KR20070023378A - System for providingg audio data in ip telephony and method thereof - Google Patents

Abstract

The present invention relates to a system and method for providing an audio sound source in an internet phone, and to a technical standard for transmitting an audio sound source through smooth call interaction using the existing Voice over Internet Protocol (VoIP) protocol. define. According to the present invention, an audio sound source providing system includes a soft switch server for performing call processing required for an audio sound source, a profile server for storing information and subscriber information about the audio sound source, a media server for extracting the audio sound source, and streaming in real time. It includes a media control server that controls a series of procedures required to provide an audio source between a softswitch server and a media server. Based on this, the ring back tone is provided through a call setup agreement step in which a message for transmitting and receiving early session information on the ring back tone is added in the existing Internet phone protocol. For the one-way call background sound and the non-call type audio sound source, the one-way call background sound and the non-call type audio sound source are provided through a media channel set with the media control server as the receiving side. Regarding the two-way call background sound, the called terminal and the calling party terminal receive the two-way call background sound through the media channel set up from the media control server.

Internet phone, audio source service

Description

SYSTEM FOR PROVIDING AUDIO SOURCES IN AN INTERNET TELEPHONE AND THE METHOD THEREOF {SYSTEM FOR PROVIDING G AUDIO DATA IN IP TELEPHONY AND METHOD THEREOF}

1 is an overall configuration diagram of a system for providing an audio sound source in an Internet telephone according to an embodiment of the present invention.

Figure 2 is a signal flow diagram for providing a ring back tone in the audio sound source providing method according to an embodiment of the present invention.

Figure 3 is a signal flow diagram for providing a one-way audio sound source in the audio sound source providing method according to an embodiment of the present invention.

Figure 4 is a signal flow diagram for providing a two-way audio sound source in the audio source providing method according to an embodiment of the present invention.

The present invention relates to a system and a method for providing a call-type audio sound source including a call connection sound and a background sound in addition to a voice call in an Internet telephone network, and more particularly in a broadband communication network (BcN) called a next-generation network. The present invention relates to a system and a method for providing a non-callable audio source including an IP radio and a jukebox as well as the callable audio source.

Broadband integrated network is a network based on a wide bandwidth communication line. Because of its large transmission capacity, a wide range of transmission lines are available, including telephone, data communication, facsimile transmission, CATV (wired TV), and two-way CATV that can transmit and receive. Simultaneously provide information and communication services.

Telecommunications operators are developing new business plans with a focus on reducing network investment management costs and generating new revenue sources through IP-based multimedia services. In terms of communication network structure, voice and data service networks have been converted to packet-based integrated networks, and existing voice-oriented services have been improved to support multimedia.

In addition, the Ministry of Information and Communication has increased the speed of 1.5 ~ 2Mbps high speed internet service to 50 ~ 100Mbps, suitable for various telecommunications and broadcasting services in future homes. The goal is to provide.

On the other hand, related services focused on various telephone supplementary services, which are non-speech sectors, have been developed in various ways in the wireline telephone industry, which was considered cumbersome and inconvenient due to the development of mobile communication and diversification of mobile phone functions. This is a reflection of the fact that value-added services are becoming more and more important given the slump in utilization of fixed-line phones due to the market saturation and the increase in mobile phone subscribers. In particular, in the cable industry, the competitiveness of value-added services rather than the classic voice service has been a factor in securing subscribers after full implementation of local phone number portability. As cable operators aiming at maximizing profits are strengthening the intelligent network segment, related service launches are becoming a hurdle, while fixed-line phones, which only deliver voice, are releasing various additional services using intelligent network services such as toll-free telephone and lifetime number.

However, in the conventional communication environment through the circuit telephone network, it is necessary to consider the limited bandwidth of the circuit telephone network and the transmission quality according to the line characteristics of the analog method, so that only the narrowband information that can be transmitted to the circuit telephone network is sampled. Delivered. Therefore, the entire signal could not be transmitted. Of course, even if only the narrowband information is sampled and delivered in the case of a voice signal, the sensitivity of the listener to the loss is not high. However, in the case of an audio signal based on an additional service, if only the narrowband information is sampled and delivered, the listener is not aware of the loss. The degree of sound quality you feel is considerable.

This is pointed out as a very big problem that causes the dissatisfaction of the subscribers due to the deterioration of the audio service quality due to the sound quality in the situation that various applications or application services using the telephone network are bursting.

Therefore, solving the problem of degradation of sound quality of audio source is very big problem in communication service when reflecting the situation of the present industry that provides various audio services in addition to voice calls.

Accordingly, the present invention has been made to solve the above problems, the basic voice call using an Internet phone based on Voice over Internet Protocol (VoIP) in a broadband integrated network that is positioned as the next generation network In addition, in providing various audio sources, the technical standard for smooth interlocking and sound data transmission is defined by using the existing Voice over Internet Protocol (VoIP) protocol, which is sufficient for the conventional narrowband telephone network. To solve the problem of the sound quality that could not be solved to provide a high quality audio source of the original sound level.

In order to achieve the technical problem as described above, according to the characteristics of the present invention, the audio source providing system is a variety of audio including a non-callable audio source including a call-type audio source and a jukebox, IP radio, including a call connection sound, a call background sound A system for providing a sound source to an Internet telephone, the system comprising: a profile server configured to store and manage a profile of an audio sound source providing service subscriber and setting information of various audio sources; a signaling protocol of an internet telephone on an audio channel separate from the Internet voice call Soft switch server for processing the signaling for providing the audio sound source through the Internet telephone call, including the transmission and reception of call-related messages required for audio sound source using; A media server extracting a corresponding sound source of the subscriber to the audio application service in real time by interworking with the soft switch server; And receiving the audio application service request information from the soft switch, including the called number, and transmitting the received audio application service request information to the media server, transmitting the media channel information provided from the media server to the soft switch server, and transmitting the audio application service from the profile server. It receives the requester's setting information and audio source information and controls the streaming of the audio source through the media server so that the soft switch server and the media server are interconnected to transmit the audio source through the internet phone call using the internet phone protocol. It includes a media control server that controls a series of processing performed.

According to still another aspect of the present invention, there is provided an audio sound source providing method for a call background sound in the audio sound source providing method for providing a call connection sound when a call is connected through an Internet phone, the method comprising: a) requesting a call using the signaling protocol of the Internet phone; After determining whether the caller is a subscriber to the ringer, the softswitch server includes a message for requesting an initial session for providing the ringtone, including media channel information (IP and Port information of the media server to which the ringtone is to be sent). Transmitting to the calling terminal; b) receiving, by the softswitch server, a response message to the message requesting an initial session from the calling terminal which received the media channel information in step a), including IP and Port information of the calling terminal; And c) the soft switch server receiving the early response response message in step b), transmitting a message indicating the establishment of the early session to the caller to the calling terminal.

According to still another aspect of the present invention, there is provided a method for providing a one-way audio sound source in a method of providing an audio sound source providing a one-way call background sound, a juke box, and an IP radio through an Internet phone, including a) one-way call background sound, juke box, and IP radio. Receiving, by the softswitch server, a message regarding a request call of an audio sound source for any one from a calling terminal; b) extracting the sender's information (including the telephone number and IP address) from the message received in step a) and transmitting the information to the media control server; And c) sending, by the soft control server to the calling terminal, a message indicating that the media session is established by using the media control server as the called party.

According to still another aspect of the present invention, there is provided a two-way audio sound source providing method in which an audio sound source provides a two-way call background sound through an Internet telephone, the method comprising: a) receiving a message regarding a two-way call background sound request call from a calling terminal; Doing; b) extracting sender and called party information (including telephone number and IP address) from the message received in step a) and transmitting the same to the media control server by the softswitch server; c) the media control server sending an invitation message to the called party's telephone number received in step b); And d) under the control of the media control server, a caller terminal and a receiver terminal exchange messages connected on the media channel set by the soft switch server to establish a two-way call background sound channel.

DETAILED DESCRIPTION Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings so that those skilled in the art may easily implement the present invention. As those skilled in the art would realize, the described embodiments may be modified in various different ways, all without departing from the spirit or scope of the present invention. In the drawings, parts irrelevant to the description are omitted in order to clearly describe the present invention, and like reference numerals designate like parts throughout the specification.

In addition, when a part is said to "include" a certain component, it means that it may further include other components, without excluding other components unless otherwise stated.

In addition, the term module described herein refers to a unit for processing a specific function or operation, which may be implemented in hardware or software, or a combination of hardware and software.

An embodiment of a system and method for providing an audio sound source in the present invention Internet phone will be described in detail with reference to the drawings.

1 is an overall configuration diagram of a system for providing an audio sound source in an Internet telephone according to an embodiment of the present invention.

As shown in the figure, the overall configuration of the system for providing the audio sound source in the Internet phone is composed of the subscriber terminal 1, the audio sound source providing system (2).

Here, the audio source providing system 2 provides a call-type audio source that is provided during or during a call connection, such as a call connection sound and a call background sound, to the subscriber terminal 1 through a Broadband Communication Network (BcN). In addition, it is a system that provides non-callable audio sound sources such as jukebox and IP radio regardless of call status. At this time, between the subscriber station 1 and the broadband integrated network, xDSL (x Digital Subscriber Line, a general term for a digital subscriber line that enables high-speed data communication using a telephone line), FTTH (Fiber To The Home, optical subscriber line using an optical fiber) ) Can be connected.

The audio sound source providing system 2 includes a soft switch server 20, a profile server 21, a media server 22, and a media control server 23. In addition, the media server 22 may include a memory cache 24. In addition, the audio sound source providing system 2 preferably includes a billing processing server 25, an automatic response server 26, a web server 27. The web server 27 may include an XML generator 28.

Hereinafter, the above-described components will be described in more detail.

The soft switch server 20 performs a call processing function such as generation and termination of a call required for an audio sound source providing service including an internet voice call. That is, it replaces the existing exchange and controls various call processing and additional functions at the top layer of the network structure. In addition, it provides gateway functions for various signaling and provides additional services for media gateways and IP endpoints. It is responsible for providing end-to-end real-time connectivity and switching call control to other network elements. In the present invention, in particular, it defines an early-session for the ring back tone on the signaling protocol of the Internet phone, not a separate protocol for the audio source, and the one-way call background sound, the call message for the jukebox and the IP radio, and the call message for the two-way call background sound. And so on. In this case, the signaling protocol configures a call setup flow by adding a message for transmitting and receiving Early Session information on a ring back tone between the soft switch server 20 and the subscriber terminal in a call setup agreement phase of a telephone call. Provides the transmission of the ring back tone through, but does not affect the existing voice call, additional services. In addition, if the received number received using the Internet telephony protocol includes a service identification number, a delimiter, and a sound source identification number, the call using the Internet telephony protocol with the media control server (described below) as the called party. The media channel is started in the setup agreement phase.

The profile server 21 stores and manages a profile in which subscriber information, including a subscriber's (or caller's) phone number, and an audio source for a call-type audio service are designated. Then, the caller receives the caller's telephone number from the soft switch server 20, searches for a profile corresponding thereto, and provides setting information of the audio sound source to the soft switch server 20.

The media server 22 is an abbreviation of DBMS (DataBase Management System) for an audio sound source, and functions as a software for efficiently storing, updating, deleting, reading (searching) the sound source data, and the like. That is, the audio source specified in the profile server 21 is extracted, a media channel for audio source transmission is prepared, and the extracted sound source is streamed in real time to the subscriber terminal 1 through the media channel. In addition, jukeboxes or IP radios play live recorded non-calling audio sources. In this case, real time media transmission protocol (RTP, Real-Time) is performed using MP3 as payload described in RFC-3119 (A More Loss-Tolerant RTP Payload Format for MP3 Audio) as well as non-call audio sound source. Transfer Protocol). In addition, when simultaneously streaming the same sound to multiple terminals at the same time, including two-way background sound, jukebox, and IP radio, the same sound source is transmitted by sending a stream of a single audio source to all connected media channels in the form of time series data. Can listen. In addition, in the case where it is desired to charge the audio sound source, the physical file size (MP3) of the sound source transmitted in a predetermined time period (for example, 10 seconds) at the time of transmission of the audio sound source that is charged with the billing processing server. File based) can be transferred to the billing processing server, or a certain file size (100Kbyte or 1Mb) can be sent to the billing processing server for transmission.

The memory cache unit 24 is for storing the audio sound source to reduce the audio sound source extraction time of the media server 22 for the audio sound source frequently used by the audio application service subscribers to provide a faster time. That is, in order to improve the response processing speed of the audio source in consideration of the fact that the audio source is stored in the HDD (Hard Disk Drive) but the frequently requested sound source is stored in the memory of the system. .

The media control server 23 performs a control function of the audio sound source providing service. That is, when the soft switch server 20 detects a call processing event necessary for providing an audio sound source by interworking with the soft switch server 20 and the media server 22, the media control server 23 informs the media control server 23 of the media. The control server 23 is a process required for providing audio sound source (to be described later, in the case of providing a unidirectional call background sound or a non-calling audio sound source, the IP address and port information of the call background sound requester terminal are transmitted to the media server. In this case, the caller requests a background voice call to the called party's terminal. In addition, the subscriber terminal 1 is connected to the automatic answering server 26 and the web server 27 via an audio telephone source connected to the automatic answering server 26 connected via an Internet telephone call, or connected through a broadband integrated network. A series of processes of transmitting the audio source selected by the web server 27 through the media server 22 is controlled. In addition, if the user wants to charge the audio sound source, the phone number of the subscriber who requested the audio sound source, the corresponding sound source information, and the transmission start time are transmitted to the billing processing server 25. It transfers to the process server 25. In addition, the non-call type audio sound source service request is set as a UA (User Agent) on the session establishment protocol and becomes a called party for the request. In addition, with respect to the bidirectional call background sound, a call connection message requesting the initiation of a media channel may be transmitted to the called party, and the same call background sound may be transmitted to the media channels at both ends of the called / calling party while all call connection setting steps are completed. At this time, since the sequence of media sent to each media channel of both ends is the same, there is no time difference in the sound source heard at both ends.

The billing processing server 25 is linked to the media server 22 and the media control server 23, the audio sound source usage information (subscriber's phone number, used sound source information, transmission start and end time) for each subscriber on a daily and monthly basis , Physical size of audio source, transmission volume, etc.) are collected and calculated through batch process, and monthly charging is possible.

The automatic answering server 26 is an interface for a subscriber to set up a call-type audio source (call connection sound, call background sound) or request a non-call-type audio source (jukebox, IP radio). That is, the service for the audio source can be set up by receiving the subscriber's telephone key input, and the subscriber's setting information or information about the requested audio source is transmitted to a media control server, and under the control of the audio source, the audio source is the media server 22. Stream). At this time, it is also provided with a two-stage dialing function, the call connection is set in the first stage (1-Stage), such as a connection ring tone information of the user and disconnects. And, when the subscriber wants to listen to the sound source immediately, such as a non-calling audio sound source, the connection with the automatic answering server connected to the first stage (1-Stage) is disconnected and the media control server 23 is connected to the Internet telephone. It is possible to listen to the sound source through call interlocking on 2-stage.

The web server 27 is for performing an interface for setting or driving an audio sound source through a menu screen and a keypad displayed on the display unit of the subscriber terminal 1. At this time, the hypertext transfer protocol (HTTP) may be used for communication with the subscriber terminal 1, and the setting information of the audio source may be changed and stored in association with the profile server 21. Also, the media server 22 can stream the audio sound source information set in association with the media control server 23 to the subscriber station 1 through the media channel. In addition, the menu screen displayed on the display unit of the subscriber terminal 1, including the XML (eXtensible Markup Language) generation unit 28, passes through a file defined by XML (eXtensible Markup Language) and is in XML (eXtensible Markup Language). This can flexibly cope with changes in menu structure and service reorganization.

Hereinafter, the procedure for providing the call connection sound, the unidirectional audio sound source and the bidirectional audio sound source will be described based on the signal flow diagram based on the audio sound source providing system 2 described above. Prior to the above description, a session establishment protocol (SIP) and a session description protocol (SDP) are used as protocols ranging from a ring back tone to a call background tone and a non-callable audio source. The form is referred to as MP3 (MP3).

2 is a signal flow diagram for providing a call connection sound in the audio sound source providing method according to an embodiment of the present invention.

As shown in the figure, the ring back tone is provided by a process of adding a message for transmitting and receiving the earliest information on the ring back tone on an existing internet telephony signaling protocol. More specifically, the media channel decision on early session for providing ring back tone in MP3 type on Session Description Protocol (SDP) using the technology of The Early Session Disposition Type for the Session Initiation Protocol. Follow the method. That is, in the call setup agreement step of the telephone call, the message for the early session is added between the soft switch server and the subscriber station.

Referring to the drawing to explain the process of providing a ring back tone, first, the soft switch server receives an Invite message that is a call request message from the calling terminal (S10). At this time, the soft switch server passes the called party number to the profile processing unit of the subscriber in the received Invite message. Then, the profile processing unit determines whether the ring back tone is mapped to the called party number, and if the ring back tone is mapped, requests the mapped call ring tone information to the media server. The media server transmits its IP and port information to the soft switch server for the media channel to transmit the ring back tone. Of course, the soft switch server requests the incoming terminal an Invite message, which is a telephone call request of the calling terminal, and receives a response message (S11, S12).

The soft switch server receives the IP and Port information of the media server received as a response, constructs the media channel information, and transfers it to the caller terminal in a 183 Early Offer message (S13).

The caller terminal prepares to receive a ring back tone according to the received media channel information in the 183 Early Offer message, and transmits the IP and port information of the caller terminal to receive the ring back tone in response to the PRACK message. S14).

Receiving the PRACK message, the soft switch server transmits a 200OK message indicating the establishment of an early session to the calling terminal (S15). At this time, the IP and Port information of the caller terminal included in the PRACK message is transmitted to the media server, and the call connection sound is transmitted through the media channel set as 183 Early Offer, the caller terminal included in the PRACK message and the media server.

Subsequently, when a 200 OK message regarding a voice session for a voice call is received from the called party terminal at the soft switch server (S16), the soft switch server 20 transmits it to the calling party terminal (S17) and receives a response message thereto ( S18) The Early Session for the ring back tone ends.

3 is a signal flow diagram for providing a unidirectional audio sound source in the audio sound source providing method according to an embodiment of the present invention.

As shown in the figure, the soft switch server receives an invite message containing caller information (IP, Port of the caller terminal) as a telephone call requesting a one-way audio sound source from the caller terminal (S20).

The soft switch server transmits the caller information (IP, Port of the caller terminal) included in the received invite message to the media control server (S21).

The media control server transmits a reception response message for the caller information to the caller terminal through the soft switch server (S22, S23).

In addition, a message indicating the establishment of a media session for providing the one-way audio sound source is transmitted to the caller terminal through a soft switch server, and a response message is received therefrom (S24, S25, S26). The session is established. Through this, one-way audio source, that is, one-way call background sound, juke box, and IP audio source is provided.

However, when the non-callable audio sound source including the jukebox and the IP radio includes a service identification number, a delimiter, and a sound source identification number in an invite message requesting the audio sound source, the softswitch server receiving the message provides a call. Set the incoming call to the media control server.

Figure 4 is a signal flow diagram for providing a two-way audio sound source in the audio sound source providing method according to an embodiment of the present invention.

As shown in the figure, the soft switch server receives the invite message containing the caller information (IP, Port of the caller and caller terminal) as a telephone call requesting a two-way audio sound source from the caller terminal (S30).

The soft switch server transmits the sender and the receiver information (IP, Port of the caller and receiver terminal) included in the received invite message to the media control server (S31) and receives the response message and transmits it to the caller terminal (S33). ). At this time, when the telephone number of the caller of the background tone, that is, the caller and the other party, who open the voice channel, and the caller, and the corresponding IP address are transmitted to the media server, the call is transmitted in the form of an IP address in the User-Agent header.

The media control server transmits an invite message, which is a call message requesting a media session for a call background sound (S32), and receives a response message (S33).

Thereafter, the media control server receiving the 200 OK message indicating the establishment of the media session from the called terminal transmits the 200 OK message to the calling terminal through a soft switch (S34). From the soft switch to the destination terminal via the media control server (S35). Thus, the media channel for the call background sound is set in the caller terminal and the callee terminal period based on the media control server, so that the same audio source can be provided as the background sound during the call.

The embodiments of the present invention described above are not implemented only through systems and methods, but may be implemented through a program for realizing a function corresponding to the configuration of the embodiments of the present invention or a recording medium on which the program is recorded. Implementation may be easily implemented by those skilled in the art from the description of the above-described embodiments.

Although the embodiments of the present invention have been described in detail above, the scope of the present invention is not limited thereto, and various modifications and improvements of those skilled in the art using the basic concepts of the present invention defined in the following claims are also provided. It belongs to the scope of rights.

By the above-described configuration, a call-type audio source, juke box, and IP of a ring back tone, a call back tone, and the like are made by using the Internet telephony protocol that does not change the implementation of existing technical standards and equipment in a broadband integrated network, which is attracting attention as a next-generation network. A technical foundation can be laid to realize non-callable audio source of radio. Therefore, it enables various additional services using various audio sources except for the video.

Claims (14)

In the system for providing a variety of audio sources to the Internet phone, including a call-type audio source including a ring back tone, a background sound of the call, and a non-callable audio source including a juke box, IP radio, A profile server configured to store and manage the profile of the audio source providing service subscriber and setting information of various audio sources; A soft switch server that processes signaling for providing an audio sound source through an Internet telephone call, including transmission and reception of call-related messages necessary for providing an audio sound source using a signaling protocol of an Internet telephone on an audio channel separate from the Internet voice call; A media server extracting a corresponding sound source of the subscriber to the audio application service in real time by interworking with the soft switch server; And Receives the audio application service request information from the softswitch, including the called number, and transmits it to the media server, and transmits the media channel information provided from the media server to the softswitch server. The setup information and audio source information of the through the media server The soft switch server and the media server are connected to each other by controlling the streaming of the audio source. And an audio sound source providing system including a media control server which controls a series of processes in which an audio source is transmitted through an internet phone call using an internet phone protocol. The audio source providing system of claim 1, wherein the soft switch server defines an early session for a ring back tone on a signaling protocol of an internet telephone and connects a media channel. According to claim 2, The signaling protocol of the Internet telephone (Early Session) is defined in the signaling protocol of the call call Early Session (Early Session) information for the ring back tone in the call setup agreement step of the call between the soft switch server and the subscriber terminal Audio sound source providing system to configure the call setup flow by adding a message that can be provided through this through the ring tone. The audio sound source providing system according to claim 1, wherein the audio sound source including the call connection sound, the call background sound, the jukebox, and the IP radio is MP3. The method of claim 1, wherein the soft switch server If the received number received by using the Internet phone protocol includes the service identification number, the delimiter, and the sound source identification number, the media channel is started in the call setup agreement step using the Internet phone protocol by using the media control server as the called party. Audio sound source providing system. The method of claim 1, wherein the media server When simultaneously streaming the same sound to multiple terminals at the same time, including two-way background sound, jukebox, and IP radio, audio streams are transmitted through real-time media transmission protocols in the form of time series data to all connected media channels. system. The method of claim 1, wherein the media server Audio application service Audio sound source providing system including a memory cache for storing the audio sound source to provide a quick time to reduce the audio sound source extraction time of the media server for the audio source frequently used by subscribers. The method of claim 1, Billing information for non-callable audio sound sources that are streamed in real time through the media server in conjunction with a media control server to provide audio services for non-callable audio application services such as jukeboxes and IP radios as paid services. A billing processing server for generating and storing the requester's telephone number and the fee information of the requested audio sound source; An auto-response server providing an audio application service requester and a call-based interface via an internet telephone, including an audio application service request, an audio source selection, and the like; Using the screen and keypad of the web application of audio application service requester and internet phone, i.e., the structure, setting and inquiry of menu screen for providing audio application service request and audio source selection, etc. Providing web server; And And at least one of an XML-generating unit for a menu structure dynamic with the web server. The real-time media transmission protocol according to any one of claims 1 to 8, using a Session Initiation Protocol (SIP) and a Session Desription Protocol (SDP) as a signaling protocol. Audio source providing system comprising using (RTP, Real-Time media Protocol). In the audio source providing method for providing a ring back tone when making a call through an Internet phone, a) message-media channel information (media server to which the ringback tone is transmitted) requesting an initial call for providing the ringback tone after determining whether the caller is a subscriber to the caller using the signaling protocol of the Internet phone. Transmitting, by the Soap Switch Server, to the calling terminal. b) receiving, by the softswitch server, a response message to the message requesting an initial session from the calling terminal which received the media channel information in step a), including IP and Port information of the calling terminal; And c) the soft switch server receiving the initial response response message in step b), transmitting a message indicating the establishment of an early session for the ring back tone to the calling terminal. In the method for providing a non-callable audio source including a one-way call background sound, a juke box, IP radio through an Internet phone, a) receiving, by the softswitch server, a message regarding a request call of an audio source for a selected one of a one-way call background sound, a juke box, and an IP radio; b) extracting the sender's information (including the telephone number and IP address) from the message received in step a) and transmitting the information to the media control server; And and c) transmitting, by the soft control server, a message indicating that the media session is established to the called party by the media control server as the called party. The method of claim 11, wherein in step a) The request message of the audio sound source when the request of the non-callable audio sound source including the juke box, IP radio, characterized in that the message containing a service identification number, delimiter, sound source identification number, And a soft switch server receiving the message including the service identification number, the delimiter and the sound source identification number, setting the incoming call to the media control server. In the audio source providing method for providing a two-way call background sound through an Internet phone, a) receiving, by the softswitch server, a message regarding a bidirectional call background tone request call from a calling party terminal; b) extracting sender and called party information (including telephone number and IP address) from the message received in step a) and transmitting the same to the media control server by the softswitch server; c) the media control server sending an invitation message to the called party's telephone number received in step b); And d) under the control of the media control server, the caller terminal and the called terminal exchange a message that is connected on the media channel set by the soft switch server, the method for providing an audio sound source comprising the step of establishing a two-way call background sound channel. The method of claim 13, wherein in step b) An audio source characterized in that the caller requesting the two-way call background tone, the telephone number of the caller opening the voice channel, and the corresponding IP address are transmitted to the media control server by specifying an IP address in the User-Agent header. How to Provide.

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