KR20090018330A - Multi voip phone relay method and apparatus using a singleness id - Google Patents

Abstract

A multi VoIP phone relaying method and an apparatus thereof using a single identification by applying single VoIP ID to a relaying apparatus are provided to perform intercommunication between VoIP phones and to prevent confusion with different VoIP phone due to maintaining the connection state of the VoIP phone. A plurality of VoIP phones(100) supports a VoIP service. VoIP server(300) provides a VoIP service by connecting internet/ public switched telephone network/mobile radio communications network. A modem(500) transmits and receives the VoIP server and data. A relay equipment(700) relays the VoIP service provided from the VoIP server to the VoIP phone. An AP(Access Point) wirelessly transmits and receives data with VoIP phones. If a user uses the VoIP phone and presses a send button after inputting a phone number, the VoIP phone transmits on-hook signal including identifying information to the relay equipment.

Description

Multi VoIP phone relay method and apparatus using a singleness ID

The present invention relates to a VoIP (Voice of IP) service, and more particularly, a single VoIP ID can be given to a relay device relaying VoIP services to a plurality of VoIP phones to provide a VoIP service. The present invention relates to a VoIP phone relay method and apparatus.

The proliferation of personal computers and advances in communication technologies have made it easier for most people to use the Internet at home and at work. Users are provided with various communication and multimedia services as well as searching for various information through the Internet, and the representative communication service is VoIP.

VoIP is a communication technology that provides a voice call between computers through the Internet, and is a communication service technology that enables voice calls to be made in general telephone networks by converting voice data into Internet protocol data packets. The cable can be used by multiple people at the same time, expandable and much cheaper than traditional phones. The deployment can be as simple as a computer, such as a dial pad, or an IP phone or a dedicated router sold by a professional network equipment company. Hereinafter, a computer or IP phone used for such a VoIP service will be referred to collectively as a VoIP phone.

Each of the VoIP phones has a unique ID, that is, a telephone number is given by a provider providing VoIP service, and the VoIP phone can make a voice call like a conventional Public Swithced Telephone Network (PSTN) through the assigned telephone number.

Conventional VoIP phones have a one-to-one correspondence with telephone numbers given by VoIP service providers. In other words, in case of owning and using a VoIP phone with a unique telephone number for each individual, it is excellent as an alternative service of mobile communication service. However, many phones provided by PSTN, which are generally used in homes, It is not suitable to replace the home telephone because it does not provide the function of using a single phone number. In other words, there is a disadvantage in that a number of VoIP phones and the number of the number exists in the same home because each person has to purchase a VoIP phone and be given a phone number for each VoIP phone. In addition, each time a new VoIP phone is purchased, there is a disadvantage in that a service provider is given a phone number.

The present invention has been made to solve the above problems, and to provide a multi-VOIP phone relay method and system using a single ID that can simultaneously use a plurality of VoIP phones with only one unique ID in the VoIP service.

Furthermore, the present invention provides a method and system for relaying a multi-VOIP phone using a single ID, which can prevent confusion with other VoIP phones by maintaining a call state of a VoIP phone in a call.

Furthermore, it is to provide a multi-VOIP phone relay method and system using a single ID that can make internal calls between VoIP phones.

Multi-VOIP phone relay method using a single ID according to an aspect of the present invention described above is made in a relay device connected to a VoIP server for providing a VoIP (Voice of IP) service to relay VoIP services to a plurality of VoIP phones, A) receiving a call start signal including identification information from the VoIP phone in the standby state and connecting the corresponding VoIP phone directly to the VoIP server to process the call; and during a call, a call start output from another VoIP phone except the VoIP phone. B) ignoring the signal to maintain the call state.

In addition, the multi-VOIP phone relay apparatus using a single ID according to an aspect of the present invention is a phone identification information storage unit for storing a plurality of VoIP (Voice of IP) phone identification information, and phone identification information according to the registration request of the VoIP phone A phone state management module for storing identification information in a storage unit, a call state detection module for receiving and outputting a call signal including a call start or call end signal including identification information from a VoIP phone registered by the phone management module; According to the call signal output by the call state detection module, the VoIP phone is directly connected to or disconnected from the VoIP server providing the VoIP service, but the call processing ignores the call signal of the other VoIP phone and maintains the call state. It consists of modules.

According to a characteristic aspect of the present invention, a multi-VOIP phone relay method and apparatus using a single ID according to the present invention receives a call signal transmitted from a VoIP server in the case of an incoming call and transmits the incoming signal to all stored VoIP phones and starts a call. It directly connects the VoIP phone and the VoIP server that sent the signal for the first time.In the case of outgoing call, it receives the call ID and call start signal from the VoIP phone, sends the call signal to the VoIP server, and directly connects the VoIP phone and the VoIP server. To handle the call.

In addition, the multi-VOIP phone relay method and apparatus using a single ID according to a characteristic aspect of the present invention stores the call state in a call state flag when the VoIP phone and the service network is in a call, starting a call from another VoIP phone When the signal is received, the call state information of the call state flag is checked to maintain the call state by ignoring the call request signal when the call is in progress.

According to a further aspect of the present invention, a method and apparatus for multi-VOIP phone relay using a single ID according to the present invention receives an extension call signal including identification information of a received VoIP phone from a VoIP phone, thereby connecting a transmitting VoIP phone and a receiving VoIP phone. To provide an extension service.

In the multi-VoIP phone relay method and apparatus using a single ID according to the present invention, a unique ID provided by a VoIP server, that is, a VoIP device is stored and connected to a VoIP server, rather than being assigned to a VoIP phone. By processing the service to directly connect the VoIP phone selected by the user and the VoIP server, there is an advantage that can use a plurality of VoIP phones through a single phone number without having to receive a phone number for each VoIP phone.

In addition, the multi-VOIP phone relay method and apparatus using a single ID according to the present invention ignores the call request output from the other VoIP phone during the incoming or outgoing call, and transmits the call to the corresponding VoIP phone to notify that the call is currently As a result, it is possible to prevent cross talk with other VoIP phones.

In addition, the multi-VOIP phone relay method and apparatus using a single ID according to the present invention has an advantage that can easily make an internal call by connecting the transmitting VoIP phone and the receiving VoIP phone only by the extension number input.

The foregoing and further aspects of the present invention will become more apparent through the preferred embodiments described with reference to the accompanying drawings. Hereinafter, the present invention will be described in detail to enable those skilled in the art to easily understand and reproduce the present invention.

1 is a schematic diagram schematically showing a multi VoIP phone relay system using a single ID according to an embodiment of the present invention. As illustrated, the multi-VOIP phone relay system using a single ID according to the present invention is connected to a plurality of VoIP phones 100 supporting VoIP services and VoIP connected to the Internet / PSTN network / mobile communication network to provide VoIP services. The server 300, the modem 500 for transmitting and receiving data to and from the VoIP server, the relay device 700 for relaying the VoIP service provided from the VoIP server to the VoIP phone 100, the modem 500 and the relay device ( 700 is connected to the VoIP phone (100) is configured to include an AP (Access Point) 900 for transmitting and receiving data wirelessly.

The VoIP phone 100 may be, for example, a personal computer such as a desktop or laptop equipped with an application supporting VoIP service, a VoIP dedicated phone, a mobile terminal supporting VoIP service, and the relay device 700. It is selectively connected with the VoIP server 300 to provide a call service to the user. In order to use the VoIP phone 100, it is necessary to register its own identification information in the relay device 700. The identification information is assigned to the VoIP phone 100 by the unique serial number of the VoIP phone 100 or the relay device. It can be an address.

Using the VoIP phone 100 selected by the user, for example, when the user picks up a handset of a VoIP-only phone or enters a number like a mobile communication terminal or selects a phone book and then presses a call button, the VoIP phone 100 can The call start signal including the identification information is transmitted to the relay device 700, and the relay device directly connects the corresponding VoIP phone 100 and the VoIP server 300 with reference to the identification information to provide a VoIP service.

The VoIP server 300 is connected to the PSTN network, the Internet, and the mobile communication network to enable the VoIP phone 100 user to use services of the Internet, PSTN network, and mobile communication network. In the present invention, the VoIP server 300 is preferably implemented as a SIP (Session Initiation Protocol) server, hereinafter the VoIP server 300 to the SIP server 300 for ease of description according to an embodiment of the present invention It will be called.

SIP is an application layer signaling protocol in which intelligent terminals to communicate on the Internet identify each other, find their location, and specify a procedure for creating or deleting a multimedia communication session with each other. It is a request / response structure that controls the creation, modification, and termination of multimedia service sessions such as Internet-based conferencing, telephone, voice mail, event notification, and instant messaging.

SIP uses a SIP URL similar to an e-mail address to distinguish each user, so that the service is provided without being dependent on the IP address. It is text-based, developed using many parts of HTTP and SMTP intact, so it is easy to implement, and has the flexibility and extensibility to create various services in combination with many other protocols used on the Internet.

The SIP server 300 transmits a service request of the VoIP phone 100 directly connected through the relay apparatus 700 to, for example, a gateway connecting the Internet to the PSTN network or the mobile communication network using the SIP protocol. It provides multimedia services such as Internet-based conferencing, telephone, voice mail, event notification, instant messaging by communicating with various types of terminals belonging to the Internet, PSTN network and mobile communication network.

For example, the modem 500 may be an Ethernet modem installed for Internet use at home, and the AP 900 may be a separate modem 500 when an access port for Internet use is installed in a wall of a building or apartment. Can be connected directly to the Internet without When the modem 500 is installed, the modem 500 is connected to the SIP server 300 through the Internet, receives a service request transmitted from the VoIP phone 100 or the relay device 700, and transmits the service request to the SIP server 300. The service request signal such as, for example, a call signal transmitted from the SIP server 300 is received and transmitted to the relay apparatus 700.

The AP 900 is an access point for transmitting and receiving data wirelessly with the VoIP phone 100 registered in the relay device 700, and is installed between the modem 500 and the relay device 700. The modem 500 or the relay device 700 is connected by wire and connected to the Internet network. In addition, the relay apparatus 700 may be wirelessly connected to the AP 900. The AP 900 includes a time division multiplexer (TDM) and a data line termination device (DCE). On the other hand, AP 900 is required when the VoIP phone is implemented as a cordless phone for the convenience of the user, and if the VoIP phone 100 is a wired phone to be implemented to be directly connected to the relay device 700 without the AP 900. Can be.

In addition, the AP 900 allows the SIP server 300 and the VoIP phone 100 to be directly connected without passing through the relay device 700 according to a control signal of the relay device 700 through a routing function. The routing function of the AP 900 may be a network address translation (NAT) or network address port translation (Network Address Port Translation) technique that is already widely used in the wireless AP (900).

The relay device 700 selectively connects one of a plurality of registered VoIP phones 100 with the SIP server 300, so that the relay device 700 can receive the VoIP service provided by the SIP server 300. In order to receive a SIP service, a unique ID such as, for example, a phone number provided by a SIP service provider must be given. As described above, the SIP server 300 uses a SIP URL similar to an e-mail address as a unique ID to distinguish each user, and the SIP URL is matched with a phone number and can be easily used as a PSTN. Hereinafter, a unique ID will be referred to as a phone number for ease of explanation according to an embodiment of the present invention.

Description of such a relay device will be described in more detail with reference to FIG. 2 is a block diagram schematically illustrating a relay apparatus according to an exemplary embodiment of the present invention. As shown, the relay device 700 according to the present invention, the ID storage unit 710 that stores the phone number given from the SIP server 300, and requests the registration of the phone number to the SIP server 300 and SIP ID registration processing module 720 for receiving a phone number given by the operator of the server 300 and stores in the ID storage unit 710, the SIP server 300 and the AP connected to the Internet through the modem 500 ( A communication module 730 including a wired communication unit 731 for transmitting and receiving data to and from the 900 and a wireless communication unit 732 for transmitting and receiving data to and from the registered VoIP phone 100;

A phone identification information storage unit 740 for storing a plurality of VoIP phone identification information, a phone management module 750 for storing identification information in the phone identification information storage unit 740 according to a registration request of the VoIP phone 100; A call state detection module 760 for receiving and outputting a call signal including a call start or call end signal including identification information from the VoIP phone 100 registered by the phone management module 750 and call state detection; According to the call signal output by the module 760, the VoIP phone 100 is directly connected to or disconnected from the VoIP server 300 providing the VoIP service, and the call signal of the other VoIP phone 100 is disconnected during the connection. It is configured to include a call processing module 770 to ignore the connection state.

The ID storage unit 710 may be implemented as, for example, an EEPROM, a flash memory, or a hard disk capable of reading and writing. The ID storage unit 710 stores a phone number assigned by the SIP service provider by the ID registration processing unit 720.

When at least one VoIP phone 100 is registered by the phone management module 750, the ID registration processing module 720 is one of the VoIP phones 100 registered with the registration request of the VoIP service through the modem 500. The unique number of the VoIP phone 100 is transmitted to the SIP server 300. The SIP server 300 receives a registration request and a unique number of the VoIP phone 100, gives a phone number, and provides the ID to the given ID registration processing module 720. When the ID registration processing module 720 is given a phone number from the SIP server 300, the ID registration processing module 720 stores the corresponding phone number in the ID storage unit 710.

The communication module 730 is connected to the modem 500 and the AP 900 by wire and the wired communication unit 731 and the VoIP phone 100 that can be implemented as a wired communication modem 500 for transmitting and receiving data, for example It is configured to include a wireless communication unit 732 that can be implemented as a wireless communication modem 500 for transmitting and receiving data in a wireless communication protocol, such as RF (Radio Frequency), Bluetooth (Blue Tooth), ZigBee (ZigBee). The communication module 730 includes all of the software for driving the hardware in addition to the hardware for the wired and wireless communication described above.

The phone identification information storage unit 740 may also be implemented as, for example, EEPROM or flash memory capable of reading and writing, and stores identification information of the VoIP phone 100 registered by the phone management module 750. The identification information of the VoIP phone 100 may be a unique serial number of the VoIP phone 100 or a local address assigned to the VoIP phone 100 by the phone management module 750.

The phone management module 750 registers and manages the VoIP phone 100 by storing identification information of the corresponding VoIP phone 100 in the phone identification information storage unit 740 according to a registration request transmitted from the VoIP phone 100. To perform. When a user inputs a designated command to register the new VoIP phone 100 with the relay device 700, the VoIP phone 100 receives the command and manages the phone of the relay device 700 through the wireless communication unit 732. The module 750 sends a phone registration request including its serial number. The phone management module 750 receives the phone registration request including the serial number from the VoIP phone 100 and stores the serial number in the phone identification information storage unit 740 or assigns a separate local address to be used internally. The VoIP phone 100 is registered by storing the identification information storage unit 740.

In addition, the phone management module 750 receives the deletion request transmitted from the VoIP phone 100 and deletes the identification information of the corresponding VoIP phone 100 stored in the phone identification information storage unit 740. Deletion of 100) is processed.

The call state detection module 760 receives a call start signal including identification information transmitted from the registered VoIP phone 100 and transmits it to the call processing module 770. The call termination signal is received and output to the call processing module 770, thereby notifying that the call is terminated.

The call processing module 770 receives a call start signal output by the call state detection module 760 to directly connect the SIP server 300 and the corresponding VoIP phone 100 to provide a VoIP service, and to establish a VoIP connection. Receiving a call termination signal from the phone 100 and disconnects the VoIP phone 100 and the SIP server 300 to process the VoIP service termination.

The call processing module 770 receives the call signal transmitted through the VoIP server 300 and transmits the ring tone to all the VoIP phones 100 stored in the phone identification information storage unit 740, and the call state detection module ( An incoming call processing unit 771 that directly connects the VoIP phone 100 that has transmitted the call start signal first detected by the 760 and the VoIP server 300 to process an incoming call, and an outgoing call number from the call state detection module 760. Receiving a call start signal and transmits a call signal to the VoIP server 300, and directly connected to the VoIP phone 100 and the VoIP server 300, characterized in that it comprises a call processing unit 772 for processing the call. .

When the call signal is received from the SIP server 300 through the modem 500, the call processing unit 771 transmits the call signal to all the VoIP phones 100 stored in the phone identification information storage unit 740. When the user checks the ringtone ringing from all the VoIP phones 100 and picks up the handset of one of the VoIP phones 100, the VoIP phone 100 transmits a call start signal including its own identification information to a call state detection module ( 760). The call state detection module 760 checks the VoIP phone 100 that transmitted the initial call start signal and outputs the received call processor 771. The incoming call processing unit 771 checks the VoIP phone 100 transmitting the call start signal transmitted from the call state detection module 760 through the identification information, so that the VoIP phone 100 and the SIP server 300 are directly connected. The control signal is output to the AP 900 as much as possible. The AP 900 is connected to the VoIP phone 100 and the SIP server 300 according to the control signal to make an incoming call.

When the call is terminated and the user puts down the handset of the VoIP phone 100 in a call, the VoIP phone 100 outputs a call termination signal including identification information to the state detection module, and the call state detection module 760 It receives this and outputs it to the incoming call processing unit 771. The incoming processing unit 771 outputs a control signal for releasing the connection between the VoIP phone 100 and the SIP server 300 to the AP 900 according to the call termination signal, and the AP 900 outputs the VoIP phone according to the control signal. 100) and the SIP server 300 is disconnected to handle the termination of the incoming call.

The outgoing processor 772 processes the outgoing call by allowing the VoIP phone 100 and the SIP server 300 to be directly connected to the outgoing call of the VoIP phone 100. When the user selects one of the plurality of VoIP phones 100 and picks up the handset and inputs a calling number, the VoIP phone 100 receives a call start signal including a calling number and identification information. To send). Accordingly, the call state detection module 760 transmits a call start signal to the call processor 772, and the call processor 772 outputs a call signal including the corresponding call phone number to the SIP server 300. When the outgoing processor 772 outputs the call signal, the VoIP phone 100 and the SIP server 300 are directly connected to the AP 900 to output a control signal to make an outgoing call. The AP 900 is connected to the VoIP phone 100 and the SIP server 300 to make an outgoing call.

When the call is terminated and the user puts down the handset of the VoIP phone 100 in a call, the corresponding VoIP phone 100 outputs a call termination signal including the identification information to the call state detection module 760 and the call state detection module. 760 receives this and outputs it to the originating processor 772. The call processor 772 outputs a control signal for disconnecting the VoIP phone 100 and the SIP server 300 to the AP 900 according to the call termination signal, and the AP 900 calls the VoIP phone according to the control signal. The connection between the 100 and the SIP server 300 is released to process outgoing call termination.

Therefore, the relay apparatus 700 according to the present invention stores the own ID provided by the SIP server 300, that is, the phone number, rather than giving the VoIP phone 100, and stores the connection with the SIP server 300. The VoIP service 100 is directly connected to the selected VoIP phone 100 and the SIP server 300 by processing the provided VoIP service, and a plurality of VoIP numbers are provided through a single phone number without having to receive a phone number for each VoIP phone 100. Has the advantage of using the phone 100.

According to an additional aspect of the present invention, the call processing module 770 according to the present invention ignores a call start signal according to an origination request from another VoIP phone 100 during a call state, thereby making a call of the VoIP phone 100 currently in a call. Keep your status. Accordingly, the call processing module 770 according to the present invention includes a call state flag 773 for storing call states of the VoIP phone 100 and the VoIP server 300, and a call start transmitted from the call state detection module 760. Receiving a signal to check the call state information of the call state flag 773, and if the call includes a call state maintaining unit 774 to maintain the call state by ignoring the call start signal of the corresponding VoIP phone (100).

The call state flag 773 stores the call state when the VoIP phone 100 and the SIP server 300 are connected and connected by the call processing unit 771 or the call processing unit 772. For example, when the VoIP phone 100 and the SIP server 300 are connected to the call processing unit 771 or the call processing unit 772 and the call is started, the call state flag 773 stores '1'. When the call is terminated again and the connection between the VoIP phone 100 and the SIP server 300 is released, '0' is stored in the call state flag 773, so that the current call state can be confirmed.

The call state maintaining unit 774 disregards the outgoing request transmitted from the VoIP phone 100 other than the VoIP phone 100 currently in the call to maintain the current call state. Accordingly, when the call start signal is transmitted from the call state detection module 760, the call state maintaining unit 774 checks call state information stored in the call state flag 773. When the call state is stored as '0', the call start signal is transmitted to the call processor 772 so that the call request of the corresponding VoIP phone 100 can be processed.

Meanwhile, when the call state information is stored as '1', it is determined that the current call is in progress and ignores the call start signal transmitted to the call state detection module 760 to maintain the current call state. In addition, the call state maintaining unit 774 disregards the call start signal and maintains the current call state, and at the same time, for example, sends a message, guide voice, and a graphic signal to the VoIP phone 100 that has requested the call. Notify via Accordingly, the user of the VoIP phone 100 can know that another VoIP phone 100 is currently in a call.

According to a further aspect of the present invention, the relay device 700 according to the present invention provides an extension call service between registered VoIP phones 100. Accordingly, the relay device 700 receives the extension call signal including the identification information of the receiving VoIP phone 100 to which the call is desired from the VoIP phone 100 and connects the transmitting VoIP phone 100 to the receiving VoIP phone 100. It further comprises an extension processing module 780 for processing an extension call between the VoIP phone (100).

When the user picks up the handset of the VoIP phone 100 and identifies the identification information of the VoIP phone 100 that the user wants to make an extension call with, for example, an extension number, the VoIP phone 100 transmits the extension number to the call state detection module 760. Outputs an extension call request signal comprising a. The call state detection module 760 receives the corresponding extension call request signal and outputs it to the extension processing module 780. The extension processing module 780 transmits the ring tone to the VoIP phone 100 corresponding to the extension number through the extension number included in the extension call request signal, and the VoIP phone 100 and the receiving VoIP phone 100 requesting the extension call. ) Outputs a control signal to the AP 900 to be connected directly. Accordingly, the AP 900 establishes a connection between the two VoIP phones 100 to make an extension call. On the other hand, when an extension call termination signal is transmitted from the call state detection module 760 from one VoIP phone 100 during an extension call, the extension processing module 780 receives the extension call termination signal and extends the call to the AP 900. The control signal is output to disconnect the two VoIP phones 100 in a call. Accordingly, the AP 900 processes the extension call termination by releasing the connection between the two VoIP phones 100.

Hereinafter, a multi-VOIP phone 100 relaying process using a single ID made in the above-described relay device 700 will be described. Accordingly, the multi-VOIP phone relay method using a single ID according to the present invention is connected to a VoIP server 300 that provides a VoIP (Voice of IP) service and relays a plurality of VoIP phones 100 to a VoIP service. In step (a) of receiving a call start signal including identification information from the VoIP phone 100 in a standby state and directly connecting the VoIP phone 100 with the VoIP server 300 to process a call, and during a call And b) maintaining a call state by ignoring a call start signal output from another VoIP phone 100 except for the VoIP phone 100.

3 is a flowchart schematically illustrating a process of registering and deleting a VoIP phone according to an exemplary embodiment of the present invention. As shown, the ID registration processing module 720 of the relay device 700 according to the present invention, if at least one VoIP phone 100 is registered by the phone management module 750, SIP through the modem 500 For example, the registration request including the serial number of the VoIP phone 100 registered with the server 300 is transmitted (S101). Accordingly, the SIP server 300 processes registration by assigning a phone number to the VoIP phone 100 according to the registration request, and transmits the assigned phone number to the relay device 700 through the modem 500. The relay device 700 receives the transmitted phone number and stores the ID in the ID storage unit 710 to complete registration of the SIP service (S103).

On the other hand, when a user inputs a designated command to register the newly added VoIP phone 100 to the relay device 700, the VoIP phone 100 transmits a phone registration request including its serial number (S105). . Accordingly, the phone management module 750 of the relay device 700 stores the serial number transmitted according to the phone registration request as phone identification information as identification information of the corresponding VoIP phone 100. In addition, the phone management module 750 registers the VoIP phone 100 by assigning and storing a separate local address. The local address assigned to the VoIP phone 100 is provided to the VoIP phone 100 after registration (S107).

In addition, when a user inputs a designated command to delete the already registered VoIP phone 100 from the relay device 700, the VoIP phone 100 transmits a registration deletion request including the identification information stored at the time of registration (S109). ). Accordingly, the phone management module 750 of the relay apparatus 700 completes the deletion process by deleting the identification information of the corresponding VoIP phone 100 from the phone identification information storage unit according to the registration deletion request (S101).

4 is a flowchart schematically illustrating a call origination and calling process according to an exemplary embodiment of the present invention. As shown, when a call signal is received from the SIP server 300, the relay device 700 processes the call signal (S201), and transmits an incoming signal according to the call signal to all registered VoIP phones 100 ( S203). Accordingly, all VoIP phones 100 registered in the relay device 700 ring.

When the user recognizes that a call has been received and selects one of the plurality of VoIP phones 100 to pick up the handset, the VoIP phone 100 transmits a call start signal including its identification information to the relay device 700. Transfer to the call processing module 770 (S205). The incoming processing unit 771 of the call processing module 770 checks the VoIP phone 100 that initially transmitted the corresponding call start signal through the identification information, and the corresponding VoIP phone 100 and the SIP server 300 to the AP 900. ) Outputs a control signal to be directly connected (S207). The AP 900 makes an incoming call by directly connecting the VoIP phone 100 and the SIP server 300 according to a control signal.

In addition, when the user picks up the handset of the VoIP phone 100 for phone call through the VoIP phone 100 and inputs an outgoing phone number, the VoIP phone 100 relays the call start signal and the input outgoing phone number. A call processing module 770 of step 700 is transmitted (S209, S211). The outgoing processor 772 of the call processing module 770 receives the call start signal and the outgoing phone number and transmits a call signal including the outgoing phone number to the SIP server 300 (S213). After transmitting the call signal, the originating processor 772 checks the VoIP phone 100 transmitting the call signal through the identification information, and the VoIP phone 100 and the SIP server 300 directly to the AP 900. The control signal is output to be connected (S215). The AP 900 makes an outgoing call by directly connecting the VoIP phone 100 and the SIP server 300 according to a control signal.

On the other hand, if a call start signal is transmitted from another VoIP phone 100 during incoming or outgoing call (S217), the call state maintaining unit 774 of the relay device 700 checks the call state stored in the call state flag 773 and If the call state is currently on the phone, the call start signal is ignored to maintain the current call state (S219). In addition, the call state maintaining unit 774 transmits, for example, a message, a guide voice, and a display signal to the VoIP phone 100 that has requested the call, and the VoIP phone 100 outputs the same, thereby allowing the user to present another VoIP phone ( 100) allows the call to be known (S221).

When the call is ended and the user puts down the handset of the VoIP phone 100 in a call, the VoIP phone 100 outputs a call termination signal including identification information to the call state detection module 760 of the relay device 700. Then, the call state detection module 760 receives this and outputs it to the incoming processing unit 771 or the outgoing processing unit 772 (S223). The incoming processing unit 771 or the originating processing unit 772 outputs a control signal to the AP 900 for disconnection between the VoIP phone 100 and the SIP server 300 according to the received call termination signal, and the AP 900. To terminate the call by disconnecting according to the control signal (S225).

5 is a flowchart schematically illustrating an extension call process according to an exemplary embodiment of the present invention. As shown, when the user selects one of the plurality of VoIP phones 100 and enters, for example, the extension number of the VoIP phone 100 to which he / she wants to make an extension call, the VoIP phone 100 receives the input. The extension call signal including the extension number of the VoIP phone 100 is transmitted to the relay device 700 (S301). The extension processing module 780 of the relay device 700 receives the extension call signal and transmits the incoming signal to the receiving VoIP phone 100 corresponding to the extension number, and the VoIP phone 100 that requests the extension call. The receiving VoIP phone 100 outputs a control signal to the AP 900 so as to be directly connected (S303). Accordingly, the AP 900 establishes a connection between the two VoIP phones 100 to make an extension call.

On the other hand, if an extension call termination signal is received from the VoIP phone 100 or the receiving VoIP phone 100 requesting the extension call during the extension call (S305), the extension processing module 780 is the two VoIP during the extension call to the AP (900) Outputs a control signal to disconnect the phone 100. Accordingly, the AP 900 processes the extension call termination by releasing the connection between the two VoIP phones 100 (S307).

The present invention has been described above with reference to the preferred embodiments, but is not limited thereto, and is interpreted by the appended claims intended to cover many modifications that will be apparent to those skilled in the art without departing from the scope of the present invention. Should be done.

1 is a schematic diagram schematically showing a multi VoIP phone relay system using a single ID according to an embodiment of the present invention.

2 is a block diagram schematically illustrating a relay apparatus according to an exemplary embodiment of the present invention.

3 is a flowchart schematically illustrating a process of registering and deleting a VoIP phone according to an exemplary embodiment of the present invention.

4 is a flowchart schematically illustrating a call origination and calling process according to an exemplary embodiment of the present invention.

5 is a flowchart schematically illustrating an extension call process according to an exemplary embodiment of the present invention.

*** Description of the symbols for the main parts of the drawings ***

VoIP Phone 300. VoIP Server

500. Modem

700. Repeater

710. ID storage unit 720. ID registration processing module

730. Communication module

731. Wired communication unit 732. Wireless communication unit

740. Phone identification information storage unit 750. Phone management module

760. Call state detection module

770. Call Handling Module

771. Incoming processing unit 772. Outgoing processing unit

773. Call state flag 774. Call state maintainer

780. Extension Processing Module

900. Access Point

Claims (16)

It is made in a relay device connected to a VoIP server that provides VoIP (Voice of IP) service and relays VoIP services to multiple VoIP phones. A) receiving a call start signal including identification information from the VoIP phone in a standby state and directly connecting the corresponding VoIP phone to the VoIP server to process a call; And b) maintaining a call state by ignoring a call start signal output from another VoIP phone except the VoIP phone during a call. The method according to claim 1, wherein the identification information And a unique serial number of the VoIP phone or a local address given to the VoIP phone by the relay device. The method of claim 2, wherein step a) comprises: In the case of an incoming call, receiving the call signal transmitted from the VoIP server and transmitting the incoming signal to all VoIP phones, and directly connecting the corresponding VoIP phone to which the first call start signal is transmitted with the VoIP server, In the case of an outgoing call, a multi-VoIP phone relay using a single ID receives a call ID and a call start signal from the VoIP phone, sends a call signal to the VoIP server, and directly connects the VoIP phone to the VoIP server. Way. The method of claim 3, wherein step b) B1) storing call state information in a call state flag; B2) receiving a call start signal output from another VoIP phone except the VoIP phone in a call;  B3) notifying that the call is in progress when the call state flag is in a call state. The method according to claim 4, And c) receiving a call termination signal from the VoIP phone to disconnect the corresponding VoIP phone from the VoIP server. The method according to claim 5, D) receiving an extension call signal including received VoIP phone identification information from the VoIP phone; And e) processing an extension call by connecting a transmitting VoIP phone and a receiving VoIP phone according to the extension call signal. The method according to claim 6, And requesting a unique ID from the VoIP server and storing the ID assigned from the VoIP server. The method according to claim 7, And g) storing identification information of the corresponding VoIP phone according to the registration request of the VoIP phone. The method according to claim 8, And h) deleting identification information of the corresponding VoIP phone according to the deletion request of the VoIP phone. A phone identification information storage unit for storing a plurality of VoIP (Voice of IP) phone identification information; A phone management module for storing identification information in the phone identification information storage unit according to a registration request of the VoIP phone; A call state detection module for receiving and outputting a call signal including a call start or call end signal including identification information from the VoIP phone registered by the phone management module; According to the call signal output by the call state detection module, the VoIP phone directly connects to or disconnects from the VoIP server providing the VoIP service, while maintaining the call state by ignoring the call signal of the other VoIP phone during the connection. Multi VoIP phone relay device with a single ID containing a processing module. The method according to claim 10, wherein the identification information And a unique serial number of the VoIP phone or a local address given to the VoIP phone by the phone management module. The method according to claim 11, wherein the call processing module The VoIP phone and the VoIP server receiving the call signal transmitted through the VoIP server and transmitting an incoming signal to all the VoIP phones stored in the phone identification information storage unit, and first transmitting a call start signal by the call state detection module. And an incoming call processing unit for directly connecting; Receiving an outgoing telephone number and a call start signal from the call state detection module to send a call signal to the VoIP server, and comprises a call processing unit for directly connecting the VoIP phone and the VoIP server. Multi VoIP Phone Repeater. The method according to claim 12, wherein the call processing module A call state flag for storing call state information of the VoIP phone and the VoIP server; Receiving a call start signal transmitted from the call state detection module to check the call state information of the call state flag, and if the call further includes a call state maintaining unit for maintaining the call state by ignoring the call start signal of the corresponding VoIP phone Multi VoIP phone relay device with a single ID, characterized in that. The method of claim 13, wherein the call state maintaining unit When the call state flag is a call state, the multi-VoIP phone relay apparatus with a single ID, characterized in that notifying the VoIP phone that the call start signal is busy. The method according to claim 14, wherein the phone management module And the identification information of the corresponding VoIP phone is deleted from the phone identification information storage unit according to the deletion request received from the VoIP phone. The method according to claim 15, And a station processing module for receiving an extension call signal including identification information of the receiving VoIP phone from the VoIP phone and connecting the transmitting VoIP phone with the receiving VoIP phone. .

Images