JP2005333312A - Load decentralization system of media gateway device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide the load decentralization system of a media gateway device which is capable of preventing call loss. <P>SOLUTION: In a system which realizes telephone service between a fixed telephone network 8 and an IP network 2 by controlling media gateway devices 4 by a call agent device 3 to provide VoIP voice communication connecting the fixed telephone network 8 and IP network 2, a representative VoIP function part 20 which detects loads on the media gateway devices 4 and decentralizes the load on a media gateway device 4 when the media gateway device 4 enters a high load state is provided between the call agent device 3 and media gateway devices 4. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

The present invention relates to a load distribution system for a media gateway device.
In order to provide VoIP (Voice Over IP) voice communication that connects a fixed telephone network (PSTN) and an IP (Internet Protocol) network, a call agent device (abbreviated as CA device) takes the lead. By controlling a gateway device (abbreviated as MG device), a telephone service between the fixed telephone network (PSTN) and the IP network is realized.

As a signaling protocol for the call agent device to control the media gateway device, there is a MEGACO (Media Gateway Control) protocol, which is standardized as follows.
・ H. 248 (MEGACO): Media gateway device control protocol standardized by ITU-T (Telecommunication Standardization Division of International Telecommunication Union) TD-1002: MG control protocol standardized by TTC (Information and Communication Technology Committee) -RFC3015: Media gateway device control protocol standardized by IETF (standardization organization that examines technologies used on the Internet) The present invention does not depend on a call agent device, and the load distribution and accommodation by the media gateway device alone This is a technique for dynamically changing the connection of a line.

  The media gateway device uses a call agent device and a MEGACO protocol to make a call connection between a fixed telephone and an IP telephone. The positioning of the call agent device and the media gateway device is as follows.

Call agent device: A device that determines a connection path between a fixed telephone network (PSTN) and an IP network by negotiating with a PSTN switch using a common line signal. Media gateway device: For a path specified by a call agent device. FIG. 14 is a block diagram showing a connection configuration example in the MEGACO communication system. In the figure, reference numeral 1 denotes an IP telephone terminal. In the figure, 050-2222-1111 indicates a telephone number, and (192.168.10.1) indicates an IP address. Reference numeral 2 denotes an IP network connected to these IP telephone terminals 1. Reference numeral 3 denotes a call agent device # 0 connected to the IP network 2. Reference numeral 4 denotes a media gateway device connected to the call agent device # 0. In this example, two media gateway devices 4 are provided, # 0 and # 1. In the media gateway device 4, 4a is a VoIP function unit, and 4b is a time switch (TSW) connected to the VoIP function unit 4a.

  Reference numeral 5 denotes a space switch (SPSW) connected to the media gateway device 4, 6 denotes a time switch (TSW) connected to the space switch 5, and 7 denotes a line termination device connected to the time switch 6. In these line terminators 7, # 0 is connected to TSW # 2, and # 1 is connected to TSW # 3. Reference numeral 8 denotes a fixed telephone network (PSTN) connected to these line terminators 7, and 9 denotes an exchange # 0 connected to the fixed telephone network 8. Reference numeral 10 denotes a telephone terminal (fixed telephone) connected to the exchange # 0. A is a common line signal connected between the call agent device # 0 and the exchange # 0.

  As described above, the media gateway apparatus 4 performs call connection between the fixed telephone 10 and the IP telephone 1 using the call agent apparatus 3 and the MEGACO protocol. The call agent device 3 determines a connection path between the fixed telephone network (PSTN) 8 and the IP network 2 by negotiating with the PSTN switch 9 using a common line signal (call control signal). The media gateway device 4 performs a call connection between the fixed telephone network (PSTN) 8 and the IP network 2 for the path designated by the call agent device 3.

Using the system shown in FIG. 14, VoIP call setting processing between the fixed telephone 10 having the telephone number 045-222-1111 and the IP telephone having the telephone number 050-2222-1111 will be described below.
(CS1) The call agent device 3 uses the exchange 9 and the common line signal to select a line and a channel to be used between the telephones to be connected. For example, in order to connect between 045-222-1111 and 050-2222-1111, channel number 0 with line number 0 is selected.
(CS2) The call agent device 3 determines the media gateway device to be communicated from the line number and channel number set in CS1, based on the media gateway device information management data stored therein. FIG. 15 is a diagram showing media gateway device information management data inside the call agent device. As shown in the figure, it is composed of MG information and accommodation line information.

  The MG information includes an MG number and an IP address for MG, and the IP address when the media gateway device is, for example, media gateway device # 0 is 192.168.0.1. The accommodation line information includes an accommodation line number and an accommodation channel number, and the accommodation channel number is 0 to 1 when the accommodation line number is 0, for example.

For example, since line 0 and channel 0 are accommodated in the media gateway device # 0, communication with the media gateway device # 0 is determined.
(CS3) The media agent device 4 determined by the call agent device 3 requests the corresponding channel connection using the MEGACO protocol.
(CS4) In the media gateway device 4, the internal VoIP function unit 4a receives the corresponding channel connection request, sets the connection channel number specified in the call management table managed by the VoIP function unit 4a, A response of channel connection completion is sent to the agent device 3. FIG. 16 is a diagram showing a call management table in the VoIP function unit 4a.

  Access the call information number to obtain the call information at the corresponding address. The call information includes a connection channel number, a subscriber IP address, and voice information used (CODEC). The connection channel number is set in CS4, and the subscriber IP address and voice information used are set in CS6.

Note that the interconnection between the VoIP function unit, the TSW, the SPSW, and the line terminating device (hereinafter referred to as a TST connection) is fixedly set using a fixed connection table at the time of initial setting. FIG. 17 is a diagram showing a fixed connection table in the VoIP function unit. As shown in the figure, it consists of MG information and accommodation line information. The MG information is composed of an MG number and a VoIP function unit, and the accommodated line information is composed of a TST structure and a connection line termination device. For example, when the TST configuration is TSW # 0-SPSW-TSW # 2, the connection line termination device is the line termination device # 0.
(CS5) The call agent device 3 notifies the media gateway device of the subscriber IP address and the used voice information (CODEC) connected to the call-connected channel, and requests connection to the IP network. For example, if the telephone number is 050-2222-1111, the IP address notifies 192.168.10.1 as shown in FIG.
(CS6) In the media gateway device 4, the internal VoIP function unit 4a connects the designated subscriber IP address and the voice path according to the used voice information (CODEC), and the line received by the CS4 Do the link. Further, the specified subscriber IP address and voice information (CODEC) to be used are set in the call management table (see FIG. 16) managed inside the VoIP function unit, and response to voice call connection completion is sent to the call agent device 3. To do.

With the above flow, call connection between the PSTN-IP network is performed.
Next, the basic operation for the TST connection will be described. The T-S-T connection is a technique for transmitting virtual channel information acquired from the network termination device 7 to the SPSW 5 via the TSW and reconstructing virtual channel information received on the TSW side which is a reception unit. . The media gateway device 4 performs semi-fixed channel connection at the time of initial setting as shown in FIG. FIG. 18 shows a TST connection diagram. This figure shows an outline of the operation and time slot assignment in a state where semi-fixed channels are connected. Here, 15 indicates a time slot corresponding to a channel in the line of the line network termination device 7.

As a conventional system of this type, for example, there is known a detour method that is effective when a device for determining / managing a PBX trunk destination configured by a plurality of VoIP trunks is used (for example, Patent Document 1). reference).
Japanese Unexamined Patent Publication No. 2003-169079 (5th page, 6th page, FIG. 1)

  The configuration of the line (STM0 / STM1 (52M, 156M) / E1 / T1) accommodated in the media gateway device 4 is fixedly assigned to the media gateway device 4. The call agent device 3 recognizes the line and channel accommodated in the media gateway device 4 and makes a call connection request to the channel under the initiative of the call agent device 3.

  Therefore, the call agent device 3 must communicate with a specific media gateway device 4 in order to acquire one channel. When the load is concentrated on the channel accommodated in one media gateway device 4 due to the above-described restrictions, the load on the VoIP function unit 4a of the corresponding device increases, which hinders service.

  Specifically, in the configuration shown in FIG. 14, when the processing capability of the VoIP function unit 4a in the media gateway device 4 is an upper limit of 30000 BHCA (number of calls / time), 50000 BHCA is set for the channel in the line number 1. When a call connection request is made, regardless of the load state of the VoIP function unit # 0, the load of the VoIP function unit # 1 increases rapidly, resulting in a call loss of 20000BHCA. FIG. 19 is an explanatory diagram of problems in the conventional system. The same components as those in FIG. 14 are denoted by the same reference numerals.

  In FIG. 19, if the load is concentrated on the line number 1, 30000 BHCA or more is not accepted and the call is lost. At this time, if the VoIP function unit # 0 has a low load, call loss should be prevented by entrusting the processing of 20000BHCA overflowing to the VoIP function unit # 0.

  The present invention has been made in view of such problems, and an object of the present invention is to provide a load distribution system for a media gateway device that can prevent call loss.

  In the present invention, a representative VoIP function unit for managing a plurality of VoIP function units 4 a is newly provided to perform load distribution in the media gateway device 4. FIG. 1 is a principle block diagram of the present invention. The same components as those in FIG. 14 are denoted by the same reference numerals. In the figure, reference numeral 20 denotes a representative VoIP function unit designed to distribute the load to the media gateway device 4 between the call agent device 3 and the media gateway device 4. The representative VoIP function unit 20 is a part that characterizes the present invention. In this system, the call agent device # 0 is not conscious of the presence of the representative VoIP function unit 20, and performs an operation conscious of being connected to the media gateway device 4 as usual.

The representative VoIP function unit 20 has the following functions.
A function for receiving a signal from the call agent device 3 as an accepting unit of a plurality (or a single) media gateway device. A load state of the subordinate VoIP function unit 4a is determined and a call connection is requested to an appropriate VoIP function unit 4a. Function / Function to dynamically allocate channel to subordinate VoIP function unit 4a With such a function, load distribution control closed inside the media gateway device can be performed without changing the processing of the call agent device 3. Specifically, in the configuration shown in FIG. 14, when the processing capability of the VoIP function unit 4a in the media gateway device 4 is the upper limit of 30000BHCA, a call connection request for 50000BHCA is made to the channel in the line number 1. However, the load is distributed to the VoIP function unit 4a by 25,000 BHCA, and no call loss occurs. This is because the number of calls processed by each VoIP function unit 4a is 25000, which is within the upper limit of 30000.

As shown in FIG. 1, even if the load is concentrated on the line number 1, the representative VoIP function unit 20 distributes the load to the VoIP function unit 4a of each media gateway device 4, so that no call loss occurs. Thus, according to the present invention, call loss can be prevented.
(1) According to the first aspect of the present invention, in order to provide VoIP voice communication connecting the fixed telephone network and the IP network, the call agent apparatus controls the media gateway apparatus, whereby the telephone between the fixed telephone network and the IP network In a system for realizing a service, when a load of a media gateway device is detected between the call agent device and the media gateway device and a high load state occurs in any of the media gateway devices, the load on the media gateway device A representative VoIP function unit is provided to distribute the above.

Here, a call agent device is a call agent device, a media gateway device is a media gateway device, and VoIP voice communication is (Voice Over IP) voice communication. It is.
(2) In the invention according to claim 2, when the representative VoIP function unit receives a MEGACO signal, a load is averaged on all connected subordinate VoIP function unit management units using a round robin method. It is characterized by being dispersed.

Here, MEGACO refers to Media Gateway Control (Media Gateway Control). In the case of network load distribution, round robin means that a plurality of systems having the same configuration are prepared and processing requests are allocated in order.
(3) In the invention according to claim 3, when the representative VoIP function unit receives the MEGACO signal, the load is averagely distributed to all connected subordinate VoIP function unit management units using a hash function. It is characterized by making it.

Here, the hash function is an arithmetic technique for generating a fixed-length pseudorandom number from a given original text.
(4) In the invention according to claim 4, when the representative VoIP function unit receives the MEGACO signal, the MEGACO signal is transmitted using all or any items of the CC usage rate, the channel usage rate, and the digital signal processing device usage rate. It is characterized by sorting.

Here, the CC usage rate is the usage rate of the CPU of the corresponding VoIP, and the digital signal processing device usage rate is the usage rate of the device connecting the lines.
(5) The invention according to claim 5 is characterized in that, when the representative VoIP function unit receives a MEGACO signal, the selected line is dynamically connected and released.

(1) According to the first aspect of the present invention, when a high load state is generated in any of the media gateway devices by the representative VoIP function unit, this is detected and the load is distributed to the media gateway devices on average. Therefore, the call loss does not occur.
(2) According to the invention described in claim 2, the representative VoIP function unit can distribute the load on the average to the subordinate VoIP using the round robin method.
(3) According to the invention described in claim 3, the representative VoIP function unit can distribute the load on the average to the subordinate VoIP using a hash function.
(4) According to the invention described in claim 4, the representative VoIP function unit can distribute the MEGACO signal using the CC usage rate, the channel usage rate, or the digital signal processing device usage rate.
(5) According to the fifth aspect of the present invention, when the representative VoIP function unit receives the MEGACO signal, dynamic and quick line control can be performed by dynamically connecting and releasing the selected line. it can.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
FIG. 2 is a diagram illustrating a configuration example of the representative VoIP function unit. The same components as those in FIG. 1 are denoted by the same reference numerals. In the figure, a portion surrounded by a broken-line square is the representative VoIP function unit 20. In the signal flow, a broken line and an arrow indicate a data reference relationship, and a solid line and an arrow indicate a function call relationship. In the figure, reference numeral 12 denotes a MEGACO signal receiving unit that receives a MEGACO signal from the call agent device 3. The MEGACO signal receiving unit 12 receives a MEGACO signal from the call agent device 3 and distributes it to the subordinate VoIP function unit 4a. Reference numeral 16 denotes a representative call information management table for managing a link between a call accepted by the MEGACO signal accepting unit 12 and a call processed by the subordinate VoIP function unit 4a.

  FIG. 3 is a diagram showing a configuration example of the representative call information management table 16. The representative call information is accessed by the representative call information number. The representative call information number is an internal number for specifying a call in the representative VoIP function unit 20. As shown in the figure, the representative call information accessed by this representative call information number includes an external public IP address, an internal communication IP address, a call information number, VoIP side TSW connection information, SPSW connection information, and a line terminator side TSW. Consists of connection information. Here, the call information number is an internal number for specifying a call in the subordinate VoIP function unit 4a.

  In FIG. 2 again, 13 is a MEGACO signal conversion that converts the MEGACO signal into a signal that can be executed by the subordinate VoIP function unit 4a or a signal that can be notified to the call agent device 3, and notifies the subordinate VoIP function unit 4a or the call agent device 3 Part. The MEGACO signal converter 13 accesses the representative call information management table 16 and refers to the data.

  Reference numeral 14 denotes a subordinate VoIP function unit management unit that manages the use line and outgoing call information of the subordinate VoIP function unit 4a. Reference numeral 17 denotes a subordinate VoIP function unit management table that receives access from the subordinate VoIP function unit management unit 14. FIG. 4 is a diagram illustrating a configuration example of the subordinate VoIP function unit management table 17. The subordinate VoIP function unit management table 17 includes subordinate VoIP function unit management information that is accessed from the number of subordinate VoIP function units by the VoIP function unit number.

  The subordinate VoIP function unit management information includes a subordinate VoIP availability status, an internal communication IP address, the number of lines used, a CC usage rate, and a digital signal processing device usage rate. Here, the IP address for internal communication has been set with a fixed value assigned to the function unit. In the example of FIG. 1, the VoIP function unit # 0 is 10.22.0.1, and the VoIP function unit # 1 is 10.22.0.2.

  In FIG. 2 again, reference numeral 15 denotes a line management unit for managing the availability of the mounted line and failure recovery. The line management unit 15 manages a correspondence table of channels, media gateway devices 4, and representative call information numbers that the representative VoIP function unit 20 shows to the call agent device 3. Correspondence between the MG data provided to the call agent device 3 corresponding to the channel number and the internal management information can be confirmed.

  FIG. 5 is a diagram showing a configuration example of the line state management table 18. The line state management table 18 is composed of accommodated line information, MG information, and corresponding representative call information numbers. The accommodated line information includes a line number, a channel number, and a line state. Here, it is assumed that the line status can be managed as “free”, “in use”, “block”, “failure”, and “failure block”. The MG information is composed of an MG number and an external public IP address. For example, for the media gateway device # 0 with the MG number, the external public IP address is 192.168.0.1. The external public IP address indicates the IP address of the MG recognized by the call agent device 3. The corresponding representative call information number is an internal number for specifying a call in the representative VoIP function unit.

  FIG. 6 is a diagram showing a sequence of events that occur in the representative VoIP function unit. 2 and 1 are denoted by the same reference numerals. In the figure, the range indicated by the arrow is the representative VoIP function unit 20. First, a line state change notification is notified from the VoIP function unit 4a to the line management unit 15 (S1). This line state change notification is notified at the time of line state change notification. The line management unit 15 sets the line state notified to the line state management table 18 (S2). In this case, “vacant”, “blocked”, “failure”, and “failure blocked” are set in the line state of the accommodated line information in the line state management table 18. Thereafter, the line management unit 15 returns a response to the VoIP function unit 4a (S3).

  FIG. 7 is a diagram showing a subordinate VoIP state change notification event sequence. 2 and 1 are denoted by the same reference numerals. The range indicated by the arrow in the figure is the representative VoIP function unit 20. First, the CC usage rate is notified from the VoIP function unit 4a to the subordinate VoIP function unit management unit 14 (S1). This CC usage rate is notified when a certain period or a certain CC usage rate is reached. Upon receiving the CC usage rate notification, the subordinate VoIP function unit management unit 14 sets the notified value to the CC usage rate in the subordinate VoIP function unit management table 17 (S2). Then, the subordinate VoIP function unit management unit 14 returns a response to the VoIP function unit 4a (S3).

  Next, the VoIP function unit 4a notifies the subordinate VoIP function unit management unit 14 of the digital signal processing device usage rate (S4). In this case, notification is made when a certain period or a part of the digital signal processor usage rate is reached. The CC usage rate and the interface may be the same. The subordinate VoIP function unit 14 that has received the digital signal processing device usage rate notification sets the value notified to the digital signal processing device usage rate in the subordinate VoIP function unit management table 17 (S5). Next, a response is returned to the VoIP function unit 4a (S6).

  Next, the VoIP function unit 4a notifies the subordinate VoIP function unit management unit 14 of the availability of VoIP (S7). The availability of VoIP is notified when the state changes. The subordinate VoIP function management unit 14 that has received the notification of availability of VoIP sets the value notified to the subordinate VoIP availability status in the subordinate VoIP function management table 17 (S8). Thereafter, a response is returned to the VoIP function unit 4a (S9).

  FIG. 8 is a sequence diagram when a MEGACO signal is received. 2 and 1 are denoted by the same reference numerals. In the figure, the range indicated by the arrow is the representative VoIP function unit 20. First, a MEGACO signal is issued from the call agent device 3 (S1). When the MEGACO signal receiving unit 12 receives this signal, the MEGACO signal receiving unit 12 notifies the line management unit 15 of a line state read signal (S2). The line management unit 15 collects line states from the line state management table 18 (S3). The line management unit 15 notifies the MEGACO signal receiving unit 12 of the line state (S4).

  The MEGACO signal receiving unit 12 determines the line state. If it is empty / in use, the process proceeds to step S6. If it is a failure / blocking state, the process jumps to step S20. Next, when the line state is idle / busy, it is determined whether or not the call is a new call (S6). If it is a new call, the process proceeds to step S7. The MEGACO signal receiving unit 12 acquires a representative call information number (S7). Then, the subordinate VoIP function unit determination notification is transmitted to the subordinate VoIP function unit management unit 14 (S8). A VoIP function unit to be communicated is selected from the subordinate VoIP function unit management table 17 (S9). The logic to be selected will be described later. Next, the subordinate VoIP function unit management unit 14 returns a response to the MEGACO signal reception unit 12 (S10).

  Upon receiving this response, the MEGACO signal receiving unit 12 sets subordinate VoIP function unit information to be communicated in the representative call information management table 16 (S11). Next, the MEGACO signal accepting unit 12 notifies the line managing unit 15 of a connection line setting request (S12). The line management unit 15 receives this notification and makes an SPSW connection request to the TSW / SPSW (S13). The TSW / SPSW returns a response to the line management unit 15 (S14).

  When receiving this response, the line management unit 15 notifies the TSW / SPSW of the TSW connection request again (S15). The TSW / SPSW returns a response to this notification to the line management unit 15 (S16). Upon receiving this response, the line management unit 15 makes the line state management table 18 in use and sets a number for the representative call information (S17). Next, the line management unit 15 notifies the MEGACO signal reception unit 12 of a response (18).

  Upon receiving this response notification, the MEGACO signal receiving unit 12 sets the connection information of SPSW and TSW in the representative call information management table (S19). Next, the MEGACO signal reception unit 12 notifies the MEGACO signal conversion unit 13 of a received signal (S20). Next, the MEGACO signal conversion unit 13 acquires the subordinate VoIP function unit information of the communication destination from the representative call information management table 16 (S21). Then, a MEGACO signal is sent to the VoIP function unit 4a (S22). At the same time, the MEGACO signal conversion unit 13 returns a response to the MEGACO signal reception unit 12 (S23).

  FIG. 9 is a sequence diagram when the MEGACO signal is transmitted. 1 and 2 are denoted by the same reference numerals. The range indicated by the arrow in the figure is the representative VoIP function unit 20. First, a MEGACO signal is transmitted from the VoIP function unit 4a to the MEGACO signal conversion unit 13 (S1). Upon receiving the MEGACO signal, the MEGACO signal conversion unit 13 acquires a representative call information number from the line management unit 15 (S2). Next, the line management unit 15 returns the representative call information corresponding to the channel to the MEGACO signal conversion unit 13 (S3, S4).

  The MEGACO signal conversion unit 13 checks whether the call is a new call (S5). If the call is a new call, the call information number is set in the representative call information management table 16 (S6), and the process skips to step S16. If it is an existing call in step S5, a transmission signal determination is made (S7), and if it is a subtract reply, a connection line deletion request is notified to the line management unit 15 (S8). . Upon receiving the connection line deletion request, the line management unit 15 makes an SPSW connection request to the TSW / SPSW (S9). The TSW / SPSW returns a response to the line management unit 15 (S10).

  In response to this response, the line management unit 15 notifies the TSW / SPSW of a TSW connection request (executed twice on the VoIP side and the line terminating device side) (S11). TSW / SPSW returns a response to this (S12). Next, the line management unit 15 clears the line status management table 18 and clears the representative call information number (S13). The line management unit 15 notifies the MEGACO signal conversion unit 13 of a signal to that effect (S14). The MEGACO signal conversion unit 13 clears the information in the representative call information management table 16 (S15). In this case, the information of the corresponding call agent device 3 is held in advance as separate data.

  According to the present invention, when the representative VoIP function unit 20 receives the MEGACO signal, the load can be distributed on the average to all of the connected subordinate VoIP function unit management units 14 using the round robin method. it can. Further, when the representative VoIP function unit 20 receives the MEGACO signal, the load can be averagely distributed to the subordinate VoIP using a mesh function.

  FIG. 10 is a diagram showing a subordinate VoIP selection logic flow. First, the subordinate VoIP function part number to be used is set to 0 (S1). Next, step S10 and step S20 are looped by the number of subordinate VoIPs. The used subordinate VoIP function unit checks whether it can be used (S11). If it cannot be used, the process skips to step S20. If it can be used, a = CC usage rate * m, b = line usage rate * n, and c = digital signal processor usage rate * o are calculated (S12 to S14).

  Each coefficient is applied to various information in the subordinate VoIP function unit management table 17. The coefficient can be determined by the maintenance person. If the coefficient does not need to be selected, 0 is given, so that the corresponding element does not affect the load distribution.

  Next, the total value of a + b + c is compared with the information of the scheduled VoIP function unit (S15). As a result of the comparison, if the VoIP is scheduled to be used, the process skips to step S20. If it is the current VoIP, the subordinate VoIP function part number scheduled to be used is updated (S16). As described above, steps S10 to S16 are repeated for the number of subordinate VoIPs (S20). Next, the subordinate VoIP function number being selected is used (S2).

  As described above, according to this embodiment, the representative VoIP function unit 20 can distribute MEGACO signals using the CC usage rate, the channel usage rate, or the digital signal processing device usage rate.

  FIG. 11 is a diagram showing a TST connection image at the time of load distribution. At the time of load distribution, TST connection is dynamically performed. A TST connection image in which channel 0 and channel 1 accommodated in line 0 are load-balanced connected is shown. As shown in A, the connected channels are dynamically set for each call setup. B is a time slot corresponding to a channel in the line of the line terminating device.

  FIG. 12 is a diagram showing a system configuration example when there are three VoIP function units. The same components as those in FIG. 1 are denoted by the same reference numerals. This embodiment shows an example in which three media gateway devices 4 are provided. Basically, even if three media gateway devices 4 are provided, the basic operation does not change. When the call agent device 3 transmits a MEGACO signal similar to the conventional one, the representative VoIP function unit 20 allocates the call distribution to the media gateway devices 4 from # 1 to # 3.

  FIG. 13 is a diagram illustrating a system configuration example in a situation where there is one VoIP function unit. The same components as those in FIG. 1 are denoted by the same reference numerals. This embodiment shows a case where two call agent devices 3 are provided, # 0 and # 1, and one media gateway device 4 is provided. When the load is very small, only one media gateway device 4 is needed, and the built-in VoIP function unit 4a can cover it.

  In the above-described embodiment, when the representative VoIP function unit 20 receives the MEGACO signal, the selected line can be dynamically connected and released. In this way, dynamic line control can be performed dynamically by dynamically connecting and releasing the selected line.

It is a principle block diagram of the present invention. It is a block diagram which shows the structural example of a representative VoIP function part. It is a figure which shows the structural example of a representative call information management table. It is a figure which shows the structural example of a subordinate VoIP function part management table. It is a figure which shows the structural example of a line | wire state management table. It is a figure which shows the sequence of the event which generate | occur | produces in a representative VoIP function part. It is a figure which shows a subordinate VoIP state change notification event sequence. It is a sequence diagram at the time of MEGACO signal reception. It is a sequence diagram at the time of MEGACO signal transmission. It is a figure which shows a subordinate VoIP selection logic flow. It is a figure which shows the TST connection image at the time of load distribution. It is a figure which shows the system structural example in case there are three VoIP function parts. It is a figure which shows the system configuration example in the condition where there is one VoIP function unit. It is a block diagram which shows the example of a connection structure in a MEGACO communication system. It is a figure which shows the media gateway apparatus information management data inside a call agent apparatus. It is a figure which shows the call management table in a VoIP function part. It is a figure which shows the fixed connection table in a VoIP function part. It is a TST connection diagram. It is explanatory drawing of the problem of a conventional system.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 IP telephone terminal 2 IP network 3 Call agent apparatus 4 Media gateway apparatus 4a VoIP function part 4b TSW (time switch)
5 SPSW (space switch)
6 TSW (Time switch)
7 Line termination equipment 8 PSTN (fixed telephone network)
9 Switch 10 Fixed telephone 20 Representative VoIP function section

Claims (5)

In a system for realizing a telephone service between a fixed telephone network and an IP network by a call agent device controlling a media gateway device in order to provide VoIP voice communication connecting the fixed telephone network and the IP network.
The load of the media gateway device is detected between the call agent device and the media gateway device, and when a high load state occurs in any of the media gateway devices, the load is distributed to the media gateway device. A load distribution system for a media gateway device, comprising a representative VoIP function unit.   2. The representative VoIP function unit, when receiving a MEGACO signal, distributes a load on an average to all connected subordinate VoIP function unit management units using a round robin method. Media gateway device load balancing system.   2. The load according to claim 1, wherein when the representative VoIP function unit receives a MEGACO signal, a load is averagely distributed to all connected subordinate VoIP function unit management units using a hash function. Media gateway device load balancing system.   2. The MEGACO signal is distributed using all or arbitrary items of a CC usage rate, a channel usage rate, and a digital signal processing device usage rate when the representative VoIP function unit receives a MEGACO signal. Media gateway device load balancing system.   2. The load distribution system for a media gateway device according to claim 1, wherein when the representative VoIP function unit receives a MEGACO signal, the selected line is dynamically connected and released.

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