WO2014104614A1

WO2014104614A1 - Method and device for performing ip address-based telephone conversation through plurality of heterogeneous communication networks

Info

Publication number: WO2014104614A1
Application number: PCT/KR2013/011368
Authority: WO
Inventors: 이승준; 송준석
Original assignee: 주식회사에어플러그
Priority date: 2012-12-28
Filing date: 2013-12-09
Publication date: 2014-07-03
Also published as: KR101403525B1

Abstract

According to the present invention, a device allows a user to perform IP address-based telephone conversation, that is, VoIP telephone conversation, by using a plurality of communication networks. The device establishes a first telephone conversation path in a first wireless communication network for a requested VoIP telephone conversation and additionally establishes a second telephone conversation path in a second wireless communication network for the VoIP telephone conversation in a state in which the first telephone conversation path is established. In addition, the device transmits voice packets only through any one telephone conversation path selected from both the first telephone conversation path and the second telephone conversation path in a state in which both the telephone conversation paths of the first telephone conversation path and the second telephone conversation path are established, and transmits, by using the voice packets, voice data which is the same as the voice data carried on a voice packet transmitted through the any one telephone conversation path through the other one of the both telephone conversation paths when quality of the any one telephone conversation path is not good.

Description

Method and apparatus for performing IP address based call through a plurality of heterogeneous communication networks

The present invention relates to a terminal for making a Voice over Internet Protocol (VoIP) telephone conversation over a communication network, in particular a wireless communication network, a server supporting the call, and a method performed in the terminal and the server.

With the development of mobile network technology and its stability, the end-to-end IP address based call (hereinafter also referred to as “VoIP call”) can provide good quality. It became.

For the VoIP call, as illustrated in FIG. 1, a general purpose VoIP server 10 supporting such a call intervenes to establish a call path between the terminals 1a and 1b through the mobile communication network 11a. FIG. 2 briefly illustrates an example of a call setup process performed between both terminals 1a and 1b for establishing a call path. In general, when the calling terminal 1a requests call setup (S11), The general-purpose VoIP server 10 confirms the IP address assigned to the terminal from the information (for example, an e-mail address) for specifying the destination terminal 1b included in the request, and confirms the confirmed IP. The call setup request is made to the address (S12). Of course, this call establishment request (S12) may be made by the invited server to the called terminal 1b by the general-purpose VoIP server 10 sending an INVITE message to another server. Then, after the connection request for the call establishment request and acknowledgment of the connection request, a two-way communication path is established between the terminals 1a and 1b. That is, IP addresses of packets to which voice messages are to be transmitted to each other are allocated to the called party and the sender, respectively. Thereafter, the general-purpose VoIP server 10 relays the packets to which the allocated IP address is added to each other, thereby making a VoIP call S15 between the terminals 1a and 1b.

Unlike the method of supporting the call in the mobile communication network 11a, the VoIP call made as described above does not exclusively allocate the channel for the call path on the radio signal. -based) ”). Instead, only when data for a call is generated, the data is transmitted using a channel shared for data service with other mobile communication terminals. Therefore, there is not always enough bandwidth available for the call. If the available bandwidth of the radio signal is not sufficient, the transmission delay time of the packet carrying the voice message is increased, the packet loss rate is also high, and the call quality is reduced.

However, since it is generally more advantageous in terms of cost than using a line-based call, a mobile subscriber using a VoIP call using a mobile terminal may suffer some degradation in call quality. I tend to use it. It is more cost-effective for users because of the carrier's telecommunications policy, the data service adopts a lower rate system than the circuit-based call, and also provides unlimited data service at a fixed rate. This is because they operate a fee system that can be used.

Thus, users will prefer VoIP calls, which will naturally gradually reduce the usage time by circuit-based calls. However, when the change in the usage method for the call becomes prominent, the load on the operation communication network becomes greater from the mobile communication provider's point of view. Considering this aspect, it is very likely that the increase in network load caused by VoIP calls will not be tolerated by mobile operators. And even if the mobile communication network evolves and its physical service speed is increased, as long as the current service fee system imposed on the user continues, the VoIP call will not be supported beyond the call quality level currently supported by the mobile communication network. Of course, even in the present state, if the amount of data associated with a VoIP call increases, the overall load due to such a call can be suppressed by the mobile operator, so that individual call quality may be further reduced.

Nevertheless, VoIP calls using data services are very attractive to the user in terms of cost and would be beneficial to the user if this type of call is available with improved quality. And if the call quality can be improved in the VoIP call without increasing the load on the mobile network as a whole (without totally burdening the amount of data required for the call quality), Will be beneficial.

It is an object of the present invention to provide a method and apparatus for enabling a VoIP call of improved quality by selectively using a wireless data network in addition to a mobile communication network.

Another object of the present invention is to provide a method and apparatus for selectively using a plurality of wireless communication networks in order to support VoIP calls with stable quality, which is advantageous for the user in terms of cost.

It is still another object of the present invention to provide a method and apparatus for enabling a VoIP call that can suppress the cost burden of using a data service in a range that does not degrade the quality of the VoIP call.

It is still another object of the present invention to provide a method and apparatus for selectively providing a VoIP call with improved quality in consideration of a user's use condition of a mobile communication service.

The object of the present invention is not limited to the object explicitly stated above, and of course includes the purpose of achieving an effect that can be derived from the specific and exemplary description of the present invention.

In accordance with an aspect of the present invention, an apparatus for making an IP address-based call using a plurality of communication networks, establishes a first call path on a first wireless communication network for a requested IP address-based call, and A controller configured to additionally open a second call path on a second wireless communication network for the IP address-based call with the first call path opened, and both call paths between the first call path and the second call path In the opened state, confirming the call quality for the call path while transmitting voice packets to only one call selected by both call paths, and based on the checked call quality, to the other of both call paths, And a communication unit configured to transmit the same voice data as the voice data carried in the voice packet transmitted in any one call through the voice packet.

According to an embodiment of the present invention, the device is implemented in a server for making a call with a called party by processing a call made from a terminal for an IP address-based call, or making an IP address-based call. It is implemented in a terminal capable of making or receiving a call.

In an embodiment in which the device is implemented in a terminal, when the second call path is additionally established, the call part is the same data block and the sequence number of the block as the voice packet transmitted to the first call path to the second call path. It is further configured to transmit a voice packet having a. The second wireless communication network has a shorter wireless communication network, for example, a Wi-Fi wireless LAN network, in which service areas are scattered, and a serviceable distance in each service area is shorter than that of the first wireless communication network. Further, in the present embodiment, the second call path, under the condition that the terminal is the called party of the IP address-based call, the control unit makes a new call path to an external server as the second wireless communication network becomes available. It may be additionally established by requesting a connection.

In one embodiment according to the present invention, the selected one of the communication paths, the mutual comparison between the two communication paths with respect to the value of the communication characteristics indicating the call quality of the series of voice packets respectively received through the two communication paths It is a channel with a better communication characteristic through. In this embodiment, the value of the communication characteristic indicating the call quality is obtained by the measurement of the call part, or is transmitting and receiving a voice packet with the call part via the first call path or the second call path. It may be obtained by the measurement on the other side and reported to the communication unit. Also, in this embodiment, the value of the communication characteristic may be a transmission delay time. In this case, the same sequence number received in duplicate of each of the two communication paths, with the call path showing a better communication characteristic value. Is the path on the average that is received at the time of receipt between the pairs of data blocks.

According to various embodiments of the present disclosure, the checking of the call quality for the call path may be based on a transmission delay time or a packet loss rate for a packet of a test block transmitted through the first call path or the second call path. Communication characteristic information indicating a call quality, obtained by measuring a packet loss rate for voice packets received through one of the communication paths, or a terminal from which the IP address-based call is originated or received. Call from the information obtained for the service area from the external server, or by providing the external server with information about the service area in which the originating or destination terminal of the requested IP address based call is currently located. It may be to check the quality.

According to an embodiment of the present invention, when the checked call quality satisfies a predetermined requirement, the call unit may include voice packets carrying voice data so that the same voice data is not duplicated and transmitted to both of the calls. It is further configured to transmit the divided into the two currencies. In the present embodiment, the predetermined requirement includes a first requirement that all of the call quality of the two currencies be equal to or greater than the first predetermined reference value. The predetermined requirement may further include a second requirement that a specific mode should be designated by the user for IP address-based calls. If the predetermined requirement includes both the first requirement and the second requirement, if any of the two call paths has a lower call quality, the call unit transmits the voice data repeatedly to both call paths. The first reference value is a value that designates higher quality than the second reference value. If the predetermined requirement includes only the first requirement, the communication unit is further configured to duplicately transmit voice data to the two communication paths when any one of the two communication paths falls below the first reference value. Can be. In addition, in the present embodiment, the predetermined requirements include a requirement that all fluctuations in the call quality of the two currency paths be within a predetermined limit, and a reference value predetermined in any one of the two currency paths. Includes requirements to be: The call unit may be further configured to dynamically change a quality value, which is a reference of the fluctuation range, over time, and determine whether the call quality is within the limited range based on the dynamically changing quality value. .

In one embodiment according to the present invention, the communication unit is further configured to transmit a voice packet only to a call path that is equal to or greater than the reference value when any one of the two call paths exceeds a predetermined reference value in call quality.

In one embodiment according to the present invention, when the call unit is designated as having a difference in the use for the first wireless communication network and the second wireless communication network, both of the two communication paths are different in terms of the call quality. If it is above a specified threshold, it is configured to transmit the voice packet only to a call established on a wireless communication network which is more cost-effective for the user in the subsequent use of the data service.

According to another aspect of the present invention, a method for making an IP address-based call using a plurality of communication networks may include establishing a first call path on a first wireless communication network for a required IP address-based call. Transmitting and receiving a voice packet, establishing a second call path on a second wireless communication network for the IP address-based call while the first call path is established, and the first call path and the first call path. Confirming a call quality for at least one of the call paths while transmitting voice packets to either of the two call paths, and based on the confirmed call quality, to another of the two call paths, And transmitting the same voice data through the voice packet as the voice data carried in the voice packet transmitted in any one call.

In one embodiment according to the present invention, the step of establishing the second call path, the terminal of the calling party or the called party of the requested IP address-based call can transmit and receive signals with the second wireless communication network It proceeds as it enters the state.

According to another aspect of the present invention, a program supply apparatus includes communication means capable of transmitting and receiving data to and from the outside through communication, and storage means including an application executed in the communication terminal transmitted through the communication means. It is configured to include. When the application is executed in the communication terminal, the application opens a first call path on a first wireless communication network for a requested IP address-based call, and in a state in which the first call path is opened, When a second wireless communication network becomes available, a function of additionally establishing a second call path on the second wireless communication network for the IP address-based call, and the two call paths between the first call path and the second call path are established. In the state, while transmitting the voice packets to the outside of only one of the selected calls in the two channels, the call quality for the call path is confirmed, and based on the confirmed call quality to the other of the two paths, Program code for performing the function of transmitting the same voice data to the outside via the voice packet, which is carried in the voice packet transmitted to any one call. It is configured to include them.

In the above-described apparatus, method and program supply apparatus, each of the two communication paths is established in accordance with a communication protocol for the occupation of physical communication resources of logical communication entities at both ends of sending and receiving the voice packet. Corresponds to session.

At least one embodiment of the present invention described above, or described in detail with reference to the accompanying drawings, to enable a VoIP call with a more stable or improved quality by using a plurality of wireless communication networks complementarily do. As a result, the VoIP call user according to the present invention can increase the satisfaction in the call.

In addition, when a wireless communication network, such as a Wi-Fi network that is relatively inexpensive or inexpensive to the user, supports an appropriate quality, VoIP calls are made mainly on the wireless communication network, thereby making VoIP calls. Allow users to take advantage of the cost. At the same time, the mobile operator can efficiently distribute network resources by converting the load on the cellular network to another communication network by the VoIP call, which is not beneficial to the circuit-based call.

1 and 2 illustrate a simple service environment in which a VoIP call is made through a mobile communication network, and a simple example of a signal exchange procedure for making a VoIP call.

3 illustrates a configuration of a terminal capable of performing a VoIP call through a plurality of heterogeneous communication networks according to an embodiment of the present invention.

4 illustrates a configuration of a server capable of supporting VoIP calls through a plurality of heterogeneous communication networks according to an embodiment of the present invention.

FIG. 5 is an example of a signal exchange procedure involved when a VoIP call is made as a calling part by a terminal capable of performing a VoIP call using a plurality of heterogeneous communication networks, according to an embodiment of the present invention. Is shown,

FIG. 6 exemplarily shows that, in the server of FIG. 4, components of a VoIP calling unit designated to be in charge of processing for one VoIP call and necessary information for call processing are set in each component. ,

7 and 8 briefly illustrate the structure of a voice packet carrying an audio frame for a VoIP call,

9 is a diagram schematically illustrating a process of dually transmitting the same voice packet to a plurality of wireless communication networks when another wireless communication network is available during a VoIP call using a single wireless communication network, according to an embodiment of the present invention.

FIG. 10 illustrates a process of relatively measuring a transmission delay time of a voice packet in each communication network in order to determine a comparative advantage in call quality for a plurality of wireless communication networks according to an embodiment of the present invention. Drawing for

FIG. 11 shows information on each time point of transmission and reception in a test block because a test block transmitted to grasp an absolute transmission delay time required for voice packet transmission between entities involved in a VoIP call is loopbacked at a receiving end. Shows an example in which

12 is a diagram illustrating the dual transmission of voice packets to another wireless network in order to ensure the stability of the call quality when the call quality of the wireless communication network currently used for the VoIP call path is lowered according to an embodiment of the present invention. The drawing is shown as

FIG. 13 is a diagram illustrating a method for switching a VoIP call to another wireless communication network while simultaneously transmitting a voice packet to another wireless communication network in a transient state in which a connection with the wireless communication network currently being used as a VoIP call path is released. Diagrammatically showing what is being done,

FIG. 14 is a diagram schematically illustrating voice packets being divided and transmitted to each wireless communication network when each current call quality supported by the plurality of wireless communication networks is stable, according to an embodiment of the present invention.

15 is a diagram schematically showing that voice packets are transmitted in duplicate or divided into each communication network, based on the call quality of each wireless communication network, according to an embodiment of the present invention.

FIG. 16 illustrates an example of a signal exchange procedure involved when a VoIP call is made as a called part by a terminal capable of performing a VoIP call using a plurality of heterogeneous communication networks, according to an embodiment of the present invention. It is shown.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

3 illustrates a configuration of an apparatus that can use IP address-based calls through a plurality of wireless communication networks according to an embodiment of the present invention. Referring to the configuration of the apparatus 100, a signal processor 21 for properly amplifying an analog signal and converting between an analog signal and a digital signal, and compressing digital voice data into audio frames and compressing the voice in an audio frame A vocoder 22 for extracting data and decompressing the data, a main control unit 20 for performing call processing for VoIP calls and obtaining control operations or information necessary for selective use of multiple calls; And a VoIP call section constituted by a distribution section 23 and an arrangement section 24. The distribution unit 23 configures audio frames including compressed voice data into packets for VoIP calls (hereinafter referred to as "voice packets"), and distributes them to the established call paths. The arranging unit 24 performs a function of arranging voice packets received from the established call path. Here, the "call path" is specified by an IP address and a port number respectively assigned to logical communication entities at both ends of transmitting and receiving voice packets, and according to the communication protocol for the occupation of physical communication resources between the two communication entities. Corresponds to the session established for both communication entities.

In addition, the apparatus 100 includes a data communication unit 25 performing a function according to protocols to be observed for data communication using a physical resource of a communication network, and a cellular mobile communication network in a cellular manner. (11a), for example, a cellular interface unit that encodes and modulates data according to a method specified by a 3G or 4G communication network (hereinafter referred to as a "cellular network"), and demodulates a radio signal to decode the encoded data. 26a and a high-speed wireless data network, for example, a Wi-Fi type wireless LAN network (hereinafter referred to as a "Wi-Fi network"). And a Wi-Fi interface unit 26b for demodulating the encoded data.

Each component illustrated in the configuration of the apparatus 100 may be implemented in hardware, or may be implemented in software, and of course, the hardware and software may be implemented in combination. In the following description, a software-implemented part, for example, an application, is stored in a mass storage means provided in a specific server, in a component of the apparatus 100 for a VoIP calling method selectively using a plurality of wireless communication networks. It can be executed after being downloaded to and installed on the communication terminal according to a normal on-line purchasing process performed by communication terminals interconnected through the communication network by means of communication also provided with the specific server. In this case, the communication terminal should be provided with ordinary hardware resources, operating systems, and the like, capable of executing program codes of software. When the installed software part is executed, the communication terminal includes the device 100 illustrated in FIG. 3 as a component together with the hardware resource provided.

In addition, the configuration of the device 100 is just one example for implementing a method of using a IP address-based call through a plurality of heterogeneous communication networks according to the present invention, the apparatus for performing a VoIP call is necessarily Figure 3 It does not have to be configured as For example, accessing a wireless communication terminal, such as a cellular network and a Wi-Fi network, which includes hardware, firmware and / or software that collectively or separately execute the functions performed by each component of FIG. 3. It may be a smart phone, a pad computer, or a notebook computer. Accordingly, in accordance with the present invention, an apparatus in which a method of using an IP address-based call through a plurality of heterogeneous communication networks is implemented may include other components (for example, a keypad, a display panel, GPS module, etc.) may be included. Hereinafter, for convenience of description, the term "VoIP terminal" is used as a meaning encompassing the device 100 of FIG. 3 or a wireless communication terminal of any type or any name including the components illustrated in FIG. do.

The server of the communication network processing the VoIP call call requested by the VoIP terminal 100 according to the present invention is configured as shown in FIG. 4 is a server 300 capable of supporting an IP address-based call through a plurality of wireless communication networks according to the present invention, hereinafter referred to as "dual-VoIP server". The binary-VoIP server 300 is connected to the VoIP terminal 100 through a wireless communication network, for example, a communication node and a gateway of a cellular network and a Wi-Fi network, It is possible to exchange data with a terminal communicating wirelessly with the wireless communication network. Of course, the binary-VoIP server 300 may be connected to a local network including the gateway.

The binary-VoIP server 300 includes a

network connection part

31a, 31b for performing physical connection with a communication network or a communication line and data exchange through the connection, respectively, and a VoIP call request from a terminal. Subscriber information necessary to open and manage a VoIP call path (e.g., identification information for the subscriber, preferred VoIP usage mode, plan information for the cellular network, and the like) And a subscriber db (30a) that contains a dynamically allocated IP address, etc. for that subscriber. The

network connections

31a and 31b are shown logically divided into first and second network connections in the drawings in order to clarify the flow of signals for VoIP calls and to help understand the operation. It can be configured integrally on the same hardware. In some cases, a network connection for exchanging data with a VoIP terminal and a network connection for exchanging data with a relay device on a communication network for a VoIP call may be provided as separate devices. In the description of the present invention, it is assumed that the first network connection portion 31a is provided for the former, and the second network connection portion 31b is provided for the latter.

On the other hand, the call control unit 30 may include a hardware element, or may be implemented only in software. In the latter case, the binary-VoIP server 300 includes a processor, which is naturally a hardware resource for executing it. Here, the processor refers to an entity configured in such a manner as to include a CPU, a memory resource, and a necessary peripheral for executing the given instruction codes. In addition, the call control section 30, to support the VoIP call between the two terminals, a VoIP call at the time necessary part _{(3 k, k = 1,2,3,} ..) ( namely, the call unit (3 _k Activate the elements belonging to)). The VoIP call part (3 _k) is to be a software executed by the processing and the "activation" term in this case are means you run a call to either "OK" or the execution code block, the software . If, in case that the VoIP call part (3 _k) is composed of only the hardware, the term of the "activation" is, the "to turn" on their hardware for such variables, or parameters needed to process packets for the VoIP call it means. The meaning of this "active" term is equally applied to the internal components of each VoIP call part (3 _k). In the illustrated binary-VoIP server 300 configured in FIG. 4, a function of performing VoIP call call processing and quality control of the call, and exchanging voice packets for VoIP call while selectively using a plurality of wireless communication networks It is not necessarily assumed that the components for the function, etc., are configured as one resource type, that is, software or hardware, and the implementing type or the type of the resource used and the like may be illustrated for the purposes of the present disclosure. It may be different. Therefore, the scope of analysis should not be excluded by the claims of the present invention because of different types of configurations or types of resources.

The VoIP call unit 3 _k , k = 1, 2, 3,... Is a combination of a transmitter and a receiver responsible for transmitting and receiving voice packets corresponding to each session for a VoIP call path with a terminal (hereinafter, ' The term "transmitter and receiver" is used to collectively refer to "transmitter" and "receiver", and the notation 's' or 'r' added to distinguish between 'transmit' and 'receive' is also indicated. For example, the notation of "3c _k " is added to the transmitter and receiver which are referred to as "3sc _k " and "3rc _k ". ((3sc _k , 3rc _k ), (3sw _k , 3rw _k )}, and an array unit 3dps _k for selectively arranging voice packets received and transmitted by the receiving unit 3rc _k or 3rw _k , and another device (VoIP packet relay device or other binary-VoIP). A distribution unit 3dtb _k for selectively distributing voice packets received from the server or the like) to the transmitter 3sc _k or 3sw _k . Foregoing the VoIP call unit activation of (3 _k) is that VoIP call part (3 _k) of transceiver pairs contained in (3c _{_k,} 3w _k), and the arrangement unit (3dps _k), and distribute the minutes ( 3dtb _k ) means activation.

Hereinafter, between the calling party and the called party terminal of the VoIP terminal 100 by mutual operation between the VoIP terminal 100 and the binary-VoIP server 300 illustrated in FIGS. 3 and 4, respectively. It will be described in detail with respect to the process of making a VoIP call according to the present invention.

5, the VoIP terminal 100 establishes a terminal 110 (hereinafter referred to as "called terminal") to which the VoIP terminal 100 is to be called, establishes a VoIP call path, and conducts a VoIP call through the established call path. The signal exchange flow for the process is shown as an example. In the example of FIG. 5, the called terminal 110 may or may not be a VoIP terminal including the configuration as shown in FIG. 3. In the former case, when the operation for establishing a VoIP call is completed, the VoIP call unit 3 _j , j ≠ i, 3 _i of the configuration illustrated in FIG. 4 is also transmitted to the called terminal 110. In case of an active VoIP call part for the terminal, the terminal is allocated and used.

First, the main controller 20 of the VoIP terminal 100 configures a VoIP connection request message by designating a called party according to a VoIP call request from a user through an appropriate input device. 25) request that transfer. The packet carrying this request message is, of course, destined for the binary-VoIP server 300 (more specifically, the binary-VoIP in the IP address field of the packet carrying the request message, for example a TCP packet). The destination is designated by describing the IP address of the server 300.), as illustrated in Fig. 5, an identifier call_ID ₀ (hereinafter abbreviated as "call identifier") for a requesting VoIP call. , The IP address IP11 which is the calling party identification information, the called party identification information dst_ID ₂ , and the local port number Lpn ₁₁ assigned to the call path to be opened. The data communication unit 25 is a wireless communication network that is currently connected, that is, a wireless communication network assigned an access IP address. (For convenience of description of the present invention, it is assumed that a wireless communication network requiring a first VoIP call is a cellular network. By sending the VoIP connection request message over an interface, e.g., the cellular interface 26a, the request message is sent to the binary-VoIP server 300 in routing by nodes establishing a communication network. It is reached (S401).

In one embodiment according to the present invention, the VoIP connection request message, information on the type of wireless communication network (for example, 3G or 4G cellular network, Wi-Fi network, etc.) to establish a VoIP call path Included. In order to include network information in such a call, the main control unit 20 obtains the information of the wireless communication network currently connected to the data communication unit 25 before constructing the VoIP connection request message. If a plurality of wireless communication networks, for example, a cellular network and a Wi-Fi network are both connected and receive information on the plurality of communication networks, the VoIP connection request message is selected to select one of the communication networks and indicate the communication network. When the data communication unit 25 transmits the VoIP connection request message to the data communication unit 25, the communication network selected by the data communication unit 25 is designated. If this designation is made, the data communication unit 25 transmits a VoIP connection request message to the designated wireless communication network. In addition, the above-mentioned calling party IP address IP11 also checks the IP address allocated from the currently connected wireless communication network or from a communication network selected from a plurality of wireless communication networks from the data communication unit 25 to perform the VoIP. Used for connection request messages.

Meanwhile, the VoIP connection request message reaching the binary-VoIP server 300 is transmitted by the call control unit 30 through the first network connection unit 31a. Then, the call control unit 30 extracts all or part of the information included in the received VoIP connection request message and registers it as one call entry in the VoIP call management list managed by the call controller 30. The entry listed in the VoIP call management list necessarily includes the aforementioned call identifier. The call controller 30 checks the called party identification information dst_ID ₂ included in the VoIP connection request message, and if the identification information is an IP address, that is, dst_ID ₂ = IP2, the destination is sent to the IP address. The VoIP connection request message is configured in the manner described above, and is transmitted through a network connection unit for transmitting data to the IP address, for example, the second network connection unit 31b (S402). Of course, the connection request message includes a local port number Lpn ₃₁ assigned by the call controller 30 to the call path. If the other party's identification information dst_ID ₂ of the received VoIP connection request message is not an IP address, for example, other types of identification information such as an e-mail address, the corresponding identification is performed in the subscriber db 30a of the user. It searches the information, finds the IP address listed in association with the identification information, and uses the IP address. The VoIP connection request message transmitted through the second network connection unit 31b is transmitted to the called terminal 110 via another relay device.

The called party terminal 110 displays the information necessary for the incoming response to the user in an appropriate manner according to the reception of the VoIP connection request message, and when the user accepts the incoming call, the " accepted " Message is transmitted to the sender of the connection request message, that is, the binary-VoIP server 300 (S403). The acceptance message for the VoIP call configured by the called terminal 110 includes a call identifier (call_ID ₀ ) for the call, and also includes a port number Rpn ₂₁ assigned by the called party as a remote port number. do. The acceptance message from the called terminal 110 arrives at the binary-VoIP server 300 and is transmitted to the call controller 30 by the second network connection 31b.

The call control unit 30, based on the call identifier (call_ID ₀ ) of the received acceptance message, grasps the information of the VoIP connection request message that it received and stored as an entry in the VoIP call management list. By constructing an acceptance message for the connection request from the grasped information and transmitting it as a response to the connection request to the calling terminal requesting the VoIP call, that is, the VoIP terminal 100 (S404), the VoIP call path becomes a cellular network. Open at. The acceptance message at this time includes a port number Rpn ₃₂ assigned by the call control unit 30 as a remote port number, and the acceptance message is transmitted through the first network connection unit 31a to pass through the cellular network. To the VoIP terminal 100. In addition, the call control unit 30 activates one of the VoIP call part (3 _L) to the processing in charge of the VoIP call with the transmission of the permission message and the information that can specify the activated which VoIP call unit, For the VoIP call currently in progress, the end-to-end connection is recorded in the corresponding information field of the entry listed in the VoIP call management list. Then, the notification of the port number assigned for its activation in which the VoIP call part (3 _L), IP address of the calling party and the called party associated with the acceptance VoIP call to, and the VoIP call connection by the component, Fig. Set 51 on each component as illustrated in FIG. When the port number and the like are set, the receiving unit 3rc _L and the distribution unit 3dtb _L intended to receive voice packets from the outside are assigned to the corresponding port numbers Lpn ₁₁ to the respective

network connection units

31a and 31b. , Rpn ₂₁ ) requests the creation of a receive socket. In addition, the transmitter 3sr _L also requests the first network connection unit 31a to generate and request a transmission socket for packet transmission. When the transmission socket is generated, the transmitter 3sr _L notifies the distribution unit 3dtb _{L that the} signal is “sendable.” do.

According to an embodiment of the present invention, when the VoIP connection request received from the terminal includes the information of the network in the call, in activating one VoIP call unit, the call control unit 30, the call path The IP address and port number are set by specifying a transceiver that is designated to handle the transmission and reception of data through the communication network corresponding to the network information. Since it is assumed that the VoIP terminal 100 requested the initial connection request of the VoIP call through the cellular network, FIG. 6 is provided to each of the transceivers 3c _L designated to handle the transmission and reception of data via the cellular network. The port number is set. Of course, for each of the pairs of transceivers 3c _k and 3w _{k of} each VoIP call unit 3 _k , k = 1, 2, 3,. have. In this case, the array unit (3dps _k) and the distribution of the VoIP call part (3 _k) if intended for the processing of data for adding transceiver which sets the current IP address and port number via which the wireless communication network (3dtb _k Will be notified).

On the other hand, the accept message of the connection request sent to the VoIP terminal 100 is received by the cellular interface 26a via a cellular network according to the assumption of the previously used communication network, and the cellular interface 26a is received. The acceptance message is extracted from the received signal and transmitted to the data communication unit 25. The main control unit 20 that has received the acceptance message from the data communication unit 25 indicates the local port number Lpn ₁₁ of the VoIP connection request message sent by the data communication unit 25 and the IP address IP3 of the destination of the message. The distribution unit 23 is notified and set, and the remote port number Rpn ₃₂ included in the received acceptance message is notified to the arranging unit 24 and set. Further, information on the wireless communication network to be used is also notified to the distribution section 23 and the arrangement section 24, respectively. Of course, the wireless communication network to be used is the communication network that transmitted the VoIP connection request message (previously assumed to be a cellular network). The main control unit 20 then records, as the VoIP call information, information including the call mode and the call identifier for the currently accepted VoIP connection. The call mode is divided into "single path" using a single communication network and "multipath" using multiple communication networks. In the case of a single path, the type of communication network used as the path is also recorded. In addition, the main controller 20 indicates that the VoIP connection request has been accepted, by displaying on the output device provided in the VoIP terminal 100, for example, a displayer (not shown), allowing the user to make a VoIP call. You will be notified that it has been opened. The arranging unit 24 intended to receive a voice packet from the outside, when the remote port number is set together with the notification of the communication network to be used from the main control unit 20, the data communication unit as the set remote port number (Rpn ₃₂ ) Ask 25 to create a reception socket for the cellular network.

A user who recognizes the establishment of a VoIP call from the information displayed on the displayer inputs his or her own voice through a microphone, and the input voice signal is transmitted to the audio frame by the vocoder 22 via the signal processing unit 21. Is converted into a sequence. Each audio frame is composed of compressed voice data of a certain length of time, for example, several tens of msec, and includes information on compression of audio in a frame header. An audio frame sequence is applied to the distribution unit 23, and the distribution unit 23 configures each piece of the entire or divided frame of each audio frame as a voice packet, and the packet is UDP (User Datagram). Protocol). As illustrated in FIG. 7, in the header of each voice packet, an IP address IP3 and a local port number Lpn ₁₁ of a destination notified and set by the main control unit 20 are written (601), In the data field (payload) of the packet, the sequence number 61 for the portion of the audio frame carried in the packet is written. The sequence number is a value assigned to the corresponding audio frame part in the order inputted from the vocoder 22. In one embodiment according to the present invention, as illustrated in FIG. 8, a call identifier 62 initially assigned to a currently established VoIP call path may be inserted into a data field (payload). In the present embodiment, the call identifier is notified from the main control unit 20. The structure of the packet illustrated in FIGS. 7 and 8 is a simplified diagram showing only information elements of a packet or data field necessary for explanation and understanding of the present invention, and naturally includes information elements for configuration of a conventional packet and data field. It is configured.

On the other hand, the distribution unit 23 generates a transmission socket by requesting the data communication unit 25 for a wireless communication network, that is, a cellular network that has been notified for use, for transmitting voice packets constituting its own. When the voice packets are configured as described above, the voice packets are transmitted to the data communication unit 25 through the generated socket for the cellular network, and the transmission is requested. In another embodiment according to the present invention, when generating the transmission socket, by providing the IP address and the port number of the destination, the distribution unit 23 constitutes only a data field (in some embodiments, a call identifier is included. And a voice packet in which the data communication unit 25 adds an IP address and a port number associated with the transmission socket to the data block received through the transmission socket. The transmission is made upward through the interface 26a.

In another embodiment according to the present invention, the voice packet may be configured in a format according to TCP (Transfer Control Protocol) rather than UDP. Of course, in this case, the IP address and port number of the side sending the voice packet will be inserted according to the format. In addition, a socket corresponding to the format is also generated between the data communication section 25 and the distribution section 23 / array section 24 in order to transmit and receive a packet in TCP format.

Voice packets transmitted upward through the cellular interface 26a arrive at the binary-VoIP server 300 via a cellular network and, by the first network connection 31a, are located in the header of the voice packets. Voice packets which are temporarily stored in the receiving socket generated for the port number Lpn ₁₁ and accumulated in the receiving socket are read by the receiving unit 3rc _L , so that the data block of each packet (i.e., data in the data field) is stored. It is extracted and delivered to the arrangement portion 3dps _L. The arranging unit 3dps _L constructs a voice packet having a packet header of an IP address and a port number set to itself for each received data block to the second network connection unit 31b through the previously-generated transmission socket. The second network connection unit 31b transmits the received voice packet to a connected normal wired communication network. Thus, by the binary-VoIP server 300, the header information 601 of the received packet is changed into the information 602 toward the called party of the VoIP call as illustrated in FIG. 7 and finally via the communication network. It is delivered to the called terminal 110. According to an embodiment of the present invention, when a data field includes a call identifier, as illustrated in FIG. 8, the call identifier 62 is removed and directed to the called terminal 110. May be converted to

The voice packets delivered to the called terminal 110 are output as voice after appropriate processing in the terminal, and the contents of the voice of the user speaking in response to the voice are processed by the called terminal 110. The means are converted into voice packets and transmitted to the communication network. The receiving terminal 110 receives and processes the voice packet and configures the input voice into the voice packet according to a conventional method, or the receiving terminal 110 is illustrated in FIG. 3. In this case, since the transmission and reception of the voice packet are performed according to the above-described process and the process described later, a detailed description of such an operation in the called terminal 110 will be omitted here. The voice packets transmitted to the communication network by the called terminal 110 are accepted by the called terminal 110 in the header of the binary-VoIP server 300 with the IP address IP3 of the connection request. The remote port number Rpn ₂₁ delivered to the other party in S403 is described. Accordingly, the voice packets transmitted from the called terminal 110 arrive at the binary-VoIP server 300, which are transmitted by the second network connection part 31b to the port number Rpn ₂₁ of the packet header. After being temporarily stored in the receiving socket created for the read socket, the distribution unit 3dtb _L monitors the receiving socket. The distribution unit 3dtb _L extracts a data block (i.e., data in a data field) from each voice packet read from the reception socket, and delivers the data block to the transmission unit 3sc _L which has notified its current transmission status. do. In the VoIP call part (3 _L) is currently active, both the transmitter (3sc _{_L,} 3sw _L) are both used, which may be a case where the notification that the "transmission state", In this case, the distributor (3dtb _L) the transmitter Either or both transmitters may be selectively used. This optional method of use is described in detail later.

On the other hand, when the data blocks in the packet are received from the distribution unit 3dtb _L according to the above-described process, the transmitter 3sc _L , with respect to each of the received data blocks, is configured with the IP address IP11 set to itself. After configuring a complete voice packet by adding a packet header including the port number Rpn32, the packet is transmitted to the first network connection part 31a through the transmission socket created at the time of activation and requested for transmission. According to an embodiment of the present invention, when the call identifier is included in the data field of the voice packet transmitted by the VoIP terminal 100, the transmitter 3sc _L also assigns the call identifier to the voice packet configured by itself. Will be inserted. Of course, this call identifier is notified from the call control unit 30. The configuration of the packet header as described above may be performed by the first network connection unit 31a. To this end, when the transmission unit 3sc _L requests the first network connection unit 31a to create a transmission socket, the transmitter 3sc _L informs of an IP address and a port number to be added to the data block transmitted through the socket. The voice packets delivered to the first network connection part 31a are transmitted downward to the VoIP terminal 100 via a cellular network and thus received by the cellular interface 26a and then in the form of voice packets. Applied to the data communication unit 25.

The data communication unit 25 temporarily stores the voice packets in the reception buffer generated for the port number based on the port number included in each header of the authorized voice packets. Then, the arranging unit 24 reads out the voice packets from the receiving buffer and temporarily stores them in data blocks, and sequentially in accordance with the sequence number described in each data block regardless of the receiving order. An audio frame portion is applied to the vocoder 22. If the portion of the audio frame is not for one complete audio frame, the next block of data is waited until the amount of buffering by the audio frame applied to the vocoder 22 is exhausted. If the temporarily stored data block has a sequence number that precedes the sequence number of the data block previously transmitted to the vocoder 22, the data block is ignored and not transmitted to the vocoder 22. The vocoder 22 decodes each part of the audio frame applied to itself and applies decompressed digital voice data to the signal processor 21, and the signal processor 21 applies the digital voice data applied thereto. After converting to an analog signal and amplified properly, the speaker or headphones can be output so that the user can listen to the voice.

By the above-described method, voice data according to a voice signal is transferred between the calling party's VoIP terminal 100 and the called party's terminal 110 through a packet and transferred to the other party, thereby making a VoIP call between both parties ( S405). Of course, the VoIP call at this time is made by a cellular network based call path according to the above assumption.

When the VoIP terminal 100 enters a serviceable area of another wireless communication network, for example, a Wi-Fi network, during a VoIP call, the Wi-Fi interface unit 26b receives the status. Will be detected. For example, it is possible to receive a radio signal from an access point (AP) of a Wi-Fi network, and the state is notified to the main controller 20 through the data communication unit 25. Then, the main controller 20 checks whether there is a VoIP call currently being made through another communication network, for example, a cellular network, from the recorded VoIP call information, and if there is a VoIP call, the data communication unit 25 The request is made to connect to the access point of the Wi-Fi network (S411). If the communication mode of the recorded VoIP call information is "single path" and the communication network used is not a Wi-Fi network, a supplementary connection request message is configured. The supplementary connection request message configured by the main controller 20 includes a call identifier (call_ID ₀ ) in the VoIP call information, an IP address (IP12) allocated from a newly connected Wi-Fi network, and a supplementary call path. (The term "complementary channel" is currently used only as an additionally established meaning for a call path for a VoIP call, but not as a meaning limited to that generated for a Wi-Fi network.) The assigned local port number (Lpn ₁₂ ) is included. As in the initial connection request message, the called party identification information dst_ID ₂ may be further included. It may also include information about the type of communication network used for the supplementary channel. The main control unit 20 requests the transmission while designating a communication network to transmit the complementary connection request message configured as described above to the data communication unit 25, and in response to the request, the data communication unit 25 sends a supplementary connection request message. Wireless transmission through the Wi-Fi interface unit 26b. The supplementary connection request message transmitted as a wireless signal arrives at the binary-VoIP server 300 via a Wi-Fi network (S412).

The supplementary connection request message reaching the binary-VoIP server 300 is transmitted to the call control unit 30 by the first network connection unit 31a. The call control unit 30 checks the call identifier (call_ID ₀ ) included in the message, and checks whether an entry including the call identifier is listed in the VoIP call management list managed by the call controller. If it is listed, newly by the port number, the complement connection request from the VoIP call part (3 _L) is assigned to (Rpn ₃₃₎ and indicated by the identification information contained in that entry of the communication network used by a call The transmitter / receiver 3w _L designated to handle transmission / reception is notified of the assigned port number Rpn ₃₃ and the originating IP address IP12 and the port number Lpn ₁₂ included in the supplementary connection request message. Set as in 6 (52). As another method of setting the IP address and port number, while setting to a valid IP address and port number from the VoIP call part (3 _L) notifies the transmitting and receiving unit that is not currently set, the data to the receiving portion via a certain communication network whether of the intended processing is notified to the VoIP array unit (3dps _L) and distribution (3dtb _L) of the call unit (3 _L). New IP address and the transmitter (3sw _L) port number is set, the receiving unit (3rw _L) is, in the same manner as that of the transmitter (3sc _L) and the receiver (3rc _L) is set such as above IP address had done, the first The transmission socket and the reception socket are respectively generated in the network connection part 31a, and the transmission part 3sw _L notifies the distribution part 3dtb _L that it is a "transmission possible state".

In the transceiver pairs 3c _L and 3w _L , a transceiver configured for data transmission and reception with a Wi-Fi network is provided with a new IP address and a port number in the state where an IP address and a port number are currently set. May be notified. This will be described later in detail, but may occur when the VoIP terminal 100 makes a supplementary connection request through the same access point or another access point after the connection with the Wi-Fi network is made and terminated during the VoIP call. . When the new IP address and the port number are notified from the call controller 30 while the currently set IP address and the port number are set as described above, the corresponding transceiver is the socket previously created for the first network connection 31a. They will close and create sockets for sending and receiving with the new IP address and port number advertised.

There may be various ways in which the call controller 30 determines which communication network the supplementary connection request message is for using as a call path. If the supplementary connection request message contains the information of the call route network, the information may be used. If there are two network types available for the VoIP call route, the information is designated for the same call identifier listed in the VoIP call management list. It can also be grasped from network information. In other words, the remaining communication network other than the communication network recorded in the entry is regarded as the communication network to be used by the complementary connection. Alternatively, the connection request message may be distinguished from each other according to the type of communication network to be used for the VoIP call. Of course, the type of the connection request message may be distinguished by a value or a flag described in an attribute field separately designated in the same connection request message.

The main controller 30 configures an acceptance message for the received supplemental connection request and transmits it through the first network connection unit 31a after setting the IP address and the like to the above-described transmission and reception unit. Unlike the original VoIP connection request, for the complementary connection request, the processes S402 and S403 for call processing with the called terminal 110 are not performed. The acceptance message includes a call identifier (call_ID ₀ ) included in the supplementary connection request message and the port number (Rpn ₃₃ ) newly allocated for the supplemental connection.

The transmitted acceptance message arrives at the VoIP terminal 100 via a Wi-Fi network (S413) and is received by the data communication unit 25 via the Wi-Fi interface unit 26b. The data communication unit 25 transmits the received acceptance message to the main control unit 20, and the main control unit 20 having received the acceptance message transmits the local port number (Lpn) of the complementary connection request message transmitted by the data communication unit 25. ₁₂ ) and the IP address IP3 of the destination of the request message is set by notifying the distribution unit 23, and the remote port number Rpn ₃₃ included in the received acceptance message is arranged in the array unit 24. The distribution unit 23 and the arranging unit 24 also notify the designated information of the wireless communication network, that is, the Wi-Fi network, to be used for the supplementary communication path. In response to the notification, the distribution unit 23 and the arrangement unit 24 request the data communication unit 25 to generate a transmission socket and a reception socket for the Wi-Fi network. Of course, the remote port number (Rpn ₃₃ ) set to itself is used for the request for creating the reception socket. The main control unit 20 then records and changes the call mode to "multipath" in the current VoIP call information.

After generating the transmission socket for the Wi-Fi network, the distribution unit 23 stores the voice packets configured for the audio frame sequence input from the vocoder 22, as shown in FIG. Dual uplink transmission. That is, the same voice data is repeatedly transmitted upward through the cellular network and the Wi-Fi network (71). Of course, the port numbers of the voice packets carrying voice data transmitted in duplicate are different from each other. In the example of the present description, the voice packet transmitted through the cellular network previously used as a call path includes the port number Rpn ₃₂ in the header of the packet, and the voice packet transmitted through the Wi-Fi network used as the new call path. Port number Rpn ₃₃ is recorded in the header of the packet. The voice packet is transmitted in UDP type as mentioned above, which, unlike the TCP type, may be lost without guarantee of its transmission on a communication network. Therefore, if the data is transmitted in duplicate through different paths, the packet loss rate is reduced as a whole because lost packets on one path can be preserved on the other path.

As described above, voice packets that are dually transmitted upward through a plurality of communication networks are different from each other by the port number recorded in the header of each packet by the first network connection part 31a of the binary-VoIP server 300. Both receivers 3rc _L and 3rw _L , which are classified and stored in the reception buffer and requested to generate each reception buffer, read voice packets from the corresponding reception buffer and transfer the data blocks to the array unit 3dps _L , respectively. do. Then, the arranging unit 3dps _L is composed of a voice packet only for the first arrival in the data blocks respectively received from both receiving units 3rc _L and 3rw _L , and is transmitted to the second network connection unit 31b. The call is transmitted to the called terminal 110 through a socket. In other words, if a data block having the same sequence number as the sequence number already transmitted to the called terminal 110 is received so that the duplicately received data block is not duplicated and delivered to the called party, the data block is described in the communication network described below. It is used to determine the quality of the product and then discarded.

On the other hand, when the data block is received from both receivers 3rc _L and 3rw _L , the arrangement unit 3dps _L periodically determines which path, i.e., the packet via the wireless communication network, is better in voice quality. do. In order to determine the relative comparative advantage, the array unit 3dps _L is received from both receivers 3rc _L and 3rw _{L at} each time, for a predetermined time T _QchkInt , as illustrated in FIG. 10. The difference T _{jit_k} between the arrival times of the data blocks of the same sequence number is determined. Then, from the sum of the identified time difference T _{jit_k} , it is determined which data block received from which receiving unit arrives on average faster. Further, since the array unit 3dps _L knows which communication network each of the receivers 3rc _L and 3rw _L is, the receiving unit delivering a data block that arrives on average is faster to receive and process. This is judged to present better call quality. In the example of FIG. 10, the sum of the identified time differences T _{jit_k} will be minus, so that the wireless communication network designated for the receiver 3rw _L , for example, the Wi-Fi network, is better at this point. It is determined that the call quality.

In another embodiment according to the present invention, the relative call quality for each wireless communication network may be determined from other reception characteristics, for example, packet loss rate. The array unit 3dps _L grasps the ratio of the number of missing sequence numbers in the sequence numbers of the data blocks received from the receivers 3rc _L and 3rw _{L for} each predetermined time T _QchkInt . Then, it is determined that the communication path through which the receiver having the smaller ratio processes data, that is, the wireless communication network provides relatively higher quality in the VoIP call.

In one embodiment according to the present invention, it is possible to determine which wireless communication network has a relatively better quality for VoIP calls by combining the evaluation according to the transmission delay time and the evaluation by packet loss rate.

The arranging unit 3dps _L informs the distribution unit 3dtb _L of the network information when determining the wireless communication network which has the current comparative advantage in call quality by any of the above-described methods. Alternatively, by providing the information and the network information for the relative quality determination identified for each of the wireless communication network to said distributor (3dtb _L) in consideration of the distribution (3dtb _L) a variety of conditions and parameters in the VoIP call It is also possible to determine which network is more advantageous to the user. The same applies to the absolute quality check for the wireless communication network described below.

On the other hand, the distribution unit (3dtb _L ), if there is a notification of the network information according to the determination of the relative comparative advantage, or if there is a predetermined number or more of network information notification, the relative wireless communication network, or a predetermined number or more notifications relative To the wireless communication network notified more, so that voice packets received from the called terminal 110 are transmitted downward. In other words, each data block extracted from the voice packets received by the second network connection unit 31b is transmitted to a transmission unit for which transmission processing for the corresponding wireless communication network is designated. If a plurality of VoIP call paths are established between the VoIP terminal 100 and the binary-VoIP server 300, if the Wi-Fi network has a better call quality than the cellular network, the incoming call is received. The data blocks of voice packets received from the side terminal 110 are switched to the transmitter 3sw _L by the distribution unit 3dtb _L so as to pass through the Wi-Fi network. In this case, the IP address IP12 and the port number Rpn ₃₃ set in the transmitting unit 3sw _L are added as headers to each data block, and the VoIP terminal 100 is connected via the Wi-Fi network in the form of a voice packet. Is transmitted downward. If, until now, the cellular network is still in a better state of call quality, the delivery destination is not switched and delivery to the transmitter 3sc _L continues. Of course, at some point through repeated call quality evaluation, the destination may be switched to use the Wi-Fi network.

As described above, the operation of evaluating the call quality in each wireless communication network through the received voice packet and selectively transmitting the voice packets using the wireless communication network according to the evaluation may be performed by a plurality of VoIP call paths. In the opened state, it is continuous. In this state, as illustrated in FIG. 4, a cellular network call path 414a and a Wi-Fi network call path 414b are connected between the binary-VoIP server 300 and the VoIP terminal 100. Alternatively, or both paths are used in combination, voice packets are transferred between the calling party and the called party (S414).

According to another embodiment according to the present invention, the VoIP call part (3 _L) in provided in place of, that the VoIP terminal 100 is directly measured to measure the quality of the call path to each wireless communication network, the current call Quality variable information for the formed wireless communication network may be used. The quality variable includes, in addition to the transmission delay time and the packet loss rate mentioned above, the data reception speed, the strength of the received signal, etc. which the terminal can measure. A method of measuring or confirming values for the quality variables by the terminal will be described later.

In another embodiment according to the present invention, quality variable information may be obtained from a specific external server. In the present embodiment, when the VoIP terminal 100 makes a supplementary connection request, a specific information field in the request message informs the current location, for example, a service area of a cellular network or a Wi-Fi network. The binary-VoIP server receiving identification information (such as base station ID or access point MAC address) or longitude coordinates (if a GPS module is obtained from the satellite signal to obtain longitude coordinates) and transmitting the supplementary connection request is received. call control section (300) (30), complementary connecting the service area identification information and transmits a permission message for a request to the VoIP terminal 100, the VoIP call part (3 _L), which processes the VoIP call The distribution unit (3dtb _L ) of will be notified. Then, the distribution unit 3dtb _L acquires and uses the information on the call quality of the corresponding service area of each wireless communication network while providing the service area identification information to the predetermined external server through the data communication unit 25. do. Of course, in this case, a socket other than a socket for transmitting and receiving voice packets, for example, a TCP type socket is created between the data communication unit 25 and the socket is used. In this embodiment, the VoIP terminal 100 changes the service area when the service area is changed for a cellular network which is a wireless communication network capable of mobile communication (i.e., handover between service areas). Identification information is provided to the binary-VoIP server 300. The cellular interface unit 26a detects the change of the service area, and the main control unit 20 that has confirmed the fact through the data communication unit 25 receives the changed service area identification information from the cellular interface unit 26a. after confirming, adding a service area identity, a call identifier (call_ID ₀₎ of the currently connected to the VoIP call is transmitted to the two won -VoIP server 300 by configuring the VoIP call environment information message. This environmental information message is transmitted to the call control unit 30 similarly to the VoIP call processing messages, and the call control unit 30 extracts service area identification information from the received environmental information message, and also the call identifier of the message. It specifies a VoIP call part (3 _L) applicable on the basis of (call_ID ₀₎ is notified to the distributor (3dtb _L) of the VoIP call part (3 _L). Then, the distribution unit 3dtb _L periodically acquires quality variable information related to the area by using the service area identification information notified of the change. The distributor (3dtb _L) without obtaining a quality variable information said call control section (30) may be periodically obtained by the information to be provided to the distributor (3dtb _L). In the present embodiment, the external server receives a communication quality (for example, communication characteristic values such as data reception speed, transmission delay time, packet loss rate, etc.) grasped during data service usage in each service area from numerous wireless communication terminals. This is registered in the database, and when there is a request for the quality variable information as described above, the quality information statistically confirmed for the corresponding service area (each communication characteristic value, for example, or a quality combined with each weighted value thereof) Scores).

In the state where a plurality of VoIP call paths are established, while only one Wi-Fi network is used as a transmission path to the VoIP terminal 100 for voice packets received from the called terminal 110, The allocation 3dtb _L configures a series of test blocks to confirm the absolute transmission quality of the VoIP call of the path, and provides the transmission unit 3sw _L to which the transmission process to the Wi-Fi network is designated. Perform periodically. The test blocks provided in this way are each configured by the transmitter 3sw _{L into} test packets of the same format as the voice packet (for example, UDP), and the VoIP terminal 100 through the Wi-Fi network. Is sent to. The test block carried by the test packet includes type information indicating that the block is a test block and time information at the time of transmission. When a plurality of test blocks are transmitted instead of a single test block, a number indicating the order may be included in each block.

A series of test packets (configured or transmitted as described above) may be received by the Wi-Fi interface unit 26b of the VoIP terminal 100 and communicated with the data communication unit 25. It is transmitted to the array unit 24 through the Wi-Fi network receiving socket generated in the. If it is determined that the data block carried by the packet is a test block from the type information, the arranging unit 24 records the current time as reception time information following the transmission time information described in the test block, and then the distribution unit ( 23, and designates the transmission to the received wireless communication network, that is, the Wi-Fi network. According to this transmission designation, when the transmission time point is available for transmission, the distribution unit 23 adds the reception time information to the received test block following the reception time information, and configures the data in the form of a packet. Transfer to the communication unit 25 through the Wi-Fi transmission socket. The test packet configured as described above is received by the receiving unit 3rw _L designated to receive the packet via the Wi-Fi network through the first network connection unit 31a again through the binary-VoIP server 300. The test block is passed to the array section 3dps _L. It identifies that it is a test block from the type information of the data block received by the array unit 3dps _L , and adds time information on the time point of receiving the test block. As such, the format of the final information of the test block circulating through the channel may be as illustrated in FIG. 11.

The arranging unit 3dps _L transfers a test block having an information format as illustrated in FIG. 11 to the distributing unit 3dtb _L , and the distributing unit 3dtb _L transmits each time information described in the received test block. Obtain the transmission delay time from For example, the average time required for each time (T _travers1 = t _r1 -t _s1 , T _travers2 = t _r2 -t _s2 ) obtained from each transmission / reception time information pair (91 ₁ , 91 ₂ ) {= (T _{From travers1} + T _travers2 ) / 2} we confirm the absolute quality of the VoIP call. In one embodiment according to the present invention, the absolute quality of the VoIP call based only on the time required on the downlink path (T _travers1 = t _r1 -t _s1 ) from the binary-VoIP server 300 to the VoIP terminal 100. You can also check The distribution unit 3dtb _L , if the absolute quality identified in the above manner, or the average value of the absolute quality confirmed in the above manner several times is lower than any predetermined lower reference value, the called party terminal 110 Rather than having the voice packets received from the downlink transmitted only to the Wi-Fi network, the current channel is transmitted down through both established wireless communication networks. That is, by repeatedly providing the data blocks carried by any voice packet received to the called terminal 110 to both transmitters 3sw _L and 3sc _L , voice packets of the same voice data are transmitted through both wireless communication networks. To be. In this process, each voice packet redundantly transmitted through both wireless communication networks may not carry an audio frame or a corresponding portion in the same format. For example, audio frames included in N voice packets transmitted over a Wi-Fi network may be transmitted in M (≠ N) voice packets over a cellular network. That is, the distribution unit 3dtb _{L configures} N data blocks to be appropriately divided and / or combined for the N data blocks to be transmitted to the Wi-Fi network and provides them to the transmitter 3sc _{L for} the cellular network. It may be. This also applies to the following description.

In another embodiment according to the present invention, instead of actively checking the absolute quality of each call through a test block, the absolute quality of voice packets transmitted and received from the other party to the corresponding call, for example, a packet loss rate. Or the like can be measured manually. Such passive quality measurement may be performed only for one of the communication paths, or in some cases, for all of the plurality of communication paths, since the voice packet should be transmitted from the counterpart to the corresponding call. The distribution unit 3dtb _L receiving the information on the quality of the call measured by the arranging unit 3dps _L in the passive manner as described above, the information is the voice currently received from the called terminal 110. If the quality of the call route used solely for the downlink transmission of the packets is lower than the predetermined lower reference value, the same voice data is transmitted through both wireless communication networks as described above. This passive quality measurement method may be performed in the arrangement unit 24 of the VoIP terminal 100, and based on the call path quality measured by the passive method, the distribution unit 23 performs a single call. It is also possible to switch from upstream transmission of voice packets over a road to dual upstream transmission of voice data over multiple call paths.

Even during dual transmission of voice packets of the same voice data through both wireless networks, the call quality of the Wi-Fi network is measured through the test block as described above (in accordance with an embodiment of the present invention, In the same way for the network, the call quality through the test block can be measured intermittently.) When the quality exceeds the lower threshold, voice packets are transmitted downlink only to a single communication network, that is, the Wi-Fi network. 12 is a schematic diagram illustrating that voice packets are delivered to the VoIP terminal 100 by using both networks as appropriate downlink paths based on the absolute quality of the Wi-Fi network, for example, transmission delay time, according to this scheme. To show. In this case, the lower reference value 1002 may be determined as a nominal time required for transmitting a packet between a server and a terminal in an average load state of a cellular network, particularly a 3G cellular network. Alternatively, the lower reference value 1002 may be set as a packet propagation time between the server and the terminal, which is shown by the cellular network on average.

As illustrated in FIG. 12, when the call quality 1001 of the Wi-Fi network determined as described above becomes less than or equal to a predetermined lower reference value 1002 (for example, a transmission delay time becomes larger than the lower reference value) In the liver (T _worse1 , T _worse2 ), the same voice packets over the cellular network (in the example of the figure, the sequence number is p, p + 1, p + 2, q + 3, q + 4, q + 5 Dual transmission of packets) complements the cellular network. That is, the call quality can be improved by shortening or supplementing the transmission delay or packet loss part due to the current low-quality Wi-Fi network by the packets dually transmitted through the cellular network.

In the Wi-Fi network, due to the limited factors such as the service area is scattered in the characteristics and the use distance is very short, the data service for the Wi-Fi network is free for users. It is made available. In addition, small businesses have installed Wi-Fi networks for business and provide them to users free of charge. Individuals can install Wi-Fi networks in their homes or offices for free or at a fixed cost. Therefore, by using only the Wi-Fi network in a state in which a plurality of call paths (that is, sessions) are established on the plurality of wireless communication networks including the Wi-Fi network, no additional burden is incurred on the user. In areas where call quality over the Wi-Fi network is considered unacceptable to the user, or below the normal sound quality level over the cellular network, the cellular network must be supplemented to compensate for the cost. This ensures that the call quality is more than determined by the cellular network, which has relatively very stable service characteristics in a wider service area. In conclusion, if the current quality of the Wi-Fi network is an appropriate level, you can enjoy the call without any cost, but if not, the call quality does not occur even if the user bears some cost. The selective use of such a wireless communication network in VoIP calls is because a plurality of communication networks that can be used as call paths have a cost difference in the use of data services. The selective use of communication paths on the plurality of wireless communication networks corresponds to operation according to the " differential mode " As described above, in the case where it is designated that there is a difference in the use of a plurality of communication networks, even if both the Wi-Fi network and the cellular network show quality above the lower reference value 1002, as described above, it is costly to the VoIP user. The voice packets are transmitted downward using only the channel established for the more advantageous Wi-Fi network. If it is designated that there is no difference, it may be possible to use only a call path established in a wireless communication network having better quality.

As discussed above, be the communication quality of Wi-Fi networks group is below a specified lower threshold value in which the downlink transmission of voice packets in duplicate through the cellular network, the VoIP call part (3 _L) to the VoIP terminal 100, , if the upstream transmission of voice packets in duplicate through FIG both network the VoIP terminal 100, as the VoIP a call unit (3 _L), the above-mentioned, shows a relatively higher quality with respect to the path of the two networks If it is determined that the Wi-Fi network is relatively better, the complementary transmission of voice packets over the cellular network may be stopped.

In the above embodiment, the absolute quality of the VoIP call over the Wi-Fi network has been described using the transmission delay time as an example, but the packet loss rate may be used as in the relative quality comparison. In this case, the distribution unit 3dtb _L does not separately generate a test block, and the loss rate (ie, data) of the voice packets transmitted upward by the distribution unit 23 of the VoIP terminal 100 through the Wi-Fi network. It is also possible to determine the complementary use of the cellular network from the loss rate by identifying the defect rate of the sequence number of the block in real time.

On the other hand, the distribution unit 23 of the VoIP terminal 100, the Wi-Fi network is newly connected to transmit the voice packets to the binary-VoIP server 300 using both communication networks as shown in FIG. If (71), if the binary-VoIP server 300 is dual transmission of the voice packets generated in the called terminal 110 by using both communication networks, the arrangement unit 24, the binary As described with respect to the VoIP server 300, it determines the superiority of the relative call quality for both communication networks and notifies the distribution unit 23, and the distribution unit 23 sends the notification to the Wi-Fi network. If this relatively better quality is achieved, then dual transmissions over both networks will be stopped and voice packets will be sent upstream using only the Wi-Fi network. Of course, if the binary-VoIP server 300 is transmitting downlink voice packets using only the Wi-Fi network, the distribution unit 23, as described for the binary-VoIP server 300 In addition, the call quality obtained by measuring the absolute transmission delay time of the Wi-Fi network using the test block or periodically checking the loss rate of the received packets measured by the array unit 24 is compared with the preset lower reference value, If the call quality of the Fi network is higher than the lower threshold, duplex transmission through the cellular network is stopped. By such a process, when the VoIP terminal 100 switches the VoIP call to the communication network by additionally making the Wi-Fi network available during the VoIP call using the cellular network, call disconnection does not occur, of course. When a VoIP call over a newly connected Wi-Fi network reaches an appropriate quality or a better quality than a VoIP call over a cellular network, the Wi-Fi network is switched to exclusive use. As a result, the user can suppress the cost incurred by the VoIP call without incurring a separate terminal operation or effort, and can experience better call quality.

Even when switching to the single use of the Wi-Fi network as described above, the distribution unit 23 continuously checks the call quality through the Wi-Fi network, and in the time period when the quality is below the predetermined lower reference value 12 as shown in FIG. 12 (the example of FIG. 12 is for a downlink path between the binary-VoIP server 300 and the VoIP terminal 100.) using both networks as an uplink path. do.

On the other hand, in any call path established in a cellular network or a Wi-Fi network, the transmission path and the reception path may have independent characteristics without being correlated with respect to quality. That is, even if the quality of the reception path is good, the transmission path may have poor quality irrespective of it. As described above, even when the transmission path and the reception path have independent quality characteristics, the quality confirmation of the communication path through the test packet as described above may be applied as it is. For example, from the time information of each reception / transmission described in the test block received from the counterpart by the binary-VoIP server 300 or the VoIP terminal 100, the quality of the transmission path to which the packet is transmitted is checked. Can be. Depending on the quality of the transmission path thus identified, the voice packets may be transmitted using one channel in both established channels, or the voice packets may be duplicated or divided in both channels using both channels. do.

As described above, when the call quality through the Wi-Fi network is less than the predetermined lower threshold value, by using the cellular network complementarily, as illustrated in FIG. 13, the VoIP terminal 100 of the user As you move away from the area where you currently have access to the Wi-Fi network (111), the resulting call quality (1101) gradually decreases (e.g., because the transmission delay is longer). Since the uplink transmission of voice packets is naturally switched to the cellular network, and the binary-VoIP server 300 is similarly switched to the cellular network for the downlink transmission according to the quality of the call, the VoIP was performed through the Wi-Fi network. The call is naturally converted to a VoIP call over the cellular network without disconnection or voice loss (1102). When the signal reception threshold of the Wi-Fi network is out of the threshold region (1103), the Wi-Fi interface unit 26b of the VoIP terminal 100 detects a signal reception impossible state, and the data communication unit 25 Is informed). Then, the data communication unit 25 releases both the transmission socket and the reception socket generated for the current Wi-Fi network, and notifies the main controller 20 of the termination of the Wi-Fi network connection. When the transmission socket is terminated, the distribution unit 23 stops the transmission of the voice packet through the transmission socket 1104 (that is, the transmission through the Wi-Fi network is stopped). As a result, the call quality check for the Wi-Fi network is also stopped (1105). On the other hand, the main controller 20 notified of the termination of the Wi-Fi network connection corrects the VoIP call information for the current call. That is, the communication mode is recorded as a "single path" and the communication network used as a cellular network.

In one embodiment according to the present invention, the distribution unit 23 of the VoIP terminal 100 periodically checks the values for the quality variables as described above to configure quality variable information to configure the binary-VoIP server. It provides the corresponding VoIP call part (3 _L) of 300. The measurement or confirmation method for the transmission delay time and the packet loss rate belonging to the quality variable is performed as described above, and other variables, for example, a value for the received signal strength, are obtained by requesting the data communication unit 25. do. Such a request is made to a wireless communication network currently established with a VoIP call path, and the data communication unit 25, when there is a request, an interface unit (the cellular interface unit 26a or the Wi-Fi) to the corresponding wireless communication network. The signal strength measurement value is read and provided to the distribution unit 23 through an approach supported by the interface unit 26b). The measurement of the data reception speed is performed when the VoIP terminal 100 is provided with a component or an execution object (such as an application or a process executed by a physical processor) for another function. When receiving data from a remote location through the cellular interface unit 26a or the Wi-Fi interface unit 26b via the data communication unit 25, The data reception speed of the communication network can be measured from the time required. The distribution unit 23 requests and grasps the data reception speed measured by the other component or the execution object according to the information exchange method. Alternatively, the distribution unit 23 may measure the reception speed from the file data received by requesting the transmission of a file of a specific size to a specific server.

When the distribution unit 23 constructs a data block using the quality variable information, the distribution unit 23 designates the block as a specific type. This is because the arrangement unit (3dps _L) of the VoIP call part (3 _L) on the basis of a specific type does not relay the corresponding data block to the destination terminal 110, spread the minute information of the data block ( 3dtb _L ) to provide. Of course, when the quality variable information does not include information indicating the type of the communication network, the arrangement unit 3dps _L provides information about the received path, that is, quality variable information that provides the type of communication network. Together with the distribution unit 3dtb _L.

In the embodiments described above, it is a VoIP call is established through a plurality of wireless network status, currently the VoIP call part (3 _L) for handling the VoIP call with the VoIP terminal 100, the VoIP terminal 100, When the voice packets were transmitted downward to both communication paths (i.e., to both wireless communication networks), the same voice data was duplicated, i.e., duplicated. In other embodiments according to the present invention, voice packets may be transmitted downlink by dividing each call path. Hereinafter, these embodiments will be described.

In one embodiment according to the present invention, the distributor (3dtb _L), the VoIP call part (3 _L), or the VoIP terminal 100 to be measured or checked by the VoIP call part (3 _L) If the fluctuation of the call quality of each call path (that is, each wireless communication network) identified from the reported quality variable information is within a certain limit, the voice packets are divided and transmitted downward. FIG. 14 is a diagram for explaining this, and a lower reference value (VoQ _ref ) for which a call quality 1201 is previously designated in a wireless communication network, for example, a Wi-Fi network, which is continuously grasped by the distribution unit 3dtb _L. If it is less than or equal to (t _drop ) since the duplication of voice packets are transmitted to the two downlink (1211), and during the transmission process, the quality of the call is continuously identified and the fluctuation range is designated for a predetermined time (T _IntTh ) or more. If the _bandwidth is within the _{bandwidth BWRef} , then, at 1202, the voice packets are divided into two calls and transmitted (1212). In this case, it is determined that the call quality of another wireless communication network, that is, the cellular network, also varies within a certain limit. Of course, the cellular network, which is another communication network, is relatively stable in communication service, so that the call quality is not continuously measured as much as that of the Wi-Fi network, and under certain conditions (for example, a change in service area) or intermittently. If the call quality at any point identified is in good condition, the volatility may be considered low. Then, if the call quality for the Wi-Fi network to be identified in the above manner deviates from the reference quality 1201 ₁ , which is the standard of the fluctuation _range , by more than the _{limited range} VoQ _BWRef (1203), the quality value at that time Accordingly, the voice packets are dual-transmitted over multiple calls or transmitted over the current single call, that is, over the Wi-Fi network. In one embodiment according to the present invention, after starting to transmit voice packets by dividing into a plurality of communication networks, the reference quality, which is the reference of the variation range, is the multiples of N times the quality values 1201 _k and k = 2 that are repeatedly determined. It is also possible to check whether the identified quality deviates from the reference quality (VoQ _BWRef ) or more by dynamically updating and setting to (3, 4, ...).

In the above-described embodiments, the binary-VoIP server 300 uses the cellular network only when the call quality of the Wi-Fi network is not satisfactory in consideration of the cost of using the user's data service. Voice packets were sent down. However, if the user does not consider the cost burden of using the cellular network, or if no additional cost is incurred in the use, the binary-VoIP server 300 does not specifically distinguish the cellular network. Both networks are treated equally. For example, when the call quality of the Wi-Fi network is lower than the predetermined lower threshold, symmetrically with using the cellular network, and when the call quality is lower than the predetermined lower threshold while transmitting voice packets mainly around the cellular network, You can also use the Wi-Fi network. Such unlimited use of the cellular network may include information that a user connects to the binary-VoIP server 300 through a separate client device, for example, a PC or a smart phone, and registers with the subscriber db 30a, For example, the decision may be made based on the above-described VoIP usage mode or plan information about the cellular network. Who performed when using VoIP mode set by the user is a "saving mode" has, VoIP call part (3 _k) of the two won -VoIP server 300 are, as described above on the assumption that the costs for use of the cellular network As in the examples, the cellular network is specially differentiated, i.e. operated by differential communication network, and in the "high quality mode", the cellular network and the Wi-Fi network are regarded as mutually equal in cost point of view, i.e. It is used to transmit voice packets by de-differentiating them. The latter may be similarly applied to the case where the plan information registered by the user in the subscriber db 30a is a plan in which no additional charge is incurred over the determined use amount.

On the other hand, the binary-VoIP server 300 is a component for receiving the subscriber information from the user as described above and processing the registration in the subscriber db (30a), for example, includes a web processing unit. The web processing unit receives the subscriber information input to the page by properly providing a web page or the like to a client connected to the binary-VoIP server 300 and registers it in the subscriber db 30a. When the VoIP call is first requested from the VoIP terminal 100, the call control unit 30 includes information in the VoIP connection request message, for example, a phone number assigned to the terminal included in the session descriptor information, By uniquely identifying the caller from the subscriber identification information such as the subscriber's e-mail address, the subscriber db 30a confirms the registered VoIP usage mode or plan information for the caller, and determines the differential or non-differentiated mode from the confirmed information. Decide Then, when after activating the VoIP call part (3 _k) for the requested VoIP call is set up to notify the determined mode to the VoIP call part (3 _k). Then, the activated VoIP call part (3 _k) is preferred to use the cellular network to transmit a voice packet in accordance with the set mode or the differential mode, such as odds. In another embodiment according to the present invention, the plan information of the subscriber may be provided by requesting another server. To this end, the call control unit 30 requests the plan information while providing subscriber identification information to the other server through the first network connection unit 31a or the second network connection unit 31b, and requests for the request. In response, the plan information for the subscriber is received and the differential or non-differential mode is used for the decision.

Distribute the branch of the VoIP call part (3 _L) for processing the current VoIP call (3dtb _L) is, when the mode such as a mode set to their odds, as illustrated in Figure 15, the VoIP terminal 100 and In a state in which multiple call paths are established, when the call quality of both call paths is equal to or more than a predetermined lower threshold value (VoQ _Ref ), voice packets received from the called terminal 110 are divided into both call paths and transmitted downward ( 1311) In either case, when the call quality is lower than or equal to the lower reference value VoQ _Ref , voice packets are repeatedly transmitted downward in both paths (1312). In the case of transmitting voice packets divided into two call paths, instead of dividing the voice packets evenly into both call paths as illustrated in the drawing, the amounts of the voice packets may be differentially transmitted according to the difference in call quality in both call paths. . For example, for N voice packets, N1 (1 <N1 <N) for the better quality of the call paths and N2 (= N-N1, N2 <N1) for the other voice paths. It can be divided into pieces so that voice packets are transmitted downward. In one embodiment according to the present invention, when the non-differential mode is due to being designated as the "high quality mode" in the VoIP usage mode, the reference value for transmitting downlink voice packets in both communication paths is the lower reference value (VoQ _Ref ). Rather, a higher good threshold (VoQ _BetterRef ) may be applied. In this embodiment, if both _channels show the quality above the _{VQ BetterRef} , the voice packets are divided and transmitted, and if any one shows the quality below the VQ _Ref , the voice packets are transferred to both _channels . Overlapping transmission, and in other cases, voice packets are transmitted using only a channel having a relatively better quality.

In one embodiment according to the present invention, when the distribution unit 3dtb _L is currently in the non-differential mode, when either of the call quality of the multiple currency paths is _equal to or higher than the high reference value VoQ _BestRef specified for high quality, For example, the voice packet may be transmitted downward only to a communication path (ie, a wireless communication network) that is higher than or equal to the higher threshold value (VoQ _BestRef ).

Meanwhile, the method using the multiple call paths described with reference to FIGS. 14 and 15 may also be performed by the distribution unit 23 of the VoIP terminal 100. The distribution unit 23 grasps the quality variable information as described above, and the VoIP usage mode or the plan information is notified from the main control unit 20 when the VoIP call is connected. The distribution unit 23 determines whether to operate in the differential mode or the non-differential mode based on the informed information. The main control unit 20 may grasp the VoIP usage mode or the plan information through information input by the user through a UI provided through an output device such as a displayer.

An embodiment according to the present invention described so far are for example, a VoIP call the part (3 _L) active in the two won -VoIP server (300) and is for the case in which the calling party to the other party connected to a VoIP call. However, the present invention is naturally applicable to the case where the called party of the VoIP call connection becomes the other party of the VoIP calling unit of the binary-VoIP server 300. Of course, in this case, it is assumed that the called terminal is a VoIP terminal including the configuration illustrated in FIG. FIG. 16 exemplarily illustrates a process in which a plurality of VoIP call paths are established by making a Wi-Fi network additionally available (S1411) when the VoIP terminal 100 becomes a called party. will be.

In order to perform the process as illustrated in FIG. 16, the call controller 30 of the binary-VoIP server 300 may request a VoIP connection request S1401 from the calling terminal 120 to the VoIP terminal 100. ) intermediate and (S1402) enable (S1404), the VoIP call unit to handle VoIP calls (3 _M) when transmitting the acceptance message to the calling terminal 120 in accordance with accepted (S1403) to the request by while, it is set to notify the VoIP call connection process, the calling party and the called party of the IP address and the VoIP call the port number portion (3 _M) obtained from. When there is a complementary connection request from the VoIP terminal 100, which is the called party (S1412), the supplementary connection request is appropriate in the corresponding VoIP call information based on the call identifier call_ID ₁ included in the connection request. If it is determined that determines whether, and appropriate sent the acceptance message to the connection request, and that the beam, and (S1413), and the IP address for the formation of additional VoIP call to the current that the VoIP call unit that is handling the call (3 _M) The port number is notified and set.

And, all of the various operations of the VoIP call unit (3 _M) also, the VoIP call part (3 _L) as described by assuming the handle VoIP call far by the outgoing call handling the VoIP call by the incoming call Can be done. For example, operations such as checking call quality for each call path, i.e., each wireless communication network, double-downward transmission or division of voice packets according to call quality, and classification of voice packet transmission methods according to a differential mode or a non-differential mode. Can be performed.

In some cases, both the calling party and the called party may exchange VoIP call processing messages directly with the binary-VoIP server 300 without intervention or relay of another binary-VoIP server. In this case, two

VoIP call units

3 _P and 3 _Q are activated in the binary-VoIP server 300 to transmit and receive voice packets corresponding to the calling party and the called party, respectively. Both

VoIP calling units

3 _P and 3 _Q may independently perform the various operations described above. In this case, the amount of exchange of voice packets between the VoIP call unit _(P 3, _Q 3) can be made over a local loop of the second network connection (31b). Of course, the second by the network routing (routing) in the external communication network via both the connection portion (31b) transmitted to the outside is received via the second network connection (31b) the VoIP call unit (3 _P or 3 _Q) May be passed on.

On the other hand, when the VoIP terminal 100 is the called party, instead of making a VoIP connection request at the calling party, it is only different from establishing a call path for the required VoIP call by accepting the received VoIP connection request. Various other operations are performed in the same manner as described above.

So far, various embodiments according to the present invention have been described using Wi-Fi as an example of a plurality of wireless communication networks. However, the present invention is not limited to the Wi-Fi network, and of course, the present invention can be applied to any wireless communication network of any name that enables data service through a wireless signal. In addition, even between cellular mobile networks, for example, 3G network and 4G (aka LTE network), the principles and concepts of the present invention described in detail as various embodiments may be applied. If a plurality of different cellular communication networks of cellular type do not have a difference in cost between each other, the above-described non-differential mode operations are performed in the VoIP terminal 100 and the binary-VoIP server 300. If there is a cost difference, as described above, the VoIP call will be made according to the given conditions in the differential mode and the non-differential mode.

Various embodiments of the present invention and the methods described in the embodiments may be selectively combined with each other in various ways as long as they are not compatible with each other.

In addition, preferred embodiments of the present invention described above are disclosed for the purpose of illustration, and those skilled in the art, within the technical spirit and the technical scope of the present invention disclosed in the appended claims below, further improved various other embodiments Changes, substitutions or additions will be possible.

Claims

An apparatus for making a call based on an IP address using a plurality of communication networks,

A first call path is established on the first wireless communication network for the requested IP address-based call, and a second call path is opened on the second wireless communication network for the IP address-based call while the first call path is opened. A controller configured to be additionally opened,

In the state in which both the first and the second call paths are established, the voice quality of the call path is confirmed while transmitting voice packets only to one of the selected call paths of the two call paths, and the confirmation An IP address configured to include a call unit configured to transmit, through voice packets, voice data identical to voice data carried in one of the two calls to the other of the two call paths, based on the call quality. Device for base call.
The method of claim 1,

The control unit and the call unit are implemented in a server configured to provide a response to the outgoing call after performing a call processing with a called party of the call to a call originating from the terminal for IP address based call. Device for IP address based call.
The method of claim 1,

The control unit and the communication unit are configured to send or receive a call for an IP address-based call, and convert an input voice signal into a voice packet and convert data of a voice packet received through a communication network into voice. Device for IP address-based call that is implemented in a terminal configured to convert to a signal.
The method of claim 3, wherein

The communication unit is further configured to transmit a voice packet having the same data block and sequence number of the block as the voice packet transmitted to the first call path to the second call path when the second call path is additionally established.

And the second wireless communication network is a wireless communication network in which service areas are scattered, and a serviceable distance in each service area is shorter than that of the first wireless communication network.
The method of claim 1,

The selected one of the communication paths, the relatively better communication characteristics through the mutual comparison between the two communication paths to the value of the communication characteristics indicating the call quality of the series of voice packets received through each of the two communication paths Device for IP address-based calls, which is the valued currency.
The method of claim 5,

The value of the communication characteristic indicative of call quality is obtained by measurement of the communication unit, or is measured by the counterpart on which the voice packet is transmitted / received with the communication unit via the first call path or the second call path. Apparatus for IP address-based call that is obtained by and reported to the call.
The method of claim 5,

The value of the communication characteristic is for the transmission delay time,

The call showing a better communication characteristic is an IP address-based call, which is an average forward call at the time of reception between pairs of data blocks of the same sequence number received in duplicate of each of the two call paths. Device for.
The method of claim 1,

The acknowledgment of the call quality for the call path may include a transmission delay time or a packet loss rate for a packet of a test block received through the corresponding call path after transmitting through the first call path or the second call path. A device for IP address based call.
The method of claim 1,

And said confirmation of call quality for a call path comprises at least a check for a packet loss rate for voice packets received through said any one call path.
The method of claim 1,

And the confirmation of call quality for a call path is confirmation of communication characteristic information indicating call quality obtained by measuring at the terminal from which the IP address-based call is made or received.
The method of claim 1,

The confirmation of the call quality for the call path is performed by providing the external server with information on the service area in which the calling party or the called party's terminal of the requested IP address-based call is currently located. And verify the call quality from the information obtained for the IP address-based call.
The method of claim 1,

The communication unit is further configured to transmit voice packets carrying voice data divided into the two calls so that the same voice data is not duplicated and transmitted in both calls when the checked call quality satisfies a predetermined requirement. A device for IP address based calling.
The method of claim 12,

And the predetermined requirement includes a requirement that all of the call qualities of the two currencies be equal to or greater than a predetermined first reference value.
The method of claim 13,

The predetermined requirement further includes a requirement that a specific mode be specified by the user for IP address based call,

The first reference value is a value that designates a higher quality than the second reference value in which the call unit duplicates and transmits voice data to both the communication paths when the call quality is lower than either of the two communication paths. Device for IP address based call.
The method of claim 13,

And the communication unit is further configured to transmit voice data redundantly to both communication paths when any one of the two communication paths falls below the first reference value.
The method of claim 12,

The predetermined requirements include a requirement that all fluctuations in the call quality of the two currencies be within a predetermined limit, and that any one of the two currencies should be less than or equal to the predetermined standard in the call quality. A device for IP address based calling.
The method of claim 16,

The communication unit is further configured to dynamically change the quality value that is the reference of the variation over time, and determine whether the call quality is within the limit based on the dynamically changing quality value. Device for calls.
The method of claim 1,

And the communication unit is further configured to transmit a voice packet only to a communication path having a value greater than or equal to a predetermined reference value in any one of the two communication paths.
The method of claim 1,

If the communication unit is designated that there is a difference in its use for the first wireless communication network and the second wireless communication network, if both of the two communication paths are equal to or greater than a predetermined reference value in the call quality, the data service is performed. And transmit the voice packet only to a call established on a wireless communication network which is more advantageous to the user in cost.
The method of claim 1,

Each of the two communication paths corresponds to a session established in accordance with a communication protocol to occupy physical communication resources of logical communication entities at both ends of transmitting and receiving the voice packet. Device for call.
In a method for making a call based on the IP address using a plurality of communication networks,

Transmitting and receiving a voice packet by establishing a first call path on a first wireless communication network for a requested IP address-based call;

Establishing a second call path on a second wireless communication network for the IP address-based call while the first call path is opened;

Confirming a call quality for at least one of the call paths while transmitting voice packets to either of the first and second call paths;

And based on the confirmed call quality, transmitting the same voice data through the voice packet to the other of the two call paths, the same voice data carried in the voice packet transmitted to the one call path. Method for base currency.
The method of claim 21,

The step of establishing the second call path is performed as the terminal of the calling party or the called party of the requested IP address-based call enters a state capable of transmitting and receiving a signal with the second wireless communication network. Method for IP address based call.
In the program supply device for providing a program stored in the storage space through a communication network,

Communication means for transmitting and receiving data with the outside through communication,

It comprises a storage means which is stored through the communication means, the application running in the communication terminal containing,

When the application is executed in the communication terminal,

Establishing a first call path on the first wireless communication network for the requested IP address-based call;

A function of additionally establishing a second call path on the second wireless communication network for the IP address-based call when the second wireless communication network becomes available with the first call path opened;

In a state in which both of the first and second call paths are established, the voice quality of the call path is checked while transmitting voice packets to only one of the selected call paths. And based on the confirmed call quality, performing the function of transmitting the same voice data to the other of the two call paths through the voice packet, the same as the voice data carried in the voice packet transmitted to the one call. Program supply comprising program codes.