JP2000253150A - Speech auxiliary device for speech system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To allow a speech auxiliary device for speech system for compensating for the deterioration of message transmitting performance caused by the deterioration of communication quality in telephone communication. SOLUTION: The deterioration of the quality of real-time speech between terminals 11A and 11B through an UDP port is determined, based on the number of times of packet collision in the Internet 14 detected by a gateway 13A. A request for the input of a message to the called terminal is reproduced to a terminal 11A connected with a telephone network 12A, and the inputted message is preserved in the gateway 13A in which the deterioration of the speech quality is determined, and transmitted through the TCP port of the Internet 14 to a gateway 13B in a non-real time. At the same time, a TCP interrupt packet is transmitted to the gateway 13B, so that a message from a terminal 11B can be preserved in the gateway 13B, and transmitted to the gateway 13A in a non-real time. Then, the message is reproduced to each corresponding terminal 11A and 11B in each gateway 13A and 13B.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a telephone call assisting device for a telephone communication system used in telephone communication accompanied by a deterioration in telephone call quality.

[0002]

2. Description of the Related Art Conventionally, general telephone communication is performed by connecting a telephone terminal on a calling side and a telephone terminal on a receiving side by a public telephone line. In recent years, in addition to the conventional telephone communication connected via the Internet, an Internet telephone system has recently reduced telephone charges by performing telephone communication via the Internet, especially for long-distance telephone calls such as international calls. Has been put to practical use.

In this Internet communication system, a gateway device for converting a communication protocol between a public telephone line and the Internet is provided, and the transmitting terminal is registered from a telephone terminal on the transmitting side via the public telephone line. When a call is made to the existing gateway device by designating the telephone number of the partner telephone terminal, a connection request packet including the telephone number of the partner terminal is generated in the gateway device, and the partner terminal is connected via the Internet. Sent to registered gateway devices. As a result, the other gateway device that has received the connection request packet from the originating gateway device,
By specifying the telephone number of the destination included in the received packet and making a call via the public telephone line, the communication between the telephone terminal on the calling side and the telephone terminal on the receiving side is switched from the public telephone line to the gateway device. ~ Internet ~ Gateway device ~ Connected via public telephone line, no matter how long distance calls, only pay the call fee on each public telephone line on the calling side and the receiving side and the registration fee for the gateway device It is possible to perform telephone communication at low cost.

[0004] The Internet network is a low-cost packet network connecting the world. The Internet network compresses and encodes voice into packets and transfers the packets so that the first telephone terminal and the second telephone terminal can communicate with each other.
As compared with a direct connection to a telephone terminal via a public telephone line, the advantage that the cost can be greatly reduced, especially for long-distance telephone communication, is obtained.

On the Internet, TCP / IP (trans
Communication is performed by a protocol called mission control protocol / internet protocol).
A transport layer protocol called UDP (user datagram protocol) which is a part of / IP is used. This U
Unlike the TCP / IP transport layer, DP / IP has high-speed protocol processing, but lacks error correction and retransmission functions, and thus has low reliability. Even if packet loss / delay occurs, retransmission is performed. There is a disadvantage that no permutation is performed.

[0006]

In such a conventional Internet telephone system, the Internet network is a communication network without bandwidth guarantee, and unless a special bandwidth securing means is taken, a call may be interrupted due to congestion. is there. Since real-time communication is required for communication, retransmission for packet loss is not performed, and processing for filling gaps with data inferred from previous and subsequent packets is performed.

If the packet loss exceeds tens of percent, noise and dropouts occur in communication intensely. If the call is continued in such a situation, there is a possibility that the contents may be erroneously heard. However, there is a problem that even if the gateway device or the user himself or herself disconnects the communication, the communication ends up halfway.

SUMMARY OF THE INVENTION The present invention has been made in view of the above-described problems, and provides a communication assistance device of a communication system capable of compensating for deterioration of message transmission performance due to deterioration of communication quality in telephone communication. The purpose is to do.

[0009]

A first aspect of the present invention is directed to a communication system for a communication system for performing a real-time communication between one terminal and another terminal via a communication network. A communication auxiliary device connected between a telephone network to which the terminal is connected and the communication network, wherein the communication quality detecting means for detecting a decrease in communication quality in the communication network, and the communication quality reduction detecting means If a decrease in call quality in the communication network is detected by
A call suspending means for suspending a real-time call via the communication network; and a terminal connected via the telephone network when the real-time call via the communication network is suspended by the call suspending means. A message acquisition means for acquiring a voice message, and a message transfer means for converting a voice message from a terminal acquired by the message acquisition means into a non-real-time protocol and transferring the protocol to a partner.

[0010] In such a communication auxiliary device of the communication system according to the first aspect of the present invention, when a decrease in the communication quality in the communication network is detected by the communication quality reduction detecting means, the communication via the communication network is performed. The real-time call is interrupted by the call interrupting means, the voice message from the terminal connected via the telephone network is obtained by the message obtaining means, converted to a non-real-time protocol by the message transfer means, and Transferred to the communication auxiliary device. Therefore, even when real-time communication becomes difficult due to the deterioration of the communication quality on the communication network, a minimum necessary message can be transmitted to the user of the partner terminal.

According to a second aspect of the present invention, there is provided a call assisting device for a call system according to the first aspect of the present invention, further comprising a message transfer unit of the call assisting device on the other side. Message storage means for storing the transferred voice message, and message playback means for playing back the transfer voice message from the other party stored by the message storage means to a terminal on the other side connected to the telephone network And characterized in that:

[0012] According to the second aspect of the present invention, the voice message transferred by the message transfer unit of the other party's call auxiliary device is stored by the message storage unit. The transferred voice message from the called party is reproduced by the message reproducing means to the terminal on the local side connected to the telephone network.
Even when real-time communication becomes difficult, the minimum necessary messages can be transmitted to the users of the partner terminals.

According to a third aspect of the present invention, there is provided a call assisting device for a call system according to the second aspect of the present invention, further comprising a real-time call through the communication network. A terminal number storage means for storing the terminal telephone number of the user and a terminal telephone number of the other party, and the message reproducing means reproduces a voice message to the terminal on the side of the telephone connected to the telephone network. When a terminal connection request for real-time communication is received from the other party's call assist device in the state where the connection is made, the connection request source terminal telephone number and the other party's terminal number stored in the terminal number storage means are received. And a connection terminal for determining whether or not the terminal telephone number of the connection request destination matches the terminal telephone number of our side stored in the terminal number storage means. A determining means, when the matching determination by the connection terminal determination means is made, characterized in that a resume call resuming means real-time call through the communication network.

[0014] In the above-described communication auxiliary apparatus of the communication system according to the third aspect of the present invention, when a real-time telephone call is made via the communication network, the terminal telephone number of the other party and the terminal telephone of the other party are used. The number and the number are stored by the terminal number storage means, and while the voice message is being played back to the terminal on the other side connected to the telephone network by the message playback means, the other party's call assistance device can perform a real-time call. If a terminal connection request for the connection is received, the terminal telephone number of the connection request source matches the terminal telephone number of the other party stored in the terminal number storage means, and the terminal telephone number of the connection request destination And the terminal number stored on the terminal number storage means is determined by the connection terminal determination means, and if a match is determined, Since the real-time call via the communication network is resumed by the call resuming means, it is possible to reconnect immediately if there is a call from the partner terminal, without having to restart the call from the terminal on the other side. .

According to a fourth aspect of the present invention, there is provided a call assistance device for a call system according to any one of the first to third aspects, further comprising: In a state where a real-time call through a network is being performed, the apparatus further includes a latest voice storage unit that constantly updates and stores a call voice for a predetermined period of time immediately before the message transfer unit, Message transfer means for converting a voice message obtained by adding a call voice for a predetermined time immediately before stored by the latest voice storage means to a voice message from the terminal on the side of the terminal to a non-real-time protocol and transferring the same to a communication auxiliary device on the other side; It is characterized by having done.

[0016] In the above-described communication auxiliary device of the communication system according to the fourth aspect of the present invention, when a real-time communication is being performed through the communication network, the communication voice for a predetermined time immediately before is constantly updated. The voice message from the terminal on our side obtained by the latest voice storage means and stored by the latest voice storage means is added with the call voice for a predetermined time immediately before stored by the latest voice storage means. Since the voice message to which the voice is added is converted into a non-real-time protocol by the message transfer means and transferred to the communication auxiliary device on the partner side, the message content can be accurately transmitted to the user of the partner terminal. Become.

[0017]

Embodiments of the present invention will be described below with reference to the drawings.

(First Embodiment) FIG. 1 is a diagram showing the overall configuration of an Internet telephone system according to a first embodiment of the present invention.

In this Internet telephone system, a public telephone line 12A, one gateway device 13A, the Internet (packet network) 14, another gateway device 13B, and a public telephone line 12A are connected between one telephone terminal 11A and the other telephone terminal 11B. The connection is made via the telephone line 12B.

FIG. 2 is a block diagram showing an electronic circuit configuration of the gateway device 13A (13B) in the Internet telephone system.

This gateway device 13A (13B)
Has a telephone line interface 21 connected to the public telephone line 12A (12B) and a packet network interface 22 connected to the Internet 14. The telephone line interface 21 has a telephone line input ring buffer 23 and a telephone line output ring. The buffer 24 is connected.

The telephone line input ring buffer 23 is connected to a frequency analysis unit 25 and a voice coding unit 26, and the output of the frequency analysis unit 25 is connected to a main control unit 27. The output is connected to the main control unit 27 through the latest sound storage ring buffer 28 and directly to the main control unit 27.

An audio decoding unit 29 is connected between the main control unit 27 and the telephone line output ring buffer 24.

On the other hand, a packet input ring buffer 30 is connected between the packet network interface 22 and the main control unit 27 and between the voice decoding unit 29 and
A packet output ring buffer 31 is connected between the packet network interface 22 and the main control unit 27 and between the voice encoding unit 26.

The packet network interface 2
2 is connected to a packet collision counter 32, and an output of the packet collision counter 32 is connected to the main control unit 27.

The main controller 27 includes a fixed voice data storage memory 33, a remote telephone number storage memory 34,
A local telephone number storage memory 35, a remote voice message storage memory 36, and a local voice message storage memory 37 are provided.

The telephone line interface 21 receives an incoming call,
It has basic functions of calling, answering, and disconnecting. In a call connection state, the input voice from the telephone line 12A (12B) is sequentially taken into the telephone line input ring buffer 23, and conversely, the telephone line output ring buffer 24 The data is sequentially transmitted as voice to the telephone line 12A (12B).

The latest voice storage ring buffer 28 is a buffer for sequentially storing voices newly coded in the voice coding unit 26, and sequentially overwrites old voices even without reading out the voices, and the current voices are stored for a predetermined period of time. Save conversation voice up to. Here, the fixed period for storing the conversation voice is 10 seconds. This latest voice storage ring buffer 2
The stored voice data in 8 can be read by the main control unit 27.

The audio encoding section 26 is a DSP (digital sig
The ITU-T (International Telecommunications U.S.A.
nion-Telecommunication Sector) 729 and the like, and a header indicating the audio data is added to the encoded data and written into the packet output ring buffer 31. In addition, the audio encoding unit 26, in accordance with an instruction from the main control unit 27 to start / stop the latest audio storage in the latest audio storage ring buffer 28,
The voice storage is written into the latest voice storage ring buffer 28.

The audio decoding unit 29 is constituted by a DSP, and is written in the packet input ring buffer 30.
Every time voice data of a packet having a header indicating voice data arrives, the voice data is subjected to voice decoding in accordance with the voice coding in the voice coding unit 26 and written into the telephone line output ring buffer 24.

The frequency analysis unit 25 is constituted by a DSP, identifies a push tone signal by built-in signal processing software, and outputs an identification result at the request of the main control unit 27.

The packet network interface 22 is constituted by an independent processor, and transfers the contents of the packet output ring buffer 31 to the Internet (packet network) 14.
To the packet input ring buffer 3.
Store sequentially to 0. Here, the destination address of the packet is simply one of the gateway devices 13A, 13A, 1
3B shall designate each other. The Internet (packet network) 14 has a communication protocol of TC.
In P / IP, voice packets for real-time communication are transmitted using the UDP transport layer, and other packets are transmitted using the TCP transport layer.

The packet collision counter 32 counts the number of times that a packet collision in the Internet (packet network) 14 is detected.
The main control unit 27 checks the counter value of the packet collision at regular time intervals to obtain congestion state change data.

In the fixed voice data storage memory 33, fixed voice message data for announcements and prompts to the user is stored in advance by a required number of types of messages.

The remote telephone number storage memory 34 stores the Internet (packet network) 14 from the gateway device 13A (13B) and the gateway device 13 on the other end.
B (13A), the telephone number of the telephone terminal 11B (11A) of the other party connected via the telephone line 12B (12A) is stored.

The local telephone number storage memory 35 stores the telephone line 11 of the gateway device 13A (13B).
Our telephone terminal 1 connected via A (11B)
The telephone number of 1A (11B) is stored.

The remote voice message storage memory 36 is connected from the gateway device 13A (13B) via the Internet (packet network) 14, the other gateway device 13B (13A), and the telephone line 12B (12A). The message message for the local telephone terminal 11A (11B) input from the remote telephone terminal 11B (11A) is stored.

The local voice message storage memory 37 stores the other party's telephone input from the telephone terminal 11A (11B) on the other side connected via the telephone line 11A (11B) on the side of the gateway device 13A (13B). Terminal 1
The message message for 1B (11A) is stored.

The main controller 27 comprises a CPU (center processor unit), a work memory, and a program memory, and controls the operation of each circuit in the gateway device 13A (13B), and is stored in advance in the program memory. The call control process according to the state transition shown in FIG. 3 is executed according to the performed call processing program.

Next, the operation of the Internet telephone system having the above configuration will be described.

FIG. 3 is a state transition diagram showing the entire operation flow in the gateway device 13A (13B) of the Internet telephone system.

In the initialization process (Ja), each part of the circuit is initialized, and the process proceeds to a standby process (Jb).

In the standby process (Jb), the incoming call is detected and the connection request packet is detected, and the telephone terminal 11A (11B) on the other side is switched to the telephone terminal 11B (11
When making a call to A), the corresponding telephone terminal 11A
When an incoming call is detected from (11B) via the telephone line 12A (12B), an incoming call process (Jc: see FIG. 4)
When a call is made from the other party's telephone terminal 11B (11A) to the other party's telephone terminal 11A (11B),
When a connection request packet is received from the gateway device 13B (13A) of the other party via the Internet (packet network) 14, the process proceeds to a transmission process (Jd: see FIG. 5).

FIG. 4 is a flowchart showing an incoming call process (Jc) in the gateway device 13A (13B) of the Internet telephone system.

When a call is made from our telephone terminal 11A (11B) to the other telephone terminal 11B (11A),
When an incoming call is detected from the corresponding telephone terminal 11A (11B) via the telephone line 12A (12B), the telephone number of the calling terminal 11A (11B), which is the caller, is extracted and the local telephone number is extracted. It is stored in the storage memory 35 and the telephone line 1
The telephone terminal 11A (11B) via 2A (12B)
(Steps c1 and c2).

Then, the telephone terminal 11A (1
1B), it is determined whether or not the telephone number of the partner terminal 11B (11A) has been designated and input as a sub-address along with the incoming call to the gateway device 13A (13B). Counterpart terminal 11
The telephone number of B (11A) is written into the remote telephone number storage memory 34 (step c3 → c4).

On the other hand, if the telephone number of the terminal 11B (11A) has not been designated and input as the subaddress, the input request message of the terminal number stored in the fixed voice data storage memory 33 in advance is displayed as "the telephone of the other party". "Please enter a number" is read out, and is sequentially decoded into audio data by the audio decoding unit 29, and the telephone terminal 11A (11B) on the local side is transmitted from the telephone line interface 21.
(Step c3 → c5).

Then, the telephone terminal 11A (1
1B) via the telephone line 12A (12B) to the telephone line input ring buffer 23 of the gateway device 13A (13B) by a push tone.
When the telephone number of B (11A) is input, the frequency analysis unit 25 analyzes a DTFM (Dual Tone Multi Frequency) signal to obtain the telephone number of the partner terminal 11B (11A). It is written (step c6).

Then, the telephone number of the partner terminal 11B (11A) stored in the remote telephone number storage memory 34 and the telephone number of the terminal 11A (11B) stored in the local telephone number memory 35 are included. A connection request packet, which is a TCP packet to which a header indicating a connection request is added, is generated, and the packet network output ring buffer 31
And is output from the packet network interface 22 to the Internet (packet network) 14 and transmitted to the gateway device 13B (13A) on the other side (step c7). Then, the process proceeds to a call process (Je) (step c8).

FIG. 5 is a flow chart showing a calling process (Jd) in the gateway device 13A (13B) of the Internet telephone system.

When a call is made from the telephone terminal 11B (11A) of the other party to the telephone terminal 11A (11B) of the other party,
When a connection request packet is received from the other gateway device 13B (13A) via the Internet (packet network) 14, the telephone number of the other terminal 11B (11A) included in the connection request packet is extracted, and the remote telephone number is extracted. The data is written to the storage memory 34 (step d).
1).

The telephone number of the destination terminal 11A (11B) to be called, which is included in the connection request packet, is extracted and written into the local telephone number storage memory 35 (step d2).

Then, a call is made to the telephone line 12A (12B) via the telephone line interface 21 with the telephone number stored in the local telephone number storage memory 35 as the call destination, and the callee, the call destination, is called. Telephone terminal 1
When the telephone line with 1A (11B) is connected, the processing shifts to the telephone call processing (Je) (steps d3, d4, d5).

In the call process (Je), the packet input ring buffer 30, the packet output ring buffer and the packet output ring buffer are transmitted to the voice encoding unit 29 and the voice decoding unit 29 in the gateway devices 13A (13B) on the communication side and the communication side, respectively. 31 is instructed to start inputting / outputting data, and a UDP packet storing audio data is transmitted to each gateway device 13A, 1A.
3B are exchanged with each other via the Internet (packet network) 14 so that the telephone lines 1 connected to both are exchanged.
A call is made between the telephone terminals 11A and 11B beyond 2A and 12B.

Then, our telephone line 12A (12B)
When a normal disconnection operation is performed on the telephone terminal 11A (11B) connected via the
(Jf), and also monitors the packet input ring buffer 30 and checks the gateway device 13B (1
When the reception of the disconnection packet based on the normal disconnection operation at the partner terminal 11B (11A) transmitted from 3A) is detected, the processing shifts to disconnection processing B (Jg). When it is determined that the call quality has decreased because the increment value of the obtained packet collision counter value at regular intervals exceeds a preset threshold value, the process proceeds to a call interruption notification process (Jh). When the input ring buffer 30 is monitored and the reception of the interruption packet based on the determination of the decrease in the communication quality at the partner gateway device 13B (13A) transmitted from the partner gateway device 13B (13A) is detected. Then, the process proceeds to a call interruption process (Ji).

In the disconnection process A (Jf), a disconnection packet, which is a TCP packet to which a header indicating a normal disconnection in the telephone terminal 11A (11B) on the local side is added, is generated and written to the packet output ring buffer 31. Gateway device 13B on the other side via the Internet 14
(13A). And the cutting process B (Jg)
Will be migrated to.

In the disconnection process B (Jg), a normal disconnection operation is performed in the telephone terminal 11A (11B) on the local side, and the disconnection packet is generated and transmitted, or the other party is monitored by monitoring the packet input ring buffer 30. When the reception of the disconnection packet based on the normal disconnection operation at the other terminal 11B (11A) transmitted from the gateway device 13B (13A) on the other side is detected, the audio encoding unit 26 and the audio decoding unit 29 Therefore, an instruction to stop the input / output operation with the packet input ring buffer 30 and the packet output ring buffer 31 is issued, and the connection with the telephone terminal 11A (11B) on the local side via the telephone line 12A (12B) is disconnected. Thereafter, the process shifts to the standby process (Jb).

In the call interruption notifying process (Jh), when it is determined that the call quality has deteriorated because the increment value of the packet collision counter value obtained by the packet collision counter 32 for each predetermined period exceeds a preset threshold value. An interrupt packet, which is a TCP packet indicating disconnection from the telephone terminal 11A (11B) on the local side, is generated and written into the packet output ring buffer 31, and is transmitted via the Internet 14 to the gateway apparatus 13B (13A) on the remote side. Sent out. Then, the process proceeds to a call interruption process (Ji).

In the call interruption processing (Ji), when the interruption packet is generated and transmitted based on the judgment of the decrease in the communication quality on our side, or by monitoring the packet input ring buffer 30, the other party's gateway device 13B (13A) When the reception of the interrupted packet based on the determination of the decrease in the communication quality is detected, the speech encoding unit 26 and the speech decoding unit 2
9, the input / output operation of the packet input ring buffer 30 and the packet output ring buffer 31 is instructed to stop, and a TCP port to which a dedicated port number is allocated is opened in the packet network interface 22, and the other gateway device is opened. Reception of voice message data transmitted from 13B (13A) is started.

This partner gateway device 13B (13
In the process of receiving the voice message data from A), the TCP packet storing the received voice message is written into the remote voice message storage memory 36 until the voice termination code is determined to be received. Then, the process proceeds to message processing (Jj: see FIG. 6).

FIG. 6 is a flowchart showing message processing (Jj) in the gateway device 13A (13B) of the Internet telephone system.

In the call interruption processing (Ji), the input / output operation is stopped based on the judgment of the decrease in the call quality on the caller or the caller, and the caller gateway device 13B (13)
When the reception of the voice message data from A) is started, a message message is input to the other party due to the interruption of the call, which is the coded voice data stored in advance in the fixed voice data memory 33 as the recording start announcement. The request "Call is interrupted due to Internet congestion. Please leave a message. Please speak." The voice decoding unit 29 sequentially decodes the data into voice data. 11A
(11B) (step j1).

Then, the telephone terminal 11A (11
B), the voice data of the message from the terminal 11A (11B) to the partner terminal 11B (11A), which is sequentially acquired by the telephone line input ring buffer 23, is encoded through the voice encoding unit 26. The converted voice data is stored in the local voice message storage memory 37 (step j2).

Then, the telephone terminal 11A (11
B) indicates that the input of the DTMF signal indicating the end of the input of the message message by the push tone is performed by the frequency analyzer 2.
5, the storage of the message in the local voice message storage memory 37 is completed, and the other party's gateway device 13B (1), which has been the communication destination via the Internet (packet network) 14, is called.
The TCP port to which the dedicated port number of 3A) is allocated is opened, and is stored in the local voice message storage memory 37 to which the encoded voice data for the immediately preceding 10 seconds stored in the latest voice storage ring buffer 28 is added. From the terminal 11A (11B) on the other side to the terminal 11 on the other side
The encoded voice message to B (11A) and the voice end code are sequentially TCP-packetized, written to the packet output ring buffer 31, and transmitted to the gateway device 13B (13A) on the other side (step j3 → j).
4).

Then, TCP from the other party to this party
Waiting for packetized message message reception (J
k).

If the telephone terminal 11A (11B) is disconnected during the execution of the message processing, the processing shifts to a callback processing (Jm: see FIG. 7).

In the reception waiting process (Jk), during the elapse of 20 seconds after the start of the reception of the voice message data transmitted from the partner gateway device 13B (13A), which was performed in the call interruption process (Ji), When the receiving and storing of the TCP packetized message from the other party is completed, the received message is reproduced (Jl:
On the other hand, if 20 seconds have elapsed without the arrival of the message in the form of a TCP packet from the other party due to the congestion of the Internet (packet network) 14, the callback processing (Jm: see FIG. 7) ).

FIG. 7 is a flowchart showing a callback process (Jm) in the gateway device 13A (13B) of the Internet telephone system.

During the execution of the message processing (Jj) for acquiring, storing, and packetizing the message from the telephone terminal 11A (11B), if the telephone terminal 11A (11B) is disconnected, or In the reception waiting process (Jk), if 20 seconds have elapsed since the message message converted into the TCP packet from the other party has not arrived due to the congestion of the Internet (packet network) 14, the process proceeds to the callback process (Jm). Then, the telephone line 12A (1) with our telephone terminal 11A (11B)
3A) is temporarily disconnected, and thereafter, the counterpart terminal 11B is disconnected.
T from (11A) to our terminal 11A (11B)
The process waits until the reception and storage of the CP packetized message message is completed (steps m1 and m2).

When the reception of the TCP packetized message message from the other party and the storage in the remote voice message storage memory 36 are completed, the local terminal 11A ( The telephone line 12A (12B) via the telephone line interface 21 with the telephone number of 11B) as a destination.
Is made (step m2 → m3).

Then, our telephone terminal 11A (11B)
When the telephone line 12A (12B) is connected again, the process proceeds to a received message reproduction process (Jl: see FIG. 8) (step m4 → m5).

FIG. 8 is a flowchart showing a received message reproduction process (Jl) in the gateway device 13A (13B) of the Internet telephone system.

After the reception waiting process (Jk), the reception of the TCP packetized message message from the other party and the storage in the remote voice message storage memory 36 are completed, or the callback process (Jm:
After the completion of the reception and storage of the message message from the other party through the process shown in FIG. 7), the telephone terminal 11A (11B) on the other party side
When the reconnection process is performed and the process proceeds to the received message playback process (Jl), the message playback start message “play back the stored message” stored in the fixed voice data storage memory 33 in advance. Is read out, sequentially decoded into audio data by the audio decoding unit 29, and transmitted from the telephone line interface 21 to the telephone terminal 11A (11B) on the local side (step l).
1).

Then, the coded message to the terminal 11A (11B) is read from the terminal 11B (11A) stored in the remote voice message storage memory 36, and the voice data is sequentially read by the voice decoder 29. And transmitted and reproduced from the telephone line interface 21 to the telephone terminal 11A (11B) on our side (step 12).

During a process of reproducing the message message of the other party to the telephone terminal 11A (11B) of the other party, when a connection request packet is received from the gateway apparatus 13B of the other party, the called party included in the connection request packet is recalled. Terminal phone number and current local phone number storage memory 3
Our telephone terminal 11A (11B) stored at 5
And the telephone number of the other party's telephone terminal 11B (11A) currently stored in the remote telephone number storage memory 34. If it is determined that 一致 一致 と 一致 移行 一致 一致 一致 ｅ 一致 通話 通話 移行 移行 移行 再 び 移行 移行 再 び 移行 移行 再 び 移行 再 び 再 び 再 び さ れ て 再 再 ス テ ッ プ ス テ ッ プ ス テ ッ プ ス テ ッ プ ス テ ッ プ ス テ ッ プ ス テ ッ プ.

On the other hand, the other party gateway device 13B
When the reproduction processing of the other party's message message to the telephone terminal 11A (11B) of the other party is completed without receiving the connection request packet from, the processing is shifted to the disconnection processing B (Jg), and the voice encoding unit 26, The audio decoding unit 29 is instructed to stop the input / output operation with the packet input ring buffer 30 and the packet output ring buffer 31, and the telephone terminal 11 on the local side via the telephone line 12A (12B).
The connection with A (11B) is disconnected, and thereafter, the process proceeds to the standby processing (Jb).

FIG. 9 is a sequence diagram showing a specific operation flow in the gateway device 13A (13B) of the Internet telephone system.

First, when the first telephone terminal 11A designates the telephone number of the second telephone terminal 11B as an incoming call subaddress and transmits the first gateway device 13A as an incoming telephone number (S1), the first telephone terminal 11A receives the first telephone device 13A as the incoming telephone number (S1). The first telephone terminal 11A is the first
(S4: steps c1 and c2).

At the same time, the first gateway device 13
In A, the called sub address becomes a remote telephone number (step c3 → c4), and a connection request is sent to the second gateway device 13B through the TCP transport layer on the Internet (packet network) 14 (step c3). S2: Step c7).

In the second gateway device 13B receiving the connection request from the first gateway device 13A,
The received remote telephone number becomes the local telephone number and the second
Is called to the telephone terminal 11B (S3: Step d)
1 to d3), connected via the telephone line 12B (S
5: Step d4).

Thus, the first telephone terminal 11A and the second
Between the two telephone lines 12A and 1B.
The communication state is established via the Internet (packet network) 14 via 3B and the Internet (packet network)
The call voice on 14 is exchanged by the UDP transport layer (S6: call processing Je).

Here, when congestion of the Internet 14 is detected in the first gateway device 13A, the TC
The interruption packet is sent to the second gateway device 13B via the P transport layer (S7: call interruption notification process Jh), and the first and second gateway devices 13A and 13B are transmitted.
In B, the exchange of the call voice is stopped (call interruption processing J
i), corresponding first and second telephone terminals 11A,
11B, a message message input request "Call is interrupted due to Internet congestion. Please leave a message. Please speak." (S8, S
9: Step j1).

When voice messages for the partner terminals are received from the first and second telephone terminals 11A and 11B to the corresponding gateway devices 13A and 13B, respectively, and stored as local messages (S10, S1).
1: Steps j2 and j3), the first and second gateway devices 13A and 13B send local voice messages to which the last 10 seconds of speech have been added to the other gateway devices on the Internet (packet network) 14. Transfer is started through the TCP transport layer (S
12, S15: Step j4).

Here, when the transfer of the voice message from the first gateway device 13A to the second gateway device 13B is completed within 20 seconds (S12: reception waiting process Jk), the second gateway device 13B causes the second gateway device 13B to perform the second transfer. The voice message from the first telephone terminal 11A is reproduced to the other telephone terminal 11B (S13: Steps 11 and l).
2), the telephone line 12B with the second telephone terminal 11B is disconnected (S14: steps 13 → 14).

On the other hand, the transfer of the voice message from the second gateway device 13B to the first gateway device 13A took more than 20 seconds due to the influence of the deterioration of the communication quality due to the congestion of the Internet 14 (S15: reception waiting process J).
k) At that time, the first gateway device 13A once disconnects the telephone line 12A for the first telephone terminal 11A (S16: step m1), and further transfers the voice message from the second gateway device 13B. Wait for the reception to be completed and the local telephone number 1
Outgoing call is made with the telephone terminal 11A of the call destination (S)
15, S17: Step m2 → m3).

Thus, the first gateway device 13
When the first telephone terminal 11A is reconnected to A via the telephone line 12A, a voice message from the second telephone terminal 11B, which is a remote voice message transferred and received from the second gateway device 13B, is reproduced. (S
18, S19: Step m4 → m5).

Here, when the user of the second telephone terminal 11B tries to contact the user of the first telephone terminal 11A again, the second gateway device 13B is called and the first telephone terminal 11B is called. When the telephone number of the terminal 11A is designated as the incoming call sub-address and a call is made (S20), the second gateway device 13B sends a call to the first gateway device 13A.
A connection request including the telephone number of the first telephone terminal 11A to be the remote telephone number is transmitted over the Internet 14 through the TCP transport layer (S21).
The second telephone terminal 1 is connected to the second gateway device 13B.
1B is connected via the telephone line 12B (S22: steps c1 to c3 → c4, c7).

Thus, the first gateway device 13A
To the first telephone terminal 11A with respect to the second telephone terminal 1
While the voice message obtained from 1B is being reproduced, a connection request from the second gateway device 13B is received, and the remote telephone number and the local telephone number included in the connection request are being reproduced. The telephone number of the first telephone terminal 11A and the second telephone terminal 11
When it is confirmed that the telephone number matches the telephone number of B, the telephone call between the first and second telephone terminals 11A and 11B via the Internet 14 is resumed, and the voice of the telephone becomes U
Exchanged through the DP transport layer (step S
23: Steps 11 to 13 → 15 to 17).

Then, at the second telephone terminal 11B,
With the end of the call, the second gateway device 13
When the telephone line 12B for B is disconnected (S24: call processing Je), a disconnection packet is transmitted from the second gateway device 13B to the first gateway device 13A through the TCP transport layer (S25: disconnection process). (A) Jf), the telephone line 12A with the first telephone terminal 11A to the first gateway device 13A is disconnected (S26: disconnection processing (B) Jg). Then, the first and second gateway devices 13A and 13B are returned to the standby state (standby processing Jb).

Therefore, according to the Internet telephone system having the above configuration, the gateway device 13A (13
Based on the number of packet collisions on the Internet (packet network) 14 counted by the packet collision counter 32 of B), between one telephone terminal 11A and the other telephone terminal 11B through the UDP transport layer of the Internet 14 When it is determined that the quality of the real-time call is reduced, the one gateway device 13A that has determined that the quality of the call is lowered sends a call to the other telephone terminal 11A connected via the telephone network 12A. A voice for requesting the input of a message message to the terminal is reproduced, the message message input from the one telephone terminal 11A is fetched and stored, and the gateway device 13B of the other party is transmitted in non-real time through the TCP transport layer on the Internet 14. Sent to. In addition, TC
The interruption packet due to the P packet is also transmitted to the gateway device 13B on the other side, so that the message message from the other telephone terminal 11B is also captured and stored in the gateway device 13B on the other side, and the non-real-time message is transmitted through the TCP transport layer. Is transmitted to the one gateway device 13A. And each gateway device 1
In 3A (13B), when the message of the other party by the TCP packet is received and stored within 20 seconds, it is reproduced to each corresponding telephone terminal 11A (11B). After the connection with the corresponding telephone terminal 11A (11B) is temporarily cut off and the receiving and saving of the message message of the other party is completed, the connection with the corresponding telephone terminal 11A (11B) is made by callback and the message message is reproduced. Therefore, even when real-time communication becomes difficult due to the deterioration of the communication quality on the Internet 14, it is possible to reliably and quickly transmit the minimum necessary message to the user of the partner terminal.

Further, according to the Internet telephone system having the above-mentioned configuration, during reproduction of a message of the other party to the corresponding terminal 11A (11B) in each of the gateway apparatuses 13A (13B), the other party's gateway apparatus by recalling from the other party's terminal 11B. When the connection request from the terminal 13B is received, the destination terminal telephone number and the called terminal telephone number included in the connection request and the real-time communication state immediately before being stored in the current gateway device 13A are set. It is determined that the other party terminal phone number matches the other party terminal phone number, and the call processing by reconnection with the other party terminal 11B via the Internet 14 is restarted. Waste can be eliminated.

Further, according to the Internet telephone system having the above configuration, each gateway device 13A (13B)
, The other party gateway device 13B of the message message fetched and stored from the corresponding terminal 11A (11B).
At the time of transmission to (13A), the conversation voice for the immediately preceding 10 seconds is also transmitted and reproduced to the partner terminal 11B (11A). Even in the case where there is a continuous transition from the end of a conversation to the incorporation of a message message, the context can be reliably transmitted to the user of the partner terminal.

In the first embodiment, the means for identifying the caller telephone terminals 11A and 11B associated with the incoming call at each of the gateway devices 13A and 13B is not limited to the one using the caller telephone number notification. This may be performed by speaker recognition processing for analyzing the characteristics of a tone signal or voice.

In the first embodiment, one telephone terminal 11A (11B) has been described as being connected to one gateway device 13A (13B). The number of telephone terminals 11A (11B) to be connected and the number of gateway devices 13A (13B) are not limited, and vary depending on the assembly of the communication network, and the address may be selected individually. It is.

As the means for detecting the congestion degree of the Internet (packet network) 14 in the first embodiment, in addition to using the packet collision counter 32, a packet loss counter, a packet delay counter, a packet data total amount counter and the like are used. Alternatively, the detection may be performed by acquiring a notification of the congestion degree from the router.

The telephone line 12A (12B) in the first embodiment may be a public telephone network or a private telephone network.
A and 12B may be lines included in the same telephone network.

Further, in the first embodiment, the gateway device 13A (1
3B), the telephone terminal 11A (11B) is reconnected via the telephone line 12A (12B), and the voice message received and stored from the gateway device 13B (13A) of the other party is reproduced and transmitted. 11A (11
The configuration may be such that the voice message is reproduced and transmitted after the incoming call reconnection to the gateway device 13A (13B) from the B) side.

In the first embodiment, a packet collision (congestion) on the Internet (packet network) 14 in the Internet telephone system is detected to determine a decrease in call quality, and a communication between the telephone terminals 11A and 11B is determined. In the above-described embodiment, the message message is stored, transferred to the other party, and played back. However, as described in the second embodiment, for example, when a mobile telephone terminal is used, when a mobile phone terminal is used, the radio wave condition deteriorates. The switching base station may judge whether the communication quality is deteriorated or disconnected, and transmit and reproduce the message message stored in advance at the beginning of the call to the partner terminal.

(Second Embodiment) FIG. 10 is a diagram showing an entire configuration of a telephone system including a mobile terminal according to a second embodiment of the present invention.

In the telephone system including the mobile terminal, a mobile telephone terminal 41 and a fixed telephone terminal 42 are connected via a telephone line 43A, an exchange 44, and a telephone line 43B.

FIG. 11 is a block diagram showing a configuration of an electronic circuit of the exchange 44 in the telephone system including the mobile terminal.

This exchange 44 is provided with telephone line interfaces 51a and 51 for communicating with at least two terminals.
1b, and a circuit block on one telephone line interface 51a side and the other telephone line interface 5
The circuit block on the 1b side is composed of exactly the same components.

One telephone line interface 51a
The circuit blocks belonging to (51b) include a telephone line input ring buffer 52a (52b), a telephone line output ring buffer 53a (53b), and a frequency analysis unit 54a (54).
b), local telephone number storage memory 55a (55b),
It is composed of a local voice message storage memory 56a (56b), and the operation of each telephone line circuit block is controlled by the main control unit 57. The main control unit 57 includes a fixed voice data storage memory 58.

A line switch 59 is provided between one telephone line block and the other telephone line block to switch or isolate a communication path between the telephone lines.

In this exchange 44, a telephone line input ring buffer 52a (52b) and a telephone line output ring buffer 53a (53b) are directly connected to the main control unit 57.
The audio data is handled as an uncompressed audio format.

The main control section 57 comprises a processor unit such as a CPU, a working memory, and a program memory.

Next, the operation of the telephone system including the mobile terminal having the above configuration will be described.

FIG. 12 is a state transition diagram showing the entire operation flow in the exchange 44 of the telephone system including the mobile terminal.

In the initialization process (Kn), each part of the circuit is initialized, and the process proceeds to a standby process (Ko).

In the standby process (Ko), the incoming call detection and the connection request event are detected, and one terminal 41 (4
When calling the other terminal 42 (41) from 2),
When an incoming call is detected from the one terminal 41 (42) via the telephone line interface 51a (51b),
The process proceeds to an incoming call process (Kp: see FIG. 13). If a connection request event from one telephone line block side to the other telephone line block side is detected by this incoming call process, an outgoing call process (Kq: FIG. 14).

FIG. 13 is a flowchart showing the incoming call processing (Kp) in the exchange 44 of the telephone system including the mobile terminal.

From one terminal 41 (42) to the other terminal 4
When making a call to 2 (41), the one terminal 41 (4)
When an incoming call is detected from 2) via the telephone line interface 51a (51b), when another incoming vacant telephone line is detected, that is, when an incoming call is detected via one telephone line 43A, the other telephone line 43B is detected. Is checked to see if it is free and selected as a calling line (step p
1) If not available, the one terminal 41 (4
The connection to 2) is rejected and the standby state is maintained.

When the other telephone line 43B (43A) is selected as the outgoing line by the detection of an incoming call from one terminal 41 (42) via the one telephone line 43A (43B), one of the terminals on the calling side is selected. The telephone number of the terminal 41 (42) is written in the one local telephone number memory 55a (55b), and the one telephone line interface 51a (51
The line to the one terminal 41 (42) is connected by b) (steps p1, p2, p3).

When the telephone number of the partner terminal 42 (41) is specified as the called sub-address following the incoming call from the one terminal 41 (42), it is selected as the calling line. The data is written into the local telephone number storage memory 55b (55a) on the other telephone line block side (step p4 → p5).

On the other hand, if the telephone number of the other party terminal 42 (41) is not specified as a subaddress, the input request message of the other party terminal number stored in the fixed voice data storage memory 58 in advance is displayed. "Please input" is read out, and one terminal 41 which is the calling source is read from one telephone line interface 51a (51b).
(42) (Step p4 → p6).

Then, the telephone number of the partner terminal 42 (41) by a push tone from the one terminal 41 (42) to the one telephone line input ring buffer 52a (52b) via the telephone line 43A (43B). Is input, the frequency analysis unit 54a (54b)
(Dual Tone Multi Frequency) The signal is analyzed to obtain the telephone number of the other terminal 42 (41), and the local telephone number storage memory 55b (55
a) is written (step p7).

Then, based on the number format of the telephone number of one terminal 41 (42) which is the caller stored in one local telephone number storage memory 55a (55b), it is determined whether the terminal is a mobile terminal. It is determined whether or not the mobile terminal is a mobile terminal. If it is determined that the mobile terminal is a mobile terminal, the guide data of the disconnection method message and the message input request stored in advance in the fixed voice data storage memory 58 are displayed. Please press * before you hang up. Please insert a voice message to tell the other party if you hang up without pressing * "and read the calling mobile terminal from one telephone line interface 51a (51b). (Steps p8 → P9).

The telephone line input ring buffer 52a (52b) sent from the one terminal 41 (42).
The local voice message storage memory 56a () until the frequency analysis unit 54a (54b) analyzes and detects the input of the DTMF signal indicating the end of the input by a push tone is sequentially obtained. 56b) (step p10).

Then, a connection request event to which a header indicating a connection request is added is given to the control system on the other telephone line block side selected as the calling line, and the process shifts to a call process (Kr) ( Step p11,
p12).

FIG. 14 is a flowchart showing a calling process (Kq) in the exchange 44 of the telephone system including the mobile terminal.

By the incoming call process (Kp), a connection request event to which a header indicating a connection request is added is given from the control system on one telephone line block side to the control system on the other telephone line block side. When the process is shifted to the calling process, the telephone number of the other terminal 42 (41), which is the called party, stored in the local telephone number storage memory 55b (55a) on the other telephone line block side is designated, and Telephone line interface 51b (5
A call is made to the other telephone line 43B (43A) via 1a), and the process stands by until the connection with the other terminal 42 (41) is completed (steps q1 and q2).

The other terminal 42 (4) via the other telephone line interface 51b (51a) of the exchange 44
When the line connection with 1) is completed, the other local telephone number storage memory 55b (55a) stores the telephone number of the other terminal 42 (41) based on the number format of the telephone number of the other terminal 42 (41). Is determined to be a mobile terminal, and if it is determined that the mobile terminal is a mobile terminal, the guide data of the disconnection method message and the message input request stored in advance in the fixed voice data storage memory 58 are referred to as "telephone If you want to hang up, press * on the push button before you hang up. Please insert a voice message that tells the other party if you hang up without pressing * "and the other telephone line interface 51b (51a) Is transmitted to the other terminal 42 (41), which is the destination mobile terminal (step q2 → q3 → q4).

The telephone line input ring buffer 52b (52a) sent from the other terminal 42 (41).
The voice data of the message message sequentially acquired in the other local voice message storage memory 56b (until the frequency analysis unit 54b (54a) analyzes and detects that the DTMF signal indicating the end of the input is input by the push tone is detected. 56a), and thereafter, the call processing (K
r) (steps q5 and q6).

In the call processing (Kr), the line switch 59
Thus, a communication path is formed between the one telephone line interface 51a and the other telephone line interface 51b, and the communication between one terminal 41 and the other terminal 42 is performed.

Then, the telephone line interface 51a
If the disconnection by the terminal 41 (42) is directly detected via (51b), the disconnection reason determination process (Ks: FIG.
5), and when a normal disconnection event is detected from the control system of one of the telephone line blocks to the control system of the other telephone line block, the process proceeds to a normal disconnection process (Kt). Further, when an abnormal disconnection event is detected from the control system on one of the telephone line block sides to the control system on the other telephone line block side, the processing shifts to remote abnormal disconnection processing (Kv: see FIG. 17). Is done.

FIG. 15 is a flowchart showing the disconnection reason determining process (Ks) in the exchange 44 of the telephone system including the mobile terminal.

When the disconnection from one or the other terminal 41 (42) is detected, the local telephone number storage memory 55a in the telephone line block on the side where the disconnection is detected.
When the number format of the telephone number stored in (55b) indicates a mobile terminal (step s1)
If it is determined that the DTMF signal analyzed last by the frequency analysis unit 54a (54b) is not a push tone corresponding to "*" indicating a normal disconnection (step s2), the other party's telephone An abnormal disconnection event is given to the control system of the line block (step s).
3), the process proceeds to a local abnormal disconnection process (Ku: see FIG. 16) (step s4).

On the other hand, the local telephone number storage memory 55a in the telephone line block on the side where the disconnection was detected.
If the number format of the telephone number stored in (55b) does not indicate that the terminal is a mobile terminal, or even if it is determined that the terminal is a mobile terminal, "*" means a normal disconnection.
If it is determined that the push tone corresponding to is finally detected by analysis, a normal disconnection event is given to the control system of the other party's telephone line block, and the process shifts to a normal disconnection process (Kt) (step). s1 or 2 → s
5).

In the normal disconnection process (Kt), the communication path between one telephone line interface 51a and the other telephone line interface 51b is disconnected by the line switch 59, and the telephone line 43A between the one and the other terminals 41 and 42 is disconnected. , 43B are disconnected, and the process returns to the standby process (Ko).

FIG. 16 shows a local abnormal disconnection process (Ku) in the exchange 44 of the telephone system including the mobile terminal.
It is a flowchart which shows.

For example, when the push tone corresponding to “*” is not detected from the mobile terminal 41 whose disconnection is detected by the disconnection reason determination processing (Ks), the control system on one telephone line block side transmits the other telephone. When an abnormal disconnection event is generated in the control system on the line block side and the process is shifted to the local abnormal disconnection processing, the one line interface 51 is operated by the line switch 59.
a and the telephone path between the other telephone line interface 51b is disconnected, and the telephone line 43 with the mobile terminal 41 is disconnected.
A is disconnected (steps u1 and u2).

Thereafter, if there is an incoming call within 10 minutes, the terminal telephone number of the caller from which the incoming call is detected and the local telephone number storage memory 55 on one telephone line block side.
It is determined whether or not the telephone number of the mobile telephone terminal 41 whose line has already been disconnected already matches the telephone number already stored in a (step u3 → u4 → u3 → u5).

Here, if the terminal telephone number of the caller whose incoming call is detected matches the telephone number of the mobile telephone terminal 41 already stored in the local telephone number storage memory 55a on one telephone line block side. If determined,
The process proceeds to the reconnection and message reproduction process (Kw: see FIG. 18) (step u5).

On the other hand, if the terminal telephone number of the caller whose incoming is detected does not match the telephone number of the mobile telephone terminal 41 already stored in the local telephone number storage memory 55a of one telephone line block. If determined,
The connection with the receiving terminal is rejected, and after 10 minutes, the process returns to the standby process (Ko) (step u5 → u7).
→ u3 → u4 → u8).

FIG. 17 shows remote abnormal disconnection processing (Kv) in the exchange 44 of the telephone system including the mobile terminal.
It is a flowchart which shows.

For example, when an abnormal disconnection event is detected from the control system on one telephone line block side to which the mobile terminal 51 is connected to the control system on the other telephone line block side, the process is shifted to the remote abnormal disconnection processing. Then, the communication path between the one telephone line interface 51a and the other telephone line interface 51b is disconnected by the line switch 59, and the guide voice for starting reproduction of the other party message stored in the fixed voice data storage memory 58 in advance. The data "the call was disconnected due to deterioration of the radio wave condition of the partner terminal. The message stored in advance is reproduced" is read out and sent to the telephone terminal 42 via the other telephone line interface 51b (step). v1, v2).

Then, the message voice message from the mobile terminal 41 stored in the one local voice message storage memory 56a is read out, sent to the telephone terminal 42 via the other telephone line interface 51b, and reproduced. (Step v3).

Then, the guide data of the message input request stored in advance in the fixed voice data storage memory 58 is read out. The data is transmitted to and reproduced from the P.42 (step v4).

[0139] As for the voice data of the message transmitted from the telephone terminal 42 and sequentially acquired by the telephone line input ring buffer 52b, the frequency analysis unit 5 indicates that the DTMF signal indicating the end of the input is input by the push tone.
4b (54a), it is stored in the other local voice message storage memory 56b until it is analyzed and detected, and thereafter, the process proceeds to the normal disconnection process (Kt) (step v).
5, v6).

FIG. 18 is a flowchart showing the reconnection and message reproduction processing (Kw) in the exchange 44 of the telephone system including the mobile terminal.

In the local abnormal disconnection processing (Ku), the mobile terminal 41 whose disconnection is detected and its telephone line 4
After 3A is disconnected, 10 from the same mobile terminal 41
When it is determined that an incoming call is received within minutes and the process is shifted to the reconnection and message playback processing, the telephone line 43A with the mobile terminal 41 is connected, and the other party's message playback start stored in the fixed voice data storage memory 58 in advance is started. Is read out and transmitted to the mobile terminal 41 via the one telephone line interface 51a (step w1).

Then, the remote abnormal disconnection processing (K
v: see FIG. 17), the message voice message from the telephone terminal 42 stored in the other local voice message storage memory 56b is read out, transmitted to the mobile terminal 41 via one telephone line interface 51a, and reproduced. (Step w2).

Thereafter, the process proceeds to the normality judgment processing (Kt), the communication path between the one telephone line interface 51a and the other telephone line interface 51b is disconnected, and the connection between the one and the other terminals 41 and 42 is established. It is disconnected (step w3).

FIG. 19 is a sequence diagram showing a specific operation flow in the exchange 44 of the telephone system including the mobile terminal.

When the first terminal 41 designates the telephone number of the second terminal 42 as the called subaddress and transmits the exchange 44 as the called telephone number (X1), the first terminal 4
1 is connected to the exchange 44 via the telephone line 43A (X2:
Steps p1 to p3).

At this time, when the exchange 44 determines that the first terminal 41 is a mobile terminal, the disconnection method and the guide voice of the message input request are displayed. Please press and hang up. Please insert a voice message to tell the other party if you hang up without pressing * ”(X3: step p8 → p9) and sent from the mobile terminal 41 in order. The acquired message voice message is stored (X4: step p10).

Then, a call is made to the telephone terminal 42 to which the call is to be made, and the telephone line 43B is connected (X5,
X6: Steps q1 and q2), a call path is formed between the mobile terminal 41 and the telephone terminal 42, and a call is started (X7: call processing Kr).

When the disconnection of one mobile terminal 41 is detected during the call processing between the one mobile terminal 41 and the other telephone terminal 42 (X8), the line disconnection with the detected mobile terminal 41 is performed. (X9: Step v)
1) A guide sound for starting the reproduction of the message of the other party "The call was disconnected due to the deterioration of the radio wave condition of the other terminal. The message stored in advance is reproduced" is sent to the other telephone terminal 42, The message voice message from the mobile terminal 41 stored in X4 is also transmitted and reproduced (X10: steps v2 and v3).

Further, a guide voice of a message input request, "Please leave a message, please speak," is transmitted and reproduced to the other telephone terminal 42 (X11: step v4). The transmitted message voice message is saved and (X1
2: Step v5), telephone line 4 with the telephone terminal 42
2 is also disconnected (X13: normal disconnection processing Kt).

On the other hand, if there is an incoming call from the same mobile terminal 41 within 10 minutes after the disconnection from the mobile terminal 41 (X14: steps u1 to u5),
The telephone line 43A with the mobile terminal 41 is connected (X1
5), a guide voice for starting reproduction of the other party's message "reproduce the stored message" and the message voice message from the telephone terminal 42 stored in the X12 are transmitted and reproduced (X16: steps w1 and w2). Then, the telephone line 43A with the mobile terminal 41 is disconnected (X17: step w3).

Therefore, according to the telephone system including the mobile terminal having the above-described configuration, when an abnormal disconnection from the mobile terminal 41 is detected in the exchange 44 due to the deterioration of the radio wave condition on the mobile terminal 41 side, While the connection with the other party's telephone terminal 42 is maintained, the message message from the other party's terminal is fetched and saved, and when there is an incoming call from the same mobile terminal 41 within 10 minutes, the connection is re-established. Thereafter, the message message captured and stored from the partner terminal 42 is played back, so that the user of the partner telephone terminal 42 at least transmits a message desired to be conveyed to the user of the mobile terminal 41 whose call has been disconnected. Not only can the user of the other party terminal 42 leave the message message necessary while the conversation contents immediately after the disconnection of the call are stored, and the user can transfer the message. The user of the terminal 41 is radio waves (call)
The time spent waiting for the recovery of the state can be effectively used.

Further, according to the telephone system including the mobile terminal having the above-described configuration, a call is made from one mobile terminal 41 to the exchange 44 by designating the telephone number of the other telephone terminal 42 as an incoming call subaddress. If it is determined that the calling terminal is a mobile terminal 41 with an unstable element in the call state based on the number form of the terminal telephone number of the calling source, When the message on the mobile terminal 41 is disconnected due to the deterioration of the radio wave condition, the message is sent to the telephone terminal 42 of the other party whose connection is maintained from the exchange 44.
Since the pre-stored message message from the mobile terminal 41 is reproduced, the time that the user of the mobile terminal 41 spends waiting for the recovery of the radio wave (communication) state can be effectively used to minimize the necessary time. The message can be conveyed reliably.

In the second embodiment, the mobile terminal 4
The explanation above has been directed to the case of abnormal disconnection caused by the deterioration of the radio wave condition in the connection with 1. However, the same message transmission function should be applied to the abnormal line disconnection between ordinary fixed telephone terminals. Can be.

The telephone line 43A (43B) in the second embodiment may be a public telephone network or a private telephone network.
A and 43B may be lines included in the same telephone network.

Further, each terminal 4 in the second embodiment is used.
When saving a voice message from
A configuration may be adopted in which the call voice for a certain period immediately before, which is stored in the latest voice storage ring buffer 28 or the like shown in the embodiment, is added and stored, and transmitted to the partner terminal.

In the second embodiment, after the line with the terminal whose disconnection has been detected is once disconnected, the voice message from the partner terminal is reproduced and transmitted after waiting for reconnection due to an incoming call from the same terminal. Although the configuration has been described, a configuration may also be adopted in which the connection with the same terminal already disconnected by the callback from the exchange 44 is performed, and the voice message from the partner terminal is reproduced and transmitted.

Further, the Internet telephone system in the first embodiment and the mobile and fixed-line telephone systems in the second embodiment may be appropriately combined.

[0158]

As described above, according to the communication assisting apparatus of the communication system according to the first aspect of the present invention, when the communication quality deterioration detecting means detects a decrease in the communication quality in the communication network, the communication quality is reduced. The real-time call via the communication network is interrupted by the call interrupting means, and the voice message from the terminal connected via the telephone network is obtained by the message obtaining means and converted to a non-real-time protocol by the message transferring means. Then, it is transferred to the communication auxiliary device of the other party. Therefore, even when real-time communication becomes difficult due to the deterioration of the communication quality on the communication network, the minimum necessary message can be transmitted to the user of the partner terminal.

Further, according to the second embodiment of the present invention, the voice message transferred by the message transfer means of the other party's call assistance device is stored by the message storage means, and this storage is performed. The transferred voice message from the called party is played back by the message playing means to the terminal on the other side connected to the telephone network. Even in such a case, the minimum necessary messages can be transmitted to the users of the partner terminals.

Further, according to the third embodiment of the present invention, when a real-time call is made via a communication network, the terminal telephone number of the other party and the terminal telephone of the other party are used. The number and the number are stored by the terminal number storage means, and while the voice message is being played back to the terminal on the other side connected to the telephone network by the message playback means, the other party's call assistance device can perform a real-time call. If a terminal connection request for the connection is received, the terminal telephone number of the connection request source matches the terminal telephone number of the other party stored in the terminal number storage means, and the terminal telephone number of the connection request destination And the terminal number stored on the terminal number storage unit is determined by the connection terminal determination unit. If a match is determined, Since the real-time call via the communication network is resumed by the call resuming means, it is possible to immediately reconnect if there is a call from the partner terminal, without having to restart the call from the terminal on our side. .

According to the fourth embodiment of the present invention, when a real-time call is being made through a communication network, the call voice for a certain period immediately before is constantly updated. The voice message from the terminal on our side obtained by the latest voice storage means and stored by the latest voice storage means is added with the call voice for a predetermined time immediately before stored by the latest voice storage means. Since the voice message to which the voice is added is converted into a non-real-time protocol by the message transfer means and transferred to the communication auxiliary device of the other party,
The message content can be accurately transmitted to the user of the partner terminal.

Therefore, according to the present invention, it is possible to compensate for the deterioration of the message transmission performance accompanying the deterioration of the communication quality in telephone communication.

[Brief description of the drawings]

FIG. 1 is a diagram showing an overall configuration of an Internet telephone system according to a first embodiment of the present invention.

FIG. 2 is a block diagram showing a configuration of an electronic circuit of the gateway device in the Internet telephone system.

FIG. 3 is a state transition diagram showing an overall operation flow in the gateway device of the Internet telephone system.

FIG. 4 is a flowchart showing an incoming call process in the gateway device of the Internet telephone system.

FIG. 5 is a flowchart showing a calling process in the gateway device of the Internet telephone system.

FIG. 6 is a flowchart showing message processing in the gateway device of the Internet telephone system.

FIG. 7 is a flowchart showing a callback process in the gateway device of the Internet telephone system.

FIG. 8 is a flowchart showing a received message reproduction process in the gateway device of the Internet telephone system.

FIG. 9 is a sequence diagram showing a flow of a specific operation in the gateway device of the Internet telephone system.

FIG. 10 is a diagram showing an overall configuration of a telephone system including a mobile terminal according to a second embodiment of the present invention.

FIG. 11 is a block diagram showing a configuration of an electronic circuit of an exchange in a telephone system including the mobile terminal.

FIG. 12 is a state transition diagram showing an overall operation flow in the exchange of the telephone system including the mobile terminal.

FIG. 13 is a flowchart showing an incoming call process in an exchange of a telephone system including the mobile terminal.

FIG. 14 is a flowchart showing a calling process in an exchange of a telephone system including the mobile terminal.

FIG. 15 is a flowchart showing a disconnect reason determination process in the exchange of the telephone system including the mobile terminal.

FIG. 16 is a flowchart showing a local abnormal disconnection process in the exchange of the telephone system including the mobile terminal.

FIG. 17 is a flowchart showing remote abnormal disconnection processing in the exchange of the telephone system including the mobile terminal.

FIG. 18 is a flowchart showing a reconnection and message reproduction process in the exchange of the telephone system including the mobile terminal.

FIG. 19 is a sequence diagram showing a specific operation flow in the exchange of the telephone system including the mobile terminal.

[Explanation of symbols]

 11A, 11B: Telephone terminal, 12A, 12B: Telephone line (telephone network), 13A, 13B: Gateway device, 14: Internet (packet network), 21: Telephone line interface, 22: Packet network interface, 23: Telephone line input Ring buffer, 24: telephone line output ring buffer, 25: frequency analysis unit, 26: voice coding unit, 27: main control unit, 28: latest voice storage ring buffer, 29: voice decoding unit, 30: packet input ring buffer 31 packet output ring buffer 32 packet collision counter 33 fixed voice data storage memory 34 remote phone number storage memory 35 local phone number storage memory 36 remote voice message storage memory 37 local voice Message case Memory 41 mobile telephone terminal 42 telephone terminal 43A, 43B telephone line (telephone network) 44 exchange, 51a, 51b telephone line interface 52a, 52b telephone line input ring buffer 53a, 53b Telephone line output ring buffers, 54a, 54b: frequency analysis unit, 55a, 55b: local telephone number storage memory, 56a, 56b: local voice message storage memory, 57: main control unit, 58: fixed voice data storage memory.

──────────────────────────────────────────────────続 き Continued on the front page (51) Int.Cl. ⁷ Identification code FI Theme coat ゛ (Reference) H04Q 11/04 H04Q 11/04 R 9A001 F term (Reference) 5K024 AA13 AA14 AA72 BB05 BB10 CC01 DD01 DD04 EE09 FF06 GG03 GG12 HH04 5K034 CC05 EE12 HH63 5K051 AA01 AA02 BB01 BB02 CC02 DD06 DD13 EE01 EE02 EE04 EE06 EE07 FF02 HH16 HH18 HH26 JJ02 JJ09 JJ13 JJ16 KK01 5K069 AA15 BA12 DB12 FA02 DB02 FA02 DB02 CB02 MM05 MM06 MM07 NN07 NN15 NN23 NN25 NN34 PP03 PP05 PP06 QQ02 QQ08 RR14 SS08 TT02 UU03 UU19 UU20 VV01 VV05 9A001 BZ03 CC02 CZ06 JJ25 LL02 LL09

Claims (7)

[Claims] 1. A communication system for performing a real-time communication between one terminal and another terminal via a communication network, wherein the communication assistance is connected between the telephone network to which the terminal is connected and the communication network. A communication quality detecting means for detecting a decrease in communication quality in the communication network; and a communication quality detection means for detecting a decrease in communication quality in the communication network when the communication quality reduction detecting means detects the decrease in communication quality in the communication network. Call interrupting means for interrupting the real-time call, and acquiring a voice message from a terminal connected via the telephone network when the real-time call via the communication network is interrupted by the call interrupting means. Message acquiring means, and a message for converting the voice message from the terminal acquired by the message acquiring means into a non-real-time protocol and transferring the same to the other party. Call aid call system comprising: the di-transferring means. 2. A message storage means for storing a voice message transferred by a message transfer means of a call auxiliary device of the other party, and a transfer voice message from the other party stored by the message storage means. 2. The apparatus according to claim 1, further comprising: a message playback unit that plays back a terminal connected to the telephone network. 3. A terminal number storage means for storing a terminal telephone number of the user and a terminal telephone number of the other party when a real-time call is made via the communication network. In the state where the voice message is being played back to the terminal on the other side connected to the telephone network, if a terminal connection request for a real-time call is received from the call auxiliary device on the other side, the connection request is received. A match between the original terminal telephone number and the terminal telephone number of the other party stored in the terminal number storage means, and the terminal telephone number of the connection request destination and the terminal telephone number of the other party stored in the terminal number storage means A connection terminal judging means for judging a match with the number; and a real-time call via the communication network is resumed if the connection terminal judgment means makes a match. The call assisting device for a call system according to claim 2, comprising: Further, in a state where a real-time call through the communication network is being performed, a latest voice storage unit that constantly updates and stores a call voice for a predetermined period of time immediately before, and the message transfer unit includes: The voice message from the terminal on the side of the terminal obtained by the message obtaining unit to which the call voice for the predetermined time immediately before stored by the latest voice storing unit is added is converted into a non-real-time protocol to communicate with the other party. 4. The call assist device of the call system according to claim 1, wherein the call assist device is a message transfer unit that transfers the message to the assist device. 5. 5. The communication network according to claim 1, wherein the communication network is the Internet, the real-time communication is performed by a UDP protocol, and the non-real-time protocol is a TCP protocol. Item 5. A call assistance device for a call system according to any one of Items 4. 6. A communication auxiliary device for a communication system connected between a telephone network to which one terminal is connected and a telephone network to which the other terminal is connected and for connecting mutual telephone lines, Disconnection detection means for detecting an abnormal line disconnection for one terminal, and message acquisition means for acquiring a voice message from the other terminal when the disconnection detection means detects a line disconnection for one terminal, Reconnecting means for reconnecting the telephone line to one terminal whose disconnection has been detected by the disconnection detecting means; and for the one terminal reconnected to the telephone line by the reconnecting means, the message obtaining means And a message playback unit that plays back the voice message from the other terminal. 7. A terminal judging means for judging that the connection target terminal is an unstable connection terminal in accordance with connection of a telephone line to the terminal, and a terminal having an unstable connection by the terminal judging means. If it is determined that the terminal is disconnected, a preliminary message acquisition unit that acquires a voice message from the terminal in advance, and if the line disconnection to one terminal is detected by the disconnection detection unit, the line disconnection is detected. 7. The communication system according to claim 6, further comprising: a spare message reproducing unit that reproduces a voice message from the terminal acquired in advance by the preliminary message acquiring unit with respect to the other terminal that is not provided. Call assistance device.