KR100809398B1 - Sms transmission method and system for multicast protocol support - Google Patents

Abstract

Disclosed are a method and system for transmitting SMS in VoIP that supports multiprotocol. In this method, a soft switch existing in a VoIP network receives a first call connection request signal for a call connection from an originating terminal to an SMS interworking server, and relays a first call setup between the originating terminal and the SMS interworking server. ; Receiving, by the SMS interworking server, SMS data from the calling terminal through a first RTP media channel opened according to the first call setup; And transmitting, by the SMS interworking server, the SMS data to the called terminal based on the first call connection request signal or the called terminal number included in the SMS data. Contains an identifier or a number that specifies the SMS interworking server. According to the present invention, each application service can easily use text messages, support multi-protocol, and minimize existing system modifications.

Description

Method and system for transmitting SMS in WIPIP supporting multiprotocol {Method and system for transmitting SMS for VoIP service supproting Multi-protocol}

1 shows an SMS transmission method and system therefor in VoIP supporting multiprotocol according to an embodiment of the present invention.

2 and 3 illustrate a method in which an SMS transmission method according to an embodiment of the present invention is applied between VoIP-based terminals.

4 illustrates a method of applying an SMS transmission method between a VoIP based terminal and another terminal according to an embodiment of the present invention.

5 illustrates contents stored in the routing DB 420 according to an embodiment of the present invention.

The present invention relates to a method and a system for transmitting an SMS. More specifically, when an ITSP (Internet Telephony Service Provider) provides an SMS service, the present invention provides an SMS service to a new SIP subscriber as well as an existing H.323 subscriber. In order to provide an efficient SMS service without modifying the existing network configuration.

In mobile communication, SMS has been spotlighted as a killer application service, and in the case of KT, SMS service is provided through local telephone service. The same SMS requirements exist in wired and wireless VoIP environments. The text message function is used in various application services. In case of directly linking mobile carriers in individual application services, it causes redundancy of development and it is difficult for integrated billing management. Development is desired. In addition to the SIP (Session Initiation Protocol) interworking method that can interoperate with existing VoIP services, various interworking technologies such as H.323, MGCP (Media Gateway Control Protocol), which are used in existing VoIP, are needed.

Many existing technologies use SMS channels to directly send SMS messages to send SMS messages. In this case, network devices such as SSW or Session Border Controller (SBC) must be created for SMS. There are disadvantages.

In the trend of standardization, we try to send SMS using MESSAGE method of SIP. Doing so will disable existing H.323 users or MGCP subscribers. This is because standardization is a standard targeting only SIP-based systems.

That is, the prior art has two main directions, a method of modifying the SMS transmission module used in the existing mobile carriers and a method of MESSAGE of SIP.

The former is a way of constructing a new network to provide SMS in VoIP. Regardless of the SSW that already exists, the SMS server is provided, and each VoIP terminal connects to the SMS server in order to transmit the SMS, and then sends and receives SMS messages in proprietary specifications. In this case, each terminal needs to modify the SMS transmission module, and a new network device for routing between terminals, including the SMS server, causes significant cost waste.

The latter method uses SIP's MESSAGE, which means that H.323 subscribers cannot use this system, so a separate SMS server for H.323 is required. This is a waste of having to build a separate system for each VoIP protocol. In addition, the existing routing equipment is mainly routed system using INVITE, so the routing part using MESSAGE needs to be modified.

Equipment for transmitting SMS to new SIP and MGCP subscribers, including existing H.323 subscribers, is required. In addition, there is a need for equipment that can efficiently provide SMS without modifying existing routing equipment for low cost.

The technical problem to be achieved by the present invention, regardless of the VoIP protocol method, can provide the SMS service while using the existing SSW (Soft Switch: hereinafter referred to as soft switch) and routing equipment such as SBC or media processing equipment as it is. There is provided an apparatus and a method.

In order to achieve the above technical problem, according to the present invention, a method for transmitting an SMS in VoIP supporting multiprotocol according to the present invention includes a first call connection request signal for a soft switch existing in a VoIP network for a call connection from an originating terminal to an SMS interworking server. Receiving a call and relaying a first call setup between the calling terminal and the SMS interworking server; Receiving, by the SMS interworking server, SMS data from the calling terminal through a first RTP media channel opened according to the first call setup; And transmitting, by the SMS interworking server, the SMS data to the called terminal based on the first call connection request signal or the called terminal number included in the SMS data. Contains an identifier or a number that specifies the SMS interworking server.

In the above method, preferably, at the call setup, the media type of the RTP media channel is set to a voice channel or a video channel.

In the above method, preferably, the transmitting comprises: a second call between the SMS interworking server and the called terminal based on a first call connection request signal or a called terminal number obtained from the SMS interworking server; Relaying settings; And sending, by the SMS interworking server, the SMS data to the called terminal through a second RTP media channel opened according to the second call setting.

In the method, preferably, the transmitting may include: transmitting, by the SMS interworking server, the SMS data to the called terminal through a gateway of a network corresponding to the called terminal, based on the called terminal number. It includes.

In the above method, preferably, the relaying of the second call setup may include: connecting, by the SMS interworking server, a call to the called terminal based on the called terminal number included in the SMS data received from the calling terminal. Transmitting a second call connection request signal for a soft switch; And receiving, by the soft switch, the second call connection request signal, and relaying a second call setup between the SMS interworking server and the called terminal.

In the above method, preferably, the first call connection request signal includes the called terminal number, and the relaying of the second call setting may include: an incoming call in which the soft switch is included in the first call connection request signal. Relaying a call setup between the SMS interworking server and the called terminal based on the terminal number.

In the above method, preferably, the SMS interworking server further comprises the step of receiving the reception result of the SMS data from the called terminal and transmitting the received reception result to the calling terminal.

In the method, preferably, the second call connection request signal includes an identifier or a number specifying the SMS interworking server.

Receiving the SMS data from the calling terminal through the gateway of the network to which the calling terminal according to the present invention for achieving the above another technical problem; Transmitting, by the SMS interworking server, a call connection request signal for connecting a call to a called terminal to a soft switch based on the called terminal number included in the SMS data; Receiving, by the soft switch, the call connection request signal and relaying a call setup between the SMS interworking server and the called terminal; And transmitting, by the SMS interworking server, the SMS data to the called terminal through an open RTP media channel according to the call setup, and the call connection request signal includes an identifier or number specifying the SMS interworking server. do.

In the above method, preferably, at the call setup, the media type of the RTP media channel is set to a voice channel or a video channel.

In the above method, preferably, the SMS interworking server further comprises the step of receiving the reception result of the SMS data from the called terminal and transmitting the received reception result to the calling terminal.

According to another aspect of the present invention, there is provided an SMS transmission system in a VoIP supporting multiprotocol according to the present invention, which receives an SMS data from a calling terminal through a first RTP media channel opened according to a first call setup. An SMS interworking server that transmits the SMS data to the called terminal based on a call connection request signal or a called terminal number included in the SMS data; And a soft switch, located in a VoIP network, for relaying the first call setup between the calling terminal and the SMS interworking server based on the first call connection request signal received from the calling terminal. The signal includes an identifier or number specifying the SMS interworking server.

In the system, when setting each call, the media type of each RTP media channel is set to a voice channel or a video channel.

In the system, the soft switch relays the second call setup between the SMS interworking server and the called terminal based on a first call connection request signal or a called terminal number obtained from the SMS interworking server, and the SMS interworking server. Transmits the SMS data to the called terminal through a second RTP media channel opened according to the second setup.

In the system, the SMS interworking server transmits the SMS data to the called terminal through a gateway of a network corresponding to the called terminal based on the called terminal number.

In order to achieve the above technical problem, the SMS transmission system in the VoIP supporting the multiprotocol according to the present invention is based on the called terminal number included in the SMS data received through the gateway of the calling terminal. An SMS interworking server generating a call connection request signal for connecting a call and transmitting the SMS data to the called terminal through an open RTP media channel according to the call connection request signal; And a soft switch for receiving the call connection request signal and relaying call setup between the SMS interworking server and the called terminal, wherein the call connection request signal includes an identifier or a number for specifying the SMS interworking server.

In the system, when the call is set, the media type of the RTP media channel is set to a voice channel or a video channel.

Hereinafter, a method and an apparatus according to the present invention will be described in detail with reference to the accompanying drawings.

The present invention has been proposed in order to solve the above problems, and can provide an SMS service regardless of the VoIP protocol method, and use only the SMS interworking server while using existing routing devices or media processing equipment such as SSW and SBC. SMS can be provided in such a way that it is possible.

To provide this, call processing for routing in each VoIP protocol is already provided through SSW, so use SSW. In addition, the data transmission for transmitting the actual SMS is to send the SMS data through the Real Time Protocol (RTP) channel established between the terminal and the SMS interworking server. Originally, the RTP media channel is used for transmitting voice or video data regardless of SIP, H.323, and MGCP. When this channel is used for SMS transmission, SMS can be transmitted regardless of the protocol. In order to be able to use it in SBC for NAT (Network Address Transition) or a user in a firewall, SMS data must be sent as if it is a voice codec or video codec used in RTP.

When the calling terminal attempts to make a call, if the called number is the actual called number, since it does not go through the SMS interworking server, the fixed interworking number is assigned to the SMS interworking server and the call is made to the SMS interworking server. The actual calling number and the actual called number are transmitted through the RTP media channel.

In order to distinguish between general call and SMS, the called terminal must send SMS interworking server to the calling number when SMS interworking server attempts SMS call with the called terminal. Thus, the called terminal can look at the calling number and distinguish whether it is an SMS call or a general call.

When sending an SMS to a mobile phone or a local phone, the procedure from the VoIP terminal to the SMS interworking server is the same as that of the incoming / outgoing VoIP call. When the SMS interlocking server sends and receives data to each mobile phone SMS GW or city telephone SMS GW, do.

In other words, when the present invention is to provide a Short Message Service (SMS) in a VoIP (Voice Over IP) service that supports multi-protocol, it utilizes the routing function of SSW (SoftSwitch), a VoIP routing device, and processes SMS messages and targets. The present invention relates to a method for efficiently providing an SMS by using an SMS interworking server that handles interworking with a network.

1 shows an SMS transmission method and system therefor in VoIP supporting multiprotocol according to an embodiment of the present invention.

Referring to Figure 1, the system according to the present invention is a configuration related to the VoIP network 100, SIP terminal 110, 112, H.323 terminal 120, MGCP terminal 130, SSW 400, SMS Interworking server 410 and routing DB 420 is included.

In addition, in the system according to the present invention, the wireless public telephone network 200 and the public telephone network 300 are taken as examples of other networks other than the VoIP network. Here, in a configuration related to the wireless public telephone network 200, and includes a mobile phone SMS gateway 210 and a mobile phone terminal 220, and in a configuration related to the public telephone network 300, the local phone SMS gateway 310 and the city The phone terminal 320.

The SIP terminals 110 and 120 generate and call SIP calls. The SIP terminals 110 and 120 include all terminals capable of generating SIP calls such as SIP soft phones, SIP hard IP phones, SIP gateways (GW), and general telephones.

The H.323 terminal 120 generates an H.323 call and makes an incoming / outgoing call. The MGCP terminal 130 generates an MGCP call and makes an incoming / outgoing call. The H.323 terminal 120 and the MGCP terminal 120 and 130 are similar to the SIP terminals 110 and 112, and the H.323 / MGCP soft phone, the H.323 / MGCP hard IP phone, and the H.323 / MGCP GW (Gateway). This includes all terminals capable of receiving H.323 / MGCP calls, such as landlines.

The VoIP network 100 transmits a VoIP signal.

SSW 400 may process a multi-protocol, and accordingly relays call setup by the SIP protocol between the SIP terminal (110, 120) and the SMS interworking server 410, the H.323 terminal 120 and SMS interworking The server 410 relays the call setting by the H.323 protocol, and the MGCP terminal 130 and the SMS interworking server 410 relays the call setting by the MGCP protocol.

SMS interworking server 410 is responsible for SMS processing. Specifically, the SMS interworking server 410 is connected to various terminals 110, 112, 120, 130 based on VoIP according to the call setup relay of the SSW 400, the connected terminal (110, 112, 120, 130) Alternatively, SMS data is transmitted and received with various gateways 210 and 310. Here, the SMS interworking server 410 performs transmission and reception using the routing DB 420 in which SMS routing information is stored.

In particular, the SMS interworking server 410 is provided with one or more specific VoIP phone number that can be received, accordingly, the VoIP-based terminal (110, 112, 120, 130) can be directly connected to the SMS interlocking server 410 . In this case, the call connection request signal required for the call connection may include or be used as the specific VoIP telephone number. An example of the specific VoIP telephone number is a dedicated number of the SMS-linked server 410 such as 070-7700-0000. Or "identifier (identification number)-called terminal number indicating the SMS service". That is, the SSW 410 receives the call connection request signal and interworks with the IP and port of the SMS interworking server 410.

The mobile phone SMS gateway 210 links the wireless public telephone network 200 with the VoIP network 100 or the public telephone network 300, and the local telephone SMS gateway 310 connects the public telephone network 300 with the VoIP network 100 or the like. Interwork with the wireless public telephone network 200.

The mobile phone terminal 220 and the local phone terminal 320 may process the corresponding text message according to the SMS data.

Referring to Figure 1, the concept of SMS transmission between the VoIP-based calling terminal and the VoIP-based called terminal according to an embodiment of the present invention. Here, the calling terminal is the SIP terminal 110, the called terminal presupposes the SIP terminal 112, assuming that the call connection request signal is a dedicated number of the SMS interworking server 410 such as 070-7700-0000 Explain.

The calling terminal transmits a first call connection request signal for call connection with the SMS interworking server 410.

The soft switch 400 in the VoIP network 100 receives the first call connection request signal and relays the first call setup between the calling terminal and the SMS interworking server 410. In this case, the protocol used for call setup is a SIP protocol, and the soft switch 400 supports the multi-protocol, such a call setup relay is possible. Here, as described above, the first call connection request signal includes a number for specifying the SMS interworking server, and the soft switch 400 relays the call setup to the SMS interworking server 410 based on this. .

The SMS interworking server 410 receives SMS data from the calling terminal through the first RTP media channel opened according to the first call setup, and utilizes the routing DB 420 based on the called terminal number included in the SMS data. When it is determined that the called terminal is a VoIP-based terminal, it generates a second call connection request signal and transmits it to the soft switch 400.

The soft switch 400 receives the second call connection request signal and relays the second call setup between the called terminal and the SMS interworking server 410. Here, the protocol used for call establishment is a SIP protocol which is a protocol used by the called terminal. Here, the second call connection request signal includes an identifier or a number for specifying an SMS service or the SMS interworking server 410. For example, 070-7700-0000 should be used so that the called terminal can later distinguish it from a general call.

The called terminal detects an SMS service related call according to the call setup relay of the soft switch 400 and is connected to the SMS interworking server 410 to open a second RTP media channel.

The SMS interworking server 410 transmits the received SMS data to the called terminal through the second RTP media channel.

Thereafter, the called terminal receives the SMS data and transmits the received result to the SMS interworking server 410 through the second RTP media channel. After that, when the object related to the SMS is achieved, the connection between the calling terminal and the SMS interworking server 410 and the connection between the called terminal and the SMS interworking server 410 are terminated.

Referring to FIG. 1, the concept of SMS transmission between a VoIP-based calling terminal and a VoIP-based called terminal according to another embodiment of the present invention will be described below. Here, the calling terminal is the SIP terminal 110, the called terminal is assumed to be the SIP terminal 112, the description is based on the assumption that the first call connection request signal "SMS identifier (for example, 77)-called terminal number" do.

The calling terminal transmits a first call connection request signal for call connection with the SMS interworking server 410.

The soft switch 400 in the VoIP network 100 receives the first call connection request signal and relays the first call setup between the calling terminal and the SMS interworking server 410. In addition, the soft switch 400 relays the second call setup between the SMS interworking server 410 and the called terminal based on the called terminal number included in the first call connection request signal.

The SMS interworking server 410 receives SMS data from a calling terminal through a first RTP media channel opened according to the first call setup, and sends the SMS data to a called terminal through a second RTP media channel opened according to the second call setup. Send the received SMS data.

Thereafter, the called terminal receives the SMS data and transmits the received result to the SMS interworking server 410 through the second RTP media channel. After that, when the object related to the SMS is achieved, the connection between the calling terminal and the SMS interworking server 410 and the connection between the called terminal and the SMS interworking server 410 are terminated.

Referring to FIG. 1, the concept of SMS transmission between a VoIP-based calling terminal and another network-based called terminal according to an embodiment of the present invention is as follows. Here, the calling terminal is a SIP terminal 110, the called terminal is a mobile phone terminal 220, the premise that the call connection request signal is a dedicated number of the SMS interworking server 410 such as 070-7700-0000 Will be explained.

The calling terminal transmits a first call connection request signal for call connection with the SMS interworking server 410.

The soft switch 400 in the VoIP network 100 receives the first call connection request signal and relays the first call setup between the calling terminal and the SMS interworking server 410. In this case, the protocol used for call setup is a SIP protocol, and the soft switch 400 supports the multi-protocol, such a call setup relay is possible. Here, as described above, the first call connection request signal includes a number for specifying the SMS interworking server, and the soft switch 400 relays the call setup to the SMS interworking server 410 based on this. .

The SMS interworking server 410 receives SMS data from the calling terminal through the first RTP media channel opened according to the first call setup, and utilizes the routing DB 420 based on the called terminal number included in the SMS data. If it is determined that the called terminal is a terminal based on a wireless public telephone network, the SMS data is transmitted to the mobile telephone SMS gateway 210 for the wireless public telephone network. Then, the mobile phone SMS gateway 210 transmits the SMS data to the mobile phone terminal 220. Thereafter, the reception result of the SMS data of the called terminal is transmitted to the SMS interworking server 410 via the mobile phone SMS gateway 210 and the VoIP network 100.

2 is a diagram illustrating a method of applying an SMS transmission method between VoIP-based terminals according to an embodiment of the present invention, wherein the calling terminal is a SIP terminal 110 and the called terminal is a SIP terminal 112. It demonstrates with reference to 1.

The user inputs an SMS message to the SIP terminal 110 and requests a call to a specific SMS interworking server number (070-7700-0000) (S220).

Next steps (S221, S222) is a step of establishing a SIP call from the SIP terminal 110 to the SMS interworking server 410 via the soft switch 400. In this case, the SMS interworking server 410 may operate as a B2BUA (Back To Back User Agent) to separately control the calling and the called party and may process multiple calls simultaneously. When the call is set up, an RTP media channel between the SIP terminal 110 and the SMS interworking server 410 is opened.

In the next step S223, SMS data is transmitted from the SIP terminal 110 to the SMS interworking server 410 through the RTP media channel. The SMS data includes the actual calling number, the actual called number and the message, and may further include the calling time, reservation time, retransmission, etc., as necessary.

The next step (S224) is to see the actual incoming number received through the RTP media channel SMS interworking server 410 inquires the routing DB 420, if the VoIP call, and determines that the call to the soft switch 410 to be. When the call is made by SSW, the calling number must be used with the number (070-7700-0000) of the SMS interworking server 410 so that the called terminal can be distinguished from the general call later.

Next steps (S225, S226) is the step of setting the SIP call to the SIP terminal 112 via the SMS interlocking server 410 via the soft switch 400. The called terminal, i.e., the SIP terminal 112 or the user of the terminal sees the calling number after the call is set and if it is the number of the SMS interworking server 410, determines that it is an SMS call and performs SMS processing (S227). Through this step, an RTP media channel for transmitting SMS data is opened between the SMS interworking server 410 and the SIP terminal 112.

The next step S228 is a step in which the SMS interworking server 410 transmits SMS data to the SIP terminal 112. The SMS interworking server 410 transmits the SMS data received from the caller SIP terminal 110 to the caller SIP terminal 112.

The next step (S229) is a step in which the called SIP terminal 112 sends a signal indicating that the SMS has been well received to the SMS interworking server 410. The SMS interworking server 410 sends the result to the caller SIP terminal 110 (S230).

If the transmission is successful, the SMS interworking server 410 terminates the call connection of each terminal 110, 112 through the soft switch 400. (S231, S232, S233, S234)

However, when setting media, the media type is set as if it is an audio channel such as G.711, G.723.1, or G.729a, or as if it is a video channel such as H.261, H.263, or H.264. For example, NAT (Natwork Addres Transistion) or Firewall users use a technology that uses a SBC (Session Border Controller) to pass. If the codec is not frequently used, other SBCs can be modified or NAT traversed. Because you have to make a device.

3 is a diagram illustrating a method of applying an SMS transmission method between VoIP-based terminals according to an embodiment of the present invention, wherein the calling terminal is a SIP terminal 110 and the called terminal is an H.323 terminal 120. This will be described with reference to FIG. 1.

Since both the calling terminal and the called terminal are terminals based on the VoIP network, the description of the procedures of S220 to S234 of FIG. 2 is applied in principle to the procedures of S320 to S334 which are the procedures of this embodiment. That is, similar to the embodiment of FIG. 2, the SMS interworking server 410 and the soft switch 400 use the same SIP protocol, and the H.323 terminal 120 and the soft switch 400 between the H. Set the call using the 323 signal. This signal conversion process is utilized because it is a unique function of the soft switch 400. Therefore, when the call is set up and terminated, the H.323 connection process between the soft switch 400 and the H.323 terminal 120 is different from the embodiment of FIG. Here, the operation of the SMS interworking server 410 can be seen that the same regardless of the terminal.

4 is a diagram illustrating a method of applying an SMS transmission method between a VoIP-based terminal and another terminal according to an embodiment of the present invention, wherein the calling terminal is a SIP terminal 110 and a called terminal mobile telephone terminal or a local telephone. Assuming the case of a terminal, it will be described with reference to FIG.

The user inputs an SMS message to the SIP terminal 110 and calls a specific SMS interworking server number (070-7700-0000) (S420).

Next steps (S421, S422) is a step of establishing a SIP call from the caller SIP terminal 110 to the SMS interworking server 410 via the soft switch 400. In this case, the SMS interworking server 410 may operate as a B2BUA (Back To Back User Agent) to separately control the calling and the called party and may process multiple calls simultaneously. When the call is set up, an RTP media channel between the SIP terminal 110 and the SMS interworking server 410 is opened.

In the next step S423, SMS data is transmitted from the SIP terminal 110 to the SMS interworking server 410 through the RTP media channel. The SMS data includes the actual calling number, the actual called number and the message, and may further include the calling time, reservation time, retransmission, etc., as necessary.

The next step (S424) is to see the actual incoming number received through the RTP media channel SMS interworking server 410 queries the routing DB 420, if the mobile phone number or local phone number, the mobile phone SMS gateway 210 or It is a step of determining to transmit to the local telephone SMS gateway (310).

Next step (S425, S426), the SMS interworking server 410 via the mobile phone SMS gateway 210 or the local call SMS gateway 310 SMS data to the destination mobile phone terminal 220 or the local phone terminal 320 It is a step of transmitting. At this time, the SMS interworking server 410 must transmit SMS data in compliance with the mobile phone SMS standard.

The next step (S427, S428) is that the recipient mobile phone terminal 220 or the local phone terminal 320 receives the SMS data, the result of receiving the SMS data through the mobile phone SMS gateway 210 or the local phone SMS gateway 310 via the SMS interworking server Sending to 410. The SMS interworking server 410 sends the result to the caller SIP terminal 110 (S429).

If the transmission is successful, the SMS interworking server 410 terminates the call connection of the caller SIP terminal 110 through the soft switch 400 (S430, S431).

The calling terminal is a terminal other than a VoIP network (wireless public telephone network or public telephone network), that is, a mobile telephone terminal 220 or a local telephone terminal 320, and the called terminal is a VoIP based terminal 110, 112, 120, 130 may be applied in the opposite direction to the embodiment of FIG. 4.

5 illustrates contents stored in the routing DB 420 according to an embodiment of the present invention. That is, the routing DB 420 of FIG. 1 is a diagram showing an embodiment of an IP port and a port of an SMS gateway or a soft switch matching the terminal number range and service provider type. As shown in the figure, IP and port are matched with one terminal number range and service provider form in the form of a lookup table.

The invention can also be embodied as computer readable code on a computer readable recording medium. Computer-readable recording media include all kinds of recording devices that store data that can be read by a computer system. Examples of computer-readable recording media include ROM, RAM, CD-ROM, magnetic tape, floppy disks, optical data storage devices, and the like, which are also implemented in the form of carrier waves (for example, transmission over the Internet). Include. 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. In addition, functional programs, codes, and code segments for implementing the present invention can be easily inferred by programmers in the art to which the present invention belongs.

Such a method and apparatus of the present invention have been described with reference to the embodiments shown in the drawings for clarity, but these are merely exemplary, and various modifications and equivalent other embodiments are possible to those skilled in the art. Will understand. Therefore, the true technical protection scope of the present invention will be defined by the appended claims.

According to the present invention, without modifying the existing routing equipment, it is possible to provide an SMS service by leaving only the SMS linked server. In other words, by using soft switch, routing can use the existing equipment and provide SMS processing and other mobile phone network or local phone network by SMS interworking server, so that service can be provided regardless of VoIP protocol including existing H.323 subscriber. In addition, interworking between providers is easy because other ITSP providers are already linked with soft switches.

Claims (16)

delete Receiving, by a soft switch existing in a VoIP network, a first call connection request signal for a call connection from an originating terminal to an SMS interworking server, and relaying a first call setup between the originating terminal and the SMS interworking server; Receiving, by the SMS interworking server, SMS data from the calling terminal through a first RTP media channel opened according to the first call setup; And Transmitting, by the SMS interworking server, the SMS data to the called terminal based on the first call connection request signal or the called terminal number included in the SMS data; The first call connection request signal includes an identifier or a number for identifying the SMS interworking server, and when the call is set up, a media type of the RTP media channel is set to a voice channel or a video channel. SMS transmission method in VoIP. The method of claim 2, wherein the transmitting step, Relaying, by the soft switch, a second call setup between the SMS interworking server and the called terminal based on a first call connection request signal or a called terminal number obtained from the SMS interworking server; And And transmitting, by the SMS interworking server, the SMS data to the called terminal through a second RTP media channel opened according to the second setting. The method of claim 2, wherein the transmitting step, The SMS interworking server transmitting the SMS data to the called terminal through a gateway of a network corresponding to the called terminal, based on the called terminal number; How to send SMS. 4. The method of claim 3, wherein the relaying of the second setting Transmitting, by the SMS interworking server, a second call connection request signal to a soft switch to connect a call to the called terminal based on the called terminal number included in the SMS data received from the calling terminal; And Receiving, by the soft switch, the second call connection request signal, and relaying a second call setup between the SMS interworking server and the called terminal; SMS transmission method in VoIP supporting multi-protocol. . The method of claim 3, The first call connection request signal includes the called terminal number. Relaying the second setting is The soft switch relays a call setup between the SMS interworking server and the called terminal based on the called terminal number included in the first call connection request signal. How to send SMS. The method of claim 2, And receiving, by the SMS interworking server, the reception result of the SMS data from the called terminal, and transmitting the received reception result to the calling terminal, SMS transmission in VoIP supporting multi-protocol. Way. The method of claim 3, wherein the second call connection request signal is SMS transmission method in the VoIP supporting the multi-protocol, characterized in that it comprises an identifier or number for specifying the SMS interworking server. Receiving, by an SMS interworking server, SMS data from the calling terminal through a gateway of a network to which the calling terminal belongs; Transmitting, by the SMS interworking server, a call connection request signal for connecting a call to a called terminal to a soft switch based on the called terminal number included in the SMS data; Receiving, by the soft switch, the call connection request signal and relaying a call setup between the SMS interworking server and the called terminal; And Sending, by the SMS interworking server, the SMS data to the called terminal through an open RTP media channel according to the call setup; The call connection request signal is an SMS transmission method for supporting a multi-protocol, characterized in that it comprises an identifier or number specifying the SMS interworking server. The method of claim 9, When the call is set, SMS transmission method for VoIP supporting multi-protocol, characterized in that the media type of the RTP media channel is set to a voice channel or a video channel. The method of claim 9, And receiving, by the SMS interworking server, the reception result of the SMS data from the called terminal, and transmitting the received reception result to the calling terminal, SMS transmission in VoIP supporting multi-protocol. Way. Receive SMS data from the calling terminal through the first RTP media channel opened according to the first call setup, and receive the SMS data based on the first call connection request signal or the called terminal number included in the SMS data. SMS interworking server to send to; And Located in the VoIP network, and based on the first call connection request signal received from the calling terminal includes a soft switch for relaying the first call setting between the calling terminal and the SMS interworking server, The first call connection request signal is an SMS transmission system in support of multi-protocol, characterized in that it comprises an identifier or number for specifying the SMS interworking server. The method of claim 12, The soft switch relays the second call setup between the SMS interworking server and the called terminal based on a first call connection request signal or a called terminal number obtained from the SMS interworking server, The SMS interworking server transmits the SMS data to the called terminal through a second RTP media channel opened according to the second setting. The SMS interworking server of claim 12, wherein The SMS transmission system of the VoIP supporting the multi-protocol, characterized in that for transmitting the SMS data to the called terminal through the gateway of the network corresponding to the called terminal based on the called terminal number. Based on the called terminal number included in the SMS data received through the gateway of the calling terminal, a call connection request signal is generated for connecting a call to the called terminal, and an RTP media channel is opened according to the call connection request signal. SMS interworking server for transmitting the SMS data to the called terminal through; And And receiving a call connection request signal and relaying a call setup between the SMS interworking server and the called terminal. The call connection request signal is an SMS transmission system in support of multi-protocol, characterized in that it comprises an identifier or a number that specifies the SMS interworking server. A computer-readable recording medium containing a program for performing the method according to any one of claims 2 to 11.

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