KR20040039548A - System and Method for Connecting Internet Phone Using Private Branch Exchange - Google Patents

Abstract

PURPOSE: An Internet phone access system using a private exchange and a method therefor are provided to connect an exterior type VoIP gateway to a private switching system through an SLT(Single Line Telephone) interface without adding a special function and to support a telephony service through the Internet in a phone-to-phone or PC-to-phone form. CONSTITUTION: A private switching system(300) having a VoIP gateway function is connected with a plurality of station subscriber terminals(400-1-400-N). Also the private switching system(300) is connected with a plurality of PCs(100-1-100-N) through an IP network. In the case of a voice call between a station subscriber and an Internet phone user in the private switching system(300), a VoIP gateway system(200), interworking with the private switching system(300) through the SLT interface, accesses the IP network. The VoIP gateway system(200) is comprised of a ring signal detection part(201), a loop signal control part(202), a DTMF transmit-receive part(203), a tone transmitting part(204), a memory part(205), a central control part(206), a VoIP part(207), a voice codec part(208), and a LAN access part(209).

Description

System and Method for Connecting Internet Phone Using Private Branch Exchange}

The present invention relates to a system and method for accessing an Internet phone using a private exchange. In particular, an external VoIP gateway is connected to a private exchange system through an SLT interface without an additional function to provide an Internet in a telephone-to-phone or PC-to-telephone format. The present invention relates to an Internet phone access system and a method using a private exchange for supporting a telephone service.

In general, with the explosive increase in the use of the Internet, the field of application has gradually expanded, and as a result, there has been a great deal of interest in Voice over Internet Protocol (VoIP) technology that enables the digitization of existing voice calls and the absorption into data networks. The VoIP gateway device is a gateway that connects a public switched telephone network (PSTN) network and an Internet protocol (IP) network, and connects communication networks of both sides to allow IP telephone users and general telephone users to communicate with each other.

Internet phone using the VoIP gateway means real-time voice conversation between PC (Personal Computer) users through IP network, and the demand is gradually increasing due to low communication rates. It is divided into PC-to-PC, PC-to-phone, and phone-to-phone systems. To date, many developments have been made on PC-to-PC and PC-to-phone systems.

Here, the PC-to-PC method is a method of connecting and transmitting a voice through a microphone and a speaker by connecting a PC and a PC. Both sides access the IP network at the same time, and then enter analog through a microphone using the same program. When the voice signal is digitized and compressed, the packetized voice is sent to the counterpart PC through the IP network, and the corresponding PC restores the packetized voice and outputs it through the speaker to make a call.

In addition, the PC-to-telephone method installs a program on a corresponding PC, connects to the Internet, and communicates with a public telephone network through a VoIP gateway installed in a specific area.

However, in the related art, an expensive interface board was required for connection between a private switching system and a VoIP gateway device in order to provide a telephone service through the Internet in the form of telephone-to-phone or PC-to-phone.

The present invention is to solve the above-mentioned general problems, the object is to connect the VoIP gateway of the external type to the private switching system by the SLT interface without any additional function to connect the Internet of the phone-to-phone or PC-to-phone type In support of the telephone service.

1 is a diagram showing an Internet phone access system using a private exchange according to the present invention.

2 and 3 are flowcharts illustrating a method for accessing an Internet phone using a private exchange according to an embodiment of the present invention.

* Description of the symbols for the main parts of the drawings *

201: ring signal detector 206: central controller

202: loop signal control unit 207: Voice over Internet Protocol (VoIP) protocol unit

203: DTMF transmission / reception section 208: Voice codec section

204: tone transmitter 209: LAN connection

205: memory unit

Internet phone access system using a private exchange according to an embodiment of the present invention for achieving the above object in the voice call between the subscriber and the Internet phone user of the private exchange system, through the private exchange system and the SLT interface And a VoIP gateway device connected to and accessing the IP network.

In addition, the Internet phone access method using a private exchange according to an embodiment of the present invention for achieving the above object is a telephone service method using a private gateway system and a VoIP gateway device interworking through an SLT interface, Detecting a ring signal and a DTMF signal transmitted from the switching system via the SLT interface; Recognizing the detected ring signal and DTMF, characterized in that it comprises the step of requesting the VoIP transmission to the corresponding IP network.

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

In the private switching system equipped with the VoIP gateway function according to an embodiment of the present invention, as shown in FIG. 1, a plurality of extension subscriber stations 400-1 to 400-N are connected, PCs 100-1 to 100-N or wired terminals (not shown) are connected.

The VoIP gateway device 200 accesses an IP network by interworking with the private switching system 300 through an SLT interface during a voice call between an extension subscriber of the private switching system 300 and an Internet phone user. As described above, the ring signal detector 201, the loop signal controller 202, the DTMF transmitter / receiver 203, the tone transmitter 204, the memory 205, the central controller 206, and VoIP A protocol section 207, a voice codec section 208, and a LAN connection section 209 are provided.

At this time, the ring signal detection unit 201 is responsible for detecting and reporting the ring signal transmitted from the private switching system connection unit 301 to the central control unit 206, the loop signal control unit 202 is the central control unit 206 Off-Hook signal transmission request from the off-hook signal to the private exchange system connection unit 301 is responsible for transmitting the off-hook signal, DTMF (Dual Tone MultiFrequency) transmitting / receiving unit 203 is a private exchange system connection ( It is responsible for receiving the DTMF information from the 301, the tone transmitter 204 generates a busy tone, pending melody, ring tone, etc. used in the private exchange system 300, the control signal applied from the central control unit 206 According to the role of stopping or transmitting the corresponding tone, the memory unit 205, the algorithm for the operation of the VoIP gateway device 200, and the destination IP connected through the IP network is registered, the central control unit ( 20 6) extracts a predefined called party IP according to the control signal applied from the central control unit, the central control unit 206 controls and arbitrates the functions of the entire building block constituting the VoIP gateway device 200, It is responsible for controlling the overall operation to provide a telephone service over the IP network in accordance with the status change reported from the private switching system connection unit 301, the VoIP protocol unit 207 is a control authorized from the central control unit 206 It is responsible for converting voice data into packet data when requesting for making a VoIP call to an IP network according to a signal and receiving a connection message from the corresponding IP network indicating that the connection with the VoIP gateway device 200 is completed. The codec unit 208 voices the packet data in accordance with a control signal applied from the central control unit 206 to the called party via the LAN connection unit 209. It serves to transmit.

Operation of the VoIP gateway device interfaced with the private exchange system according to an embodiment of the present invention will be described below with reference to the flowcharts of FIGS. 2 and 3.

First, Fig. 2 describes a call processing procedure when a call is received from the private exchange system 300.

When an event of calling an idle subscriber number among SLT (Single Line Telephone) subscribers connected to the VoIP gateway device 200 from the extension subscriber station 400-1 subscribed to the private exchange system 300 occurs ( In step S21, the private exchange system 300 transmits a ring signal to the line (step S22).

The ring signal detection unit 201 of the VoIP gateway device 200 detects the ring signal from the corresponding line and reports it to the central control unit 206 (step S23). After receiving the ring signal detection, the central control unit 206 requests the loop signal control unit 202 to transmit an off-hook signal (step S24), and the DTMF transmission / reception unit 203 receives DTMF reception from the line. Request (step S25), and also request the tone transmitter 204 to transmit a dial tone (Dial Tone) on the line (step S26).

At this time, the extension subscriber hears the dial tone and dials the called party's number and '#' using DTMF.

When DTMF is received from the extension subscriber station 400-1 (step S27), the DTMF transmitting / receiving unit 203 of the VoIP gateway device 200 receives the number and reports it to the central control unit 206 (step S27). S28).

The process unit 206 reports DTMF information from the DTMF transmitter / receiver 203 and requests the tone transmitter 205 to stop transmission of the dial tone (step S29).

The central control unit 206 reports the DTMF information from the DTMF receiving unit 203, requests the DTMF transmission / receiving unit 203 to stop DTMF reception (step S30), and the corresponding number previously defined in the memory unit 205. After extracting the called party's IP address (step S31), the VoIP protocol unit 207 requests transmission of the VoIP call (step S32). At this time, the VoIP protocol unit 207 transmits the VoIP call through the LAN connection unit 210.

Thereafter, when the VoIP protocol unit 208 receives an alarm message from the called party (step S33), the VoIP protocol unit 208 reports the alarm message to the central control unit 206 (step S34). Upon receiving this report, the central controller 206 requests the tone transmitter 204 to transmit a RingBackTone to the line. When the called party responds, the VoIP protocol unit 207 receives a connection message from the IP network and reports it to the central control unit 206. Thereafter, the central control unit 206 requests the voice codec unit 209 to packetize the voice signal input from the line (step S35), and sends the packetized data to the called party to the VoIP protocol unit 208. (Step S36).

At the same time, the voice codec unit 209 is requested to voice the packetized data received from the VoIP protocol unit 208 and send it out to the corresponding line.

3 illustrates a call processing procedure when a call is received from an IP network.

First, the VoIP protocol unit 207 receives a setup (SET-UP) message through the LAN connection unit 209 and reports it to the central control unit 206 (step S301). The central control unit 206 selects an idle line among the lines connected to the private exchange system 300 and requests the loop signal control unit 202 to transmit an off-hook signal (step S302), and the memory unit 205. After extracting the subscriber number of the private exchange system 300 previously defined (step S303), the DTMF transmitter 203 is requested to transmit the number (step S304). At the same time, it requests the VoIP protocol unit 207 to packetize an alarm message and a connect message (step S305), and requests that the voice codec unit 208 packetize and transmit it to the calling party. (Step S306), the VoIP protocol unit 207 requests that the packetized data be sent to the called party. At the same time, the voice codec unit 208 requests the voice coded data received from the VoIP protocol unit 207 to be voiced and transmitted on the corresponding line (step S307).

Although the preferred embodiments of the present invention have been described in detail above, it will be understood by those skilled in the art that the present invention may be modified or modified in various ways. Therefore, changes of the embodiments of the present invention will not be able to escape the technical scope of the present invention.

As described above, the present invention requires an expensive interface board by connecting an external VoIP gateway to a private switching system without additional functions to support a telephone service through a telephone-to-phone or PC-to-telephone Internet. There is no cost savings.

Claims (4)

In a voice call between an extension subscriber of an private switching system and an Internet phone user, an Internet phone connected to the private switching system through an SLT interface and including a VoIP gateway device for accessing an IP network is provided. Connection system. In the telephone service method using a private gateway system and VoIP gateway device interworking through the SLT interface, Detecting a ring signal and a DTMF signal transmitted from the private switching system via an SLT interface; And recognizing the detected ring signal and DTMF and requesting a VoIP call to a corresponding IP network. The method of claim 2, Detecting the ring signal and DTMF signal, A ring signal transmitted from a private switching system when an event of calling an idle subscriber number among single line telephone (SLT) subscribers connected to a VoIP gateway device occurs from an extension subscriber terminal subscribed to the private switching system. Detecting c); Requesting transmission of an off-hook signal, DTMF reception, and dial tone after reporting the ring signal; Receiving a corresponding number when a DTMF is received from the extension subscriber station, requesting to stop transmission of a dial tone and to stop DTMF reception; And extracting a called party's IP address for a predetermined number, and requesting an outgoing call of the VoIP call. The method of claim 2, Requiring VoIP transmission to the IP network, Transmitting a ring back tone to a corresponding line when receiving an alarm message from a called party; When the called party responds, converting a voice signal input from the line into packet data when receiving a connect message from an IP network; And transmitting the packet data to a called party.