JP2001230862A - Voice relay system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a voice relay server able to house a large-scale of channels that reduces a delay time to the utmost and prevents deterioration in the quality of a reproduced voice in the case of relaying a speech of a telephone set connected to a public network or a speech between mobile phones connected to a mobile network. SOLUTION: The voice relay server relaying a speech of a telephone set is installed at a connecting point between an IP network and a public network or a mobile network. When caller and called communication terminals are telephone sets, the voice relay server selects a PCM with a small processing time and a low processing load for its voice coding system, and selects a coding system used for the mobile network in order to reduce the number of times of code conversion in order to relay a speech with the mobile phone.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a multimedia communication system, and more particularly to a voice data relay system capable of relaying real-time call voice data.

[0002]

2. Description of the Related Art With the spread of the Internet, the integration of telephones into packet-switched networks has attracted attention. A VoIP (Voice over IP) service that relays calls between general telephones by carriers over the Internet has been announced, and experiments have been conducted. Has been started. When implementing a VoIP service, the gateway device performs a process of converting a call control procedure and a process of converting a voice data format. Standards organization ITU
−T, the H.323 recommendation, which summarizes each processing module as a protocol, as a configuration of multimedia communication terminals,
gaco (same as H.248 of ITU-T), sigtran, etc. are under discussion (discussed in megaco-BOF "draft-ietf-megaco-pr
otocol-03.txt (expire: 2000/2)).

Further, as a function of the Internet network side, standardization work of a QoS (Quality of Service) guarantee protocol for securing a band required for communication in advance and performing priority control (for example, RSVP (Resource reSerVation Protocol) and d)
iffserv (differenciated services), etc., and by using this, it has become possible to guarantee the quality of real-time media communications (IETF RF
C2205 “Resource Reservation Protocol (RSVP) Version
1 Functional Specification ", discussed in diffserv-WG" draft-ietf-diffserv-arch-02.txt (expir
e: 99/4) ”, discussion at rap-WG“ draft-ietf-rap-co
ps-rsvp-02.txt (expire: 99/6).

[0004]

In H.323, G.323 is used. 71
1 (PCM: 64 kbps) / G. 723.1 (5.3 / 6.3 kbp
s) / G. 726 (32/16 kbps) / G. 729 (8kbps) etc.
Multiple speech coding schemes are recommended. QoS is not guaranteed, such as dial-up PCs and current ISPs (Internet Service Providers), and network capacity is relatively severe for peak traffic.
When using a VoIP service, the band used is often reduced as much as possible by using a high-compression encoding method.
However, there is a problem that the higher the compression method, the longer the coding process takes, and the longer the delay must be. (For example, in G.723.1, there is an algorithm delay of about 70 ms in one way.) There is). Carriers are switching their current circuit-switched telephone services to packet-switched VoIP
When switching to the service, ordinary telephone users naturally assume that the same quality is guaranteed. If VoIP service is performed on a large-capacity network on the premise of the QoS guarantee function, it can be expected that delay, fluctuation, and packet loss in the network will be reduced more than at present. However, when high compression coding is used or processing corresponding to a large-scale telephone line must be performed, a processing delay is added at the gateway, so that it is not possible to sufficiently cope with the severe quality requirements of the user. Can be considered. In addition, the number of mobile phone and PHS subscribers in recent years is approaching the number of public network telephone subscribers,
There is also a need to connect. However, since the coding method used in the mobile network and the coding method recommended in H.323 are different, once the G. Re-encoding to both schemes after returning to 711 causes a problem that the sound quality is severely degraded, cannot be used for conversation, and the delay increases.

[0005] It is an object of the present invention to provide a gateway that minimizes the processing delay and prevents the quality of conversation from deteriorating.

Another object of the present invention is to provide a gateway which prevents deterioration of call quality between terminals connected to different networks.

[0007]

In order to solve the above-mentioned problems, a voice relay server of the present invention uses PCM as a voice encoding system for telephone calls between telephones connected to a public network.
The system operator sets the allowable delay time in advance, measures the delay time periodically, and checks whether the delay time of the call with the same voice relay server exceeds the allowable value when a new call request occurs It is characterized in that a means for monitoring whether or not there is a call exceeding the allowable value and rejecting the call request is provided.

[0008]

DESCRIPTION OF THE PREFERRED EMBODIMENTS First, an embodiment of the present invention will be described with reference to FIGS.

FIG. 1 shows a configuration example of a voice data relay system according to an embodiment of the present invention.

In FIG. 1, voice relay servers (hereinafter referred to as GWs) 4-1 and 4- are connected to a connection point between the packet network 1, the public telephone network 2, the mobile network 3, and the connection point between the packet network 1 and the public telephone network 2. 2
Has a plurality of routers 5-1 to 5-4 and an IP
The telephone client terminal 7 transmits a telephone 6-1 to the public network 2,
6-2, a mobile network voice relay server 8 (hereinafter referred to as a mobile GW) at a connection point between the packet network 2 and the mobile network 3, and a wireless base station 9 and a mobile terminal 10 at the mobile network 3. It is connected.

FIG. 2 shows the internal configuration of the GW 4.

In FIG. 2, a GW 4 includes a memory 21 in which a communication program and the like are stored, a CPU 20 that performs processing based on the program in the memory 21, and a storage device 22.
And input devices 2 such as a microphone, a keyboard, a mouse, and a pen
3, an output device 24 such as a speaker and a display, and a public network interface 25 for performing communication on the public network 2.
, A packet network interface 26 for performing communication on the packet network 1, hardware for converting voice data, and a communication program developed in the memory 21. Each module is connected by an internal bus. The hardware that performs the audio data conversion processing includes, as transmission processing, an A / D converter 30 that converts analog audio data into digital data, a Coder 31 that encodes digitized data, and audio data that is encoded by the Coder 31. Packetizing processing unit 32 for transmitting the packet data to the packet network, and header information indicating the address information of the communication partner, the voice coding method, and the like in the packetized voice data. For example, in the case of the IP network, an IP header, TCP A network packetizing processing unit 33 for giving a / UDP header, a network packet decomposing unit 34 for taking header information from the voice packet received by the communication network interface 26 as a receiving process, and a data excluding the network header information as a command. Data to be distributed to audio data /
Output data queue 39 in hardware for command distribution processing 35 and, in the case of voice data, voice data conversion processing
To buffer. A data header removal process 40 that obtains one data at a time from the output data queue 39 and decomposes information added to the data and audio data, and a decoding process that performs a decoding process when the audio data is encoded.
r41 and a D / A converter 42 for converting the decoded digital data (PCM) into analog audio data. However, data packetization processing 32, network packetization 33, network packet decomposition 34,
The data / command distribution processing 35, the output data queue 39, and the data header removal processing 40 may be realized by software. The communication program includes, as transmission processing, an event analysis processing 27 for analyzing an event such as a dial tone received from the public network 2, and a command packetization processing for creating a corresponding command when the event is an event related to a call setting. 28, and an outgoing call control processing unit 29 that manages information necessary for creating the command packet and communication status, receives the command information distributed by the data / command distribution processing 35 as reception processing, and analyzes the command type. Command analysis processing 37, an output data generation processing section 38 for creating data for outputting to the public network 2 a command that needs to be notified to the telephone user, and a received call control for managing the status of the call control processing. And a processing unit 36.

The storage device 22 stores at least the address of each GW 4, the number of an area for identifying the telephone 6 or the mobile phone 10 under the control of each GW 4, and the allowable number in the call relay between the GWs. A set of delay times is registered. Further, information on the connection being relayed, at least, the numbers of the calling and called telephones, the address of the receiving GW 4, the coding method, and the estimated delay time are registered in pairs.

FIG. 3 shows an example of a location management table managed by the GW 4.

In FIG. 3, reference numeral 50 denotes an address of a relay GW, 51 denotes information indicating an area of a telephone set under the control of the GW, for example, an area code, 52 denotes an encoding system of the GW, 53 indicates an allowable delay time in relaying with the GW.

FIG. 4 shows an example of a connection management table managed by the GW 4.

In FIG. 4, reference numeral 60 denotes the logical line number of the call currently in communication, 61 denotes the address of the communication terminal on the call request transmitting side, 63 denotes the address of the communication terminal on the receiving side, and 62 denotes the call of the communication terminal indicated by 63. GW4 that controls the communication terminal that is the partner
, 64 indicates an encoding system (may be a bandwidth) used for call relay, and 65 indicates an allowable delay time in call relay.

FIG. 5 shows an example of the overall flow of the processing of the voice data relay system. Here, the processing in the case where the calling and called communication terminals are telephones will be described.

Referring to FIG. 5, when a user using the telephone 6-1 as the communication terminal talks to the called telephone 6-2, the calling telephone 6-1 picks up the receiver (off-hook). Processing 70 for inputting telephone number of telephone 6-2
To connect to the GW 4-1 that controls the calling telephone 6-1. GW4 detecting off-hook of calling telephone 6-1
-1 performs a process 71 for acquiring the telephone number of the calling telephone 6-1 and the telephone number of the receiving telephone 6-2 included in the dialing tone, and performs processing 71 for acquiring the telephone number of the receiving telephone 6-2 from the location management table. Using the telephone number of No. 2 as a key, a process 72 is performed to obtain an allowable value of delay time in relaying between the address of the GW 4-2 of the receiving side that controls the receiving side telephone 6-2 and the GW 4-2 of the receiving side. . From the connection management table, it is detected whether there is a call being relayed to the destination GW4-2, and it is determined whether there is no call or the delay time of the call being relayed is equal to or less than an allowable value. Processing 73 is performed. If the call request is accepted, since the communication terminal on the receiving side is a telephone, the encoding method is determined to be PCM, and the CODER 31 and DECODER 41 are set to P.
A process 74 for setting to perform the CM process is performed. The numbers of the calling and called telephones 6 and the called GW for the connection
4-2 is stored in the connection management table, and a process 75 for transmitting a call request command to the destination GW 4-2 is performed.
A process 76 for outputting a ringing tone to the terminal 2 is performed.

The receiving GW receiving the call request command
4-2 obtains the address of the calling side GW4-1, the telephone numbers of the calling side telephone 6-1 and the called side telephone 6-2 included in the command, the voice coding method, and the like, and obtains the connection management table. The processing 77 for storing the information is performed. Further, a process 78 for making a call to the called telephone 6-2 is performed, and a process 79 for detecting whether or not the called telephone 6-2 picks up the handset is performed. -2, a process 80 for returning an ACK command for notifying that the connection with -2 has been completed. Upon receiving the ACK command, the calling-side GW 4-1 performs a voice data relay process 81. In the relay processing 81, voice packets that also serve as delay time measurement packets are periodically transmitted and received.
When an audio packet for delay time measurement is received, it returns after acquiring header information and audio data. The GW 4 receiving the returned voice packet obtains an approximate value of the delay time from the received time information and the time information included in the packet, and updates the delay time of the corresponding entry in the connection management table. A process 82 is performed to detect that the calling telephone 6-1 puts down the receiver and ends the call (on-hook).
A process 83 for acquiring the address of W4-2, transmitting a disconnection request command, and deleting the corresponding entry in the connection management table is performed. The receiving GW 4-2 that has received the disconnection request command performs a process of disconnecting the receiving telephone 6-2, and performs a process 84 of deleting the corresponding entry in the connection management table.

FIGS. 6 to 12 show an example of the processing flow of the GW4. 6 shows the flow of the entire process, FIG. 7 shows the flow of the setting process by the system administrator, and FIG. 8 shows the public network 2
9 shows the flow of processing when receiving data from the network, FIG. 9 shows the flow of processing when receiving data from the packet network 1, FIG. 10 shows the flow of relay processing between telephones, and FIG. The flow of the relay process when the communication terminal is an IP phone,
FIG. 12 shows the flow of the voice relay processing.

In FIG. 6, when the GW 4 starts a communication program stored in the storage device 22,
In step 90, it is detected whether or not an event has been input from the input device 23 such as a keyboard or a mouse from the input device 23, and if an event has been input (yes in step 90), the event is detected in step 91. Is set. If there is no event input from the administrator (step 90, no), it is detected in step 92 whether or not data has arrived from the public network 2. When data is received from the public network 2 (step 92, yes), PSTN processing for the data is performed in step 93. If there is no incoming data from the public network 2 (step 92, no), it is detected in step 94 whether there is no incoming data from the packet network 1. When data is received from the packet network 1 (step 94, yes), a packet network process for the data is performed in step 95. When the administrator has performed the GW4 operation stop processing such as terminating the communication program (step 9).
6, yes), end all processing.

FIG. 7 shows an example of the flow of the setting process performed in step 91.

In FIG. 7, in step 100, the contents of the event input in step 90 are analyzed. If the event relates to a new entry such as installation of a new GW 4 (yes in step 101), the location management is performed in step 102. Add the new entry to the table. If it is not a new entry addition but a registration entry modification event (step 103, yes), step 1
At 04, the contents of the corresponding entry in the location management table are corrected. If the event is a registration entry deletion event (step 105, yes), the content of the corresponding entry in the location management table is deleted in step 106. The input event is the addition of a new entry (step 101) and the modification of a registered entry (step 10).
3) If neither of the registration entries is deleted (step 105), the process notifies the administrator of the event re-input processing in step 107. When the processing of the input event ends, the processing result is output to the output device 24 such as a screen, and the processing ends.

FIG. 8 shows the PSTN performed in step 93.
An example of the processing flow will be described.

In FIG. 8, at step 110, the data received at step 93 is off-hook or on-hook,
If the signaling is a dial tone-like signaling (step 110, yes), it is determined in step 111 whether the signaling is off-hook. If the signaling is off-hook (step 111, yes), the dial tone is continued in step 112. Collect and analyze, and step 113
To obtain the address information of the communication terminal on the receiving side. At step 114, it is determined whether or not the address information obtained at step 113 is a telephone number. If the address information is a telephone number (yes at step 114), at step 115, the destination management table for the telephone number is obtained from the location management table. The address information of the side GW4-2 is obtained. Step 116
Then, from the connection management table, whether there is a call relayed to the destination GW4-2 obtained in the step, and if so, the maximum value of the delay time of the call and the location management table Acquires the set delay tolerance value. In step 117, the delay time acquired in step 116 is compared with the allowable delay value,
If there is no call being relayed to the GW4-2 or if the current delay time is less than the allowable value (step 117,
yes), in step 118, it is determined from the telephone number of the called telephone 6-2 whether the telephone is an existing telephone connected to the public network 2 or a mobile network connected to the mobile network 3, and if it is an existing telephone, (Step 118, yes), and in step 121, the encoding method is set to PCM (G.711). If it is a mobile phone instead of an existing phone (step 11
8, no), In step 120, the encoding system used in the mobile network 3 is acquired from the location management table and set. In step 122, a call request command to the destination GW4-2 is issued. In step 123, the information of the connection is registered in the connection management table. If the delay time acquired in step 116 is equal to or greater than the allowable value (step 117, no), step 1
At 24, a busy tone is returned to the calling telephone 6-1 to notify that the call request has not been accepted. If the address of the destination communication terminal 6-2 acquired in step 113 is not a telephone number but an IP address (step 125, yes), if there is a plurality of encoding methods of the own GW 4 in step 126, A call request command including the list is issued to the destination communication terminal 6-2, and the coding method included in the ACK command returned from the destination communication terminal 6-2 is set. If it is not an IP address (step 125, no), exception processing such as error notification is performed in step 127. If the received signaling is on hook (step 128, yes), step 1
In step 29, the information regarding the connection is retrieved and acquired from the connection management table, and a disconnection request command is issued to the destination GW4-2. In step 130, the information of the connection concerned is deleted from the connection management table. If the received data is not signaling (step 110, no), voice relay processing is performed in step 131.

In this connection, in the relay between the existing telephone and the mobile telephone, the processing for setting the coding method used in the mobile network 3 in the GW 4 connected to the packet network 1 is shown in FIG. When a call is relayed to the mobile network 3, the mobile GW 8 connected to the mobile network 3 may convert the data to PCM when transmitting data to the packet network (in this case, the voice data flowing on the packet network is PCM, mobile GW 8 Is converted to the encoding method used in the mobile network).

FIG. 9 shows an example of the flow of the packet network processing performed in step 95.

In FIG. 9, in step 140, the address information of the terminal which issued the data is obtained from the data received in step 95. In step 141, it is determined whether or not the data is a command relating to connection establishment / disconnection, and if it is a command (step 1
41, yes), and in a step 142, it is determined whether or not the command is a call request. If it is a call request (yes in step 142), it is determined in step 143 from the location management table whether the issuing terminal of the command is GW4. If it is a registered GW4-1 (yes in step 143), step 14
4, the originating GW 4 from the location management table.
From the connection management table, whether or not there is a relay between the originating GWs 4-1 and, if so, the maximum value of the current delay time in the connection is acquired from the connection management table. In step 145, step 1
Compare the delay allowable value obtained in step 44 with the current delay time,
If the delay time is equal to or less than the allowable value (step 145,
yes), in step 146, the address of the destination communication terminal 6-2 included in the call request command is obtained. In step 147, the destination communication terminal 6 acquired in step 146
It is determined whether the destination communication terminal 6-2 is an existing telephone or a mobile telephone or a communication terminal from the address of "-2". If it is an existing telephone or a mobile telephone (step 147, yes). In step 148, a telephone relay process is performed. If the delay time obtained in step 144 is equal to or greater than the allowable value (step 145, no), a disconnection request command for rejecting the connection is generated in step 149, and the originating GW4-1 obtained in step 140 is generated.
Issue to When the communication terminal that has issued the incoming call request command is a terminal that has not been registered (step 1)
43, no), at step 150, an IP telephone relay process is performed. When the received command is not a call request but an ACK command (step 151, yes), the voice relay processing of step 131 is performed. If it is a disconnection request command instead of an ACK command (step 152, ye
s) In step 153, the call to the destination telephone 6-2 is cut off and the call relay processing is terminated. In step 154, the information of the connection concerned is deleted from the connection management table.

FIG. 10 shows an example of the flow of the telephone relay processing performed in step 148.

In FIG. 10, at step 160, the encoding method included in the call request command is obtained, and at step 161 the receiving telephone 6- at step 146 is obtained.
Make a call to phone number 2. In step 162, off-hook signaling indicating that the user of the called telephone 6-2 has picked up the telephone is detected, and in step 163, the GW 4 of the call request command issuing side acquired in step 140 is acquired.
Issue an ACK command to -1. In step 164, the information of the connection concerned is registered in the connection management table, and the voice relay processing in step 131 is performed.

FIG. 11 shows the IP address used in step 150.
5 shows an example of the flow of a telephone relay process.

Referring to FIG. 11, in step 170, the encoding system included in the call request command is obtained. Get all if there are multiple. In step 171, among the at least one or more encoding schemes obtained in step 170, a scheme that matches the encoding scheme of the own GW is set as the encoding scheme, and the CODER 31 and DECODER 41 are used.
Set to. If there is more than one matching method, set the method stored earlier in the list. In step 172, the destination telephone 6 included in the call request command
-Obtain a phone number and make a call. Step 1
Step 73 detects off-hook signaling indicating that the user of the called telephone 6-2 has picked up the telephone, and
In step 4, an ACK command is issued to the GW 4-1 on the call request command issuing side acquired in step 140. In step 175, the information of the connection concerned is registered in the connection management table, and the voice relay processing in step 131 is performed.

FIG. 12 shows an example of the flow of the voice relay processing performed in step 131. Here, a flow of a voice relay process in the calling side GW 4-1 performing the relay between the telephones 6 will be described.

In FIG. 12, at step 180, the C
It is determined whether or not an interrupt indicating that data has been input from the ODER 31 has been entered. If there has been an interrupt (yes at step 180), at step 181 the CO
The encoded data is obtained from the DER 31. Step 18
In step 2, information on the connection concerned, at least the address of the destination GW4-2 and the address of the calling / destination telephone 6, are acquired from the connection management table. In step 183, an audio packet is generated using the information obtained in step 182, the current time information, and the sequence number of the data as header information. In step 184, the voice packet generated in step 183 is transmitted to the destination GW4-2 obtained in step 182. When a voice packet is received from the packet network 1 (step 18
5, yes), the voice packet is obtained in step 186, and the destination GW4 is obtained from the voice packet in step 187.
2 and the telephone number of the calling telephone 6-1. In step 188, the connection line with the calling telephone 6-1 is searched from the connection management table from the telephone numbers of the receiving GW 4-2 and the calling telephone 6-1 acquired in the step 187. In step 189, the coded voice data is extracted from the voice packet, and in step 190, the coded voice data is output to the DECODER 41 corresponding to the connection line to the calling telephone 6-1 searched in step 188.

FIGS. 13 to 16 show an example of the processing flow of the mobile GW 8. FIG. 13 shows the overall processing flow.
14 shows the flow of processing when a call request from a mobile phone is received, FIG. 15 shows the flow of processing when a call request to a mobile phone is received, and FIG. 16 shows the flow of voice data relay processing. Show.

In FIG. 13, the mobile GW 8 comprises:
When the communication program stored in the storage device 22 is started, it is detected in step 90 whether an administrator has input an event relating to the location management table from the input device 23 such as a keyboard or a mouse. (Step 90, ye
s) In step 91, a setting process for the event is performed. If there is no event input from the administrator (step 90, no), it is detected in step 200 whether or not data has arrived from the mobile network 3. When data is received from the mobile network 3 (step 200, yes),
In step 201, a mobile phone calling process for the data is performed. If there is no data incoming from the mobile network 3 (step 200, no), it is detected in step 202 whether there is no data incoming from the packet network 1. When data is received from the packet network 1 (step 202, ye
s) In step 203, a mobile phone incoming call process for the data is performed. When the administrator performs a process of stopping the operation of the mobile GW 8 such as ending the communication program (step 9).
6, yes), end all processing.

FIG. 14 shows an example of the flow of the mobile phone calling process performed in step 201.

In FIG. 14, when a packet arrives from the mobile network 3 in step 210 (step 210, ye
s) In step 211, the address information of the mobile terminal 10 that has issued the packet is obtained. In step 212, it is determined whether or not the address acquired in step 211 is a mobile phone number. If the address is a mobile phone number (yes in step 212), the packet is used for establishing a connection such as a call request or a disconnection request. If the command is a signaling command indicating information on disconnection (step 21)
3, yes), it is determined in step 214 whether the signaling command is a call request signal, and if the signaling command is a call request signal (yes in step 214), in step 215, the destination communication terminal included in the call request signal Get the address information of In step 216, it is determined whether or not the address information acquired in step 215 is a telephone number.
6, yes), in step 217, the address information of the destination GW4-2 or the mobile GW8-2 that controls the telephone number is acquired from the location management table. At step 218, from the connection management table, whether there is a call relayed to the destination GW4-2 or the mobile GW 8-2 obtained at step 217, and if there is, the maximum delay time of the call And the allowable delay value set in the location management table. In step 219, the delay time acquired in step 218 is compared with the allowable delay value, and the GW4
If there is no call being relayed with the mobile communication gateway 2 or the mobile GW 8-2, or if the current delay time is equal to or less than the allowable value (yes in step 219), in step 220, the telephone number of the called communication terminal is used as the public network It is determined whether the telephone is an existing telephone connected to the mobile network or a mobile network connected to the mobile network 3.
If the telephone is an existing telephone (yes in step 220), the encoding method is determined to be PCM (G.711) in step 221 and set in the above-mentioned coder 31 and DECODER41. If it is a mobile phone (step 220, n
o), in step 222, obtain the coding scheme used in the mobile network 3 from the location management table;
The coders 31 and 41 are set.
In step 223, the call request signal is converted into a call request command, and the called GW4-2 or mobile GW8-
Send to 2. In step 224, the information of the connection is registered in the connection management table. If the delay time acquired in step 218 is equal to or greater than the allowable value (step 219, no), a disconnection request signal is returned to the calling side mobile phone 10-1 in step 225 to notify that the call request was not accepted. I do. Step 215
If the address of the called-side communication terminal 6-2 acquired in step 2 is not a telephone number (step 216, yes), step 22
In step 6, a disconnection request signal is returned to the calling side mobile phone 10-1 to notify that the call request has not been accepted. If the received signaling is not a call request signal but a disconnect request signal (step 227, yes
s), in step 228, convert the disconnection request signal into a disconnection request command, search and acquire information on the connection from the connection management table, and
4-2 or relay to the mobile GW 8-2. At step 229, the end processing of the voice data relay is performed, and at step 230, the information of the connection concerned is deleted from the connection management table. If the received data is not signaling (step 213, no), step 22
At 3, the voice relay processing is performed. If the terminal that issued the packet received in step 210 is the destination GW4-2 or mobile GW8-2 (step 212, no), it is determined in step 231 whether the packet is an ACK command, Is satisfied (step 231, yes), the
Command is converted into an ACK signaling command used in the mobile network 3 and relayed to the calling side mobile phone 10-1;
At step 223, voice data relay processing is performed. If the disconnection request command is not an ACK command (step 233, yes), the disconnection request command is converted into a disconnection request signaling command used in the mobile network 3 in step 234 and transmitted to the calling side mobile phone 10-1. Relaying is performed, and the processes of steps 228 and 229 are performed.

Although this figure does not include the processing when the destination communication terminal is an IP telephone, the same procedure as in the case of GW4 is applied to the case where the destination communication terminal is an IP telephone. By setting the encoding method used in 3 or by converting it to PCM in the mobile GW, the call can be relayed on the path of mobile phone-mobile GW-packet network-IP telephone.

FIG. 15 shows an example of the flow of the portable telephone incoming call process performed in step 203.

In FIG. 15, when a packet arrives from the mobile network 3 in step 240 (step 210, ye
s) In step 241, the address information of the terminal that issued the packet is obtained from the packet received in step 240. In step 242, the packet issuing terminal determines from the location management table that the originating mobile GW8-
1 if it is the mobile GW 8-1 (step 242, yes), it is determined whether the packet is a signaling command used in the mobile network 3 and if it is a signaling command (step 242). 2
43, yes), in step 244, the address of the destination mobile phone 10-2 included in the signaling command is obtained, and in step 245, the signaling command is relayed to the destination mobile phone 10-2. If it is not a signaling command (step 243, no), step 2
At 46, voice data relay processing is performed. If the terminal that issued the received packet is not the mobile GW 8 (step 242, no), it is determined in step 247 whether the terminal that issued the packet is a mobile phone. If the GW 4-1 connects to the public network 2 instead of the mobile phone (yes in step 247), it is determined in step 248 whether the packet is a command related to establishment / disconnection of a connection used in the packet network 1. If the command is determined (step 248, yes), step 2
At 49, the address information of the destination mobile phone 10-2 included in the command is obtained. In step 250, the command is converted into a signaling command used in the mobile network 3 and relayed to the destination mobile phone 10-2. Step 251
In step 251, it is determined whether the command is a call request. If the command is a call request (yes in step 251), information about the connection is registered in the connection management table in step 252. If the command is not a call request but a disconnection request command (step 251, no), the voice relay processing ends in step 253, and
In step 4, the information of the connection is deleted from the connection management table. If the received packet is not a command (step 248, no), voice data relay processing is performed in step 246. If the communication terminal that issued the received packet is a mobile phone (step 24)
7, yes), it is determined in step 255 whether the packet is an ACK signaling command, and if the packet is an ACK signaling command (step 255, yes)
s) In step 246, audio data relay processing is performed. A
If it is not a CK signaling command but a disconnection request signaling command (step 255, no), the processing of steps 253 and 254 is performed.

FIG. 16 shows an example of the flow of the voice data relay processing in step 224. Here, mobile phone 1
The flow of the voice data relay process in the originating mobile GW 8-1 performing the relay between 0 is shown.

In FIG. 16, when voice data is received from the mobile network 3 in step 260, the voice data is obtained in step 261 and the address information of the destination mobile GW 8-2 is read from the connection management table in step 262. In step 263, in addition to the information obtained in step 262, header information such as time information and the sequence number of the audio data is added to the audio data, and relayed to the packet network 1. Generate voice packets. In step 264, the voice packet generated in step 263 is relayed to the destination mobile GW 8-2. If a voice packet has been received from the packet network 1 (step 265, yes), the voice packet is obtained in step 266, and the telephone number of the calling side mobile phone 10-1 is obtained from the voice packet in step 267. In step 268, the voice packet header is deleted from the voice packet and voice data is extracted. In step 269, the voice data is retrieved from the calling mobile phone 10 searched in step 267.
Relay to -1.

FIGS. 17 and 18 show examples of the packet format of each command and voice data.

In FIG. 17, reference numeral 280 indicates that the packet network is I
IP header for the P network, 231 TCP / UDP
The header or 232 stores a command or data.

Reference numeral 283 denotes a field indicating the type of voice coded data or a command related to establishment / disconnection of a connection. Reference numeral 284 denotes an identifier of the command, for example, a call request command or a disconnection request command. In the field indicating the type, 285 is the address of the calling communication terminal (telephone, mobile phone, IP telephone, etc.), 286 is the address of the called communication terminal, and 287 is a field indicating the encoding method used for the call. .

FIG. 18 shows an example of the format of audio data transmitted / received between the GWs 4, or between the GW 4 and the mobile GW 8, or between the GWs 8. 288 is a sequence number of the voice packet, 289 is time information, 290 is a field indicating whether the voice packet also serves as a delay measurement packet, and 291 is voice coded data.

FIG. 19 shows the mobile terminal 10 and the mobile GW.
8 is an example of a format of a packet transmitted / received between the P.8 and the P.8.

283 to 287 are the same as those in FIG. Three
00 is header information necessary for communication in the mobile network 3, 301 is an identifier required to manage information (for example, location information, etc.) of the receiving mobile terminal 10, and 302 is a command or data. Field.
In addition, header information and the like necessary for performing mobile communication on the mobile network 3 may be added.

According to the voice relay system of this embodiment,
Since the PCM is used as the encoding method for the relay between the telephone calls connected to the public network in the call relayed by the GW, the time required for the encoding process in the GW on the calling side and the called side can be reduced.
Further, for relaying with a mobile phone connected to the mobile network, the coding method used in the mobile network is set by the GW on the mobile network side or the GW on the public network side. Only once, it is possible to reduce the time required for the encoding process and prevent the deterioration of the voice quality due to the multiple encoding processes.

[0052]

According to the present invention, since the voice relay server uses PCM having a short processing time as a voice coding method used for relaying between telephones, the time required for the coding process in the GW on the calling side and the calling side is reduced. In addition, since the encoding process is suppressed to one at the time of relaying with the mobile terminal, it is possible to reduce the time required for the encoding system conversion process and prevent the deterioration of the voice quality due to the multiple encoding system conversions. .

[Brief description of the drawings]

FIG. 1 is an overall configuration diagram of a voice data relay device according to an embodiment of the present invention.

FIG. 2 is a configuration diagram of a voice relay server.

FIG. 3 is a diagram illustrating an example of a format of a location management table of the voice relay server.

FIG. 4 is a diagram illustrating an example of a format of a connection management table of the voice relay server.

FIG. 5 is a flowchart of the overall processing of the present system.

FIG. 6 is a flowchart of the overall processing of the voice relay server.

FIG. 7 is a flowchart of a setting process by an administrator of the voice relay server.

FIG. 8 is a flowchart of a process performed when the voice relay server receives data from the public network.

FIG. 9 is a flowchart of a process performed when the voice relay server receives data from a packet network.

FIG. 10 is a flowchart of a relay process between telephones of the voice relay server.

FIG. 11: Either the calling side or the called side of the voice relay server is IP
It is a flowchart of the relay process in the case of a telephone.

FIG. 12 is a flowchart of a voice data relay process of the voice relay server.

FIG. 13 is a flowchart of the overall processing of the voice relay server connected to the mobile network.

FIG. 14 is a flowchart of a process when the voice relay server receives a call request from a mobile phone.

FIG. 15 is a flowchart of a process when the voice relay server receives a call request to a mobile phone.

FIG. 16 is a flowchart of voice data relay processing in the mobile network of the voice relay server.

FIG. 17 is a diagram illustrating an example of a packet format of a command transmitted and received between the voice relay servers.

FIG. 18 is a diagram illustrating an example of a format of a voice packet transmitted and received between voice relay servers.

FIG. 19 is a diagram illustrating an example of a packet format of a signaling command handled by a voice relay server in a mobile network.

[Explanation of symbols]

1: packet network, 2: public telephone network, 3: mobile network, 4-
1.4-2: Voice relay server, 8: Mobile voice relay server, 6.10: Voice data communication terminal.

Continuing on the front page (72) Inventor Tohru Hoshi 1099 Ozenji Temple, Aso-ku, Kawasaki City, Kanagawa Prefecture Inside Hitachi, Ltd.System Development Laboratory (72) Inventor Ichimaro Yumoto 1099 Ozenji Temple, Aso-ku, Kawasaki City, Kanagawa Prefecture Hitachi, Ltd. 5K030 HA08 HB01 HB21 HC02 HD03 JT01 5K051 AA02 BB01 CC01 CC02 CC07 GG03 JJ13 5K067 AA14 AA23 BB02 CC08 DD54 EE02 EE16 HH05 HH11 5K101 RR05 SS08 9A001 BB04 CC10

Claims (6)

[Claims] 1. A voice relay server connected to a packet network, which responds to a call request signal received from a telephone connected to a public network, based on address information of the call partner included in the call request signal. If it is determined that the telephone is another telephone connected to the public network, a means for setting the voice encoding means to the PCM and a voice to the destination voice relay server which is in charge of the destination telephone as the other party are called. Means for transmitting a call request command including an encoding method, wherein the receiving-side voice relay server specifies the call request command in response to receiving the call connection command from the calling-side voice relay server. Means for setting the selected speech coding method. 2. The voice relay server according to claim 1, wherein a relationship among an address of each communication terminal, an address of a voice relay server in charge of said communication terminal, and voice encoding means of each voice relay server is defined. A location management table to be stored; a plurality of voice encoding means; and a mobile telephone connected to a mobile network by a terminal on the other end of the communication based on address information of the other party included in a call request signal from the telephone. If it is found out, the voice coding means used in the mobile network to which the destination terminal belongs is determined by referring to the location management table, and the voice coding method used for the call is determined by the voice coding means. And a means for setting the voice coding method in the call request command. 3. The voice relay server according to claim 1, further comprising means for connecting to a mobile network, and responding to a call connection request command received from a mobile telephone connected to the mobile network. Then, if it is determined from the address information of the call partner included in the call connection request command that the call partner is a terminal connected to the public network or the packet network, the call connection request command is converted to a call request command used in the packet network. Means for converting the ACK command into a call connection ACK command in response to reception of the ACK command from the voice relay server on the receiving side; Means for adding header information to be used in the packet network, and means for deleting header information used in the packet network from the voice data from the voice relay server on the receiving side. Voice relay server according to claim. 4. The voice relay server according to claim 1, wherein the location management table further stores an allowable value of a delay time of a call between the voice relay server and the destination voice relay server. A connection management table that stores at least the address of the voice relay server on the receiving side, the addresses of the terminals on the calling side and the called side, and the approximate value of the delay time. Means for obtaining an approximate value of the delay time; and, in response to receiving the new call request command, referencing the location management table and receiving the destination voice relay corresponding to the destination terminal specified by the call request command. A server address is obtained, and a connection during call relay between the same destination voice relay server is detected from the address information of the destination voice relay server and the connection management table. Means for acquiring the current state of the delay time, and if it is determined that the acquired delay time is equal to or less than the allowable time stored in the location management table, a new entry is made in the connection management table. And a means for transmitting a call request command to the called-side voice relay server. 5. The voice relay server according to claim 4, wherein the means for calculating the approximate value of the delay time of the voice packet is that the voice packet is a delay calculation packet as header information of each voice packet. And time information are set, and when it is determined that the voice packet received by the voice relay server on the receiving side is a packet for delay calculation, output processing of the voice packet to the receiving side terminal is performed. , Returning the voice packet to the voice relay server on the calling side, wherein the voice relay server on the calling side calculates the difference from the time information of the returned voice packet and the current time information. . 6. An IP having a plurality of voice relay servers for calling between a plurality of telephones connected to a public network or a mobile network.
In the communication method of voice data relayed in a packet network, prior to the transmission of voice data, a destination terminal is specified from a calling telephone connected to the public network to a calling voice relay server in charge of the telephone. Transmitting the called call request signal, and the calling side voice relay server responds to the reception of the call request signal, wherein the called terminal is a telephone connected to a public network or connected to a mobile network. Determining whether the mobile phone is a mobile phone, and specifying the voice encoding method, and relaying between the destination voice relay server that controls the destination terminal within a predetermined delay allowable time. Transmitting a call request command that specifies the voice-encoding information with the destination terminal after confirming whether the call has been performed; and Transmitting a voice packet including information indicating that the packet is a delay measurement packet to the destination voice relay server; and a time in the delay measurement voice packet returned from the destination voice relay server. Calculating a delay time between the destination voice relay servers from the information and feeding back the delay time to the connection management table.