KR100785792B1 - Method and system for providing service on SIP-based Internet telephony system - Google Patents

Abstract

The present invention discloses a service providing method and system therefor in an Internet telephone system using a connection establishment protocol.

According to the present invention, in the Session Initiation Protocol service, each connection establishment protocol service terminal registers its own address to call through the service and a media type, which is a separate call type to be used for the call, to the registration server. When a call service is requested to an address corresponding to a terminal providing a SIP service, a query is made to a registration server to determine whether there is a media type registered in advance by the terminal corresponding to the address. To the transcoding server that converts the contents of the call into voice and text, and the transcoding server maintains separate sessions with the terminal requesting the call service and the terminal that is the target of the call service. By converting the contents to provide a call service between the two terminals, SI If the user who connects via softphone in the P-phone system does not have a headset, or voice call is difficult depending on the user's circumstances, or if the user cannot make a phone call through voice transmission and reception, such as a hearing impairment, a character is sent and received instead of voice. Allows normal phone calls.

Description

Method and system for providing service in Internet telephony system using connection establishment protocol {Method and system for providing service on SIP-based Internet telephony system}

1 illustrates a flow of an example of a service providing method in an Internet telephone system using a connection establishment protocol according to the present invention.

2 is a flow diagram of another example of a service providing method in an Internet telephone system using a connection establishment protocol according to the present invention.

3 illustrates a media mode registration process selected by a user according to the present invention.

4 illustrates an example of a process of connecting a voice call to a counterpart who has selected a text mode according to the present invention.

5 illustrates an example of a process of connecting a text call to a counterpart who has selected a voice mode according to the present invention.

FIG. 6 is a diagram illustrating a session resetting procedure and a data transmission method for changing a usage mode for a call after call setup for an internet telephony is already completed according to FIG. 4.

FIG. 7 is a diagram illustrating a session resetting procedure and a data transmission method for changing a usage mode for a call after the call setup for the Internet telephony is already completed according to FIG. 5.

8 illustrates a call termination process for terminating a telephone call after providing a service according to the present invention.

9 illustrates an example of a configuration of a service providing system in an Internet telephone system using a connection establishment protocol according to the present invention.

Fig. 10 shows another example of the configuration of the service providing system in the Internet telephone system using the connection setting protocol according to the present invention.

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to communication over the Internet, and relates to a service providing method and system therefor in an Internet telephone system using a connection establishment protocol.

Due to the rapid increase in the number of Internet users, various popular services using the Internet are being created as the distribution of Internet services becomes popular. In line with this trend, standard technologies related to Internet telephony services have become a major issue in recent years.

The Internet telephony service started from the economical point of view that the call cost is lower than the PSTN phone, but recently, it is easily integrated with various internet multimedia services such as video telephony, multi-party conference and instant messaging service. In this regard, it is in the spotlight from the viewpoint of creating a new additional service.

In addition, the Internet telephony service is important in that various network technologies become the core technology for the evolution to the ALL IP network.

Internet telephony services based on the Session Initiation Protocol (SIP) ensure user mobility. A user using a SIP-based service can be provided with an Internet telephone service through various terminals capable of accessing the Internet anytime, anywhere.

If a user connected through a soft phone does not have a headset, or a voice call is difficult depending on the user's circumstances, or if the user has a hearing impairment, a phone call through voice transmission and reception is impossible.

Various application services using SIP can transmit and receive voice, video and text. In this case, a session for transmitting and receiving the same media is established between both ends of the telephone call using SIP. That is, only the voice call, the video call only, or the text call through text transmission and reception are possible between both parties of the call. As in the above example, one side cannot provide a call environment in which various media are mixed and serviced, such as one side talking by voice and the other side by text.

In the published IETF RFC 4117 document, the media deal with mixed phone calls between media, but in this case, the call request message is sent to the calling callee and the call is sent from the callee to the transcoding server. do. This has a problem of generating a load on the terminal used for the call.

In particular, in the case of a terminal used in a wireless environment, when making a call between mixed media as described above, there is a problem in that unnecessary data is transmitted and received from the standpoint of making a call through the wireless section. In other words, in a wireless communication environment in which a communication band is limited, a call request message is transmitted to a terminal of a calling party for a telephone call mixed between media, and the call service is converted through conversion between media from the terminal. Through the process of passing the call to the transcoding server provided, there is a problem that the bandwidth of the wireless communication is narrowed due to unnecessary data not used for a telephone call.

The technical problem to be solved by the present invention is to solve the above problems, in the Internet telephone system using a connection establishment protocol that does not burden the communication band in the terminal according to the connection establishment protocol even if mixed media are converted to each other The present invention provides a method and system for providing the service.

An example of a service providing method in an Internet telephone system using a connection establishment protocol according to the present invention for solving the above technical problem is a connection establishment protocol (Session Initiation Protocol) service, (a) each connection establishment protocol Registering, by the service terminals, their address to be called through the service and a separate call type to be used for the call; (b) if a call service is requested to an address corresponding to the terminal providing the service, determining whether there is a separate call type registered in advance by the terminal corresponding to the address; And (c) when there is a separate call type, converting a call of one counterpart according to the separate call type and providing the call to another counterpart to provide a call service.

In this case, the separate call type is to select a voice call, a text call, and a call using both voice and text, and the call conversion of step (c) is a call-to-text call conversion or a voice to voice and text or A connection setting protocol is used, characterized in that the text is converted into voice and text simultaneously.

And the side requesting a call service transmits information on the call type together with the call type used when the call is requested.

Another example of a service providing method in an Internet telephone system using a connection establishment protocol according to the present invention for solving the above technical problem is a connection establishment protocol (Session Initiation Protocol) service, (a) each connection establishment protocol Registering, by a service terminal, a media type, which is a separate call type to be used for the call and its address to be called through the service, with a registration server; (b) if the call service is requested to an address corresponding to the terminal providing the service, querying the registration server whether there is a media type registered in advance by the terminal corresponding to the address; (c) if there is a separate media type registered in advance, transmitting the call service request to a transcoding server that converts the contents of a call into voice and text; And (d) providing a call service between the two terminals by converting the call content according to the separate media type while maintaining a separate session with the terminal requesting the call service and the terminal targeted for the call service. Characterized in that it comprises a.

In the step (a), registering the media type is characterized by using a register message for registering a user's access address.

Registering the media type is characterized in that it is through a separate parameter in the Contact header included in the REGISTER message.

The separate media type is to select a voice call, a text call, and a call using both voice and text. The call conversion of step (d) may be a currency conversion between voice and text, or voice to voice and text or text. It is characterized in that the simultaneous conversion to voice and text.

The requester of the call service delivers the media type used by including information on the media type to be used in the Session Description Protocol (SDP) included in the INVITE message used when the call is requested.

When the transcoding server receives the call request message, the transcoding server queries the registration server to obtain a caller's address. The transcoding server generates an INVITE message that designates a media type previously designated by the caller.

In addition, the provision of the voice call service uses the Real-time Transport Protocol (RTP), and the connection establishment protocol is characterized in that the provision of a text message is through the MSRP or MESSAGE message.

When the terminal requesting the call service changes the media type, a message for requesting the call service is transmitted to the transcoding server, and the transcoding server receiving the call service transmits the call transmitted through the session with the terminal requesting the call service. The media is converted and provided according to the type of media, and the message requesting to change the media type is characterized by using a re-INVITE message.

An example of a service providing system in an Internet telephony system using a connection establishment protocol according to the present invention for solving the other technical problem is a system for providing a connection establishment protocol (Session Initiation Protocol) service, the service Message receiving unit for receiving a request message to call through; Inquiries to the registration server for registering the address of the terminal to call through the service and the media type, which is a separate call type to be used for the call, receives the access address as a response, and receives the address. A call information receiving unit for receiving the media type information corresponding to the access address in response; And if the media type information is not received in the response through the call information receiver, providing a call service by transmitting a message to a terminal corresponding to the access address received in the response, and converting between media when the media type information is received in the response. And a call setup unit for delivering the call request message to a predetermined transcoding server providing a call service.

Another example of a service providing system in an Internet telephony system using a connection establishment protocol according to the present invention for solving the other technical problem is a system for providing a connection establishment protocol (Session Initiation Protocol) service, the service Message receiving unit for receiving a request message to call through; Inquiries to the registration server that registers the address of the terminals making the call to the call destination through the service and the media type, which is a separate call type to be used for the call, and receives the access address and the media type in response. Information receiver; And establishing a session with the terminal corresponding to the address included in the call request message, establishing a separate session with the terminal corresponding to the call counterpart address received through the call information receiving unit, and including the media included in the call request message. And a currency converter for converting the contents of the call according to the type and media type of the call counterpart to provide a call service.

The call request message received by the message receiver transmits the media type used by the call requesting terminal by including information on the media type to be used by the SDP (Session Description Protocol) included in the INVITE message used to request the call. .

The media type converted by the currency converter is one of a voice call, a text call, and a call using both voice and text, and the currency converter converts voice to text or voice to text according to the media type. Or converting text into voice and text at the same time.

The call conversion unit provides a voice call service using the Real-time Transport Protocol (RTP), and providing a text message is through an MSRP or MESSAGE message.

When the terminal requesting the call service changes the media type during the call, the message receiving unit receives and transmits the message requesting the message to the call conversion unit, and the received call conversion unit requests the call service. A call transmitted through a session with a terminal is converted and provided according to the changed media type. In this case, a message requesting to change the media type uses a re-INVITE message.

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

1 illustrates a flow of an example of a service providing method in an Internet telephone system using a connection establishment protocol according to the present invention.

In this method, which is a service according to SIP, each connection establishment protocol service terminal registers its own address to call through the service and a separate call type to be used for the call (step 100), and corresponds to the terminal providing the service. When the call service is requested to the address to determine whether there is a separate call type registered in advance by the terminal corresponding to the address (step 110), and if there is a separate call type, a call of one party according to the separate call type is made. The conversion is provided to the other party to provide a call service (step 120).

Terminal users using the service according to SIP select and register the type of call to be used through their terminal (step 100). In this process, an address for accessing the UE is also registered.

The registration in step 100 is mainly for a separate call type, and a voice call, a text call, and a call using both voice and text may be selected and registered.

The terminal requesting a call service requests a call through a message, and the request is transmitted to a proxy server located in a counterpart area of the call. The proxy server fetches the contents registered in step 100 for the counterpart terminal of the call and checks whether there is a separate call type registered in advance for the terminal.

If there is no separate call registered in advance, the call is directly connected to the terminal, and if there is a separate call registered in advance, it is transmitted to the transcoding server. The transcoding server converts the contents delivered from the other party according to a separate call registered in advance and delivers them to both parties of the call. Eventually, in this process, a call is converted between a voice and a text to each other or a voice is converted into a voice and a text or a text is converted into a voice and a text at the same time.

At this time, when the terminal requesting the first call requests the call service for the first time, the terminal transmits the call type used by the terminal applying for the call by transmitting information on the call type, and in this case, the call types desired or supported by both terminals. Conversion services are provided according to the requirements.

Hereinafter, the present invention will be described with more detailed examples.

2 is a flow diagram of another example of a service providing method in an Internet telephone system using a connection establishment protocol according to the present invention.

In this method according to the connection establishment protocol SIP service, each SIP service terminal registers its own address to call through the service and a media type, which is a separate call type to be used for the call, at the registration server (step 200). When a call service is requested to an address corresponding to a terminal providing a call, the registration server inquires whether there is a media type registered in advance (step 210), and if there is a separate media type registered in advance, the call. The service request is transmitted to a transcoding server that converts the contents of a call between voice and text (step 220), and the transcoding server maintains separate sessions with the terminal requesting the call service and the terminal targeted for the call service. Provides a call service between two terminals by converting call contents according to the separate media type. .

Referring to the overall operation of the present invention, when the SIP proxy server of the callee, the caller, receives the SIP request message, the callee obtains the access address of the callee from the registrar server. Send a message to the access address. However, if the usermode parameter exists in the callee's access address, the message is sent to a separate transcoding server instead of the message.

The INVITE message sent to the transcoding server establishes one session between the caller and the transcoding server, and establishes another session between the transcoding server and the callee. If the caller's media information specified in the SDP information of the received INVITE message and the usermode parameter information of the callee registered in the registrar server are not the same, the media conversion is performed through the transcoding server.

Since the message is transferred directly from the proxy server to the transcoding server instead of the callee, the message processing load of the callee can be reduced, and in the case of a wireless environment, data transmission and reception of the wireless section can be reduced.

A session establishment process and a message transmission / reception procedure and the like according to the present invention will be described below with reference to FIG. 3 and the accompanying drawings. 3 illustrates a media mode registration process selected by a user according to the present invention.

The SIP client A 300 registers the media mode selected by the user in the registrar server which is a registration server.

In this case, a separate extension parameter is defined in the Contact header in the REGTISTER message used to register the user's access address, and used to register the media mode selected by the user.

The following is an example.

Contact = ("Contact" / "m") HCOLON

                  (STAR / (contact-param * (COMMA contact-param)))

contact-params = c-p-q / c-p-expires

                      / contact-extension

contact-extension = generic-param

generic-param = token [EQUAL gen-value]

The extended parameter name to be used according to the contact-params ABNF requirement of the SIP message Contact header is defined as "usermode".

The "usermode" contact-params is used to define the service mode selected by the user, that is, the media mode. If the user wants only a voice call, the usermode parameter has an onlyaudio value. If a user wants only a text call, the usermode parameter has an onlytext value. The user parameter also has an audiotext value if both voice and text calls are supported.

3 shows that the SIP client A 300 registers 129.254.100.101, which is an address for accessing the registrar server or the registration server 310, with a voice call only for a media mode to be serviced through a register message. It is. The registration server 310 responds to the registration via the 200 (OK) message to the SIP client A (300).

The function described above in the present invention may be a new supplementary service for a SIP Internet telephone system, in which case it applies only to a user who selects such supplementary service according to the present invention. At this time, according to the presence of the usermode parameter in the REGISTER message from the user terminal, it is determined whether the additional service according to the present invention.

When the SIP user selects the service mode to be used by the SIP user, the service mode selected by the user is registered in the registrar server 300 through the REGISTER message. Therefore, not only when the access address of the user is changed, but also when the service mode is changed, a REGISTER message is transmitted according to the present invention to set the service for the media mode.

As shown in the diagram of FIG. 3, for the purpose of describing the present invention, SIP Client A has selected and registered only audio mode, i.e., audio media, which is a voice service. Assume that it is selected and registered.

In the following description, the onlyaudio mode and the audio media mode, which is a voice service, are used interchangeably. We also use the terms onlytext mode and textual media as a text service.

4 illustrates an example of a process of connecting a phone call according to the present invention. This diagram illustrates a session establishment process when the SIP client A 400 attempts a voice call to the SIP client B 430 that has registered the onlytext service mode, that is, the text media mode.

The media type that the SIP client A 400 will use when making a phone call is described through the Session Description Protocol (SDP) in the INVITE message.

The INVITE message transmitted by the SIP client A 400 is transmitted to the proxy server 410 in the domain to which the SIP client B 430 belongs as a result through a plurality of proxy servers (S401).

The proxy server 420 in the domain of the SIP client B 430 obtains a connection address for accessing the SIP client B 430 from the registrar server. At this time, if there is a usermode parameter value with the access address, the value is also requested.

In this case, when the proxy server 410 requests the registrar server to request the access address of the SIP client B 430, the registrar server, which is the registration server, is associated with the address, and there is a request for a separate usermode parameter if there is a usermode parameter. It can be implemented to send information about the usermode parameter with the address even if it is not present.

The proxy server 420 sends an INVITE message sent from the SIP client A 400 to the corresponding access address of the SIP client B 430 if the usermode parameter does not exist in the registrar server, that is, if the usermode parameter is not received. If there is a parameter in the usermode, it transmits the message transmitted from the SIP client A 400 to the transcoding server 420 (S403), and the transcoding server 420 responds with a 180 (Ringing) message. (S404).

The transcoding server 420 receiving the INVITE message operates as a user agent server (UAS) for the received INVITE message. The transcoding server 420 also obtains the access address of the SIP client B 430 from the registrar server. At this time, the usermode parameter is also received as described above.

The transcoding server 420 generates and transmits an INVITE message to be sent to the SIP client B 430 based on the usermode parameter transmitted together with the access address (S405) and receives a response thereto (S406). The transcoding server 420 acts as a UAC (User Agent SIP Client) in this case.

Transcoding server 420 establishes one media session with SIP Client A 400, which is the caller, and sets up a separate media session with Callee SIP Client B 430, which is the target of the call request.

The voice is transmitted and received between the SIP client A 400 and the transcoding server 420, which have established a voice media session, via a Real Time Transport Protocol (RTP), and the MSRP or MESSAGE between the transcoding server and the SIP client B, which has established a text media session. Text messages will be sent and received via messages.

Both Message Session Relay Protocol (MSRP) and MESSAGE are methods for sending and receiving text. Unlike MESSAGE, MSRP establishes a specific session between both ends and transmits and receives text. MESSAGE does not form a session, and there is a difference in processing only the characters in MESSAGE received at that time.

 In order to drive real-time media broadcasting or videoconferencing information on the Internet or intranet, there is an increasing demand for transmitting media data in real time. In order to meet the needs of the network band band, RTP, which is a real-time transmission protocol, is used.

The call text message delivered by the SIP client B 430 to the SIP client A 400 is transmitted to the transcoding server 420 (S407), and the SIP client B 430 receives a response thereto (S408).

The transcoding server 420 converts the text message transmitted from the SIP client B 430 into audio, that is, voice, to the internal proxy server 410 because the SIP client A 400, which is the party to the call, uses the audio media. Transfer (S408, S409). The voice audio data thus transferred is transferred from the internal proxy server 410 to the SIP client A 400 (S410 and S411), and the call is transferred from the SIP client B 430 to the SIP client A 400.

The voice call content from the SIP client A 400 is converted into text by the transcoding server 420 via the internal proxy server 410 and forwarded to the SIP client B 430.

In the conventional case, since the contents to be executed in the terminal receiving the call are executed in the proxy server 410 and the transcoding server 420, it can be seen that the load on the final terminal is reduced.

FIG. 5 illustrates a session establishment process when the SIP client B 400 attempts a text call to the SIP client A 430 that has registered the audio media service in the onlyaudio service mode.

In the case of FIG. 5, the call attempt is substantially the same as that of FIG. 4 except that the call originator is the SIP client B 400 and the internal proxy server 410 is a proxy server in the domain including the SIP client A 430. Therefore, a separate description is omitted.

FIG. 6 is a flowchart illustrating a process of changing a media type used in the SIP client A 400 from voice to text after a session is established between the SIP client A 400 and the SIP client B 430 in FIG. 4. will be.

In this case, the information for changing the media type is made through the re-INVITE message, and the re-INVITE message is reflected only in the session established between the SIP client A 400 and the transcoding server 420.

The re-INVITE message referred to herein refers to the INVITE mesh of FIG. 6, and generally uses the term re-INVITE message as an INVITE message transmitted again after a session is established through INVITE.

The re-INVITE message from the SIP client A 400 is delivered to the transcoding server 420 via the internal proxy server 410 (S601, S603). In this case, the type of media session to be newly set is described in the SDP portion of the re-INVITE message transmitted, and the character media type is included in this portion and transmitted.

The transcoding server 420 transmits 180 (Ringing) and 200 (OK) response messages to the SIP Client A 400 through the proxy server 410 in response to the received re-INVITE message (S604, S602). In step S605 and S608, the SIP client A transmits an ACK message to the transcoding server 420 via the proxy server 410 as an acknowledgment message for receiving the 200 (OK) response message (S606 and S608).

After transmitting and receiving a series of INVITE / 200 (OK) / ACK messages, a media session is changed between the SIP client A 400 and the transcoding server 420, and text media are transmitted and received instead of voice media.

FIG. 7 illustrates a processing procedure when a media type used by the SIP client B 400 is changed from text to voice after a session is established between the SIP client B 400 and the SIP client A 430 in FIG. 5. It is. Since the specific processing procedure is substantially the same as that of FIG. 6, the description thereof will be omitted.

8 shows a set call termination process, and a BYE message is used for call termination. The SIP client A 400 transmits a BYE message to terminate the call with the client B 430 (S801), and the message is transmitted to the transcoding server 420 through the internal proxy server 410 (S804). . The transcoding server 420 transmits a 200 (OK) response message to the client A 400 in response to the received BYE message (S804), thereby resulting in a session between the transcoding server 420 and the client A 400. To exit. In addition, the transcoding server generates and transmits a BYE message to the client B 430 (S805), and terminates the session between the two after receiving a 200 (OK) response message from the client B 430.

 In FIG. 8, the call termination is attempted by the SIP client A, but the call termination may be attempted by the SIP client B, and the process is the same.

As a result, the SIP user using the service according to the present invention selects which media to use for a telephone call, and if the selected media does not match between Caller and Callee, media conversion is performed through a separate transcoding server. .

SIP messages sent from the caller are forwarded from the proxy server to the transcoding server, not from the callee. The routing of the proxy server is determined by the presence or absence of the usermode parameter in the callee's access address registered with the registrar server.

Therefore, in the present invention, the overall system framework and the processing procedures for the following five scenarios are presented.

1) Call connection between Caller with voice and Callee with text

2) Call connection between Caller with text and Callee with voice

3) After connecting the call through scenario 1), caller changes the preferred media from voice to text.

4) Change the preferred media from text to voice on caller side after call connection through scenario 2)

5) After the call is connected, the call is disconnected from the Caller or Callee.

9 illustrates an example of a configuration of a service providing system in an Internet telephone system using a connection establishment protocol according to the present invention.

The system 900, which provides a Session Initiation Protocol service, corresponds substantially to the internal proxy server 410 of FIGS. 4-8.

The system includes a message receiving unit 910 for receiving a request message to be called through the SIP service, an address of terminals that call the address to be connected to the received call through the service and a separate call type to be used for the call. The call information receiver 920, which queries the registration server 930 for registering the media type, receives the access address in response, and receives the information in response to the separate media type information corresponding to the received access address. It includes. If the media type information is not received in the response through the call information receiver 920, a message is transmitted to the terminal corresponding to the access address received in the response to provide the call service. When the media type information is received in the response, the call setup unit 940 transfers the call request message to a predetermined transcoding server 950 that provides a call service by converting media.

Fig. 10 shows another example of the configuration of the service providing system in the Internet telephone system using the connection setting protocol according to the present invention. This system 1000, which provides a Session Initiation Protocol service, corresponds substantially to the transcoding server 420 of FIGS.

The system 1000 includes a message receiver 1010 that receives a request message from the SIP client A 1005 to make a call through the SIP service, and an address for accessing the SIP client B 1050 that is the received call target. The call information receiving unit 1020 and the call to the registration server 1030 that registers the address of the terminal and the media type, which is a separate call type to be used for the call, and receives the connection address and the media type in response. Set a session with the terminal corresponding to the address included in the request message, and establish a separate session with the terminal corresponding to the call counterpart address received through the call information receiver 1020, and the media type included in the call request message. To convert the call content according to the media type of the call counterpart and provide the call service between the SIP Client A (1005) and the SIP Client B (1050). And a currency conversion section 1040.

Since the operation of the system has been substantially described through the description of FIGS. 4 to 8, detailed description thereof will be omitted.

In the SIP Internet telephone system, even if the caller and the callee attempt to make a telephone call with different media types, both ends of the telephone call can be made and the media type can be changed seamlessly even during the telephone call. In the present invention has been described in detail through only the SIP Internet telephone system, it can be extended to be applicable to various SIP-based multimedia applications.

As described above, the present invention improves the problem of placing a large burden on the terminal in a SIP call service, and particularly, wasting the limited bandwidth available to the terminal in a wireless environment due to the procedure for the telephone call service. The proxy server minimizes the load of the terminal by dedicating functions previously used in the terminal.

Those skilled in the art will appreciate that the present invention can be implemented in a modified form without departing from the essential features of the present invention. Therefore, the disclosed embodiments should be considered in descriptive sense only and not for purposes of limitation. Examples included in the above description are introduced for the understanding of the present invention, and these examples do not limit the spirit and scope of the present invention. For example, in the above description, the Internet is mainly used as an example of a communication network, but it is also possible to use a public telephone communication network such as a PSTN, and in addition to the above examples, various embodiments according to the present invention are possible. It will be obvious to those of ordinary skill in this technical field. The scope of the present invention is shown not in the above description but in the claims, and all differences within the scope will be construed as being included in the present invention.

In addition, it can be easily understood by those skilled in the art that each of the above steps according to the present invention can be variously implemented in software or hardware using a general programming technique.

And some steps of the invention may also be embodied as computer readable code on a computer readable recording medium. The computer-readable recording medium includes all kinds of recording devices in which data that can be read by a computer system is stored. Examples of computer-readable recording media include ROM, RAM, CD-ROM, CD-RW, magnetic tape, floppy disks, HDDs, optical disks, magneto-optical storage devices, and carrier wave (eg, Internet It also includes the implementation in the form of). The computer readable recording medium can also be distributed over network coupled computer systems so that the computer readable code is stored and executed in a distributed fashion.

According to the present invention, in the Session Initiation Protocol service, each connection establishment protocol service terminal registers its own address to call through the service and a media type, which is a separate call type to be used for the call, to the registration server. When a call service is requested to an address corresponding to a terminal providing a SIP service, a query is made to a registration server to determine whether there is a media type registered in advance by the terminal corresponding to the address. To a transcoding server that converts the contents of the call into voice and text, and the transcoding server maintains separate sessions with the terminal requesting the call service and the terminal targeted for the call service, while maintaining the call according to a separate media type. Converts the contents to provide a call service between the two terminals, S In the IP telephone system, if a user connected via softphone does not have a headset, or voice call is difficult due to the user's circumstances, or if the user cannot make a telephone call through voice transmission, for example, because the user has a hearing impairment, text transmission is performed instead of voice. Allows normal phone calls.

Different media data is converted through a separate transcoding server, and SIP messages are transferred from the proxy server to the transcoding server instead of the callee. The routing of the proxy server is determined by the presence or absence of the usermode parameter in the callee's access address registered with the registrar server. Since the message is transferred directly from the proxy server to the transcoding server, the message processing load of the callee can be reduced, and in the case of a wireless environment, data transmission and reception of the wireless section can be reduced. As a result, the present invention provides a new service for enabling a telephone call by text transmission and reception instead of voice for a telephone for a hearing-impaired person who is difficult to use a voice call service during a class or conference or a voice call service in a SIP-based Internet telephone system. Provide a way.

Claims (19)

In the Session Initiation Protocol service, (a) registering each connection establishment protocol service terminal with its own address to call through the service and a separate call type to be used for the call; (b) if a call service is requested to an address corresponding to the terminal providing the service, determining whether there is a separate call type registered in advance by the terminal corresponding to the address; And (c) if there is a separate call type, converting a call of one counterpart according to the separate call type and providing the call to another counterpart to provide a call service; Method of providing service in Internet telephone system. The method of claim 1, The separate call type is to select a voice call, a text call and a call using both voice and text, The method of providing a service in the Internet telephone system using the connection setting protocol, characterized in that the call conversion of step (c) is a call conversion between voice and text, or voice to text and text to voice and text simultaneously. . The method of claim 1, The method for requesting a call service is a method for providing a service in an Internet telephony system using a connection establishment protocol, which transmits information on a call type together with a request for a call. In the Session Initiation Protocol service, (a) registering, by the access server, each connection establishment protocol service terminal with its registration address and a media type, which is a separate call type to be used for the call, with the registration server; (b) if the call service is requested to an address corresponding to the terminal providing the service, querying the registration server whether there is a media type registered in advance by the terminal corresponding to the address; And (c) if there is a separate media type registered in advance, transmitting the call service request to a transcoding server that converts the contents of a call into voice and text; And (d) the transcoding server converting call contents according to the separate media types while maintaining a separate session with the terminal requesting the call service and the terminal targeted for the call service, and providing a call service between both terminals; A service providing method in an Internet telephone system using a connection establishment protocol, characterized in that it comprises a. The method of claim 4, wherein And registering the media type in step (a) uses a register message for registering a user's access address. The method of claim 5, And registering the media type through a separate parameter in a contact header included in a register message. The method according to any one of claims 4 to 6, The separate media type is to select a voice call, a text call and a call using both voice and text, The call conversion of step (d) is a method of providing a service in an Internet telephone system using a connection setting protocol, characterized in that the call conversion between voice and text, or voice to text and text to voice and text simultaneously. . The method of claim 4, wherein The requesting side of the call service includes an information on the type of media used by the SDP (Session Description Protocol) included in the INVITE message used to request the call and delivers the type of media used by the calling service. To provide services in an Internet phone system using the Internet. The method according to claim 4 or 8, When the transcoding server receives the call request message, the transcoding server queries the registration server to obtain an address of a call partner. A service providing method in an Internet telephone system using a connection establishment protocol, wherein a call request is generated by generating an INVITE message specifying a media type designated by a caller in advance to the obtained address. The method of claim 4, wherein A method of providing a service in an Internet telephone system using a connection establishment protocol, wherein providing a voice call service uses the Real-time Transport Protocol (RTP), and providing a text message is through an MSRP or MESSAGE message. The method of claim 4, wherein When the terminal requesting the call service changes the media type, a message for requesting the call service is transmitted to the transcoding server, and the transcoding server receiving the call service transmits the call transmitted through the session with the terminal requesting the call service. A service providing method in an Internet telephone system using a connection setting protocol, characterized by converting and providing according to the type of media. The method of claim 11, And the message requesting to change the media type uses a re-INVITE message. In a system that provides a Session Initiation Protocol service, A message receiver which receives a request message to call through the service; Inquiries to the registration server for registering the address of the terminal to call through the service and the media type, which is a separate call type to be used for the call, receives the access address as a response, and receives the address. A call information receiving unit for receiving the media type information corresponding to the access address in response; And If the media type information is not received in the response through the call information receiver, a message is provided to the terminal corresponding to the access address received in the response to provide a call service, and when the media type information is received in the response, conversion between media is performed. And a call setup unit configured to transfer the call request message to a predetermined transcoding server providing a call service by using the connection setup protocol. In a system that provides a Session Initiation Protocol service, A message receiver which receives a request message to call through the service; Inquiries to the registration server that registers the address of the terminals making the call to the call destination through the service and the media type, which is a separate call type to be used for the call, and receives the access address and the media type in response. Information receiver; And Set a session with the terminal corresponding to the address included in the call request message, and establish a separate session with the terminal corresponding to the call counterpart address received through the call information receiver, and the media type included in the call request message. And a call conversion unit for converting the contents of the call according to the media type of the call counterpart to provide a call service. The method according to claim 13 or 14, The call request message received by the message receiver delivers the media type used by the call request terminal by including information on the media type to be used by the SDP (Session Description Protocol) included in the INVITE message used to request the call. A service providing system in an Internet telephone system using a connection establishment protocol, characterized in that. The method of claim 14, The media type converted by the currency converter is one of a voice call, a text call, and a call using both voice and text. The call conversion unit is a service providing system in the Internet telephone system using a connection setting protocol, characterized in that the conversion of the voice and text or the voice to the voice and text or the voice and text at the same time, depending on the media type. The method of claim 16, In the Internet telephone system using the connection establishment protocol, the call conversion unit provides a voice call service using the Real-time Transport Protocol (RTP), and providing a text message through the MSRP or MESSAGE message. Service delivery system. The method of claim 14, When the terminal requesting the call service changes the media type during a call, the message receiving unit receives and transmits the message requesting the message to the call conversion unit, and the received call conversion unit requests the call service. A service providing system in an Internet telephone system using a connection establishment protocol, characterized in that for converting and providing a call transmitted through a session with the type of the changed media. The method of claim 18, And a message for requesting to change the media type is to use a re-INVITE message.

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