KR100741379B1 - Application switch for controlling session initiation protocol communication via at least two subscriber's lines and method of operating the application switch - Google Patents

Abstract

An application switch controlling an SIP(Session Initiation Protocol) communication by using two and more subscriber lines and an operating method thereof are provided to perform a seamless SIP call connection, more stably and efficiently, by detecting a fault at plural lines connected to an external network including an SIP server and controlling process of the SIP call connection of a communication terminal allocated at a line, where a fault is detected, via other line, where a fault is not detected. An application switch controlling an SIP(Session Initiation Protocol) communication by using two and more subscriber lines comprises the first communication port(210), the second communication port(260), a memory(220), an SIP connection controller(230), a fault detector(240) and an address updating unit(250). The first communication port(210) is connected to an external network including an SIP server via plural lines wherein the first communication port(210) has two and more global IP addresses. The second communication port(260) is connected to an internal network including one and more communication terminals. The memory(220) maintains an NAT(Network Address Translation) table where an SIP address, a private IP address and a global IP address corresponding to one an more communication terminals are recorded. The SIP connection controller(230) converts the private IP address into the global IP address and vice versa by referring to the NAT table, and controls the SIP call connection via the SIP server. The fault detector(240) detects whether a fault occurs at the plural lines. The address updating unit(250) updates the first global IP address given to the first port of the first communication port connected to the first line, where a fault is detected, by the second global IP address given to the second port of the first communication port, where a fault is not detected.

Description

APPLICATION SWITCH FOR CONTROLLING SESSION INITIATION PROTOCOL COMMUNICATION VIA AT LEAST TWO SUBSCRIBER'S LINES AND METHOD OF OPERATING THE APPLICATION SWITCH}

1 is a view showing the entire network connection of the SIP service providing system according to an embodiment of the present invention.

Figure 2 is a block diagram showing the configuration of an application switch according to an embodiment of the present invention.

3 is a diagram illustrating a structure of a NAT table according to an embodiment of the present invention.

4 is a flowchart illustrating a method of processing a SIP call according to a case where a line failure occurs during setup / teardown operation of a SIP call according to an embodiment of the present invention.

5 is a flowchart illustrating a method of processing a SIP call when a circuit failure occurs during an RTP media session according to an embodiment of the present invention.

6 is a flowchart illustrating a flow of a method for providing a SIP service of an application switch according to an embodiment of the present invention.

7 is a flowchart illustrating a flow of a method for providing SIP service by an application switch according to another embodiment of the present invention.

<Explanation of symbols for main parts of the drawings>

200: application switch 210: first communication port unit

220: memory means 230: SIP connection control unit

240: failure detection unit 250: address update unit

260: second communication port

The present invention relates to an application switch for controlling SIP (Session Initiation Protocol) communication using a plurality of lines and a method for providing a SIP service, and more particularly, to a first of a plurality of lines connected to an external network including a SIP server. When a failure is detected for the circuit, the SIP call connection of the internal communication terminal through the first circuit is updated to the second circuit where the failure is not detected, and information about the register method for the update is updated. The present invention relates to an application switch for transmitting to the SIP server and a method for providing a SIP service.

Recently, the information communication technology is moving away from the technology of transmitting only data and transmitting video and voice together. The voice communication technology through the data network is called VoIP (Voice over Internet Protocol). In other words, VoIP is a protocol that enables voice and video communication using a packet switched network, which is a relatively inexpensive and efficient IP network, instead of a circuit switched network called a telephone network.

VoIP is a communication technology that provides voice telephony services and multimedia conferencing using Internet protocols. It is creating various equipments and application services by interworking IP networks and public telephone networks. Commonly used protocols for Internet telephony are H.323, Internet Engineering Task Force (SIP), Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), etc., as proposed by the International Telecommunication Unio-Telecommunication (ITU-T). There is this.

Among these Internet telephony protocols, Session Initiation Protocol (SIP) is widely used due to various functions and easy scalability. Generally, SIP is defined in IETF RFC2543 document to establish, modify, and terminate sessions or calls for multimedia communication such as video and voice. Means an application layer control protocol. SIP is a protocol that exists on the UDP / TCP / IP layer. It is a client / server based protocol that can send and receive SIP Request message and SIP Response message. It supports both unicast and multicast sessions.

The SIP-based VoIP system is composed of a user terminal and a SIP server. The SIP server may be largely configured to include a proxy server and a register server.

The proxy server is a server having a concept similar to that of a router in an IP network. When a sender terminal attempts to access a SIP call to a receiver terminal, the proxy server sends a message for establishing a SIP call between the sender terminal and the receiver terminal. send. The proxy server is responsible for transferring a call setup message from the sender terminal to the receiver terminal based on the location on the SIP network of the receiver terminal registered with the registration server.

The register server is a server that registers the current location on the SIP network of the user terminal. The register server and the proxy server may be physically present in one device. The network range managed by the SIP server is called a domain on the SIP network. The SIP server may manage a range of specific addresses.

The user terminal is assigned a respective SIP address and an IP address. The SIP address may be set similar to the email address format of the Internet. For example, it may be set in a format such as "sip: rooney@piolink.com", where "sip:" means protocol, "rooney" means user ID, and "piolink.com" means The domain to which the user terminal belongs. In addition, the IP address is a value assigned to the user terminal and may be generally set in the format of "192.168.1.1" as in the Internet.

The SIP address and IP address of this user terminal are registered in the SIP server. The SIP server may control the SIP call connection between each terminal by using the SIP address and IP address of the registered user terminal.

On the other hand, most of the SIP Internet phones currently in use are large-scale subscribers such as corporations and government offices. In this case, the large subscriber generally maintains an internal network including a plurality of communication terminals and is connected to the external network through a plurality of lines. An application switch may be used as a connection means between the external network and the internal network.

The application switch controls the SIP call connection of each communication terminal connected to the internal network by routing the SIP call received from the outside through the plurality of lines to the internal network using the SIP address and the IP address. can do.

For the SIP call connection, the application switch may control the SIP call connection of each communication terminal by allocating one or more communication terminals of a plurality of communication terminals included in the internal network to each external network line. In this case, the application switch records and maintains information (for example, an IP address) of an external line used by each communication terminal, and each communication terminal performs a SIP call connection with an external network according to the external line information. To control it.

However, if a failure occurs in a specific line among the plurality of external lines, the communication terminal assigned to the failed external line cannot perform SIP call connection with the outside. This is because, as above, a typical application switch maintains register information that allows a SIP call connection to the communication terminal to be performed only to the failed external line, as is the SIP server. That is, the SIP server also transmits a SIP call for the communication terminal to the failing line according to the register information, so that the communication terminal is connected to the outside even if a failure occurs only on a specific line among a plurality of lines connected to the application switch. There is a problem that it is inefficient because it cannot perform SIP call connection.

According to the problems of the prior art, the development of a more stable and efficient SIP communication service is required through a plurality of multiplexed circuits that overcome the instability of the circuits.

The present invention has been made to improve the prior art as described above, detecting a failure for a plurality of circuits connected to an external network including a SIP server, and the SIP call connection of the internal communication terminal assigned to the circuit where the failure occurs; An object of the present invention is to provide an application switch capable of performing a more stable and efficient SIP call connection and a SIP service method accordingly by controlling to be performed through the circuit in which the failure does not occur.

In addition, the present invention, if a failure occurs in the external line for performing the SIP call connection during the SIP call connection of the communication terminal, so that the SIP call connection of the communication terminal is replaced by a SIP call connection through the external line without the failure. By controlling, it is an object of the present invention to provide an application switch and SIP service method according to the user can ensure a stable SIP call connection.

The present invention also provides a public IP address to each of a plurality of circuits connected to an external SIP server, and assigns a private IP address to an internal communication terminal, thereby converting the public IP address and the private IP address to each other. By performing a call connection, an object of the present invention is to provide an application switch and an SIP service method according to which an SIP network resource can be economically utilized.

In order to achieve the above object and solve the problems of the prior art, the application switch according to the present invention, the first communication port unit connected to the external network including a SIP server through the plurality of lines-the first communication port unit Is given more than one public IP address; A second communication port unit connected to an internal network including one or more communication terminals; Memory means for maintaining a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded; A SIP access control unit for controlling a SIP call connection through the SIP server by translating the private IP address and the public IP address assigned to each of the communication terminals with reference to the NAT table; A failure detecting unit detecting whether a failure occurs on the plurality of lines; A first public IP assigned to a first port of the first communication port unit to which the first line is connected among the public IP addresses recorded in the NAT table when a failure of a first one of the plurality of lines is detected; An address updating unit for updating an address to a second public IP address assigned to a second port of the first communication port unit in which a failure is not detected, wherein the SIP access controller refers to the updated second public IP by referring to the NAT table; And a register method including an address and a SIP address of the communication terminal to the SIP server.

In addition, the SIP service providing method of the application switch according to the present invention, maintaining a connection with an external network including a SIP server through the plurality of circuits, the plurality of circuits comprises a plurality of first ports One or more public IP addresses assigned to the first communication port portion; Maintaining a connection with an internal network comprising one or more communication terminals; Maintaining a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded; Detecting whether the plurality of circuits are faulty; A first public IP assigned to a first port of the first communication port unit to which the first line is connected among the public IP addresses recorded in the NAT table when a failure of a first one of the plurality of lines is detected; Updating an address with a second public IP address assigned to a second port of the first communication port unit in which a failure is not detected; And transmitting a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server by referring to the NAT table.

In addition, the SIP service providing method of the application switch according to the present invention, maintaining a connection with an external network including a SIP server through the plurality of lines-the plurality of lines includes a plurality of first ports Coupled to a first communication port portion, wherein the first communication port portion is assigned two or more public IP addresses; Maintaining a connection with an internal network comprising one or more communication terminals; Maintaining a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded; When receiving a SIP call connection request from a first communication terminal, converting the private IP address given to the first communication terminal into the public IP address by referring to the NAT table to perform the SIP call connection through the SIP server. Controlling over a first line of the plurality of lines; Detecting whether the plurality of circuits are faulty; When a failure on the first line is detected, the failure detects a first public IP address assigned to a first port of the first communication port to which the first line is connected among the public IP addresses recorded in the NAT table. Updating to a second public IP address assigned to a second port of the first communication port unit, which is not performed; Controlling the SIP call connection of the first communication terminal to be replaced with a SIP call connection via a second line connected with the second port; And transmitting a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server by referring to the NAT table.

Hereinafter, with reference to the accompanying drawings will be described an embodiment of the present invention; 1 is a view showing the entire network connection of the SIP service providing system according to an embodiment of the present invention.

SIP service providing system according to an embodiment of the present invention is a SIP server 110, the Internet network 120, ADSL modems (131 and 132), application switch 133, L2 switch 134, and communication terminal 135 And 136).

The SIP server 110 may include a SIP proxy server, a SIP redirect server, a location server, and a register server. The SIP proxy server receives SIP calls of the communication terminals 135 and 136 from the application switch 133 to control the SIP call connection with other communication terminals. When the redirect server receives the SIP call, the redirect server provides the SIP proxy server with address information of the communication terminal corresponding to the SIP call.

The register server receives and registers a register method from each communication terminal 135 and 136, thereby maintaining address information and position information for each communication terminal 135 and 136. FIG. The location server receives the register method from the register server, and when receiving an address request of a destination node of a SIP call from the SIP proxy server or the redirect server, resolves it. can do.

The configuration and operation of the above-described SIP server 110 is only an example, SIP server 110 according to the present invention may be implemented to include all the various configurations and operations of the SIP server implemented in the art.

The internet network 120 includes a cable network, an asymmetric digital subscriber line (ADSL), a very high data rate digital subscriber line (VDSL), a high-bit-rate digital subscriber line (HDSL), and an integrated services digital network (IDSN). ), Including wired and wireless LANs. However, in the present specification, according to an embodiment of the present invention, a case where the most widely used ADSL network is implemented as the Internet network 120 will be described as an example.

As described above, when the Internet network 120 is implemented in ADSL, the application switch 133 may be connected to the ADSL network through the ADSL modems 131 and 132. One ADSL modem 131 and 132 may be installed for each line connected to the application switch 133. As such, when the ADSL network is implemented as the Internet network 120, a more economical SIP service system using existing network resources installed in almost all users without installing a separate line for the SIP service. You can get the effect that can build.

The application switch 133 is connected to the Internet network 120 through a plurality of lines. The plurality of lines may be connected to the application switch 133 via each ADSL modem 131 and 132. In FIG. 1, a case where two lines are connected to an application switch 133 via two ADSL modems 131 and 132 is taken as an example.

The application switch 134 is connected to an internal network including one or more communication terminals 135 and 136 through the L2 switch 134. The L2 switch 134 performs a switching function for the SIP call received by the application switch 133 through each ADSL modem 131 and 132. That is, the L2 switch 134 may control the SIP call to be transmitted to a specific communication terminal by referring to the address information of the SIP call. The L2 switch 134 may be implemented as an independent switch including an L3 switch, or may be included in the internal configuration of the application switch 133.

In summary, the application switch 133 may be installed in a customer receiving the SIP service according to the present invention, and the application switch 133 is connected to a plurality of lines drawn from the outside. In addition, one or more communication terminals 135 and 136 may be installed in the customer, and each communication terminal 135 and 136 may be connected to the application switch 133 through a predetermined internal communication network, respectively. The internal communication network may include both a wired communication network and a wireless communication network.

That is, as described above, the application switch 133 performs SIP communication with the SIP server 110 through the plurality of external lines, and performs SIP call connection with one or more communication terminals 135 and 136 through the internal communication network. do. Ports of the application switch 133 connected to the plurality of external lines may be assigned a global IP address. In addition, one or more communication terminals 135 and 136 connected to the internal communication network may be given a private IP, respectively. Accordingly, the application switch 133 may control the SIP call connection by translating the public IP address and the private IP address. This will be described in more detail with reference to the configuration and operation of the application switch according to FIG. 2.

2 is a block diagram illustrating a configuration of an application switch according to an embodiment of the present invention.

Application switch 200 according to an embodiment of the present invention is the first communication port unit 210, memory means 220, SIP connection control unit 230, failure detection unit 240, address update unit 250, And a second communication port unit 260.

The first communication port unit 210 is connected to an external network including a SIP server through a plurality of lines. That is, the first communication port unit may include a plurality of ports 211 to 214, and each port 211 to 214 may be connected to the plurality of lines, respectively. In the present specification, for convenience of description, a case in which the first communication port unit 210 includes four ports 211 to 214 and is connected to four lines through respective ports 211 to 214 is described. An example will be described. In addition, in the present specification, the case in which the first communication port unit 210 is connected to an external server or terminal through a wired network is described, but the first communication port unit 210 is connected to an external server or terminal through a wireless communication network. It is obvious that when connected, it can also be implemented through the present invention as another embodiment.

Each port 211 to 214 included in the first communication port unit 210 is provided with a global IP address. The public IP address refers to a public IP address capable of performing SIP communication with an external server or terminal through the line connected to each of the ports 211 to 214.

The second communication port unit 260 is connected to an internal communication network including one or more communication terminals 271 to 274. That is, the second communication port unit 260 may be connected to one or more communication terminals 271 to 274 installed inside the customer where the application switch 200 is installed through a wired or wireless communication network. The internal communication network is not only a wired network for general SIP call connection, but also a wireless LAN (WLAN), Bluetooth, Ultra Wide Band (UWB), Infrared Data Association (IrDA), Home Phoneline Networking Alliance (HPNA), and SWAP (Shared). Wireless access protocol), a local area network such as IEEE1394, or the like.

Private IP addresses may be assigned to the second communication port unit 260 and each of the communication terminals 271 to 274 connected through the internal communication network. The private IP address means an IP address that is allowed to be used only in the internal communication network of the customer where the application switch 200 is installed. That is, the private IP address is exposed only through the internal communication network and cannot be connected to the external communication network through the private IP address. This can be understood similarly to the concept of an extension number assigned to each telephone inside the enterprise through a private branch exchange (PBX) in a general telephone service for a large subscriber such as an enterprise.

The memory means 220 maintains a network address translation (NAT) table in which SIP addresses, private IP addresses, and public IP addresses corresponding to each of the one or more communication terminals 271 to 274 are recorded. The NAT table structure and the respective addresses will be described in detail with reference to FIG. 3.

3 is a diagram illustrating the structure of a NAT table according to an embodiment of the present invention.

In the NAT table 300 according to an embodiment of the present invention, a SIP address, a private IP address, corresponding to each of at least one communication terminal connected to the second communication port unit 260 of the application switch 200 through an internal communication network, And the public IP address are recorded. In FIG. 3, for convenience of description, the communication terminal is a first communication terminal 271, a second communication terminal 272, a third communication terminal 273, and a fourth communication terminal 274 as shown in FIG. 2. A case where it is implemented with four communication terminals including a will be described as an example.

 The SIP address can be set according to the SIP protocol protocol, a standard VoIP protocol published by the Internet Engineering Task Force (IETF). For example, as illustrated in FIG. 3, the SIP address corresponding to the first communication terminal 271 may be set to “sip: rooney@piolink.com” and the SIP corresponding to the second communication terminal 272. The address corresponds to "sip: lena@piolink.com", the SIP address corresponding to the third communication terminal 273 is "sip: park@piolink.com", and the SIP address corresponding to the fourth communication terminal 274 is "sip". : ygs@piolink.com ". The SIP address may be set by the user selecting and inputting through the SIP server.

In addition to the SIP address, the NAT table 300 records private IP addresses and public IP addresses corresponding to the respective communication terminals. As described above, the private IP address is an IP address used only in a customer's internal communication network in which the application switch 200 is installed. Therefore, one private IP address may be assigned to each communication terminal and may not be used as an IP address for an external communication network except for the internal communication network. In addition, the public IP address is an IP address used by the application switch 200 to connect to an external communication network through a plurality of lines. One public IP address may be assigned to each of the plurality of circuits.

According to the NAT table 300 according to the embodiment of the present invention shown in FIG. 3, four communication terminals may be installed in the customer where the application switch 200 is installed, and thus four private IP addresses may be assigned. In addition, the application switch 200 may be connected to the external Internet network through a total of three lines. The three lines may be connected to port 1 211, port 2 212, and port 3 213 of the application switch 200, respectively. Accordingly, a public IP address of "211.19.0.1" may be assigned to port 1 211, a public IP address of "211.19.0.2" to port 2 212 and a "211.19.0.3" to port 3 213. Can be given.

In this case, the first communication terminal 271 and the third communication terminal 273 may be assigned a "211.19.0.1" public IP address of the port 1 211, and the second communication terminal 271 may be provided with the port 2 ( "211.19.0.2" public IP address of 211) may be assigned, and the "211.19.0.3" public IP address of port 4 (214) may be assigned to the fourth communication terminal 274. That is, each communication terminal may set a port of the first communication port unit 210 to which a line used to connect to an external SIP server or terminal and a public IP address corresponding thereto are connected.

For example, when the application switch 200 receives a request for a predetermined SIP call connection from the first communication terminal 271, the application switch 200 is "123.11.0.1, which is a private IP address of the first communication terminal 271. "To the public IP address" 211.19.0.1. "To request the SIP call connection to the external SIP server. That is, the SIP call connection of the first communication terminal 271 may be implemented through a line connected to the port 1 211 of the first communication port unit 210 of the application switch 200. In addition, the SIP call connection of the second communication terminal 272 may be implemented through a line connected to the port 2 (212) of the first communication port unit 210, the SIP call connection of the third communication terminal (273) It may be implemented through a line connected to the port 1 (211). As such, by controlling the SIP call connection of the communication terminal by converting the public IP address and the private IP address to each other, it is possible to obtain an economic effect of making the most of the limited network resources at the minimum cost.

In FIG. 2, the SIP access controller 230 converts the private IP address and the public IP address assigned to each of the communication terminals 271 to 274 with reference to the NAT table 300 to convert SIP calls through an external SIP server. Control the connection. That is, as described above with reference to FIG. 3, when the SIP access controller 230 receives a SIP call connection from the internal communication terminals 271 to 274, private IPs corresponding to the respective communication terminals 271 to 274 are provided. The SIP call connection is controlled by converting an address into a public IP address.

In addition, even when a SIP call connection is requested through the first communication port unit 210 from the SIP server, the SIP connection controller 230 converts the public IP address into a private IP address by referring to the SIP address of the requested SIP call. The SIP call may be transmitted to a communication terminal corresponding to the private IP address.

When the SIP access control unit 230 receives the SIP packet from the communication terminals 271 to 274 through the second communication port unit 260, the SIP access control unit 230 converts the source IP of the SIP packet from the private IP address to the public IP address. can do. In addition, the SIP access controller 230 converts the destination IP of the SIP packet from the public IP address to the private IP address when receiving the SIP packet from the external SIP server through the first communication port unit 210. can do.

In addition, when receiving the SIP message from the communication terminals 271 to 274 through the second communication port unit 260, the SIP access controller 230 may convert the private IP address of the SIP message into the public IP address. have. In addition, the SIP access controller 230 may convert the public IP address of the SIP message received through the first communication port 210 from an external SIP server into the private IP address.

On the other hand, the failure detection unit 240 detects the failure of a plurality of lines connected to the first communication port unit 210. That is, the failure detecting unit 240 may detect whether or not a failure occurs for each of the lines by checking the reception state of the signal flowing from each of the plurality of lines. The line failure detection operation of the failure detection unit 240 may be implemented including all the communication line status check methods of various methods generally used.

The address update unit 250, when a failure of the first line of the plurality of lines is detected, of the first communication port unit 210 to which the first line is connected among the public IP addresses recorded in the NAT table. The first public IP address assigned to the port 1 211 is updated with the second public IP address assigned to the port 2 212 of the first communication port unit 210 in which the failure is not detected. That is, when a failure is detected from the first line connected to the port 1 211 by the failure detection unit 240 and fails to transmit or receive a SIP call normally through the first public IP address assigned to the port 1 211, the address is updated. The unit 250 updates the first public IP address with a second public IP address assigned to the port 2 212 in which the failure does not occur, and records it in the NAT table.

For example, referring to FIG. 3, the address update unit 250 replaces the public IP address “211.19.0.1” corresponding to the port 1 211 with the public IP address “211.19. It can be updated to 0.2 "and recorded in the NAT table 300. Therefore, the public IP addresses corresponding to the first communication terminal 271 and the third communication terminal 273 may be updated to "211.19.0.2".

As described above, when the public IP address is updated, the SIP connection controller 230 determines that the SIP call connection corresponding to the first communication terminal 271 and the third communication terminal 273 is the updated second public IP address. Can be controlled via 211.19.0.2 ". That is, the SIP access controller 230 controls the SIP call connection between the first communication terminal 271 and the third communication terminal 273 through the port 2 212 instead of the failed port 1 211.

Accordingly, even if a failure occurs on a specific line among the plurality of lines, the SIP access control unit 230 corresponds to the line where the failure does not occur by using the public IP address of the communication terminal that has received the public IP address corresponding to the failure. By updating to the public IP address and controlling through the line, it is possible to stably control the SIP call connection of all the communication terminals 271 to 271 regardless of the failure of a specific line.

In addition, the SIP access controller 230 transmits a register method including the updated second public IP address and the SIP address of the communication terminal from the NAT table to the SIP server. That is, as in the above example, the SIP connection controller 230 is configured to include the first communication terminal 271 and the first communication IP address of the first public IP address of port 1 211 to the second public IP address of port 2 212. 3, a register method including a SIP address corresponding to the communication terminal 273 and the updated second public IP address is generated and transmitted to the SIP server.

The SIP server may transmit a SIP call corresponding to the first communication terminal 271 and the third communication terminal 273 to be generated to the application switch 200 through the second public IP address according to the register method. Accordingly, the first communication terminal 271 and the third communication terminal 273, even if a failure occurs in the first line connected to the port 1 (211), SIP through the second line connected to the port 2 (212) from the SIP server Since the call is received, stable SIP service can be provided regardless of the failure.

In addition, the update of the NAT table may be implemented during the SIP call connection of the communication terminal. For example, while the first communication terminal 271 is performing a SIP call connection with an external server or terminal through the first public IP address assigned to the port 1 211, the failure detection unit 240 is a port. It can be detected that a failure occurs from 1 (211). At this time, the address updater 250 updates the first public IP address to the second public IP address of the port 2 212 in which the failure does not occur, and records it in the NAT table. As the NAT table is updated, the SIP access controller 230 may control the SIP call connection of the first communication terminal 271 to be replaced with the SIP call connection through the port 2212. Accordingly, even if the failure occurs during the SIP call connection, the first communication terminal 271 may continue to maintain the SIP call connection according to the replacement of the SIP call connection.

So far, the configuration of the application switch and the SIP service providing method according to the embodiment of the present invention have been described with reference to FIGS. 2 and 3. Hereinafter, a method of transmitting and receiving a signal according to a SIP call connection occurring between an actual communication terminal, an application, and a SIP server according to the operation of the application switch will be described with reference to FIGS. 4 and 5.

4 is a flowchart illustrating a method of processing a SIP call according to a case where a line failure occurs during setup / teardown operation of a SIP call according to an embodiment of the present invention.

In step 410, the communication terminal transmits an INVITE message through a private IP address assigned to the application switch for the SIP call connection with a predetermined external terminal. The application switch converts the private IP address of the communication terminal into a first public IP address (Global IP1) and transmits the INVITE message to a SIP server through a first line.

In step 420, the first line fails. In step 430, the SIP server transmits a 100 Tring message, a 180 Ringing message, and a 200 OK message to the application switch through the first public IP address in response to the INVITE message received from the application switch. However, since a failure occurs in the first line, the application switch cannot receive the 100 Tring message, the 180 Ringing message, and the 200 OK message through the first public IP address. Accordingly, the communication terminal retransmits the INVITE message to the application switch through the private IP address, and the application switch transmits the INVITE message back to the SIP server through the first public IP address. However, the SIP server cannot receive the INVITE message due to the failure of the first line.

In step 430, the application switch detects that a failure has occurred in the first line. In step 440, if the communication terminal sends the INVITE message back to the application switch via the private IP address, the application switch detects that the first line has failed, and thus removes the private IP address. 2 converts to a public IP address and transmits the INVITE message to the SIP server through the second line where the failure does not occur;

The SIP server normally receives the INVITE message from the application switch via the second public IP address. Accordingly, the SIP server transmits a 100 Tring message, a 180 Ringing message, and a 200 OK message to the application switch through the second public IP address in response to the INVITE message. In response, the application switch converts the second public IP address into a private IP address of the communication terminal and transmits the 100 Tring message, the 180 Ringing message, and the 200 OK message to the communication terminal.

The communication terminal returns an ACK through the private IP address to the application switch in response to receiving the 100 Tring message, the 180 Ringing message, and the 200 OK message. The application switch may control the SIP call connection of the communication terminal by converting the private IP address into the second public IP address and returning the ACK to the SIP server.

5 is a flowchart illustrating a method of processing a SIP call according to an embodiment of a circuit failure during an RTP media session according to an embodiment of the present invention.

The Realtime Transport Protocol (RTP) is a realtime media data transmission protocol for driving real-time media broadcasting or video conferencing information on the Internet or an intranet. That is, the method for processing a SIP call illustrated in FIG. 5 may be implemented as a method for processing a SIP call according to a case where a line failure occurs during a SIP call connection of a communication terminal.

In step 510, a failure occurs in the first line during a SIP call connection through the first public IP address of the communication terminal. In step 520, the SIP server continues to send an RTP message to the application switch via the first public IP address in accordance with the connection of the SIP call. However, since a failure occurs in the first line, the application switch cannot receive an RTP message transmitted by the SIP server through the first line. In addition, the SIP server cannot receive the RTP message transmitted by the communication terminal.

In operation 530, the application switch detects that a failure occurs in the first line because transmission and reception of the RTP message is impossible.

In step 540, the application switch detects the failure, and sends an INVITE message to the SIP server through the second public IP address of the second line where the failure does not occur. When the SIP server receives the INVITE message through the second public IP address, the SIP server transmits a 200 OK message to the application switch through the second public IP address. The communication terminal and the SIP server may maintain the SIP call connection through the second public IP address as the application switch returns an ACK to the SIP server through the second public IP address.

Accordingly, the application switch may convert the private IP address of the RTP message transmitted by the communication terminal through the second line into a second public IP address and transmit the converted IP address to the SIP server. The application switch may convert the second public IP address of the RTP message received from the SIP server through the second line into the private IP address and transmit the converted IP address to the communication terminal.

The method of processing a SIP call according to a case where a line failure occurs during the SIP call connection described with reference to FIG. 5 may be implemented similarly to the concept of hand off of a mobile communication system including CDMA.

6 is a flowchart illustrating a flow of a method for providing SIP service of an application switch according to an embodiment of the present invention.

The application switch according to an embodiment of the present invention maintains a connection with an external network including a SIP server through a plurality of lines (step 611). The plurality of lines may be connected to a first communication port unit including a plurality of first ports, and two or more public IP addresses may be assigned to the first communication port unit.

The application switch maintains a connection with an internal network including one or more communication terminals (step 612). In addition, the application switch maintains a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded (step 613). ).

The application switch detects a failure of the plurality of lines by detecting a failure of the first line of the plurality of lines (step 614). If a failure is detected on the first line, the application switch is configured to a first public IP address assigned to a first port of the first communication port unit to which the first line is connected among the public IP addresses recorded in the NAT table. Is updated to the second public IP address assigned to the second port of the first communication port unit in which no failure is detected (step 615).

Thereafter, the application switch transmits a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server with reference to the updated NAT table (step ( 616)). Accordingly, the application switch can reliably perform SIP call connection by connecting to the SIP server through the second port without using the failed first port.

7 is a flowchart illustrating a flow of a method for providing SIP service by an application switch according to another embodiment of the present invention.

According to another embodiment of the present invention, an application switch maintains a connection with an external network including a SIP server through a plurality of lines (step 711). The plurality of lines may be connected to a first communication port unit including a plurality of first ports, and two or more public IP addresses may be assigned to the first communication port unit.

The application switch maintains a connection with an internal network including one or more communication terminals (step 712). In addition, the application switch maintains a network address translation (NAT) table in which a SIP address, a private IP address, and a public IP address corresponding to each of the one or more communication terminals are recorded (step 713). ).

When the application switch receives a SIP call connection request from a first communication terminal among the one or more communication terminals, the application switch converts the private IP address given to the first communication terminal into the public IP address by referring to the NAT table. The SIP call connection through a SIP server is controlled through a first line of the plurality of lines (step 714).

The application switch detects the failure of the plurality of lines by detecting the failure of the first line of the plurality of lines (step 715). As described above, when a failure of the first line is detected, the application switch is configured to assign a first port to the first port of the first communication port unit to which the first line is connected among the public IP addresses recorded in the NAT table. 1 The public IP address is updated with a second public IP address assigned to the second port of the first communication port unit in which no failure is detected (step 716).

Thereafter, the application switch sends a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server by referring to the NAT table (step 717). ). In addition, the application switch controls the SIP call connection of the first communication terminal to be replaced with a SIP call connection through a second line connected with the second port (step 718).

The SIP service providing method of the application switch according to the present invention described with reference to FIGS. 6 and 7 may be implemented in the same manner including all operations according to the configuration of the application switch described with reference to FIGS. 2 to 5.

The SIP service providing method according to the present invention can be implemented in the form of program instructions that can be executed by various computer means and recorded in a computer readable medium. The computer readable medium may include program instructions, data files, data structures, etc. alone or in combination. Program instructions recorded on the media may be those specially designed and constructed for the purposes of the present invention, or they may be of the kind well-known and available to those having skill in the computer software arts. Examples of computer-readable recording media include magnetic media such as hard disks, floppy disks, and magnetic tape, optical media such as CD-ROMs, DVDs, and magnetic disks, such as floppy disks. Magneto-optical media, and hardware devices specifically configured to store and execute program instructions such as ROM, RAM, flash memory, and the like. The medium may be a transmission medium such as an optical or metal wire, a waveguide, or the like including a carrier wave for transmitting a signal specifying a program command, a data structure, or the like. Examples of program instructions include not only machine code generated by a compiler, but also high-level language code that can be executed by a computer using an interpreter or the like. The hardware device described above may be configured to operate as one or more software modules to perform the operations of the present invention, and vice versa.

While specific embodiments of the present invention have been described so far, various modifications are possible without departing from the scope of the present invention.

Therefore, the scope of the present invention should not be limited to the described embodiments, but should be defined not only by the claims below, but also by the equivalents of the claims.

According to the application switch of the present invention and the SIP service providing method according to the present invention, detecting a failure of a plurality of circuits connected to the external network including a SIP server, and the SIP call connection of the internal communication terminal assigned to the circuit where the failure occurs; By controlling to be performed through a circuit that does not have a failure, it is possible to obtain an effect of performing a seamless SIP call connection more stably and efficiently.

In addition, according to the application switch of the present invention and the SIP service providing method according to the present invention, when a failure occurs in the external line for performing the SIP call connection during the SIP call connection of the communication terminal, the SIP call connection of the communication terminal occurs By controlling to be replaced by a SIP call connection through a non-external line, it is possible to obtain an effect that can guarantee a stable SIP call connection to the user without interruption.

The present invention also provides a public IP address to each of a plurality of circuits connected to an external SIP server, and assigns a private IP address to an internal communication terminal, thereby converting the public IP address and the private IP address to each other. By performing a call connection, it is possible to economically utilize SIP network resources.

As described above, the present invention has been described by way of limited embodiments and drawings, but the present invention is not limited to the above-described embodiments, which can be variously modified and modified by those skilled in the art to which the present invention pertains. Modifications are possible. Accordingly, the spirit of the present invention should be understood only by the claims set forth below, and all equivalent or equivalent modifications thereof will belong to the scope of the present invention.

Claims (7)

In the application switch for controlling the Session Initiation Protocol (SIP) communication using a plurality of lines, A first communication port portion connected to an external network including a SIP server through the plurality of lines, wherein the first communication port portion is assigned two or more public IP addresses; A second communication port unit connected to an internal network including one or more communication terminals; Memory means for maintaining a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded; A SIP access control unit for controlling a SIP call connection through the SIP server by translating the private IP address and the public IP address assigned to each of the communication terminals with reference to the NAT table; A failure detecting unit detecting whether a failure occurs on the plurality of lines; And A first public IP assigned to a first port of the first communication port unit to which the first line is connected among the public IP addresses recorded in the NAT table when a failure of a first one of the plurality of lines is detected; An address updating unit for updating an address to a second public IP address assigned to a second port of the first communication port unit in which a failure is not detected Including, The SIP access controller transmits a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server by referring to the NAT table. . The method of claim 1, The SIP connection controller is configured to switch the SIP call connection to the SIP call connection through the second line when the communication terminal detects a failure of the first line while performing the SIP call connection through the first line. Application switch characterized in that the control to be replaced. The method of claim 1, The SIP access controller converts the source IP of the SIP packet received from the communication terminal through the second communication port unit from the private IP address to the public IP address with reference to the address table, and converts the first IP address from the SIP server. 1. The application switch of claim 1, wherein the destination IP of the SIP packet received through the communication port unit is converted from the public IP address to the private IP address. The method of claim 1, The SIP access controller converts the private IP address of the SIP message received from the communication terminal through the first communication port unit into the public IP address with reference to the address table, and converts the second communication port unit from the SIP server. SIP application switch for converting the public IP address of the SIP message received through the private IP address. In the SIP (Session Initiation Protocol) service providing method using a plurality of lines, Maintaining a connection with an external network including a SIP server through the plurality of lines, wherein the plurality of lines are connected to a first communication port portion including a plurality of first ports, wherein the plurality of lines are connected to the first communication port portion. More than one public IP address is given; Maintaining a connection with an internal network comprising one or more communication terminals; Maintaining a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded; Detecting whether the plurality of circuits are faulty; A first public IP assigned to a first port of the first communication port unit to which the first line is connected among the public IP addresses recorded in the NAT table when a failure of a first one of the plurality of lines is detected; Updating an address with a second public IP address assigned to a second port of the first communication port unit in which a failure is not detected; And Transmitting a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server by referring to the NAT table; SIP service providing method of the application switch comprising a. In the SIP (Session Initiation Protocol) service providing method using a plurality of lines, Maintaining a connection with an external network including a SIP server through the plurality of lines, wherein the plurality of lines are connected to a first communication port portion including a plurality of first ports, wherein the plurality of lines are connected to the first communication port portion. More than one public IP address is given; Maintaining a connection with an internal network comprising one or more communication terminals; Maintaining a network address translation (NAT) table in which a SIP address, a private IP address, and a global IP address corresponding to each of the one or more communication terminals are recorded; When receiving a SIP call connection request from a first communication terminal, converting the private IP address given to the first communication terminal into the public IP address by referring to the NAT table to perform the SIP call connection through the SIP server. Controlling over a first line of the plurality of lines; Detecting whether the plurality of circuits are faulty; When a failure on the first line is detected, the failure detects a first public IP address assigned to a first port of the first communication port to which the first line is connected among the public IP addresses recorded in the NAT table. Updating to a second public IP address assigned to a second port of the first communication port unit, which is not performed; Transmitting a register method including the updated second public IP address and the SIP address of the communication terminal to the SIP server by referring to the NAT table; And Controlling the SIP call connection of the first communication terminal to be replaced by a SIP call connection via a second line connected with the second port; SIP service providing method of the application switch comprising a. A computer-readable recording medium having recorded thereon a program for executing the method of any one of claims 5 and 6.

Images