KR101169433B1 - Apparatus and Method for Cancelling Line Echo In VoIP System - Google Patents

Abstract

The present invention relates to an apparatus and a method for preventing line echo in a VoIP system, generating and transmitting a VoIP voice signal to a PSTN general terminal, and analyzing a voice packet received from the PSTN general terminal to check whether line echo is generated. By transmitting the voice signal transmission gain value at the time of the line echo and applying it to the voice communication between the IP terminal and the PSTN general terminal, the transmission of the VoIP terminal capable of generating the line echo at the beginning of the interworking between the VoIP system and the PSTN. By obtaining the threshold of gain in advance through measurement, it is possible to guarantee a certain level or more of VoIP call quality when interworking with the network.

Description

Apparatus and Method for Canceling Line Echo In VoIP System

1 is a view showing an interworking relationship between a telephone network and a PSTN through the Internet to which the present invention is applied.

2 is a block diagram of a line echo prevention apparatus of a VoIP system according to the present invention and a connection relationship with a VoIP network.

3 shows a line echo prevention operation flow of the VoIP system according to the present invention.

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

100: VoIP gateway 200: line echo protection device

210: call connection unit 220: voice generation unit

230: packet analyzer 240: voice analyzer

250: control unit 300: general telephone terminal

The present invention relates to an apparatus and method for preventing line echo in a VoIP system.

Voice over Internet Protocol (VoIP) service, commonly referred to as Internet telephony, is a new type of telephony service that transmits or receives voices without distinction of calling rights by transmitting calls in packet units over the Internet.

The purpose of using VoIP can be seen in many ways, but the purpose can be examined with the following advantages.

First, investment in network facilities is reduced due to the integration of telephone and data networks. It is possible to reduce the network equipment investment cost by not installing a separate telephone network for the voice communication network and a separate network for the network. Second, the management cost is reduced and efficiency is improved due to the integrated network. Because data and voice are processed using a single network, two effects can be achieved compared to the current dual network configuration, which is to reduce management costs and improve efficiency. Third, it is easy to integrate internet-based multimedia services. By using the same network for voice and data, it is easy to configure many additional services that were difficult to use in existing circuit-switched telephones such as video conferencing.

When VoIP service is used, a means for finding and signaling a counterpart to communicate with is required. The types of VoIP signaling include ITU-T's H.323 and IETF's Session Initiation Protocol (SIP).

So far, many H.323-based VoIP services have been developed. However, SIP, which has recently emerged, is easy to parse, compile, is scalable, and text-based. have.

VoIP end-point devices (gateways, IP phones, PCs, etc.) perform mutual voice communication by continuously exchanging RTP packets between senders and receivers via packetized voice over the RTP protocol on the IP network. Exchanging RTP packets incurs traffic load on the IP network, and it is an important factor that determines the performance of the entire equipment due to the processing of RTP packets processed through the socket interface of the media processing section for the VoIP equipment. Works.

Here, the Real-Time Transport Protocol (RTP) provides an end-to-end transmission service for transmitting real-time data such as audio or video using a multicast or unicast network. RTP has no concept of connection and generally operates on the upper layer of User Datagram Protocol (UDP) and uses UDP's multiplexing and checksum services.

One of the biggest problems in delivering voice over the Internet is quality of service (QoS) and packet loss.

In particular, in the case of an IP telephone-to-normal telephone type VoIP call, line echoes occur frequently in the IP telephone. These line echoes do not have a standardized value for the conditions under which they occur, and the administrator must manually respond to each network environment where line echoes occur. In addition, as the loss plan of the VoIP network is changed to reduce the line echo in a mixed network environment, the loss plan of the entire VoIP network is changed, but to what extent and how much change is made. The reality is that it is not understood.

In order to solve the above problem, the line echo of the VoIP system to maintain the gain level of the line echo does not occur by using the gain value at the time when the line echo occurs in the VoIP phone connected to a single line An object of the present invention is to provide a prevention device and a method thereof.

According to an aspect of the present invention, there is provided a method for preventing line echo generation of a VoIP system, the method comprising: generating a VoIP voice signal and transmitting the generated VoIP voice signal to a PSTN general terminal; Checking whether a line echo is generated by analyzing a voice packet received from the PSTN general terminal; And storing the voice signal transmission gain value at the time when the line echo occurs and applying it to voice communication between the IP terminal and the PSTN general terminal.

The line echo generation prevention method of the VoIP system, if the line echo does not occur as a result of the check whether the line echo occurs, generating the VoIP voice signal by increasing the gain value of the voice signal and transmitting to the PSTN general terminal and the PSTN The method may further include repeating the checking of the occurrence of the line echo by analyzing the voice packet received from the general terminal.

The step of applying the voice communication between the IP terminal and the PSTN general terminal generates a VoIP voice signal based on a voice signal transmission gain value less than the voice signal transmission gain value at the time when the line echo occurs.

In accordance with another aspect of the present invention, an apparatus for preventing line echo generation of a VoIP system includes: a voice generator for converting a voice into a packet signal and transmitting the voice to a PSTN general terminal; A packet analyzer extracting a voice packet by analyzing a packet received from the PSTN general terminal; A voice analyzer which receives the extracted voice packet and analyzes the voice packet to check whether a line echo is generated; And a controller for storing a voice signal transmission gain value at the time point at which the line echo occurs when the generation of the line echo is detected from the voice analyzer.

When the generation of the line echo is not detected from the voice analyzer, the controller controls the voice generator to increase the gain value of the voice signal to generate the VoIP voice signal and transmit it to the PSTN general terminal.

The controller may further control the voice generator to stop generating the VoIP voice signal when a line echo is detected from the voice analyzer.

The apparatus for preventing line echo generation of the VoIP system may further include a call connection unit that generates a VoIP call and connects the PSTN general terminal.

Hereinafter, a preferred embodiment according to the present invention will be described in detail with reference to the drawings.

1 shows an interworking relationship between a telephone network and a PSTN through the Internet to which the present invention is applied.

Referring to FIG. 1, it can be seen that the gateway 100 connects an Internet network to which a public switched telephone network (PSTN), which is an existing telephone network, and an Internet telephone are connected. In the existing telephone network mainly based on the wired network, circuit switching is mainly used for the purpose of delivering voice, whereas the Internet network is mainly the technology for multimedia transmission such as packet switched video and voice. As such, a gateway as shown in FIG. 1 is required to interwork two networks having different communication methods.

Referring to the form of VoIP end-to-end service with reference to Figure 1, the voice service between the communication community having the type of IP phone-to-IP phone that the communication through the IP protocol through the Internet network, the communication ID or IP address through the Internet network PC to PC type, IP protocol to interwork Internet network and PSTN network and IP phone to landline phone to provide voice service between incoming or regular wired / wireless phone, regular phone of PSTN network For example, there is a form of a landline-to-landline connection with a landline of another PSTN network.

The components of VoIP system are largely divided into application layer, signal layer, and media layer, and each layer performs communication using the same protocol as the other party. The application layer performs the creation and execution of services, intelligent call processing, service management, and the signal layer performs call processing, call conversion, resource management, and media control. The media layer is responsible for the processing and delivery of actual data, transformation, quality assurance, tone generation, and the like.

2 shows a block configuration of a line echo prevention apparatus of a VoIP system according to the present invention and a connection relationship with a VoIP network.

As shown in FIG. 2, the line echo prevention apparatus 200 according to the present invention is connected to a general telephone terminal 300 through the VoIP gateway 100. In FIG. 2, the line echo prevention apparatus 200 is a transmitting side, and the general telephone terminal 300 is a receiving side. The line connected to the general telephone terminal 300 is a single line, and consists of one data line and one power line.

The VoIP gateway 100 connects the PSTN network and the IP network to enable communication. In the VoIP network, H.323 and Session Initiation Protocol (SIP) are generally used as signaling protocols. Here, the gateway refers to a device that interconnects two or more different types or the same types of communication networks in a broad sense, and transmits and receives information between them. In a narrow sense, protocols are different in each layer of the OSI basic reference model. It is understood as a device for performing protocol conversion by interconnecting a plurality of communication networks. That is, the VoIP gateway 100 performs a protocol conversion by interconnecting the PSTN and the Internet.

 The VoIP agent 200 includes a call connection unit 210, a voice generator 220, a packet analyzer 230, and a voice analyzer 240.

The call connection unit 210 generates a VoIP call and performs a call connection with the general telephone terminal 300 via the VoIP gateway 100.

When the call is connected and the user's voice is input to the voice generator 220, the voice generator 220 converts the voice into a packet signal and transmits the voice to the VoIP gateway 100. The voice signal transmitted to the VoIP gateway 100 is finally received by the general telephone terminal 300 of the called party, converted into an analog signal, and transmitted to the user.

The packet analyzer 230 analyzes a packet received from the general telephone terminal 300, extracts only a voice packet, and outputs the voice packet to the voice analyzer 240. The voice analyzer 240 receives a voice packet from the packet analyzer 230 and measures a peak gain of the voice packet.

On the other hand, the controller 250 controls the voice generator 220 to generate a voice while increasing the gain value of the voice signal, and generates the voice at the time when line echo occurs in the voice signal received from the PSTN network. Stop and record the maximum transmission (Tx) gain of the voice signal generated by the voice generator 220. In this case, the Tx gain value of the voice signal is actually applied to the voice signal generation of the VoIP terminal in which the VoIP call is made. Since the maximum transmit gain is the voice gain value at the time of the line echo, the voice transmission is made smaller than this value. The gain will have to be set.

3 shows a line echo prevention operation flow of the VoIP system according to the present invention.

First, a call is connected (S301), and a voice is generated (S302). The generated voice is transmitted to the other general telephone terminal through the VoIP gateway. When a packet is received from the counterpart terminal (S303), it is checked whether a line echo occurs from the received packet (S304). If the line echo does not occur, the voice gain value is adjusted to a value increased from the initial voice gain value (S305) to generate and transmit the voice (S302). Then, steps S302 to S305 are repeated until a line echo occurs by analyzing the packet received from the counterpart terminal.

If the occurrence of the line echo is detected while repeating the above process (Yes in S304), the voice generation is stopped (S306), and the voice gain value at the time of the line echo generation is stored (S307).

The present invention, by obtaining the threshold value of the transmission gain of the VoIP terminal that can generate a line echo in the early stage of the interworking between the VoIP system and the PSTN, can ensure the call quality of the VoIP at a certain level or more when the network interworking. By securing the transmission gain that generates line echoes of single lines interworking with VoIP, and reconstructing scenarios by dataizing the countermeasures against line echo prevention, it is possible to shorten the construction time when the single line interworking of VoIP network is performed later. It can effectively cope with the problem of a drop in call volume caused by a change in VoIP loss plan that is changed to prevent echo.

Claims (8)

In the line echo prevention method of the VoIP system which works with PSTN, Generating and transmitting a VoIP voice signal to a PSTN general terminal; Checking whether a line echo is generated by analyzing a voice packet received from the PSTN general terminal; And A method of preventing line echo generation in a VoIP system comprising storing a voice signal transmission gain value at a time point at which a line echo occurs and applying it to voice communication between an IP terminal and a PSTN general terminal. The method of claim 1, If a line echo does not occur as a result of checking whether the line echo occurs, Generating the VoIP voice signal by increasing the gain value of the voice signal and transmitting the voice signal to the PSTN general terminal, and analyzing the voice packet received from the PSTN general terminal to check whether a line echo is generated. How to prevent line echo generation in a VoIP system that includes 3. The method of claim 2, The step of applying to the voice communication between the IP terminal and the PSTN general terminal, And generating a VoIP voice signal based on a voice signal transmission gain value less than the voice signal transmission gain value at the time when the line echo occurs. In the line echo generation apparatus of the VoIP system which works with a PSTN, A voice generator for converting a voice into a packet signal and transmitting the voice to the PSTN general terminal; A packet analyzer extracting a voice packet by analyzing a packet received from the PSTN general terminal; A voice analyzer configured to analyze the extracted voice packet and check whether line echo is generated; And And a control unit for storing a voice signal transmission gain value at the time when the line echo is generated, when the occurrence of the line echo is detected from the voice analyzer. 5. The method of claim 4, The control unit, When the generation of the line echo is not detected from the voice analyzer, the voice generator increases the gain value of the voice signal to generate the VoIP voice signal and transmit the generated voice signal to a PSTN general terminal. Line echo generation device. 5. The method of claim 4, The control unit, And detecting the occurrence of the line echo from the voice analyzer, controlling the voice generator to stop generating the VoIP voice signal. 5. The method of claim 4, The control unit, And providing a voice signal transmission gain value stored at the time point of the line echo to the VoIP system so as to be applied to the voice communication between the IP terminal and the PSTN general terminal. 5. The method of claim 4, The apparatus of claim 1, further comprising a call connection unit for generating a VoIP call and connecting the PSTN with a general terminal.

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