KR20040072001A - The system and method for SIP message converting for linking up with PBX - Google Patents

Abstract

PURPOSE: A system and method for converting an SIP(Session Initiation Protocol) message for interworking with a private exchange are provided to process a call with many supplementary functions provided by a private exchange by exchanging various interface messages. CONSTITUTION: As an origination IP phone sends an Ack message to an SIP server, a speech path is opened(228), and the SIP server transmits a connection Ack_Ind message to an SIP driver(230). The SIP driver informs a private exchange that a destination IP phone and the origination IP phone are busy by transmitting polarity(232). When the call is terminated, the IP phone sends a bye message to the SIP server(234) and the SIP server sends a termination(Release_Ind) message to the SIP driver(236). When the SIP server receives an OK message from the origination IP phone(238), the SIP server sends an OK message to the SIP driver(240) and the SIP driver informs the private exchange that the call has been terminated through a hook-on(hon) message(242).

Description

System and method for SIP message converting for linking up with PBX}

The present invention relates to a call processing message converting system and a method of an internet protocol telephony exchange system for controlling an IP terminal as a legacy terminal by transmitting and receiving an IP protocol and user information. In particular, the present invention relates to the interworking between VoIP and a private exchange using the existing SIP protocol.

Today, due to the rapid dissemination of the Internet and the demand for various services, the IP network is dramatically developed in terms of performance and service, and accordingly, the demand for more various services is continuously emerging. For example, the transmission of voice signals using an IP network (or Voice over Internet Protocol). The voice signal transmission through the IP network has become a big role of the IP network along with the data transmission, and the demand for various voice signal transmission functions has increased accordingly.

In accordance with the requirements described above, the integration request between the legacy legacy telecommunications (Vegacy Telecommunication) and VoIP has been further researched and developed for interworking with the communication network. For example, IP-based private exchanges such as Internet Protocol PBX (IP-PBX) and IP-Centrex are typical. Therefore, a telephone terminal that can be used in an IP network has the same form as a telephone terminal of an existing public switching telephone network (PSTN) and can be operated in the same manner to meet various requirements.

Currently, the IP-phone protocol that has been generalized is a media gateway control protocol used between an H.323 or media gateway recommended by ITU-T and a media gateway controller for controlling it. : Session Initiation Protocol (SIP), a multimedia communication standard that supports MGCP) and data, voice and video, is being used.

Typically, the IP network and the PSTN are separated from each other, and terminals accessible to each network are treated differently with different protocols and characteristics. Terminals connected to the PSTN generally include analog telephones, digital telephones, and modems as subscriber terminals. Terminals in the trunk connection portion include E1 / T1, primary rate interface (PRI), and loop. ) And number 7 signaling. IP terminals for IP network access include H.323 terminals, MGCP terminals, and SIP terminals. In order to transmit and receive voice using an IP terminal and an IP network, an IP line must be connected through a LAN and a gate keeper.

SIP is a protocol designed to make it easier for multiple people to invite the other party to a conference or phone call, rather than to control the user when they are in a video conference or phone call. SIP is not tied to any protocol stack and is defined as text based such as HTTP, so it is easy to extend and easy to use. SIP uses a conventional E-MAIL address and provides mobility through location-specific messages, making it a simple protocol for receiving calls from a designated location.

On the other hand, in order for the IP network and the PSTN networks to interoperate with each other, the subscribers of the IP network and the PSTN must have a gateway in some form, and in order to occupy these gateways, an independent system configuration for occupying between networks is required. Should be. This means that the IP network and the PSTNs have independent number systems and terminal management systems.

When a legacy legacy terminal accommodated in the PSTN calls another generic legacy terminal, the preset prefix is dialed according to the location (domestic, long distance / oversea, wired / wireless) of the called party's terminal. Try to call. However, if the other party to call is an IP terminal, the call must be dialed with a prefix that occupies the gateway connected to the corresponding exchange.

IP terminals connected to the Internet also dial a preset prefix according to the other party's location (local or other region), and when attempting a call to the PSTN, occupies a gateway connected to the switching system controlling the corresponding terminal. You must dial the prefix to do this.

However, since the call control of the PSTN terminal and the IP network terminal is a system independent of each other, the conventional technology as described above must exist in two systems to form a voice call path between the IP terminal and the legacy terminal. There is discomfort. Therefore, independent functions may be performed on the interworking of the two systems, and integrated functions between the networks and terminal management systems are impossible. This problem causes considerable difficulty in the integration of IP terminals in the area using the existing PSTN network or in the same building. In other words, when the private exchange system based on IP was implemented, the private exchange could not handle the IP terminal as the subscriber, so many extensions provided by the private exchange could not be used.

Therefore, an object of the present invention is to control the private exchange by sending and receiving various interface messages through the SIP driver of the IP phone registered as the extension of the private exchange, so that VoIP call processing as well as many additional functions provided by the private exchange. In order to be able to perform processing, it is to provide interworking between VoIP and private exchange using SIP protocol.

1 is a system block diagram in accordance with the present invention.

2 is a call processing message converting control flowchart of the present invention.

***** Explanation of symbols for main parts of drawing *****

10: PSTN 12: Internet

14 private exchange 16 legacy terminal

18: IP terminal 20: switching hub

22: SIP server 24: database

26: gateway 28: database

30: exchange matching unit 32: call processing module

DETAILED DESCRIPTION In the following detailed description of preferred embodiments of the present invention, when it is determined that detailed descriptions of related well-known functions or configurations may unnecessarily obscure the subject matter of the present invention, detailed descriptions thereof will be omitted.

1 is a system block diagram according to the present invention. Referring to FIG. 1, a private exchange 14 which basically switches to a call of at least one legacy terminal 16 is connected to a PSTN 10 via a trunk line trunk card (not shown).

Although the embodiments of the present invention are not described in detail, such trunk line trunk cards are well known in the art for interfacing E1 / T1, PRI / BRI / No.7 signaling, and the like. It is also very well known to connect by incoming / outgoing call of at least one legacy terminal, for example an analog telephone, a digital telephone or the like. Therefore, embodiments of the present invention will be described with reference to a call connection service between a legacy terminal and an IP terminal, a call connection service between IP terminals, and an IP trunk (VoIP connection through a gateway) from the legacy terminal / IP terminal.

The private exchange system according to the embodiment of the present invention is largely connected to the PSTN 10 and performs the general call processing service of the legacy terminals 16 and the IP terminal 18 connected to the IP network 12. ) And a SIP server 22.

The private exchange 14 has a gateway card 26 for interconnecting different types of communication networks. The private exchange 14 and the gateway card 26 located therein, the SIP server 22 and the plurality of IP terminals 18 are connected to the IP network 12 to switch the plurality of LAN ports. It is connected to the LAN terminal port of the hub 20. This hub 20 uses a conventional one.

The private exchange 14 further includes a function of executing IP terminals 18 connected to the IP network 12 in addition to the call processing services of the conventional legacy terminals 16 in the same manner as the call processing of the extension subscribers. This operation will be easily understood by the following description.

The private exchange 14 is a phone number, port number, IP address of the IP terminals 18 connected to the IP network 12 through the hub 20 and the phone number of the legacy terminals accommodated therein A plurality of IP terminals connected through the hub 20 and the gateway 26 for forming a basic call channel between the database 28 storing the subscriber information such as the network 12 and the IP network 12 and the PSTN network 10. An exchange interface 30 for receiving call information transmitted from the mobile station 18 or for receiving call information to the IP terminal, and a call processing module 32 for controlling a call service between the legacy terminal 16 and the IP terminal 18. ). The switching interface 30 and the gateway 26 include a LAN to enable LAN communication through the switching hub 20.

The IP terminals 18 generate and transmit or receive user information such as hook state information, digits, and various tone messages including ID identification codes therein, and generate various tones. It is connected to the interface and the IP network has a communication module that performs the VoIP function according to the SIP protocol procedure (procedure). In this case, the user interface is used to generate various tones in response to a message or a command provided from a private exchange or to drive a display device. It means to generate various digit tones corresponding to. This user information is interfaced with the private exchange 14 via the LAN.

As described above, the IP terminals 18 exchange user information on the IP identification code, for example, a telephone number and a port number, which are provided from the private exchange 14 at the time of hooking off, through the LAN. Provided by 30. User information data directly transmitted and received with the private exchange 14 through the exchange interface 30 includes the ID identification code, dial on / off message, digit information, and various information for controlling the terminal.

The SIP server 22 connected to the private exchange 14 and the IP terminals 18 through the hub 20 is configured to provide the IP terminals in response to IP terminal access information and call processing information based on the SIP protocol procedure. Control the IP endpoint that is located.

In the private exchange system of the present invention configured as described above, the telephone number and various subscriber information are assigned to the legacy terminals 16 to each of the plurality of IP terminals 18 connected to the IP network 12 through the hub 20. To make it easy to add / change / delete subscriber information. For example, a maintenance terminal (not shown) is used to load and manage information such as a telephone number, a port number, and an IP address corresponding to each of the IP terminals 18 on the database 28. This maintenance is as usual. Therefore, the present invention configured as shown in FIG. 1 can provide a function that can be maintained in one integrated system in a manner maintained in two existing independent systems. That is, it is processed as a terminal in the same system regardless of whether it is an IP terminal or a legacy terminal.

As described above, in the present invention, various SIP-related messages sent from the IP terminal 18 to the SIP server 22 are converted into necessary messages of the private exchange through a SIP driver, which is a separate module implemented in the SIP server, and then the private exchange. The converted message is sent to (14) to act as an extension of the private exchange 14. This not only enables the legacy phone 16 and the IP phone 18 to be interoperable, but also enables the implementation of many additional functions provided by the existing private exchange 14. That is, the IP phone 18 registered as the extension of the private exchange 14 can be controlled by the private exchange 14 by exchanging various interface messages through the SIP driver 22, thereby adding the VoIP self call processing as well as adding the private exchange. The call processing of the function can be performed.

As described above, the IP phones 18 registered in the DB 28 of the private exchange have all the items owned by the extension subscribers of the private exchange 14, such as port numbers and telephone numbers, respectively. At this time, when the SIP server 22 is initialized, all the subscriber items are downloaded to the SQL DB 24 linked with the SIP server 22. Thereafter, the call can be made by connecting the IP phone 18 registered in the SIP server 22. In this case, the IP phone 18 should be able to input the self_IP of the IP phone, the IP of the SIP server 22, and the IP of the private exchange that can interface with the private exchange.

The present invention refers to various necessary information such as port number and telephone number of the corresponding IP phone 18 stored in SQL DB 24 when transmitting SIP message according to IP phone operation. It is characterized by converting the generated message into a message that can be processed by the exchange. The converted message is transmitted to the private exchange as a TCP / IP socket message, while the message coming down from the private exchange is converted into a SIP message and sent to the SIP server, thereby enabling the message exchange according to the present invention.

2 is a call processing message converting control flowchart according to an embodiment of the present invention, which shows call connection services between IP terminals.

First, when any one of the IP terminals 18 shown in FIG. 1 hooks off to call a neighboring IP terminal 18 and inputs a digit, the originating IP terminal 18 determines the identification ID and the digit. User information including the (digit) information is transmitted to the call processing module 32 through the exchange interface 30. In the above, the user information refers to a status message such as a telephone number assigned to the IP terminal 18, a port number of the private exchange 14, a hook off / on, and a bell of the IP terminal 18 at the private exchange 14 or the like. It includes a plurality of control messages for driving various indicators and the like. In addition, among the user information, the own port number, telephone number, etc. may be downloaded and stored when the system is initialized.

The originating IP terminal 18, which is sent as described above, transmits the INVITE message to the SIP server 22 in step 202 of FIG. At this time, the information displayed in the INVITE message includes a port number and digits (phone number of the other party) according to the user's selection of a key button located in the IP terminal 18. Meanwhile, the SIP driver is a separate module logically implemented in the SIP server 22 and is divided into an outgoing call processor CLG and an incoming call processor CLD.

In step 204, the SIP server 22 receiving the INVITE message transmitted from the source IP terminal 18 transmits the INVITE request message (INVITE req) to the source call processor (CLG) inside the SIP driver. At this time, the outgoing call processing unit CLG in the idle state transmits the received INVITE request message INVITE req to the call processing module 30 through the exchange interface in the private exchange 14 in step 206. The originating call processing section of the call processing module 32 in the private exchange 14 analyzes the INVITE request message (INVITE req) and performs the Hof processing of the corresponding IP terminal 18 and includes it in the INVITE request message (INVITE_req). The digit is processed and the seizure message is transmitted to the incoming call processor in the private exchange 14.

The incoming call processor in the private exchange 14 occupies the call in response to the reception of the occupied message, and transmits the occupancy complete message [PBX Seizure] to the incoming call processor (CLD) in the SIP driver in step 208 of FIG. By the SIP message conversion as described above, the private exchange 14 can know which of the plurality of IP terminals has called and the called IP terminal. The incoming call processing unit in the private exchange 14 performing the step 208 continues to transmit a ring-on message to the destination IP terminal through the switching interface 30. The receiving IP terminal receiving the ring-on message informs that the incoming call is driven by driving an internal ringer.

Meanwhile, the incoming call processor CLD in the SIP driver receiving the occupancy completion message [PBX Seizure] from the private exchange 14 sends the message [PBX Seizure] to the originating call processor CLG and enters the occupancy state (210). ). PBX Seizure is a message sent when the incoming call is normal. If the incoming call is not made, a Reject message is sent.

The originating call processor in the SIP driver that receives the occupancy completion message [PBX Seizure] sends an INVITE response signal (INVITE resp), which is a signal indicating an INVITE message success (INVITE_Success), to the SIP server 22 (212). The server 22 transitions to the CLG Proc state by sending an INVITE message to the destination IP phone (214).

Thereafter, the IP phone of the called party sends a ringing message to the SIP server 22 (216), and the SIP server 22 sends an alert message (Alerting_ind) to the SIP driver (218). Upon receiving this message, the SIP driver informs the private exchange 14 that the incoming IP phone is in a ringing state with a system identification information (sys_id) message (220), and indicates that the calling IP phone is in the ring back state. The private exchange 14 is informed of the identification information (sys_id) message (222). On the other hand, the ringing state refers to a message that the called terminal is ringing.

At this time, the outgoing call processor CLG and the incoming call processor CLD in the SIP driver transition to the CLG alert state and the CLD alert state, respectively. Alert state means the ring and ringback state.

After receiving the OK message from the destination IP phone (224), the SIP server sends a Connect (Connect_ind) message to the SIP driver (226). At this time, the SIP driver only transitions its own state so that the outgoing call processor transitions to the ringback state (RBACK), and the incoming call processor transitions to the ring state (RING).

The actual call path is opened by sending an Ack message from the originating IP phone 18 to the SIP server (228), and the SIP server sends a connection Ack_Ind message to the driver (230). The SIP driver notifies the private exchange 14 by transmitting polarity that the incoming / outgoing IP phone is busy (232). Polarity is a signal that is the reference for charging by reversing the polarity of the signal. When the call is terminated, the ending IP phone sends a Bye message to the SIP server (234), and the SIP server sends a Release_Ind message to the SIP driver (236). At this time, the SIP driver 22 transitions its state from the connection state CONNECT to the termination state RLS. When the SIP server receives an OK message from the IP phone 18 (238), the SIP server sends an OK message to the SIP driver (240), and the driver receives the hook-on message indicating that the call has ended to the private exchange (14). Notified via (242). After termination, the state returns to the idle state.

As described above, in the present invention, the Invite message is processed in the existing SIP server to the SIP driver to allow the SIP driver to interoperate with the private exchange to allow and restrict the call. In addition, various additional messages from private exchanges can be processed, enabling the services of various additional functions provided by private exchanges.

The technical spirit of the present invention has been described above with reference to the accompanying drawings, but this is by way of example only and not intended to limit the present invention. In addition, it is obvious that any person skilled in the art can make various modifications and imitations without departing from the scope of the technical idea of the present invention.

According to the present invention, by using the SIP driver, the limitation that the extension of the private exchange was not available due to the use of the gateway as the trunk line in the existing private exchange enables the IP phone to be treated the same as the existing extension. Many of the features of the private exchange that could not be provided by the API were implemented.

Claims (6)

In the call processing message converter system of the private exchange system, An IP terminal performing a VoIP function; A private exchange performing a call processing service in response to user information and call connection information generated from a terminal; A SIP server having a SIP driver for allowing the IP terminals and the private exchange system to interwork to perform admission and restriction on a call; Call processing message converter system comprising a. The call processing of claim 1, wherein the SIP server analyzes an INVITE message transmitted from an originating IP terminal and transmits port information and digit information of the IP terminal to a call processing module in the private exchange. And a call occupancy message transmitted from the module and user control information of the called IP terminal to the called terminal. 2. The call processing message converter system of claim 1, wherein the SIP driver transmits a system identifier to a call processing module in the private exchange in response to an alert message sent from the destination IP terminal. 2. The call processing message converter system of claim 1, wherein the SIP driver transmits a polarity message to a call processing module in the private exchange in response to a connection message being transmitted from the destination IP terminal. The call processing message converter system of claim 1, wherein the SIP driver transmits a call to the call processing module through a hook-on message in response to receiving an OK message from the IP terminal terminated at the end of the call. . A private exchange system comprising at least one IP terminal performing a VoIP function, a SIP server controlling an Internet protocol endpoint, a legacy terminal and a database storing information on the plurality of IP terminals, and a private exchange performing a call processing service. CLAIMS What is claimed is: 1. A method for converting call processing messages in a method; The SIP server transmits an INVITE request message to a call processing module in the private exchange in response to an input of an INVITE message including its port information transmitted from the originating IP terminal and digit information corresponding to the telephone number of the other party. Process; Transmitting, by the call processing module, an INVITE response message to the SIP server in response to receiving an occupation message corresponding to a digit included in the INVITE request message; In response to a ringing message sent from the called IP terminal to the SIP server, the SIP server sends the ringing message to the originating IP terminal and sends an alert message to the SIP driver, wherein the SIP driver sends the alert. Transmitting system identification information to the call processing module in response to the message; Transmitting an access Ack_ind message from the SIP server receiving the access message from the originating IP terminal to the SIP driver, and transmitting polarity from the SIP driver to the call processing module in response to the received access Ack_ind message; The OK message is transmitted from the SIP server receiving the OK message from the IP terminal terminated at the end of the call to the SIP driver, and the SIP driver receiving the OK message hooks on to the call processing module that the call is terminated. Informing through; Call processing message converting method comprising a.