KR20180105512A - Method and apparatus for providing call by using voice over internet protocol network - Google Patents

Abstract

The present disclosure relates to a method and apparatus for providing a call between a calling terminal and a called terminal using a VoIP network. The communication providing apparatus according to an embodiment of the present disclosure monitors the availability of at least one user terminal using the traffic volume of at least one user terminal using a Voice over Internet Protocol (VoIP) network, The call destination terminal is informed that the call destination terminal is in communication with the caller terminal when the call destination terminal is in a call state, Otherwise, a call connection is established between the calling terminal and the called terminal.

Description

TECHNICAL FIELD [0001] The present invention relates to a method and apparatus for providing a call using a VoIP network,

The present disclosure relates to a method and apparatus for providing a call between a calling terminal and a called terminal using a VoIP network.

Recently, due to the rapid development of communication, computer network and semiconductor technology, not only various services using wireless communication networks have been provided, but the demand of users has been increasing day by day, and the global wireless Internet service market has been exploding Trend. Accordingly, a service provided by a mobile communication system using a wireless communication network is being developed not only as a voice service, but also as a multimedia communication service for transmitting various data. Recently, with the increase of smart phones and the demand for data traffic, mobile operators are investing equipment and technology considering the system part and the influence to accommodate the increased data traffic in various ways.

Long Term Evolution (LTE) is a network for realizing requirements of high data rate, low-latency, and packet optimized radio access for an access network , And is designed to accommodate high-speed rich media while ensuring backward compatibility with existing 3GPP / non-3GPP access networks. LTE is an All-IP-based network that excludes existing circuit-switched based communications, enhancing the quality of service (QoS) management functions to provide real-time services (eg, voice communications, Thereby improving the efficiency of network resources by providing differentiated QoS for real-time services (e.g., web browsing, Store and Forward data transmission). We also extended the bandwidth for wireless communications by introducing smart antenna technology (MIMO: multiple input multiple output).

In an EPC (Evolved Packet Core) network, which is an LTE core network, an eNodeB, an MME, a MME, a Serving Gateway, and an S-GW < -> P-GW (Packet Data Network-Gateway) to perform call processing for voice and data processing. In the EPC network, a control message such as call setup and release is recognized as an IP packet and transmitted to the P-GW or received from the P-GW and transmitted to the user equipment (UE).

The calls used in the Voice over Internet Protocol (VoIP) or Voice over LTE (VoIP) service, which can provide voice / video call service using the packet exchange method using the IP network, A VoIP call may be disconnected or stopped when a call of a call exchange method using a VoIP service is introduced.

Accordingly, there is a need for a method that can prevent a VoIP call from being disconnected or stopped even if a call is exchanged during a call using a VoIP service.

A method of providing a call using a Voice over Internet Protocol (VoIP) network according to an embodiment of the present disclosure includes monitoring a call availability of the at least one terminal using a traffic volume of at least one terminal using a VoIP network; Receiving a call request from an originating terminal; Checking whether a destination terminal of the call request is busy using the result of the monitoring; And establishing a connection between the calling terminal and the called-party terminal when the called-party terminal is in a busy state, and establishing a connection between the called-terminal and the called-party terminal when the called-party terminal is not in a call do.

Also, in the method of providing a call according to an embodiment of the present disclosure, the step of checking whether the destination terminal is busy may include determining that the destination terminal is busy when the amount of traffic of the destination terminal is within a predetermined range for a predetermined period of time And determining that the call destination terminal is not in a busy state when the traffic volume of the destination terminal is not within the predetermined range for the predetermined time.

Also, in the method of providing a call according to an embodiment of the present disclosure, the step of monitoring the availability of the at least one terminal may comprise: separating the traffic volume of the destination terminal into traffic related to a voice call, a video call, And monitoring.

Also, in the method of providing a call according to an embodiment of the present disclosure, monitoring the availability of the at least one terminal may include determining whether a traffic amount of the at least one terminal is within the predetermined range over a threshold time step; And notifying the terminal of information on the entity that caused the traffic when the traffic amount is included within the predetermined range over the threshold time.

Also, in the method of providing a call according to an embodiment of the present disclosure, monitoring the availability of the at least one terminal may include monitoring the availability of the at least one terminal, ; And storing information on availability of each of the at least one terminals based on the set information.

Also, the apparatus for providing a call using a VoIP network according to an embodiment of the present disclosure may monitor the availability of the at least one terminal using the traffic volume of at least one terminal using the VoIP network, And notifies the calling terminal that the call destination terminal is in a call state, and if the call destination terminal is in a call state, A processing unit for establishing a call connection between the calling terminal and the called terminal; And a transmitting / receiving unit for receiving a call request from the calling terminal.

In addition, in the communication providing apparatus according to the embodiment of the present disclosure, the processing unit determines that the destination terminal is busy when the traffic volume of the destination terminal is within a predetermined range for a predetermined time, And determines that the call destination terminal is not in a busy state if it is not within the predetermined range for a predetermined time.

In addition, in the call providing apparatus according to the embodiment of the present disclosure, the processing unit divides the traffic volume of the call destination terminal into traffic related to voice call, video call, and SMS transmission / reception, And monitoring.

Also, in the communication providing apparatus according to the embodiment of the present disclosure, the processing unit checks whether the traffic amount of the at least one terminal is included within the predetermined range over the threshold time, and if the traffic amount is within the predetermined range It notifies the terminal of the information on the entity that caused the traffic.

Also, in the call providing apparatus according to the embodiment of the present disclosure, the processing unit may monitor the availability of the at least one terminal to set information on the monitoring result to be in a busy state or not in a busy state, And stores information on the availability of each of the at least one terminal in the storage unit.

According to the embodiment of the present disclosure, it is possible to prevent a phenomenon of interruption or stoppage during a call using a Voice over Internet Protocol (VoIP) service, thereby ensuring VoIP service quality.

1 is a diagram illustrating a configuration of a mobile communication network according to an embodiment of the present invention.
FIG. 2 is a diagram illustrating an exemplary configuration of an EPC network according to an embodiment of the present disclosure.
3 is an exemplary view showing a configuration of a call providing apparatus according to an embodiment of the present disclosure;
4 is an exemplary view showing a monitoring result of the traffic amount of the user terminal 40 according to the embodiment of the present disclosure.
5 is a flow chart illustrating the procedure of a call provision method according to an embodiment of the present disclosure;

The embodiments of the present disclosure are illustrated for the purpose of describing the technical idea of the present disclosure. The scope of the claims according to the present disclosure is not limited to the embodiments described below or to the detailed description of these embodiments.

All technical and scientific terms used in the present disclosure have the meaning commonly understood by one of ordinary skill in the art to which this disclosure belongs unless otherwise defined. All terms used in the disclosure are selected for the purpose of more clearly illustrating the disclosure and are not chosen to limit the scope of the rights under the present disclosure.

As used in this disclosure, expressions such as " comprising, "" having," "having, " and the like, unless the context requires otherwise, (open-ended terms).

The expressions of the singular forms described in this disclosure may include plural meanings unless the context clearly dictates otherwise, and the same applies to the singular expressions set forth in the claims.

As used in this disclosure, expressions such as " first ", "second ", and the like are used to distinguish a plurality of components from each other and do not limit the order or importance of the components.

The term "part " as used in this disclosure refers to hardware components such as software or an FPGA (field-programmable gate array), ASIC (application specific integrated circuit). However, "part" is not limited to hardware and software. "Part" may be configured to reside on an addressable storage medium, and may be configured to play back one or more processors. Thus, by way of example, and not limitation, "part (s) " may include but are not limited to elements such as software components, object oriented software components, class components and task components, Microcode, circuitry, data, databases, data structures, tables, arrays, and variables of program code. The functions provided within the component and the "part " may be combined into a smaller number of components and" parts " or further separated into additional components and "parts ".

As used in this disclosure, the expression "based on" is used to describe one or more factors affecting an action or an action of a decision, judgment, as described in the phrase or sentence in which the expression is contained, It does not exclude any additional factors that affect the decision, act of judgment or action.

In the present disclosure, when an element is referred to as being "connected" or "connected" to another element, it is to be understood that the element may be directly connected to or connected to the other element, But may be connected or connected via other components.

Hereinafter, embodiments of the present disclosure will be described with reference to the accompanying drawings. In the accompanying drawings, the same or corresponding components are denoted by the same reference numerals. In the following description of the embodiments, description of the same or corresponding components may be omitted. However, even if the description of the components is omitted, such components are not intended to be included in any embodiment.

In one embodiment, the mobile communication network includes a 2G wireless communication network such as Global System for Mobile communication (GSM), Code Division Multiple Access (CDMA), a Long Term Evolution Network (LTE), a wireless Internet such as WiFi, (E.g., a 3G mobile communication network such as WCDMA or CDMA2000, a High Speed Downlink Packet Access (HSDPA), or a High Speed Downlink Packet Access (HSUPA) (High Speed Downlink Packet Access) Speed uplink packet access), 4G mobile communication network currently in service, etc., macro base stations (macro eNodeBs), micro base stations (Pico eNodeBs, Home-eNodeBs) and user terminals But is not limited to, any other mobile communication network included as an element. Hereinafter, an E-UTRAN (Evolved Universal Terrestrial Radio Access Network) as a radio access network of LTE will be mainly described.

1 is a diagram illustrating a configuration of a mobile communication network according to an embodiment of the present invention.

As shown in FIG. 1, the mobile communication network may include one or more network cells, and may include a Heterogeneous Network (HET) environment in which different kinds of network cells may be mixed in a mobile communication network. The mobile communication network includes miniature base stations (Pico eNodeB, Home-eNodeB, relay, etc.) 11 to 15, 21 to 23, and 23, which manage small-sized network cells (e.g., 'small cells' such as picocells, femtocells, (Macro eNodeBs) 10, 20 and 30 for managing a wide range of cells (for example, 'macro cells' (MSS) 60, an S-GW (Serving Gateway) 80, a P-GW (Packet Data Network) gateway 90 and an HSS (Home Subscriber Server) (100). Each component shown in Fig. 1 is illustrative, and the components of the mobile communication network in which the present invention can be implemented are not limited to those shown in the drawings.

For example, the macro base stations 10, 20 and 30 can be used in an LTE network, a WiFi network, a WiBro network, a WiMax network, a WCDMA network, a CDMA network, a UMTS network, But is not limited to, the characteristics of the macro cell base station that manages the base station.

The micro base stations 11 to 15 and 21 to 23 and 31 to 33 can be used in an LTE network, a WiFi network, a WiBro network, a WiMax network, a WCDMA network, a CDMA network, a UMTS network, but is not limited to, the features of a pico base station, an indoor base station or a femto base station, or a relay that manages a cell having a radius of about m to several tens of meters.

The micro base stations 11 to 15 and 21 to 23 and 31 to 33 and the macro base stations 10 and 20 and 30 independently control the SON server 50, the MME 60, the S-GW 80, (90), HSS (100), and the like.

The user terminal 40 may be a mobile Internet network such as a GSM network, a 2G wireless communication network such as a CDMA network, a wireless Internet network such as an LTE network and a WiFi network, a WiBro network, and a WiMax network, But are not limited to, the characteristics of the terminal.

The management server (O & M server) 70, which is a network management apparatus of the micro-base station, is responsible for configuration information and management of the micro base stations 11 to 15, 21 to 23 and 31 to 33 and the macro base stations 10, 20 and 30 . The management server 70 can perform all the functions of the SON server 50, the MME 60, and the HSS 100. [ SON server 50 may include any server that performs macro / micro base station installation and optimization and functions to provide basic parameters or data necessary for each base station. The MME 60 may include any entity used to manage the mobility of the user terminal 40 and the like. In addition, the MME 60 performs a function of a base station controller (BSC), and performs resource allocation, call control, handover control, and handover control on base stations (pico eNodeB, Home- eNodeB, macro eNodeB, Voice and packet processing control, and the like. The HSS 100 is a kind of database for service / authentication of the subscriber.

In one embodiment, one management server 70 can perform all the functions of the SON server 50, the MME 60, and the HSS 100, and the SON server 50, the MME 60, The base station 100 may manage one or more macro base stations 10, 20 and 30 and one or more micro base stations 11 to 15, 21 to 23 and 31 to 33.

When the mobile communication network is assumed to be an LTE network, the LTE network is interworked with an inter-RAT network (WiFi network, WiBro network, WiMax network, WCDMA network, CDMA network, UMTS network, GSM network, etc.). (LTE network, WiFi network, WiMax network, WCDMA network, CDMA network, UMTS network, GSM network, etc.) when one of the inter-RAT networks (e.g., WiBro network) is the mobile communication network. (WiFi network, WiMax network, WCDMA network, CDMA network, UMTS network, GSM network, etc.) are shown apart from one another in the drawing, it is assumed that it is overlaid.

2 is a diagram illustrating an exemplary configuration of an EPC network according to an embodiment of the present invention.

The E-UTRAN 25 is an LTE radio access network composed of eNBs (evolved NodeB, 25-1, ..., 25-n, Network) to transmit data and control information. In addition, when a terminal using an LTE system moves to a conventional 2G / 3G mobile communication network and uses a voice service, a paging request for a circuit switch (Fallback) to receive a voice service, a short message service ) Message to the UE 40 and a direct connection to a target cell capable of a CS service.

2, "LTE-Uu" represents an air interface between the E-UTRAN 25 and the UE 40, "S1- MME" represents an interface between the MME 60 and the E-UTRAN 25, "S1-U" represents an interface between the S-GW 80 and the E-UTRAN 25, "S11" represents an interface between the S-GW 80 and the MME 60, "S5 / SGI "represents the interface between the P-GW 90 and the S-GW 90, and" SGi "represents the interface between the IP network 110 and the P-GW 90. And "S6a" may represent the interface between the HSS 100 and the MME 60.

The UE 40 and the eNBs 25-1 to 25-n of the E-UTRAN 25 communicate via the RRC (Radio Resource Control) protocol and the eNB 25- A broadcasting message to a cell area to be controlled by the RRC message is defined as an RRC message. The RRC message may include control messages from the Non-Access Stratum (NAS) protocol, which control messages are transparently transferred to the UE 40 or the core network without being read in the E-UTRAN 25 .

The eNB 25-n is an end point for the radio signal of the E-UTRAN 25 and the control signal is interlocked with the MME 60 via the S1-MME interface and the data traffic is transmitted to the S- (80). The S-GW 80 performs a buffering function for the anchor and downlink traffic for mobility in the E-UTRAN 25. [ The P-GW 90 is a connection point of the external IP network 110, performs IP allocation and billing for a mobile subscriber, and traffic control functions for user data. The P-GW 90 can be connected to the IP server 120 through the IP network 110 and can transmit traffic related information of at least one user terminal connected to the IP network 110 from the IP server 120 .

The IP network 110 provides an IMS (IP Multimedia Subsystem) service for the UE 40 in the EPC network, a Policy & Charging Rule Function (PCRF), an IMS nodes (for example, a Proxy Call Session Control Function , An Interrogating Call Session Control Function (I-CSCF), a Serving Call Session Control Function (S-CSCF), and an Application Function Function (AF)).

UE 40 sends and receives a call control message for multimedia services using IMS nodes and the Gm interface through an EPC bearer (provided by E-UTRAN / S-GW / P-GW).

The E-UTRAN 25 provides a radio communication function to the UE 40 and performs a function of managing radio resources for the radio communication function.

The MME 60 may receive authentication information for authenticating the UE 40 from the HSS 100 and perform authentication of the UE 40. [ The MME 60 also manages the mobility of the UE 40 and the eNB 25-n at the upper level of the eNB 25-n and controls the setting and release of an Evolved Packet System (EPS) session and a bearer The same call control function can be performed. Mobility and session control with the UE 40 is handled by the NAS protocol at the NAS (Non-Access Stratum) layer located in the control plane of the UE 40 and the MME 60, And the MME 60 communicate with each other through the NAS message. The NAS function is largely divided into EMM (EPS Mobility Management) and ESM (EPS Session Management) functions. In addition, the MME 60 may be directly connected to the IP network 110 through the S-GW 80 and the P-GW 90. The call processing control signal of the eNB 25-n is transmitted to the S-GW 80 via the MME 60 and is transmitted to the P-GW 90 in response to the call processing control signal, Lt; / RTI &gt; The EMM is a sublayer located in the NAS layer, and the EMM procedure is performed, so that the UE 40 has seven EMM states and the MME 60 has four EMM states. In order for the UE 40 and the MME 60 to send and receive NAS messages, a signaling connection must be created between the UE 40 and the MME 60 so that the NAS message can be transmitted. This is called an ECM (EPS Connection Management) connection . The ECM connection is a logical connection and consists essentially of an RRC connection established between the UE 40 and the eNB 25-n and an S1 signaling connection established between the eNB 25-n and the MME 60. That is, the ECM connection is turned on / off means that the RRC connection and the S1 signaling connection are both turned on / off. When an ECM connection is established, an RRC connection is seen from the UE 40, and an S1 signaling connection is established from the MME 60. [ The ECM connection has ECM-Connected and ECM-Idle status depending on whether the NAS signaling connection, ECM connection, or not. According to the EMM procedure, the ECM-Connected state and the ECM-Idle state are frequently moved. This change process is referred to as a state transition.

The S-GW 80 acts as a gateway between the 3GPP network and the E-UTRAN 25 and performs the handover between the eNBs 25-n and the UE 40 between the 3GPP network and the 3GPP network (inter-3GPP) And can perform a mobility anchor function for providing the service. The S-GW 80 can transmit to the P-GW 90 a job necessary for call processing according to the control signal of the eNB 25-n.

The P-GW 90 may allocate an IP address to the UE 40 and apply different QoS policies to the UE 40. [ In addition, the P-GW 90 acts as a gateway to a PDN (Packet Data Network), allowing the UE 40 to access a data network such as the Internet or the Internet to receive services.

Although the S-GW 80 and the P-GW 90 are illustrated as being separated and communicating at the S5 / S8 interface as an embodiment, the S-GW 80 and the P- single gateway).

The HSS 100 may manage authentication information for authenticating the UE 40, location information of the UE 40, and the profile of the UE 40. [ The profile of the UE 40 may include QoS rating information (e.g., priority, maximum available bandwidth, etc.) for each service item to which each UE 40 subscribes. In one embodiment, the authentication information for authenticating the UE 40 and the profile of the UE 40 may be transferred from the HSS 100 to the MME 60 when the UE 40 connects to the network.

The PCRF (not shown) manages policies and rules for charging and allows the P-GW 90 and the S-GW 80 to provide appropriate QoS for the UE 40 and use Allows you to perform the billing function for the bearer.

The IMS node (not shown) is made up of nodes such as P-CSCF, I-CSCF, S-CSCF, and AF in detail and the UE 40 provides multimedia services such as VoIP give.

3 is an exemplary view showing a configuration of a call providing apparatus according to an embodiment of the present disclosure;

3, the communication providing apparatus 300 may include a processing unit 310, a storage unit 320, a transmission / reception unit 330, and a system bus 340. [ In an embodiment, the processing unit 310, the storage unit 320, and the transmission / reception unit 330 may be connected to each other via the system bus 340. Also, the call providing apparatus 300 may be included in the P-GW 90 and may be provided separately from the P-GW 90.

The processing unit 310 may monitor the availability of at least one user terminal using the traffic volume of the at least one user terminal 40 using the VoIP network. In one embodiment, the monitoring results for at least one user terminal of the processing unit 310 may be stored in the storage unit 320. When receiving the call request from the specific calling terminal, the processing unit 310 can check whether or not the called terminal of the received call request is in a call using the result of the monitoring of at least one user terminal. The processing unit 310 can notify the calling terminal that the destination terminal is busy via the transmission / reception unit 330 when the destination terminal is busy. If the destination terminal is not busy, the processing unit 310 cancall the call between the source terminal and the destination terminal You can configure it to connect.

When the amount of traffic of the destination terminal is within a predetermined range for a predetermined time, the processing unit 310 determines that the destination terminal is busy. If the amount of traffic of the destination terminal is not included within a predetermined range for a predetermined time, It can be determined that the terminal is not busy. For example, the predetermined time may include a time range of 2 seconds to 200 seconds, and the predetermined range may include a traffic amount of 10 bytes to 30 bytes, but is not limited to this example. 4 is an exemplary view showing a monitoring result of the traffic amount of the user terminal 40 according to the embodiment of the present disclosure. As shown in FIG. 4, since the traffic amount of the specific user terminal 40 (including the destination terminal) is monitored, the user terminal 40 exists between Range A and Range B within the predetermined range in the period 1, It can be determined that the user is in a call using the VoIP network. In the periods 2 and 3, it can be determined that the traffic is not busy because the traffic amount exceeds a predetermined range or has a value smaller than a predetermined range.

In addition, the processing unit 310 may monitor the availability of at least one user terminal 40 by dividing the traffic volume of the destination terminal into traffic related to voice call, video call, and SMS transmission / reception, respectively. In one embodiment, the traffic-related information received from the IP server 120 through the IP network 110 may include an identification factor according to the service type. That is, the traffic related to the voice call includes the identification factor of A, the traffic related to the video call includes the identification factor of B, and the traffic associated with the text message such as SMS may include the identification factor of Z . Accordingly, the processing unit 310 can monitor the traffic for each service type for each application by checking the identification factor of the received traffic information and calculate the monitoring result.

In addition, the processing unit 310 checks whether the traffic amount of at least one user terminal 40 that is monitoring is included within a predetermined range exceeding the threshold time, and if the traffic amount of at least one user terminal 40 If the user terminal 40 is included within a predetermined range longer than the threshold time, the user terminal 40 can be notified of information on the entity that caused the traffic. Accordingly, the system administrator or the user of the user terminal 40 can take appropriate action when a garbage pattern that can be received during a call is received for a malicious purpose.

In addition, the processing unit 310 may monitor the availability of at least one user terminal 40 to set information on the monitoring result to be in a busy state or not in a busy state, 40 may store information on the availability of each call in the storage unit 320.

The storage unit 320 may store the traffic volume monitoring result for at least one user terminal 40. [ In one embodiment, the storage unit 320 may store the traffic volume monitoring result by application for each user terminal 40, and may store the traffic volume monitoring result for each application type per application. As an example, the storage unit 220 may be a ROM (Read Only Memory), a RAM (Random Access Memory), a CD (Compact Disc) -ROM, a magnetic tape, a floppy disc, Optical data storage, or carrier waves (e.g., transmission over the Internet), but are not limited to such implementations.

Transmitter / receiver 330 can send and receive signals with components that are communicatively coupled to implement the call presentation method of the present disclosure. That is, the transmitting / receiving unit 330 can receive the traffic-related information from the IP server 120 through the IP network 110 and transmit the received traffic-related information to the processing unit 310. Also, the transceiver 330 can transmit and receive a signal to connect a call between the calling terminal and the called terminal.

5 is a flow chart illustrating the procedure of a call provision method according to an embodiment of the present disclosure;

Although process steps, method steps, algorithms, and the like have been described in this flowchart in a sequential order, such processes, methods, and algorithms may be configured to operate in any suitable order. In other words, the steps of the processes, methods and algorithms described in the various embodiments of the present disclosure need not be performed in the order described in this disclosure. Also, although some of the steps are described as being performed asynchronously, some of these steps may be performed concurrently in other embodiments. Also, an illustration of a process by way of illustration in the drawings is not intended to imply that the illustrated process excludes other variations and modifications thereto, and that any of the illustrated processes or steps thereof may be used in various embodiments of the present disclosure Does not imply that it is necessary for more than one, and does not mean that the illustrated process is preferred.

As shown in FIG. 5, in step S510, the availability of at least one user terminal can be monitored using the traffic volume of at least one user terminal using the VoIP network. For example, referring to FIGS. 1 to 4, the call providing apparatus 300 may receive information about traffic of at least one user terminal 40 connected to be communicable using the VoIP network through the IP network 110 And monitor the availability of at least one user terminal 40 using information on the received traffic.

In step S520, a call request may be received from the calling terminal. For example, referring to FIGS. 1 to 4, a call request can be received from a calling terminal for communication with another user terminal (called terminal).

In step S530, it may be determined whether or not the destination terminal of the call request is busy using the monitoring result. For example, referring to FIG. 1 to FIG. 4, the call providing apparatus 300 determines whether a call target terminal including information in a call request is busy using the traffic amount monitoring result of at least one user terminal 40 Can be determined.

If it is determined in step S540 that the destination terminal is busy, it can be notified to the calling terminal that the destination terminal is busy. For example, referring to FIG. 1 to FIG. 4, the call providing apparatus 300 checks whether or not a call is being made using the traffic volume monitoring result of a call destination terminal included in the call request. If the call destination terminal is busy The caller terminal can notify that the call destination terminal is busy.

In step S550, if the call destination terminal is not in a call, a call between the caller terminal and the call destination terminal may be set to be connected. For example, referring to FIG. 1 to FIG. 4, the call providing apparatus 300 confirms whether or not a call is being made using the traffic volume monitoring result of the call destination terminal included in the call request, It is possible to establish a call connection between the calling terminal and the called terminal.

Although the method has been described through particular embodiments, the method may also be implemented as computer readable code on a computer readable recording medium. A computer-readable recording medium includes all kinds of recording apparatuses in which data that can be read by a computer system is stored. Examples of the computer-readable recording medium include a ROM, a RAM, a CD-ROM, a magnetic tape, a floppy disk, an optical data storage device, and the like, and may be implemented in the form of a carrier wave (for example, transmission over the Internet) . In addition, the computer-readable recording medium may be distributed over network-connected computer systems so that computer readable codes can be stored and executed in a distributed manner. And, functional programs, codes, and code segments for implementing the above embodiments can be easily inferred by programmers in the technical field to which this disclosure belongs.

Although the present disclosure has been described in connection with some embodiments herein, it should be understood that various modifications and changes may be made thereto without departing from the spirit and scope of the present disclosure as will be apparent to those skilled in the art to which this disclosure belongs. something to do. It is also contemplated that such variations and modifications are within the scope of the claims appended hereto.

11 ~ 15, 21 ~ 23, 31 ~ 33: micro base stations 10, 20, 30: macro base station
40: User terminal 50: SON server
60: MME 80: S-GW
90: P-GW 100: HSS
110: IP network 120: IP server
300: a call providing device 310:
320: Storage unit 330: Transmitting /
340: System bus

Claims (10)

A method for providing a call using a VoIP (Voice over Internet Protocol)
Monitoring a call availability of the at least one user terminal using a traffic amount of at least one user terminal using a VoIP network;
Receiving a call request from an originating terminal;
Checking whether a destination terminal of the call request is busy using the result of the monitoring; And
Establishing a connection between the calling terminal and the called-party terminal when the called-party terminal is in a busy state, and establishing a connection between the called-terminal and the called-party terminal when the called-party terminal is not in a call , A method of providing a call. The method according to claim 1,
The step of confirming whether or not the called terminal is in a call includes:
When the amount of traffic of the destination terminal is within a predetermined range for a predetermined time, the destination terminal is determined to be busy, and if the amount of traffic of the destination terminal is not within the predetermined range for the predetermined time, The method comprising the steps of:
How to provide a call. The method according to claim 1,
Wherein the monitoring of the availability of the at least one user terminal comprises:
And monitoring the traffic volume of the destination terminal by dividing the traffic volume into traffic related to voice call, video call, and SMS transmission / reception, respectively.
How to provide a call. The method according to claim 1,
Wherein the monitoring of the availability of the at least one user terminal comprises:
Confirming whether the traffic volume of the at least one user terminal is included within the predetermined range over a threshold time; And
And notifying the user terminal of information on the entity that has caused the traffic when the traffic volume is within the predetermined range over the threshold time.
How to provide a call. The method according to claim 1,
Wherein the monitoring of the availability of the at least one user terminal comprises:
Monitoring the availability of the at least one user terminal and setting the information about the monitoring result to be in a busy state or not in a busy state; And
And storing information on availability of each of the at least one user terminals based on the set information.
How to provide a call. A communication providing apparatus using a VoIP (Voice over Internet Protocol) network,
Monitoring a call availability of the at least one user terminal using a traffic amount of at least one user terminal using a VoIP network and determining whether a call target terminal of the call request received from the caller terminal is in a call using a result of the monitoring And notifies the calling terminal that the called terminal is in communication with the called terminal, and when the called terminal is not in communication, sets the calling terminal to connect the call between the called terminal and the called target terminal ; And
And a transmission / reception unit for receiving a call request from the calling terminal,
Call providing device. The method according to claim 6,
Wherein,
When the amount of traffic of the destination terminal is within a predetermined range for a predetermined time, the destination terminal is determined to be busy, and if the amount of traffic of the destination terminal is not within the predetermined range for the predetermined time, Judging that it is not in the middle,
Call providing device. The method according to claim 6,
Wherein,
And monitoring traffic availability of the at least one user terminal by dividing traffic amounts of the at least one user terminal into traffic related to a voice call, a video call, and an SMS transmission / reception,
Call providing device. The method according to claim 6,
Wherein,
The method comprising: confirming whether a traffic volume of the at least one user terminal is included within the predetermined range over a threshold time and, when the traffic volume is within the predetermined range over the threshold time, Lt; / RTI &gt;
Call providing device. The method according to claim 6,
Wherein,
Wherein the at least one user terminal monitors the availability of the at least one user terminal to set information on the monitoring result to be in a busy state or not in a busy state and to transmit information on the availability of each of the at least one user terminals based on the set information Storing in a storage unit,
Call providing device.

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