. 6A) through the Internet network through the input / output buffer 30 using the UDP protocol. Output to call server (CS). Here, the access permission request message is configured to include the information elements, such as shown in Table 5. Therefore, the description thereof is omitted here.

Then, when the call server CS receives the access permission request message S53, the call server CS sends an access confirmation confirmation message (S54 in Fig. 7A) through the Internet network to the Internet telephone ( To IP1). In this case, the access permission confirmation message is configured to include information elements, such as shown in Table 6. Therefore, description thereof is omitted.

Thereafter, upon receiving the connection permission confirmation message S54 from the call server CS, the central processing unit 28 of the calling party Internet telephone IP1 does not feel bored of the calling user. 20 to output the virtual ringing tone to the first codec 18 (S15), so that the user can make a call connection with the other party's Internet telephone through the voice input / output device 10 or 12 or 14. Make sure you know it's being tried.

Further, the central processing unit 28 of the calling Internet telephone IP1 analyzes the access permission confirmation message S54 and then inputs and outputs a setup message S56 using the TCP protocol. It outputs to the internet network through and transmits it to the called Internet phone (IP2). Here, the setup message (S16) may be configured to include the information elements, such as the above Table 7. Therefore, description thereof is omitted.

Further, in the user-to-user information element shown in Table 8, the source information field designates whether the type of originating endpoint, that is, the type of Internet telephone IP1 is a gateway or a terminal. In this case, this field is set to a terminal. The Overlap Send field indicates whether or not an overlap transmission is used at the sender, and this field is set to "true" in this setup message (S56 in FIG. 7A). Therefore, the Setup Acknowledge message must be followed for this Setup Message S56. The other fields are the same as the contents described in connection with Table 8 above.

Thereafter, the central processing unit 28 of the called party's Internet phone IP2 receiving the setup message S56 analyzes the setup message S56 and transfers the Internet phone with itself from the calling party's Internet phone IP1 to the gateway. You realize that you want to use. Subsequently, the central processing unit 28 of the called Internet phone IP2 sends an access request message (S57 in FIG. 7A) through the Internet network through an input / output buffer using the UDP protocol. Output to call server (CS). In this case, the access permission request message is the same as described in Table 9 and the description thereof will be omitted.

Then, when the call server CS receives the access permission request message S57, the call server CS sends an access confirmation confirmation message (S58 in Fig. 7A) through the Internet network to the Internet telephone IP2. To send). Here, since the access permission confirmation message is the same as that described in Table 10 and the description thereof, a description thereof will be omitted.

Then, upon receiving the connection permission confirmation message S58 from the call server CS, the central processing unit 28 of the destination Internet phone IP2 analyzes the connection permission confirmation message S58, and then sets up the setup approval message (S58). The setup acknowledgment message (S59) is output to the internet network through the input / output buffer using the TCP protocol and transmitted to the calling Internet telephone (IP1). Here, the setup acknowledgment message is a call signaling message transmitted only when the called party's Internet telephone IP2 is not already performing the gateway function before receiving the setup message S56. For example, the ITU-T Recommendation H. The format of the setup acknowledgment message specified in 225.0 (specifically, a setup acknowledgment message such as ITU-T Recommendation Q.931) may be used. An example of the format may include an information element as shown in Table 21 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call Message type Indicates the type of message.

As shown in Table 20, the Protocol Discriminator information element and the Call Reference information element are the same as described in Table 7. The message type information element indicates a function of a message, and is set here as a setup approval message.

Thereafter, the central processing unit 28 of the called-side Internet telephone IP2 controls the data connection arrangement 24 to be hooked off for the call to the public telephone network PSTN (S60).

On the other hand, when the central processing unit 28 of the calling Internet phone IP1 receives the setup approval message S59, it analyzes it and confirms that the called Internet phone IP2 can be used as a gateway. An information message (S61 in FIG. 7B) is output to the Internet network through the input / output buffer using the TCP protocol, and transmitted to the destination Internet phone IP2. Here, the information message S61 is a call signaling message for transmitting a telephone number for the general telephone NP21 to be called via the called Internet telephone IP2, and is referred to, for example, in ITU-T Recommendation H.225.0. The format of a prescribed Information message (specifically, an information message such as ITU-T Recommendation Q.931) may be used. An example of the format may include an information element as shown in Table 22 below.

Information element name function Protocol discriminator Indicates the version supported by the protocol. Call standard Indicates a locally unique value for the call Message type Indicates the type of message. Called party number The number of the called party is indicated. User-to-user Information on each of the called party and calling party is indicated.

As shown in Table 21, the Protocol Discriminator information element and the Call Reference information element are the same as described in Table 7. The message type information element indicates a function of a message and is set as an information message. The called party number information element describes the telephone number of the general telephone NP21 to be called as the called party's number.

The user-to-user information element may be configured to include fields, such as the following Table 23. More specifically, for example, information defined in ITU-T Recommendation Q.931. UUIE (User-to-User Information Element) in a message can be used.

Field name Explanation Protocol identifier Indicates the version supported by the protocol. Call identifier Indicates a globally unique call identifier.

Since the protocol identifier field and the call identifier field in Table 23 are the same as those described in Table 12, description thereof will be omitted.

Then, when the called party's Internet telephone IP2 receives the information message S61, the central processing unit 28 analyzes the message and then stores the telephone number string of the general telephone NP21 extracted from the called number information element. By using the digital signal processing unit 20 to control and dial to generate a DTMF tone signal corresponding to the character string, the generated tone signal is transmitted to the second codec 22, the data connection array 24, and the telephone line connection unit ( 26) to be delivered to the exchange of the public telephone network (PSTN) (S62).

In addition, the information messages S61 'and S61 "in Fig. 7B are for providing additional information if necessary other than the telephone number of the ordinary telephone NP21.

Here, as a method of transmitting the telephone number of the general-purpose telephone NP21 of the called party, the telephone number input by the calling party user is first described in a single information message (for example, S61). A method of transferring from the IP phone (IP1) to the called Internet phone (IP2), and the second number of the telephone number (for example, 3456-1234) entered by the calling user (for example, 3, 4). 1, 5, 6, 1, 2, 3, and 4 for each information message (for example, eight information messages), each of which is called from the calling Internet phone IP1 to the called Internet phone IP2. Consider how to deliver it. In the case of using the second method, the above information messages S61, S61 ', ..., S61 "may be used.

Thereafter, the central processing unit 28 of the called Internet phone IP2 outputs the Call Proceeding message S63 to the Internet network through the input / output buffer using the TCP protocol to send the calling Internet phone IP1. To send. Here, the call progress message S63 is formed in the same format as the call progress message S19 of FIG. 6B and an example thereof is the same as in Table 11, and thus description thereof will be omitted. In addition, the user-to-user information element may include a field as shown in Table 12. In the call progress message S19, the called party endpoint, that is, the Internet, may be configured. Although the Destination Information field, which specifies whether the type of the telephone IP2 is a gateway or a terminal, is set as a terminal, in the call progress message S63, the destination endpoint, that is, the Internet telephone (IP2) Because the type is a gateway, the destination information field is set to the gateway.

Thereafter, the central processing unit 28 of the called Internet phone IP2 outputs an Alerting message (S64 in FIG. 7B) through the Internet network to the Internet network through the input / output buffer using the TCP protocol. Send to the calling Internet phone (IP1). Here, the alert message S64 is formed in the same format as the alert message S21 of FIG. 6B, and an example thereof is the same as in Table 13, and description thereof will be omitted. In addition, the user-to-user information element of the alert message S64 may include a field as shown in Table 14, and description thereof will be omitted. In the user-to-user information element, the destination information field is set to a gateway.

In the following description, it is assumed that the user of the ordinary telephone NP21 receives a call for the sake of simplicity. When the telephone is received by the ordinary telephone NP21 in this manner, the digital signal processing unit 20 detects the hook-on state of the ordinary telephone NP21 and notifies the central processing unit 28.

Thereafter, the central processing unit 28 of the destination Internet telephone IP2, in response to the detection input of the hook-on state of the ordinary telephone NP21 from the digital signal processing unit 20, connects via the Internet network. message (S65 in Fig. 7B) is output to the internet network through the input / output buffer using the TCP protocol, and transmitted to the calling Internet telephone IP1.

Here, since the access message S65 has the same format as the access message S23 of FIG. 6B and an example thereof is the same as in Table 15, description thereof will be omitted. In addition, since the user-to-user information element of the access message S65 includes a field as shown in Table 16, description thereof will be omitted. In the user-to-user information element, the destination information field is set to a gateway.

Then, the originating Internet telephone IP1 and the called Internet telephone IP2 which have received the access message S63 mutually perform a signaling operation S66 as defined in, for example, ITU-T Recommendation H.245. . Since the H.245 signaling (S66) is the same as the call signaling (S24) of Figure 6b it will be omitted.

The call (S67 in Fig. 7B) is made between the calling party and the called party Internet telephone IP1 (IP2) in the state where the call channel is established by the above-described H.245 signaling (S24). At this time, the central processing unit 28 of the calling Internet telephone IP1 controls the digital signal processing unit 20 to terminate the generation of the virtual ring tone. Thereafter, the user's voice signal is converted into a digital signal at the first codec 18, compressed and encoded by the digital signal processor 20, and then input to the central processing unit 28. At this time, since the central processing unit 28 knows that the call is made through the Internet (first Internet call mode), the packetized voice data obtained by packetizing the compressed coded voice signal using the RTP protocol is loaded into the TCP / IP buffer. The input / output buffer 30 outputs the packet voice data to the data network connection unit 42 in serial. Accordingly, the output packet voice data is transmitted to the called party's Internet telephone IP2 through the Internet network.

On the other hand, the voice data signal from the ordinary telephone NP21 via the destination Internet telephone IP2 is carried in TCP / IP in the form of a packet and reaches the input / output buffer 30 in the form of serial data through the Internet. The buffer 30 transfers the input serial data to the central processing unit 28 in packet units. The central processing unit 28 decomposes the RTP protocol packet data, checks the calling identification code (IP address) and the called identification code (IP address), and confirms that the packet voice data is transmitted from the called partner's Internet telephone. After confirming, the application port number is confirmed, and the decompressed compressed coded voice signal is transmitted to the digital signal processor 20. Accordingly, the digital signal processing unit 20 decodes the compressed coded signal and outputs the decoded signal to the first codec 18. Thereafter, the decoded voice signal of the other party is converted into an analog signal in the first codec 18 and output to the voice input / output device 10 or 12 or 14 through the multiplexer 16. By this process, the call is made from the calling Internet phone IP1 to the general telephone NP21 via the called Internet phone IP2 via the Internet network.

Thereafter, when the user of the destination general telephone NP21 ends the call, the end tone is detected by the digital signal processing unit 20 of the destination Internet telephone IP2 and applied to the central processing unit 28 (S68).

Accordingly, the central processing unit 28 of the called Internet phone IP2 controls the data access arrangement 24 to be in the hook-on mode and calls the call signal channel to which it is currently connected. In order to terminate the process, the connection completion message S70 is output to the calling Internet phone IP1 through the input / output buffer using the TCP protocol, and is transmitted to the calling Internet phone IP1.

Here, the connection release completion message S70 is formed in the same format as the connection release completion message S27 of FIG. 6C and an example thereof is the same as in Table 17, and thus description thereof will be omitted. In addition, since the user-to-user information element of the connection release completion message S70 is configured to include a field as shown in Table 18, description thereof will be omitted.

Thereafter, the central processing unit 28 of the called party's Internet telephone IP2 transmits the connection release completion message S70, and thereafter, a connection request message through the Internet network (S71 in Fig. 7C). By using the UDP protocol, and outputs to the Internet through the input and output buffer to the call server (CS). Here, the connection release request message S71 has the same format as the connection release request message S28 of FIG. 6C, and an example thereof is the same as in Table 19, and thus description thereof will be omitted.

Thereafter, when the call server CS receives the connection release request message S71, the call server CS sends a disconnection confirmation message (S72 in Fig. 7C) through the Internet network to the Internet phone IP2. To send). Here, the connection release confirmation message S72 is formed in the same format as the connection release confirmation message S29 of FIG. 6C, and an example thereof is the same as in Table 20, and thus description thereof will be omitted.

Accordingly, the central processing unit 28 of the called party's Internet telephone IP2 analyzes the format of the Disconnect Confirmation message, confirms that a request for resource release or return has been made, and sets the call mode. Quit.

On the other hand, the central processing unit 28 of the originating Internet phone IP1 receiving the disconnection completion message S70 analyzes the format of the disconnection complete message S73 and receives a call by the called party Internet phone IP2. After confirming the disconnection, the connection request message (S73 in FIG. 7C) is output through the I / O buffer through the I / O buffer to the call server CS through the Internet network.

Thereafter, when the call server CS receives the connection release request message S30, the call server CS transmits a disconnection confirmation message S74 to the Internet telephone IP1 via the Internet. do.

Here, since the connection release request message S73 and the connection release confirmation message S74 are the same as the connection release request message S71 and the connection release notification confirmation message S72, description thereof will be omitted.

Accordingly, the central processing unit 28 of the calling Internet phone IP1 analyzes the format of the Connection Confirmation message and confirms that a request for resource release or return has been made. 20 to output the virtual end tone to the first codec 18 (S75), so that the user recognizes that the Internet telephone connection is terminated through the voice input / output device 10 or 12 or 14. Do it. Thereafter, when the calling user operates the voice input / output device 10 or 12 or 14 in the call termination state, the operating state (hook on state) (S76 in Fig. 7C) is detected by the call intention detecting unit 34 to perform central processing. Is applied to the unit 28. Therefore, the central processing unit 28 of the calling Internet telephone IP1 ends the call mode.

On the other hand, in the above example, the case where the called party terminates the call has been described. However, when the calling party terminates the call, the disconnection completion message is transmitted from the calling Internet phone IP1 to the called Internet phone IP2. The same procedure as in the above example is performed except that the virtual end tone output occurs from the called Internet phone IP2 to the called user.

In addition, in the above embodiment, only the case where the call signaling messages are transmitted between the Internet telephones without passing through the call server CS has been described, but these messages are transferred between the Internet telephones via the call server CS. It may be easily understood from the above description that it may be configured as possible.

2.3. Call request from other internet phone (3rd internet call mode)

The third Internet call mode is a call by a call request from the other party's Internet phone IP2 to the Internet phone IP1, and the called party's Internet phone IP2 in the first Internet call mode described with reference to Figs. 6A to 6C. It is the same as the operation performed by (that is, the operation of (2.1) described above), and a description thereof will be omitted.

3. Gateway call mode

A gateway by a call request with a landline telephone (NP21) using an Internet telephone (IP1) as a gateway, or a call with a landline telephone (NP21) using the called Internet telephone (IP2) as a gateway. The call mode will be described.

3.1. Call request from general phone to other Internet phone (1 Gateway call mode)

Hereinafter, referring to Figs. 1, 8 and 9A to 9C, a call connection to another Internet telephone (IP2) from an ordinary telephone (NP12 in Fig. 1) connected via a PSTN via an Internet telephone (IP1 in Fig. 1) is shown. It demonstrates. Here, as described with reference to Fig. 5, it is assumed that the Internet telephones IP1 and IP2 have completed registration procedures with the call server CS, respectively.

First, the ringing signal (S80 in FIG. 9A) input through the telephone line connection unit 26 is detected by the data connection arrangement unit 24, and at this time, the data connection arrangement unit 24 detects the detection of the ring signal by the central processing unit 28. Notice). The central processing unit 28 controls the data connection arrangement 24 to be in the hook-off mode (S81 in FIG. 9A), and also provides a voice guidance message stored in memory means (whether it is a call request with itself or another). For checking whether the call is made with an Internet telephone or with another general telephone via another Internet telephone), the digital signal processor 20, the second codec 22, the data connection arrangement 24, and the telephone line. It outputs to the connection part 26 (S82 of FIG. 9A). According to the voice prompt message, when the user dials (i.e., operates the keypad of the outgoing telephone NP12) (S83 in Fig. 9A), a DTMF tone signal is generated, and this tone signal is transmitted through the data connection arrangement 24. The codec 22 is input to the digital signal processor 20 to convert the digital signal. The digital signal processing unit 20 detects this digitally converted tone signal, determines which key signal corresponds to which key input, and notifies the central processing unit 28. The central processing unit 28 according to the determination result of the digital signal processor 20 determines the type of connection request of the other party's telephone (NP12) (call request with its own Internet telephone (ie, general telephone call mode) or other Internet telephone). And a call request (ie, a second gateway call mode) with a general telephone via another internet phone. At this time, assume that the connection request of the other party's telephone NP12 is a call request (first gateway call mode) with another Internet telephone IP2.

At this time, the central processing unit 28 sets the first gateway mode without operating the bell signal generator (not shown).

Then, the central processing unit 28 of the Internet telephone IP1 sends an access request message (S84 in Fig. 9A) through the Internet network through the I / O buffer 30 using the UDP protocol. Output to call server (CS). In this case, the access permission request message is configured to include, for example, an information element as shown in Table 5 above.

In Table 5, the endpoint identifier information element is an identifier assigned by the call server CS to the gateway which is an endpoint of the Internet telephone through the registration confirmation message S4 of FIG. Further, the destination information information element is the telephone number of the destination Internet telephone IP2 dialed by the user of the ordinary telephone NP11. The source information information element is also the telephone number of the gateway, which is the endpoint of the source Internet phone (IP2). These information elements are the same as described in Table 5.

Thereafter, when the call server CS receives the access permission request message S84, the call server CS sends an access confirmation confirmation message (S85 in Fig. 9A) to the Internet network using the UDP protocol. To the Internet phone (IP1).

In this case, the access permission confirmation message (S85) is configured to include the information elements and the like shown in Table 6 above. Therefore, description thereof is omitted.

Thereafter, upon receiving the connection permission confirmation message S54 from the call server CS, the central processing unit 28 of the calling Internet phone IP1 does not feel bored of the calling user (i.e., the user of the general telephone). For this purpose, the digital signal processing unit 20 is controlled to output a virtual ringing tone to the second codec 18 (S86), whereby the ringing tone is connected to the data connection arrangement 24 and the telephone line connection 26. ) Is transmitted to the calling party's telephone (NP11) through the public telephone network (PSTN). This allows the calling user to know that a call connection with the called Internet phone IP2 is being attempted.

Further, the central processing unit 28 of the calling Internet telephone IP1 analyzes the access permission confirmation message S85, and then inputs and outputs a setup message (S87 in Fig. 9A) using the TCP protocol. It outputs to the internet network through the buffer 30 and transmits it to the called Internet phone (IP2).

Here, the setup message S87 is sent as an example of the format sent by the calling Internet phone IP1 to inform the general telephone NP21 via the called Internet phone IP2 that it wants to set up a connection. Information elements such as Table 7 may be included.

In Table 7, the content other than the user-to-user information element is the same as that in Table 7. The user-to-user information element may comprise the fields shown in Table 8. Here, the source information field is set to the gateway, and the overlap send field indicates whether or not to use the overlap transmission at the calling side. In this setup message (S87 of FIG. 9A), this field is " falls " FALSE) ". Therefore, the Setup Acknowledge message does not follow this Setup message S87. The other fields are the same as the contents described in connection with Table 8 above.

Thereafter, the central processing unit 28 of the called Internet phone IP2 receiving the setup message S56 analyzes the setup message S87 and transfers the Internet phone with itself to the gateway at the calling Internet phone IP1. You realize that you want to use.

Subsequently, the central processing unit 28 of the called Internet phone IP2 sends an access request message (S88 in FIG. 9A) through the Internet network through an input / output buffer using the UDP protocol. Output to call server (CS).

The access permission request message S88 may be configured with the information elements shown in Table 9, for example. Here, the source information information element indicates the telephone number of the gateway as an endpoint for the originating Internet telephone IP1, and the description of the other information elements is the same as that described with reference to Table 9.

Thereafter, when the call server CS receives the access permission request message S88, the call server CS sends an access confirmation confirmation message (S89 in FIG. 9A) to the Internet network using the UDP protocol. To the Internet phone (IP2).

Since the access permission confirmation message S89 is the same as that described in Table 10 and the description thereof, a description thereof will be omitted.

Then, upon receiving the connection permission confirmation message S89 from the call server CS, the central processing unit 28 of the called party's Internet telephone IP2 analyzes the connection permission confirmation message S89, and then the call progress message ( The Call Proceeding message (S63) is output to the internet network through the input / output buffer using the TCP protocol, and transmitted to the calling Internet telephone (IP1).

Since the call progress message (S91 of FIG. 9B) has the same format as the call progress message S19 of FIG. 6B, and an example thereof is the same as that of Table 11, description thereof will be omitted. In addition, the user-to-user information element may include a field as shown in Table 12 above.

Thereafter, the central processing unit 28 of the called party's Internet telephone IP2 operates the bell signal generator (not shown) to notify the called user of the telephone call (S92). This allows the called user to recognize that he or she has a phone call.

At the same time, the central processing unit 28 of the called Internet phone IP2 outputs an Alerting message (S93 in FIG. 9B) via the Internet network to the Internet network through the input / output buffer using the UDP protocol. To the calling Internet phone (IP1).

Since the alarm message S93 has the same format as that of the alarm message S21 of FIG. 6B and an example thereof is the same as that of Table 13, description thereof will be omitted. In addition, the user-to-user information element of the alert message S93 may include a field as shown in Table 14, and description thereof will be omitted.

In the following description, it is assumed that the user of the ordinary telephone NP21 receives a call for the sake of simplicity. When the telephone is received by the ordinary telephone NP21 in this manner, the digital signal processing unit 20 detects the hook-on state of the ordinary telephone NP21 and notifies the central processing unit 28.

Thereafter, the central processing unit 28 of the destination Internet telephone IP2, in response to the detection input of the hook-on state of the ordinary telephone NP21 from the digital signal processing unit 20, connects via the Internet network. message (S95 in Fig. 9B) is outputted to the internet network through the input / output buffer using the UDP protocol, and transmitted to the calling Internet telephone IP1.

Here, since the access message S95 has the same format as the access message S23 of FIG. 6B and an example thereof is the same as in Table 15, description thereof will be omitted. In addition, since the user-to-user information element of the access message S95 is configured to include a field as shown in Table 16, description thereof will be omitted.

Then, the originating Internet telephone IP1 and the called Internet telephone IP2 which have received the access message S95 mutually perform a signaling operation S66 as defined in, for example, ITU-T Recommendation H.245. . Since the H.245 signaling (S66) is the same as the call signaling (S24) of Figure 6b it will be omitted.

In the state where the call channel is established by the H.245 signaling (S24) as described above, a call is made between the general telephone NP11 and the destination Internet telephone IP1 IP2 using the calling Internet telephone IP1 as a gateway. (S97 in Fig. 9B) is made. At this time, the central processing unit 28 of the calling Internet telephone IP1 controls the digital signal processing unit 20 to terminate the generation of the virtual ringing tone. Thereafter, the voice signal of the user of the calling party general telephone NP11 inputted through the telephone line connection section 26 and the data connection arrangement section 24 is converted into a digital signal at the second codec 22 and the digital signal processing section 20 is executed. Are compressed and then input to the central processing unit 28. At this time, since the central processing unit 28 knows that the call is made through the Internet network (first gateway call mode), the packetized voice data obtained by packetizing the compressed coded voice signal using the RTP protocol is loaded into the TCP / IP buffer. The input / output buffer 30 outputs the packet voice data to the data network connection unit 42 in serial. Accordingly, the output packet voice data is transmitted to the called party's Internet telephone IP2 through the Internet network.

On the other hand, the voice data signal from the ordinary telephone NP21 via the destination Internet telephone IP2 is carried in TCP / IP in the form of a packet and reaches the input / output buffer 30 in the form of serial data through the Internet. The buffer 30 transfers the input serial data to the central processing unit 28 in packet units. The central processing unit 28 decomposes the RTP protocol packet data, checks the calling identification code (IP address) and the called identification code (IP address), and confirms that the packet voice data is transmitted from the called partner's Internet telephone. After confirming, the application port number is confirmed, and the decompressed compressed coded voice signal is transmitted to the digital signal processor 20. Accordingly, the digital signal processor 20 decodes the compressed coded signal and outputs the decoded signal to the second codec 22. Then, the decoded voice signal of the other party is converted into an analog signal in the second codec 22 and sent to the calling party general telephone NP11 through the data connection array 24, the telephone line connection unit 26, and the public telephone network PSTN. Is sent. By this process, a call is made with the called Internet phone IP2 through the Internet using the Internet telephone IP1 as a gateway in the calling general telephone NP11.

Then, when the user of the called Internet phone IP2 ends the call, the state (i.e., the hook-on state) is detected by the call intention detecting unit 34 and applied to the central processing unit 28 (S98 in Fig. 9C). ).

Accordingly, the central processing unit 28 of the called Internet phone IP2 completes disconnection from the calling Internet phone IP1 in order to terminate the call for the currently connected Call Signal Channel. Complete) outputs the message (S99) to the internet network through the input / output buffer using the UDP protocol and transmits it to the calling Internet telephone (IP1).

Since the connection release completion message S70 has the same format as the connection release completion message S27 of FIG. 6C and an example thereof is the same as in Table 17, description thereof will be omitted. In addition, since the user-to-user information element of the connection release completion message S70 is configured to include a field as shown in Table 18, description thereof will be omitted.

Thereafter, the central processing unit 28 of the called Internet phone IP2 transmits the disconnection completion message S70, and then a disconnection request message through the Internet network (S100 in FIG. 9C). By using the UDP protocol, and outputs to the Internet through the input and output buffer to the call server (CS). Here, the connection release request message S100 is formed in the same format as the connection release request message S28 of FIG. 6C, and an example thereof is the same as in Table 19, and thus description thereof will be omitted.

Thereafter, when the call server CS receives the connection release request message S101, the call server CS sends a disconnection confirmation message (S101 in FIG. 9C) via the Internet network to the Internet phone IP2. To send). Here, since the connection release confirmation message S101 has the same format as the connection release confirmation message S29 of FIG. 6C and an example thereof is the same as in Table 20, description thereof will be omitted.

Accordingly, the central processing unit 28 of the called party's Internet telephone IP2 analyzes the format of the Disconnect Confirmation message, confirms that a request for resource release or return has been made, and sets the call mode. Quit.

On the other hand, the central processing unit 28 of the originating Internet telephone IP1 receiving the disconnection complete message S99 analyzes the format of the disconnection complete message S99 and receives a call by the called party's Internet telephone IP2. After confirming the disconnection, the connection request message (S102 in FIG. 9C) is output through the I / O buffer through the I / O buffer to the call server CS through the Internet network.

Thereafter, when the call server CS receives the disconnection request message S102, the call server CS transmits a disconnection confirmation message S103 to the Internet telephone IP1 via the Internet. do.

Here, since the connection release request message S102 and the connection release confirmation message S103 are the same as the connection release request message S100 and the connection release notification confirmation message S101, description thereof will be omitted.

Accordingly, the central processing unit 28 of the calling Internet phone IP1 analyzes the format of the Connection Confirmation message and confirms that a request for resource release or return has been made. 20 to output the virtual end tone to the second codec 18 (S75) so that the data connection arrangement 24, the telephone line connection 26, and the public telephone network PSTN are controlled. It is to be transmitted to the calling side general telephone (NP11) (S104).

Therefore, when the user of the calling party general telephone NP11 recognizes that the Internet telephone connection is terminated and operates his / her telephone in the call termination state, the operating state (hook on state) is detected by the digital signal processing unit 20 and the central processing is performed. Is applied to the unit 28. Therefore, the central processing unit 28 of the calling Internet telephone IP1 controls the data connection arranging unit 24 to be hooked on and terminates the first gateway call mode.

On the other hand, in the above example, the case where the called party terminates the call has been described. However, when the calling party terminates the call, the disconnection completion message is transmitted from the calling Internet phone IP1 to the called Internet phone IP2. The same procedure as in the above example is performed except that the virtual end tone output occurs from the called Internet phone IP2 to the called user.

In addition, in the above embodiment, only the case where the call signaling messages are transmitted between the Internet telephones without passing through the call server CS has been described, but these messages are transferred between the Internet telephones via the call server CS. It may be easily understood from the above description that it may be configured as possible.

3.2. Call request from general telephone to other general telephone through second internet telephone through second internet telephone (second gateway call mode)

1, 8 and 10a to 10c, a general telephone (NP12 in FIG. 1) connected through a PSTN is connected to another Internet telephone (IP2) via an Internet telephone (IP1 in FIG. 1). The call connection to the general telephone NP21 will be described. Here, as described with reference to Fig. 5, it is assumed that the Internet telephones IP1 and IP2 have completed registration procedures with the call server CS, respectively.

First, since the ringing signal S110, the hook hop S111, and the voice guidance S112 in FIG. 10A are the same as those in steps S80 to S82 of FIG. 9A, a description thereof will be omitted. According to this voice prompt S112, the user's dialing (i.e., operating the keypad of the outgoing telephone NP12) (S113 in Fig. 10A) requests a call with the other general telephone NP21 via another Internet telephone IP2. Assume a second gateway call mode). At this time, the central processing unit 28 sets the second gateway mode without operating the bell signal generator (not shown).

Then, the central processing unit 28 of the Internet telephone IP1 sends an access request message (S114 in Fig. 10A) through the Internet network through the input / output buffer 30 using the UDP protocol. Output to call server (CS). In this case, the access permission request message is configured to include, for example, an information element as shown in Table 5 above.

In Table 5, the endpoint identifier information element is an identifier assigned by the call server CS to the gateway which is the endpoint of the Internet phone IP1 through the registration confirmation message S4 of FIG. Further, the destination information information element is the telephone number of the destination Internet telephone IP2 dialed by the user of the ordinary telephone NP11. The source information information element is also the telephone number of the gateway, which is the endpoint of the source Internet phone (IP2). These information elements are the same as described in Table 5.

Thereafter, when the call server CS receives the access permission request message S114, the call server CS sends an access confirmation message (S115 of 10a) to the Internet network using the UDP protocol. To the Internet telephone (IP1).

Here, the access permission confirmation message (S115) is configured to include the information elements and the like shown in Table 6 above. Therefore, description thereof is omitted.

Then, upon receiving the connection permission confirmation message S115 from the call server CS, the central processing unit 28 of the originating Internet telephone IP1 performs a virtual ringing tone as in step S86 of Fig. 9A. Transfer to the calling general telephone (NP11). This allows the calling user to know that a call connection with the called Internet phone IP2 is being attempted.

Further, the central processing unit 28 of the calling Internet telephone IP1 analyzes the connection permission confirmation message S115, and then inputs and outputs a setup message (S117 in FIG. 10A) using the TCP protocol. Output to the Internet network through the 30 and transmits to the called Internet phone (IP2).

Here, the setup message S117 is sent as an example of the format sent by the calling Internet phone IP1 to inform the general telephone NP21 via the called Internet phone IP2 that it wants to set up a connection. Information elements such as Table 7 may be included.

In Table 7, the content other than the user-to-user information element is the same as that in Table 7. The user-to-user information element may comprise the fields shown in Table 8. Here, the source address field indicates an alias address (i.e. phone number) of the gateway which is the endpoint for the originating Internet telephone IP1, the source information field is set to the gateway, and the destination address field is the destination Internet telephone. Indicates an alias address (i.e., a telephone number) of a gateway that is an endpoint for (IP2), and the Overlap Send field is set to "TRUE". Therefore, the Setup Acknowledge message must be followed for this Setup message S117. The other fields are the same as the contents described in connection with Table 8 above.

Thereafter, the central processing unit 28 of the called party's Internet phone IP2 receiving the setup message S117 analyzes the setup message S117 and transfers the Internet phone with itself to the gateway at the calling party's Internet phone IP1. You realize that you want to use.

Subsequently, the central processing unit 28 of the called Internet phone IP2 sends an access request message (S118 in FIG. 10A) through the Internet network through an input / output buffer using the UDP protocol. Output to call server (CS).

The access permission request message S118 may be configured with the information elements shown in Table 9, for example. Here, the endpoint identifier information element is an identifier assigned by the call server CS to the gateway which is the endpoint for the Internet telephone IP2 through the registration confirmation message S4 'of FIG. The destination information information element is also the telephone number of the gateway which is the endpoint for the Internet telephone IP2. Further, the source information information element indicates the telephone number of the gateway as an endpoint for the originating Internet telephone IP1, and the description of the other information elements is the same as that described with reference to Table 9.

After that, when the call server CS receives the access permission request message S119, the call server CS sends an access confirmation confirmation message (S119 in Fig. 10A) to the Internet network using the UDP protocol. To the Internet phone (IP2).

The access permission confirmation message S89 may be configured with the information elements shown in Table 9, for example. Here, the destination type information element indicates the type of endpoint of the destination Internet phone IP2, which is set as a gateway. Other information elements are the same as the contents described in relation to Table 9, and description thereof is omitted.

Thereafter, upon receiving the connection permission confirmation message S119 from the call server CS, the central processing unit 28 of the destination Internet telephone IP2 analyzes the connection permission confirmation message S119, and then sets up the setup permission message (S119). The Setup Acknowledge message (S120) is outputted to the internet network through the input / output buffer using the TCP protocol and transmitted to the calling Internet telephone (IP1). Here, the setup approval message (S120) may be configured to include the information elements, such as shown in Table 21 above. Since the information elements of the setup approval message (S120) are the same as those described with reference to Table 21, description thereof will be omitted.

Thereafter, the central processing unit 28 of the called Internet phone IP2 controls the data connection arrangement 24 to be hooked off for a call to the public telephone network PSTN (S121).

On the other hand, when the central processing unit 28 of the calling Internet phone IP1 receives the setup approval message S122, it analyzes it and confirms that the called Internet phone IP2 can be used as a gateway. An information message (S122 in FIG. 10B) is output to the Internet network through the input / output buffer using the TCP protocol, and transmitted to the destination Internet phone (IP2). In this case, the information message S61 may include an information element such as Table 22 described above. The contents other than the user-to-user information element in Table 22 are the same as those in Table 7. In addition, the user-to-user information element may also be configured to include the fields shown in Table 23, the contents of which are the same as those described with reference to Table 23. FIG.

Then, when the called party's Internet telephone IP2 receives the information message S122, the central processing unit 28 analyzes the message and then decodes the telephone number string of the general telephone NP21 extracted from the called party's number information element. In this case, dialing is performed as in step S62 of FIG. 7B (S123).

In addition, the information messages S122 'and S122 "in Fig. 10B are for providing additional information if necessary other than the telephone number of the ordinary telephone NP21. Here, the purpose of the information messages S122' and S122" is used. May be similarly used to transmit the telephone number of the destination general telephone NP21 as described in connection with the information messages S61 ', S61 "in FIG. 7B.

Thereafter, the central processing unit 28 of the called party's Internet telephone IP2 outputs a Call Proceeding message S124 to the Internet network through the input / output buffer using the TCP protocol to send the calling Internet telephone IP1. To send. Here, the call progress message S124 is formed in the same format as the call progress message S19 of FIG. 6B, and an example thereof is the same as in Table 11, and thus description thereof will be omitted. In addition, the user-to-user information element may include a field as shown in Table 12, and the like. In the call progress message S19, the called party endpoint, that is, the Internet, may be configured. A Destination Information field specifying whether the type of the telephone IP2 is a gateway or a terminal is set as a gateway.

Thereafter, the central processing unit 28 of the called Internet phone IP2 outputs an Alerting message (S125 in FIG. 10B) through the Internet network to the Internet network through the input / output buffer using the TCP protocol. Send to the calling Internet phone (IP1). Here, the alert message S125 is formed in the same format as the alert message S21 of FIG. 6B, and an example thereof is the same as in Table 13, and thus description thereof will be omitted. In addition, the user-to-user information element of the alert message S64 may include a field as shown in Table 14, and description thereof will be omitted. In the user-to-user information element, the destination information field is set to a gateway.

In the following description, it is assumed that the user of the ordinary telephone NP21 receives a call for the sake of simplicity. When the telephone is received by the ordinary telephone NP21 in this manner, the digital signal processing unit 20 detects the hook-on state of the ordinary telephone NP21 and notifies the central processing unit 28.

Thereafter, the central processing unit 28 of the destination Internet telephone IP2, in response to the detection input of the hook-on state of the ordinary telephone NP21 from the digital signal processing unit 20, connects via the Internet network. message (S126 in Fig. 10B) is outputted to the internet network through the input / output buffer using the TCP protocol, and transmitted to the calling Internet telephone IP1.

Here, since the access message S126 has the same format as the access message S23 of FIG. 6B and an example thereof is the same as in Table 15, description thereof will be omitted. In addition, since the user-to-user information element of the access message S126 is configured to include a field as shown in Table 16, description thereof will be omitted. In the user-to-user information element, the destination information field is set to a gateway.

Thereafter, the originating Internet telephone IP1 and the called Internet telephone IP2 which have received the access message S126 mutually perform a signaling operation S127 defined in, for example, ITU-T Recommendation H.245. . Since the H.245 signaling S127 is the same as the call signaling S24 of FIG. 6B, a description thereof will be omitted.

The call (S128 in Fig. 10B) is made between the calling party and the called party's Internet telephone IP1 (IP2) while the call channel is established by the H.245 signaling (S127). At this time, the central processing unit 28 of the calling Internet telephone IP1 controls the digital signal processing unit 20 to terminate the generation of the virtual ring tone.

Thereafter, the voice signal of the user of the calling party general telephone NP11 inputted through the telephone line connection section 26 and the data connection arrangement section 24 is converted into a digital signal at the second codec 22 and the digital signal processing section 20 is executed. Are compressed and then input to the central processing unit 28. At this time, since the central processing unit 28 knows that the call is made through the Internet network (first gateway call mode), the packetized voice data obtained by packetizing the compressed coded voice signal using the RTP protocol is loaded into the TCP / IP buffer. The input / output buffer 30 outputs the packet voice data to the data network connection unit 42 in serial. Accordingly, the output packet voice data is transmitted to the called party's Internet telephone IP2 through the Internet network.

On the other hand, the voice data signal from the ordinary telephone NP21 via the destination Internet telephone IP2 is carried in TCP / IP in the form of a packet and reaches the input / output buffer 30 in the form of serial data through the Internet. The buffer 30 transfers the input serial data to the central processing unit 28 in packet units. The central processing unit 28 decomposes the RTP protocol packet data, checks the origination identification code (IP address) and the terminating identification code (IP address), and the packet voice data identifies the other party's Internet telephone (IP2) called by the packet. After confirming that the transmission is via the general telephone NP21, the application port number is confirmed, and the decompressed encoded audio signal is transmitted to the digital signal processing unit 20. Accordingly, the digital signal processor 20 decodes the compressed coded signal and outputs the decoded signal to the second codec 22. Then, the decoded voice signal of the other party is converted into an analog signal in the second codec 22 and sent to the calling party general telephone NP11 through the data connection array 24, the telephone line connection unit 26, and the public telephone network PSTN. Is sent. By this process, a call is made with the called Internet phone IP2 through the Internet using the Internet telephone IP1 as a gateway in the calling general telephone NP11.

Thereafter, when the user of the destination general telephone NP21 ends the call, the end tone is detected by the digital signal processing unit 20 of the destination Internet telephone IP2 and applied to the central processing unit 28 (Fig. 10C). S129).

Accordingly, the central processing unit 28 of the called Internet phone IP2 controls the data access arrangement 24 to be in the hook-on mode, and controls the currently connected call signal channel for the call signal channel. In order to terminate the call, a Release Complete message (S131) is output to the calling Internet phone IP1 through the I / O buffer using the TCP protocol and transmitted to the calling Internet phone IP1. .

Here, the connection release completion message S70 is formed in the same format as the connection release completion message S27 of FIG. 6C and an example thereof is the same as in Table 17, and thus description thereof will be omitted. In addition, since the user-to-user information element of the connection release completion message S70 is configured to include a field as shown in Table 18, description thereof will be omitted.

Thereafter, the central processing unit 28 of the called Internet phone IP2 transmits the disconnection request message (SS132 in Fig. 10C) after transmitting the disconnection complete message S131. Using the output through the input and output buffer to the Internet network and transmits to the call server (CS). Here, the connection release request message S132 is formed in the same format as the connection release request message S28 of FIG. 6C, and an example thereof is the same as in Table 19, and thus description thereof will be omitted.

Thereafter, when the call server CS receives the connection release request message S132, the call server CS generates a connection confirmation message (S133 in FIG. 10C) using the UDP protocol. To the Internet phone (IP2). Here, the connection release confirmation message S72 has the same format as the connection release confirmation message S29 of FIG. 6C, and an example thereof is the same as that of Table 20, and thus description thereof will be omitted.

Accordingly, the central processing unit 28 of the called party's Internet telephone IP2 analyzes the format of the Disconnect Confirmation message, confirms that a request for resource release or return has been made, and sets the call mode. Quit.

On the other hand, the central processing unit 28 of the originating Internet phone IP1 receiving the disconnection completion message S99 analyzes the format of the disconnection complete message S131 and receives a call by the called party Internet phone IP2. After confirming the disconnection, the connection request message (S134 of FIG. 10C) is output through the I / O buffer through the I / O buffer through the Internet network and transmitted to the call server CS.

Thereafter, when the call server CS receives the connection release request message S134, the call server CS transmits a disconnection confirmation message S135 to the Internet telephone IP1 through the Internet network. do.

Here, the connection release request message (S134) and the connection release confirmation message (S134) can be configured in the same format as the connection release request message (S132) and the connection release notification confirmation message (S133) described above, Omit.

Accordingly, the central processing unit 28 of the calling Internet phone IP1 analyzes the format of the Connection Confirmation message S135 and confirms that a request for resource release or return has been made. By controlling the digital signal processing unit 20 to output a virtual end tone to the second codec 18 (S75), the data connection arrangement unit 24, the telephone line connection unit 26, and the public telephone network (PSTN). To be transmitted to the calling party's general telephone (NP11) (S104).

Therefore, when the user of the calling party general telephone NP11 recognizes that the Internet telephone connection is terminated and operates his / her telephone in the call termination state, the operating state (hook on state) is detected by the digital signal processing unit 20 and the central processing is performed. Is applied to the unit 28. Therefore, the central processing unit 28 of the calling Internet telephone IP1 controls the data connection arranging unit 24 to be hooked on and terminates the first gateway call mode.

On the other hand, in the above example, the case where the called party terminates the call has been described. However, when the calling party terminates the call, the disconnection completion message is transmitted from the calling Internet phone IP1 to the called Internet phone IP2. The same procedure as in the above example is performed except that the virtual end tone output occurs from the called Internet phone IP2 to the called user.

In addition, in the above embodiment, only the case where the call signaling messages are transmitted between the Internet telephones without passing through the call server CS has been described, but these messages are transferred between the Internet telephones via the call server CS. It may be easily understood from the above description that it may be configured as possible.

4. Dual channel call mode

Next, with reference to Figs. 1 and 11, a dual channel call mode using an Internet telephone employed in the present invention will be described.

In this call mode, the digital signal processing unit 20 according to the present invention has two channel methods, and the signal transmission / reception of the first codec 18 and the digital signal processing unit 20 and the second codec 22 This is possible because the signal transmission / reception of the digital signal processing unit 20 is performed in different time slots.

In Fig. 1, an Internet telephone IP1 is connected to another Internet telephone IP2 or a regular telephone NP22 via the telephone IP2, while the other telephone is connected via the Internet telephone IP1 from the ordinary telephone NP12. A dual call channel can be formed by a call request with IPn or a general telephone NP2n via this telephone IPn. In this case, the operation of the Internet call mode described in the above section (2) and the gateway call mode described in the above section (3) are performed simultaneously, and can be understood by the combination of the above descriptions. Is omitted.

5. Other

5.1. The other party's Internet phone that you want to call

When a call request is made to the call server, the call server receives a message indicating that the other party's Internet phone is busy or a message indicating that the other party's Internet phone is busy from the other party's Internet phone during call connection signaling with the other party's Internet phone. Accordingly, the central processing unit of the calling Internet phone controls the digital signal processing unit to generate a virtual busy tone notifying the user that he is busy.

5.2. When the other party's Internet phone is busy or already in gateway mode, if the other party wants to access the call through the user from the general phone, the other server's Internet phone is busy or the gateway call is made by the call server. Receives a message indicating that the mode is in progress, or receives a message indicating that the call is in progress or the gateway call mode is being performed from the other Internet telephone during call connection signaling with the other Internet telephone. Accordingly, the central processing unit of the calling Internet phone controls the digital signal processing unit so as to generate and output a virtual busy tone to notify the calling phone that the called party requested by the call is busy.

On the other hand, the specific embodiments exemplified above are merely to aid the understanding of the present invention and are not intended to limit the scope of the present invention. Therefore, the present invention is not limited to the above specific embodiment. As long as it can be easily modified and modified by those skilled in the art, the modifications and modifications are included within the scope of the present invention as long as the features of the present invention are used.

As described above, according to the present invention, a complete telephone-to-phone Internet phone in which the first Internet phone can talk to a general telephone through a public telephone network and can also talk to a second Internet phone through an Internet network. The system can be implemented in high quality.                     

According to the present invention, a general telephone connected to a first internet telephone through a public telephone network and a second internet telephone connected through an internet network without providing a separate internet phone gateway between the internet network and a public payphone network; Can enable the call. Accordingly, it is possible to implement a high quality internet phone call through an internet telephone installed in an office using a regular telephone while on the road.

Further, according to the present invention, the first Internet telephone is connected to the third Internet telephone at the time of a call connection between a general telephone connected to the first Internet telephone through a public telephone network and a second Internet telephone connected through the Internet. Since two-channel call function can be used to make a call, an internet phone installed in an office can make a high quality internet phone call through the internet phone while another person uses a regular phone while talking with another internet phone. .                     

Claims (9)

Call paths with the other party's Internet telephones through the Internet network, Call paths between the analog voice input / output means and the ordinary telephones requesting the call by means of call requests from public telephones through the public telephone network, and Call requests from general telephones through the public telephone network A plurality of internet telephones having a gateway function for forming a communication path between the general telephone requested by the telephone and the counterpart internet telephone via the internet network, and exchanging voice signals with the counterpart via the internet network; It is possible to connect to the plurality of Internet telephones through an internet network, and to manage the plurality of Internet telephones, receive a call connection grant message from the calling Internet telephone to determine whether the call is connected, and send the access permission message to the calling Internet telephone. And at least one call server for transmitting. The method of claim 1, The Internet telephone includes an analog voice input / output means, a first signal conversion means having a first codec, a code / decoding means having a digital signal processing unit, a central processing unit (MCU), and a digital network for converting serial data and packet data. Call intent detection means for detecting call intention based on connection means, data storage means, key input means, display means, operating state of analog voice input / output means, and data connection for separating and synthesizing telephone line connection portion and analog voice signal A second signal conversion means having a public telephone network connection means having an arrangement unit, a second codec for converting an analog signal inputted from the public telephone network connection means into a digital signal, and controlling each of the devices to the call server CS. Calls through call signaling such as registration processing of Internet phones and call requests to other Internet phones A packetizer processing for packetizing the speech coded signal compressed by the digital signal processor and a depacketizer processing function for decomposing the packetized speech coded signal inputted through the digital network connection means. And a central processing unit for controlling each device in accordance with an input of a to form the call path. A communication method in a system having a plurality of Internet telephones connected through an Internet network to exchange voice signals with a counterpart, and a call server managing the plurality of Internet telephones through the Internet network, When the user connects to the Internet telephone using a regular telephone, the central processing unit of the Internet telephone recognizes this; Requesting a specific key input by the central processing unit to select one of a normal telephone call mode and an internet telephone call mode; A first gateway call mode in which the connected ordinary telephone is in communication with the called Internet telephone via the calling Internet telephone in the connected ordinary telephone; Requesting a specific key input for selecting one of the second gateway call modes for talking with another general telephone through the called Internet telephone; In the case of the key input of the first gateway call mode, the central processing unit requests input of an identification number of the other party's Internet telephone. In the case of a key input of the second gateway call mode, the central processing unit requesting input of an identification number of the counterpart Internet telephone and a telephone number of the counterpart general telephone; Detecting a tone signal by a key input in the ordinary telephone and setting the first gateway call mode or the second gateway call mode according to the detection result; . The method of claim 3, And the central processing unit has a telephone number of the inputted counterpart phone and forms a call path with the counterpart phone through the internet network and the public telephone network requested by the call. The method of claim 3, If the determination result of the key input signal is a general telephone call mode, the central processing unit further includes the step of setting a general telephone call mode and operating a ring signal generator to notify that a call connection request is made to its Internet telephone. Internet telephony communication method characterized in that. The method of claim 3, The central processing unit detecting and recognizing the operation state of the Internet telephone by the call intention detecting unit and outputting a voice message requesting a specific key input for selecting a general telephone call mode and an internet telephone call mode already stored; Internet telephony communication method further comprising. The method of claim 3, The central processing unit uses UDP (User Datagram Protocol) in call signaling processing with a call server (CS), Transmission Control Protocol (TCP) in call signaling processing with other Internet telephones, and Real-time for transmitting voice data. Internet telephony method using Transport Protocol. The method of claim 3, After the central processing unit of the originating Internet telephone IP1 sets the Internet call mode, the central processing unit outputs an access request message through the internet network to the internet network and sends a call server CS. Transmitting to; Receiving, by the call server CS, an access permission request message, and transmitting an access confirmation message to an internet telephone IP1 through an internet network; The central processing unit of the calling Internet phone IP1 analyzes the access permission confirmation message, and then outputs a setup message indicating that it wants to set up a connection to the called Internet phone. Transmitting to IP2; The central processing unit of the called Internet telephone (IP2) is configured to receive and analyze the setup message; The central processing unit of the called Internet telephone (IP2) is configured to output an access request message (Admission Request message) through an internet network to the call server (CS); Receiving, by the call server CS, an access permission request message, and transmitting an access confirmation message to an internet phone IP2 through an internet network; When the central processing unit of the called Internet phone IP2 receives the access permission confirmation message from the call server CS, the central processing unit analyzes the access permission confirmation message S18, and then sends a call proceeding message to the Internet network. And outputting the transmission to the calling Internet telephone (IP1). The method of claim 6, The central processing unit of the called party's Internet telephone (IP2) further includes a step of outputting an alerting message (Alerting message) indicating that the bell is ringed to the Internet network and transmitting the message to the calling party's Internet telephone (IP1). Internet telephony method.

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