KR100342157B1 - sound transmission system using UDP protocol for PC-to-Phone method - Google Patents

Abstract

The present invention relates to a PC-to-phone remote voice call system using the UPD protocol, and more particularly, in a voice call system using the Internet and the PSTN 400 network, a server connected to the Internet by the call client 100 ( Requesting connection to the reception number of the called party 300 to the first step (S1); A second step (S2) of attempting to connect to the called party (300) by the received number requested by the server (200); A third step (S3) of receiving a connection request from the called party and sending a connection response signal to the server 200; A fourth step (S4) in which the server 200 sends a connection completion message to the call client 100 according to the transmitted response signal; A fifth step (S5) of preparing a buffer for recording and reproducing voice in real time and in both directions, wherein the call client 100 to which the completion message has been sent is received; A sixth step (S6) of storing the voice data input to the call client 100 in a buffer and transmitting the voice data to the server 200 using a UIP protocol; The seventh step S7 of transmitting the voice data of the calling client 100 transmitted to the server 200 to the voice compression board 500, and transmitting the voice data to the telephone receiver 300 through the PSTN 400 network. ; The voice of the called party 300 is transmitted to the voice compression board 500 of the server 200 through the PSTN 400 network, and the server 200 transmits the transmitted voice using a UIP protocol. An eighth step S8 of transmitting to 100; Receives the voice of the talker 300 transmitted from the server 200 in the voice playback buffer to play the voice, and reproduces a predetermined amount of the voice recording buffer received from the talk client 100 as real-time voice data, And a ninth step (S9) of preventing the accumulation of voice data in the buffer by deleting the contents of the buffer so as to be able to use the reproduced buffer next time and placing it at the end of the queue. The second step S2 includes transmitting a request impossible message from the server 200 to the talk client 100 when the connection attempt fails due to the PSTN 400 network, and the talk client 100. ) Or when transmitting the voice data recorded in 500Byte ~ 2000Byte unit over the network to transmit the voice from the server 200 in real time. It is attached to the sex data packet, and the server 200 side is compared to the PC to the far-end voice call system using the UDPI protocol, characterized in that to reproduce in real time. Accordingly, there is an advantage that the user can talk without sound collision even when using the sound of the computer game.

Description

Sound transmission system using UDP protocol for PC-to-Phone method}

The present invention relates to a system for making a telephone call without sound collision while using a computer sound card, and to a system for making a telephone call without collision with a sound used in a computer game in a voice call system using the Internet and a PSTN network. will be.

In general, Internet phones are divided into PC-to-PC, Phone-to-Phone, and PC-to-Phone.

First of all, the PC-to-PC method relates to a method in which PCs connected on a network can communicate with each other. The above method is a direct call method consisting of a microphone, a sound card, a speaker, and software installed directly on the PC to connect a telephone, and an indirect system consisting of a microphone, a sound card, a speaker, and a server connecting a telephone. There is a form of currency. This approach has the problem that all users who wish to make calls must be in front of the computer. In addition, there is another problem that the computer sound function cannot play a computer game, an MP3 player, or the like, except for the function of listening to and speaking the phone voice during the call of the above method.

Next, the phone-to-phone method is a method that allows users to make a call between users by accessing an Internet line using a general telephone between users, and has to separately provide a problem and a telephone that cannot be talked with a conventional telephone. The Internet phone general telephone has another problem that the telephone itself is expensive.

Next, there are two types of PC-to-Phone methods: paid and free. The paid method is to visit the website of the ISP company, download the program, install the program on the user's computer, enter the phone number to be dialed and press the call button. That's the way it can be. When connected in the above manner, the user talks through a microphone, which is a peripheral device of the computer, and receives a sign through the speaker. In this manner, the user is connected through an ISP company, and the ISP company is connected to a PSTN network, which is a general telephone network, to a general telephone. Therefore, the user pays the fee for using the telephone to the ISP. And, in the case of the free method, a user has recently entered a method of accessing an ISP and signing up as a member, and then calling his / her phone by entering a phone number to be called. The above scheme is a system for sending a voice data packet using a Transmission Control Protocol / Internet Protocol (TCP / IP). In addition, the above-described method has a problem in that a sound used in a computer game and a sound used in a voice call collide with each other. Therefore, there is another problem in that the sound use program used in the computer must be terminated when the user wants to make a call.

According to the present invention for solving the above problems, according to the PC-to-phone remote voice call system using the UPI protocol according to the present invention, the server 200 system equipped with the voice compression board 500 is the Internet network And another purpose is to provide a system that can call, call, using a normal phone and a mobile phone using the PSTN (400) network.

In order to prevent the accumulation of voice data in the buffer, the real-time playback of the queue on the buffer, the deletion, and continuous use in the form of a circular queue, so that voice can be delivered without a sound collision even if a telephone call is made while using computer sound. Another purpose is to provide an existing system.

1: The overall configuration of the PC-to-phone remote voice call system using the UD protocol according to the present invention.

2 is a message flow diagram of a PC-to-phone remote voice call system using the UDPI protocol according to the present invention.

3 is a voice recording flowchart of a PC to phone remote voice call system using the UD protocol according to the present invention.

4 is a voice recording reproduction diagram of a PC to phone remote voice call system using the UD protocol according to the present invention.

<Description of Symbols for Major Parts of Drawings>

100: Song client 200: Server

300: receiver 400: PSTN

500: voice compression board S1: first step

S2: second stage S3: third stage

S4: fourth step S5: fifth step

S6: Sixth Step S7: Seventh Step

S8: 8th step S9: 9th step

The present invention for achieving the above object, the voice call system using the Internet and the PSTN network, the first step for the call client to connect to the server connected to the Internet by the recipient's receiving number; A second step of attempting to connect to the called party with the received number requested by the server; A third step in which the called party receiving the connection request sends a connection response signal to the server; A fourth step of sending, by the server, a connection completion message to the call client according to the transmitted response signal; A fifth step of preparing a buffer for recording and reproducing the voice in real time and in both directions, wherein the call client to which the completion message has been sent is transmitted; A sixth step of storing the voice data input to the calling client in a buffer and transmitting the voice data to the server using a UDPI protocol; A seventh step of transmitting the voice data of the calling client transmitted to the server to a voice compression board, and transmitting the voice data to the caller's telephone through a PSTN network; An eighth step of transmitting the voice of the called party to a voice compression board of the server through a PSTN network, wherein the server transmits the transmitted voice to a calling client using a UIP protocol; Receives the voice of the called party transmitted from the server in the voice reproducing buffer, reproduces the voice, regenerates a predetermined amount of the voice recording buffer received in the talk client 100 as real-time voice data, and then executes the replayed buffer. Clearing the contents of the buffer so that it can be used to the end of the queue to prevent the accumulation of voice data in the buffer; 9th step; PC-to-phone method using the UIP protocol It relates to a voice call system.

In addition, the second step may be configured to include transmitting a request impossible message from the server to the calling client when the connection attempt fails due to the PSTN network.

In addition, when transmitting the voice data recorded in 500Byte ~ 2000Byte unit over the network in order to transmit the voice from the calling client or the server in real time, the voice data packet is sent to the voice data packet, and the server side sequentially PC-to-phone remote voice call system using the UD protocol, characterized in that the playback in real time compared to.

Accordingly, in the voice call system using the Internet and the PSTN network, the user can make a call without sound collision even while using computer sound.

Hereinafter, with reference to the accompanying drawings, it will be described in detail the PC to phone remote voice call system using the UI protocol according to the present invention.

1 is an overall configuration diagram of a PC-to-phone remote voice call system using the UPD protocol according to the present invention, Figure 2 is a message flow diagram of a PC-to-phone remote voice call system using the UPD protocol according to the present invention. .

First, step 1 (S1) will be described. The first step (S1) is a step in which the calling client 100 requests the connection to the server 200 connected to the Internet to the receiving number of the called party 300 to call.

Next, the second step (S2) will be described. Attempting to connect to the called party 300 using the received number requested by the server 200.

Next, step 3 (S3) will be described. The receiver 300 who has received the connection request sends a connection response signal to the server 200.

Next, step 4 (S4) will be described. The server 200 sends a connection completion message to the call client 100 according to the transmitted response signal.

Steps 1 to 4 are conventional techniques in the PC-to-Phone method, so detailed description thereof will be omitted.

Next, step 5 (S5) will be described. The call client 100, to which the completion message is sent, prepares a buffer for recording and reproducing voice in real time and in both directions. A step of preparing to make a phone call according to the connection complete message sent in step 4. That is, a recording buffer for recording the voice transmitted from the talk client 100 to the talker 300 and a playback buffer for reproducing the voice transmitted from the talker 300 to the talk client 100 are prepared. Thereafter, the voice of the calling client 100 or the called party 300 waits for input.

3 is a voice recording flowchart of a PC to phone remote voice call system using the UPD protocol according to the present invention.

Next, six steps (S6) will be described. The voice data input to the call client 100 is stored in a buffer, and is transmitted to the server 200 by using a UIP protocol. When the voice of the calling client 100 is input through a microphone, which is a computer peripheral device, in step 5 (S5), the sound card is input, and the user's voice is captured through the sound capture stream. All voices are copied through a wave stream and stored through a memory stream. The stored user voice is transmitted to the UDP protocol having the port number 2020 and sent to the server 200 through the network.

4 is a voice recording reproduction diagram of a PC-to-phone remote voice call system using the UDPI protocol according to the present invention.

Next, step 7 (S7) will be described. The voice data of the calling client 100 transmitted to the server 200 is transmitted to the voice compression board 500 and transmitted to the telephone receiver 300 through the PSTN 400 network. The server 200 receives the voice of the user of the calling client 100 transmitted from the port number 2021 in the form of a stream. The input stream is stored as a memory stream and copied back into a wave stream. At this time, the voice playback is started from the time point when the size of the wave stream (Wave Stream) exceeds 4,500Bytes (0.5 seconds), and is stored as a Wave Stream if the size does not exceed 4,500Bytes. When the wave stream is reproduced beyond 4,500 bytes, the wave stream is output to the output terminal of the sound card, and the call client 100 is connected to the receiver of the receiver 300 through the PSTN 400 network. The voice of the side is transmitted.

Next, eight steps (S8) will be described. The voice of the called party 300 is transmitted to the voice compression board 500 of the server 200 through the PSTN 400 network, and the server 200 transmits the transmitted voice using a UIP protocol. The step of transmitting to 100. In the eighth step S8, as in the seventh step S7, the server 200 receives the voice of the called party 300 transmitted from the port number 2021 in the form of a stream. The input stream is stored as a memory stream and copied back into a wave stream. At this time, the voice playback is started from the point of time when the size of the wave stream (Wave Stream) exceeds 4,500Bytes (0.5 seconds), and is stored as a Wave Stream if the size does not exceed 4,500Bytes. When the wave stream is reproduced beyond 4,500 bytes, the wave stream is output to the output terminal of the sound card, and the user of the talk client 100 hears the voice of the talker 300.

Next, step 9 (S9) will be described. Receives the voice of the talker 300 transmitted from the server 200 in the voice playback buffer to play the voice, and reproduces a predetermined amount of the voice recording buffer received from the talk client 100 as real-time voice data, This step is to prevent the accumulation of audio data in the buffer by deleting the contents of the buffer so that it can be used later.

Hereinafter, an embodiment using the PC-to-phone remote voice call system using the UIP protocol according to the present invention will be described.

First, a user connects to the server 200 using the call client 100 system. Here, the calling client 100 is connected to the server 200 by executing an Internet browser in a state connected to the network, inputting a URL. In this case, the server 200 receives user information and searches a user database stored in the server 200 to confirm the user. After the checking process, the user enters a phone number to which he or she wants to connect in the calling client 100 system, and requests the server 200 to make a telephone connection. Accordingly, the server 200 attempts to make a telephone call by sending a ring tone to the called party 300 to the telephone number input by the calling client 100. In addition, the receiver 300 recognizes the ringing tone, and in this case, the receiver responds to the ringing tone by picking up the handset or pressing a handset button. The response signal is transmitted to the server 200, and the server 200 sends a response signal to the talk client 100. Accordingly, the call client 100 system records a user voice and allocates a buffer for reproducing the voice of the talker 300.

First, voice recording goes through the following process. The user of the talk client 100 inputs his / her voice through a microphone, which is a peripheral device of a computer, and the voice is input to an input terminal of a sound card and captures the user's voice through a sound capture stream. The captured voice is copied through a wave stream and stored through a memory stream. The stored user voice is transmitted to the UDP protocol having the port number 2020 and sent to the server 200 through the network.

In the voice playback, the server 200 receives the voice of the receiver 300 transmitted from the port number 2021 in the form of a stream. The input stream is stored as a memory stream and copied back into a wave stream. At this time, the voice playback is started from the point of time when the size of the wave stream (Wave Stream) exceeds 4,500Bytes (0.5 seconds), and is stored as a Wave Stream if the size does not exceed 4,500Bytes. When the wave stream is played back more than 4,500 bytes, the wave stream is output to the output terminal of the sound card and delivers the voice to the outside through a speaker connected to the output terminal.

In addition, the above-mentioned voice recording and voice reproduction delete the voice data stored in the buffer at the moment of recording and reproducing on the buffer in the talk client 100 system, and moves to the last position of the buffer prepared for recording and reproducing. Positioning the buffer prevents the accumulation of voice data in the buffer so that computer sound and voice do not collide with each other.

Next, the voice of the user of the calling client 100 recorded by the server 200 is transmitted to the telephone of the receiver 300 through the PSTN 400 network through the voice compression board.

As described above, the voice of the caller 300 is input through the PSTN 400 network, transferred to the server 200 through the voice compression board 500, and the user of the call client 100 through the process of playing the voice. Is spoken to you.

As described above, in order to prevent the accumulation of voice data in the buffer, by real-time playing the queue on the buffer, deleting it and continuing to use it in the form of a circular queue, even if a telephone call is performed while using computer sound, the voice is not crashed. There is an advantage that can be delivered.

Claims (3)

In the voice call system using the Internet and PSTN network, A first step of a calling client connecting to a server connected to the Internet using a called party's incoming number; A second step of attempting to connect to the called party with the received number requested by the server; A third step in which the called party receiving the connection request sends a connection response signal to the server; A fourth step of sending, by the server, a connection completion message to the call client according to the transmitted response signal; A fifth step of preparing a buffer for recording and reproducing the voice in real time and in both directions, wherein the call client to which the completion message has been sent is transmitted; A sixth step of storing the voice data input to the calling client in a buffer and transmitting the voice data to the server using a UDPI protocol; A seventh step of transmitting the voice data of the calling client transmitted to the server to a voice compression board, and transmitting the voice data to the caller's telephone through a PSTN network; An eighth step of transmitting the voice of the called party to a voice compression board of the server through a PSTN network, wherein the server transmits the transmitted voice to a calling client using a UIP protocol; Receives the voice of the called party transmitted from the server in the voice playback buffer to play the voice, and reproduces a predetermined amount of the voice recording buffer received from the talk client as real time voice data, and uses the finished buffer for the next time. And a ninth step of preventing the accumulation of the voice data in the buffer by deleting the contents of the buffer and placing the buffer at the end of the queue. . The method of claim 1, The second step is a PC-to-phone long distance voice call system using a UPD protocol, characterized in that the server if the connection attempt fails due to the PSTN network to transmit a request impossible message from the server. The method of claim 1, To transmit the voice data recorded in 500Byte ~ 3200Byte units through the network to send the voice data from the calling client or the server in real time, the time when the voice data is sent is pasted to the voice data packet, and the time is sequentially compared on the server side. PC-to-phone remote voice call system using the UPD protocol, characterized in that to play in real time.