JP2005020676A - Telephone communication method and apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a telephone communication method and apparatus which realize a call-waiting service regardless of the kind of a protocol packaged in a telephone terminal. <P>SOLUTION: In the telephone communication method for exchanging a telephone call signal in the form of a voice packet via a packet data network, a plurality of telephone terminals are connected to repeat the telephone call signal. An interrupt signal from a telephone terminal of an interrupt source to a telephone terminal of an interrupt incoming destination which is any one of the plurality of telephone terminals, is received, the interrupt signal is transferred to the telephone terminal of the interrupt incoming destination, and connection set by the first relay step is held in response to a hooking signal from the telephone terminal of the interrupt incoming destination. The telephone terminal of the interrupt source and the telephone terminal of the interrupt incoming destination are connected to repeat a telephone call signal, and a holding tone signal is transmitted in the form of a voice packet to a telephone terminal other than the telephone terminal of the interrupt incoming destination in the plurality of telephone terminals. The telephone communication apparatus carries out the above method. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

[0001]
BACKGROUND OF THE INVENTION
The present invention relates to a telephone communication method and apparatus for transmitting / receiving a telephone call signal via a packet data network such as IP (Internet Protocol).
[0002]
[Prior art]
If a telephone terminal connected to a public telephone network (hereinafter referred to as PSTN: Public Subscriber Telephone Network) is subscribed to call waiting, it will receive calls from other telephone terminals via PSTN even during a call. be able to.
On the other hand, a telephone service via an IP network called a VoIP (Voice over Internet Protocol) telephone service has been put into practical use. In such a service, calls can be made not only between dedicated IP phone terminals, but also between the IP phone terminal and a normal phone terminal connected to the PSTN, with each as a source or destination. You can also. In order to realize VoIP, various protocols, data compression, and address search are required. As such a protocol, ITU-T recommendation H.264 shown in Non-Patent Document 1 is used. There are protocols such as H.323 or SIP (Session Initiation Protocol). Regarding SIP, Non-Patent Document 2 is referred to.
[0003]
ITU-T recommendation Assuming the VoIP protocol according to H.323, it is conceivable to provide a call waiting service even in an IP telephone service. In that case, ITU-T Recommendation H.264 shown in Non-Patent Document 3 is used. Standards relating to additional services such as hold / transfer specified in 450 may be used.
[0004]
[Non-Patent Document 1] ITU-T recommendation 323
[0005]
[Non-Patent Document 2] IETF Standard Specification RFC2543
[0006]
[Non-Patent Document 3] ITU-T Recommendation 450
[0007]
[Problems to be solved by the invention]
However, in order to perform call transfer by the call waiting service using the VoIP protocol, it is necessary that both of the opposing IP telephone terminals in a call have a function based on such a standard. This is the same not only when the telephone terminal that interrupts and receives a call requesting an interruption is an IP telephone terminal, but also when it is a normal telephone terminal via PSTN. Since telephone communications must always ensure that communication with unspecified destination telephone terminals is possible in advance, it is hoped that a call waiting service will be realized regardless of whether or not the standard protocol is followed. It is rare.
[0008]
The present invention has been made in view of such problems, and its object is to provide telephone communication that enables a call waiting service to be realized regardless of the type of protocol implemented in a telephone terminal connected to a packet data network. It is to provide a method and apparatus.
[0009]
[Means for Solving the Problems]
A telephone communication method according to the present invention is a telephone communication method for transmitting and receiving a telephone call signal in the form of a voice packet through a packet data network, and is a first method for connecting a plurality of telephone terminals and relaying a telephone call signal. An interrupt signal transfer step for receiving an interrupt signal for any one of the plurality of telephone terminals from the interrupt source telephone terminal and transferring the interrupt signal to the interrupt destination telephone terminal; In response to the hooking signal from the interrupt destination telephone terminal, the connection set by the first relay step is suspended, and the telephone call signal is connected between the interrupt source telephone terminal and the interrupt destination telephone terminal. A second connection step for relaying a voice signal to a telephone terminal other than the interrupt destination telephone terminal among the plurality of telephone terminals. Characterized in that it comprises a holding tone transmission step of transmitting at Tsu DOO mode, a.
[0010]
A telephone communication apparatus according to the present invention is a telephone communication apparatus that is connected to a packet data network and transmits / receives a telephone call signal in the form of a voice packet via the packet data network, and connects between a plurality of telephone terminals. A first connection means for relaying a telephone call signal; an interrupt signal for an interrupt destination telephone terminal of any of the plurality of telephone terminals is received from an interrupt source telephone terminal; and the interrupt signal is sent to the interrupt destination telephone terminal In response to a hooking signal from the interrupt signal transfer means for transferring and the interrupt destination telephone terminal, the connection set by the first relay means is suspended, and the interrupt source telephone terminal, the interrupt destination telephone terminal, A second connection means for connecting a telephone call signal and relaying a telephone call signal; and telephone terminals other than the interrupt destination telephone terminal among the plurality of telephone terminals In, characterized in that it comprises a holding tone transmission means for transmitting a holding tone signal in the form of a voice packet.
[0011]
DETAILED DESCRIPTION OF THE INVENTION
Embodiments of the present invention will be described in detail with reference to the accompanying drawings.
FIG. 1 shows the entire system configuration including the VoIP relay device 20. Here, the VoIP relay device 20 is connected to the public telephone line network 10 (hereinafter referred to as PSTN 10) and the IP network 30. A plurality of analog telephone terminals are connected to the PSTN 10, and two analog telephone terminals 11a and 11b are shown here as an example. The IP network 30 is a packet data network based on an IP protocol such as the Internet or an intranet. A plurality of IP telephone terminals are connected to the IP network 30. Here, three IP telephone terminals 31a to 31c are shown as an example. The IP telephone terminals 31a to 11c may be terminals dedicated to IP telephones or normal personal computers capable of voice input / output. A SIP proxy server 50 is further connected to the IP network 30. The SIP proxy server 50 has a function of accepting a call connection request in VoIP, searching for a partner terminal based on the request, and performing arbitration such as connection authentication with an appropriate partner terminal.
[0012]
The hardware configuration of the VoIP relay device 20 includes an MPU (Micro Processor Unit) 21 that is a normal central processing unit, a storage unit 22 that is a normal memory that stores a program executed by the MPU 21, and a DSP (Digital Signal). Processor) 23, office line accommodation unit 24, subscriber line accommodation unit 25, and network control unit 26, which are connected to each other by a bus 29. The VoIP relay device 20 can be configured as a computer device having an IP telephone relay function. Therefore, it may further include components such as a display and operation buttons that perform basic functions as a computer device, for example, a display operation function.
[0013]
The DSP 23 of the VoIP relay device 20 is a dedicated digital signal processor that is connected to the path 29 and processes a voice signal that is a call signal. The voice analog signal is PCM encoded or decoded (CODEC function), and voice compression / decompression. Responsible for processing voice packets by function. The DSP 23 further has a function of generating a holding tone for the call waiting service and outputting this as a RTP (Real-Time Transport Protocol) packet.
[0014]
The office line accommodating unit 24 is connected to the PSTN 10, serves as an interface with the telephone exchange equipment included in the PSTN 10, and has a function of executing a call control sequence including call connection and release based on call origination or call reception. As a result, the incoming calls of the analog telephone terminals 11a to 11b via the PSTN 10 are received by the office line accommodating unit 24, and the interrupt incoming calls are also received by the office line accommodating unit 24.
[0015]
The subscriber line accommodation unit 25 is connected to the telephone terminal 41 via a normal telephone call line. As a modification, the subscriber line accommodation unit 25 may have an exchange function that is connected to a plurality of telephone terminals and exchanges the plurality of telephone terminals. In addition, a configuration in which the telephone terminal 41 is integrated with the VoIP relay device 20 to be a single device is possible.
The network control unit 26 is connected to the IP network 30 by an Ethernet (registered trademark) cable, and is a MAC (Media Access Control) controller that performs a communication control function of the Ethernet (registered trademark) physical layer and a communication control of the data link layer. It has a function.
[0016]
In the MPU 21, a control program that performs basic operation control of the VoIP relay device 20 operates, and software that implements an IP telephone function is executed. The configuration of the software includes a state management unit 211 that manages the state of the IP telephone function, a subscriber line management unit 212 that controls communication with the telephone terminal 41, and a station line that controls communication of the accommodation line via the PSTN 10. It comprises a management unit 213, a ground management unit 214 responsible for VoIP communication control based on the SIP (Session Initiation Protocol) protocol, and a voice processing unit 215 mainly executed by the DSP 23 (see FIG. 2). Execution of these software may be appropriately shared between the MPU 21 and the DSP 23.
[0017]
The SIP protocol will be described again with reference to FIG. The SIP protocol is a protocol located in an application layer higher than the IP protocol responsible for communication control in the network layer, and uses the RTP located in the network layer and the UDP (User Datagram Protocol) located in the transport layer, Call control such as call connection, disconnection and maintenance in the telephone, and stream control of RTP / UDP packets.
[0018]
Call control by SIP is described in ITU-T Recommendation H.264. Based on the high-speed connection procedure in H.323, messages are exchanged by UDP packets. First, the originating IP telephone terminal, for example, the IP telephone terminal 31b requests connection to the SIP proxy server 50 using the INVITE message. This message includes identification information such as a telephone number or an IP address of a partner terminal, for example, the IP telephone terminal 31c, and information such as a codec and a port number used during communication. Next, the SIP proxy server 50 searches for the IP telephone terminal 31c of the connection partner. If the requested connection partner is not a user managed by the SIP proxy server 50, the SIP proxy server 50 inquires of another server (for example, a SIP proxy server of another user group not shown) to exchange information and searches for a partner. If the partner is found, the SIP proxy server 50 transmits a connection request to the IP telephone terminal 31c of the connection partner. This is also done by an INVITE message.
[0019]
When the IP telephone terminal 31c in the example of FIG. 1 permits the connection, a 100 Trying message is sent from the IP telephone terminal 31c to the originating IP telephone terminal 31b via the SIP proxy server 50. Following the transmission of the 100 Trying message, a 180 Ringing message indicating that the calling state has been established is further sent to the IP telephone terminal 31b to notify the operator that there is an incoming call.
[0020]
When the operator of the incoming IP phone terminal 31c performs an operation such as picking up the handset and becomes ready for a call, the IP phone terminal 31c transmits an OK message notifying that the call is ready. When an ACK message is returned from the originating IP telephone terminal 31b, the call is started. After the connection, the IP telephone terminal 31b and the IP telephone terminal 31c directly communicate with each other. As in the case of H.323, it is performed by a UDP packet using RTP.
[0021]
Note that the above call connection sequence is replaced with the ITU-T Recommendation H.264 based on the IP protocol instead of the SIP protocol. It may be realized by a protocol based on H.323.
FIG. 3 shows a processing procedure at the time of call waiting service of the VoIP relay device 20 shown in FIG. The sequence of this figure will be described with reference to the components shown in FIG. In the VoIP relay device 20, the subscriber line accommodation unit 25, the station line accommodation unit 24, the state management unit executed by the MPU 21, and the network control unit 27 operate in cooperation.
[0022]
First, an IP telephone terminal, for example, an IP telephone terminal 31a, makes an IP telephone call via the telephone terminal 41 via the subscriber line accommodating part 25 and the network control part. That is, the state management unit (MPU) 21 sets up a VoIP session between the telephone terminal 41 and the IP telephone terminal 31a (step S11). As a result, the subscriber line accommodating unit 25 performs voice analog signal relay processing for a call from the telephone terminal 41 (step S12). The network control unit 27 performs voice packet relay processing for the call from the IP telephone terminal 31a (step S13). Between them, the audio signal is processed by the DSP 23.
[0023]
At this time, if an interrupt incoming call from the analog telephone terminal 11a occurs, the office line accommodating unit 24 receives the PSTN incoming call (step S14). In response to the incoming call notification from the office line accommodation unit 24, the state management unit (MPU) 21 issues an IIT (incoming call display tone) transmission instruction in the same manner as a normal telephone exchange procedure (step S15). The IIT is sent to the subscriber line accommodating unit 25 and reproduced by the telephone terminal 41, so that the operator of the telephone terminal 41 can recognize that the incoming call has arrived.
[0024]
When the operator of the telephone terminal 41 that has recognized the incoming call is accepting the incoming call, the operator performs a hooking operation on the handset. Alternatively, an operation button for accepting an interrupt incoming call is operated. In response to this, the subscriber line accommodation unit 25 receives a hooking signal (step S16).
This hooking signal is notified to the state management unit 21. In response to this, the state management unit 21 interrupts the VoIP session so far and issues a hold sound transmission instruction (step S17). In response to this, the network control unit 27 sends the music on hold RTP packet generated by the DSP 23 to the IP telephone terminal 31a (step S18).
[0025]
The interruption of the VoIP session stops the exchange of voice packets based on the RTP protocol while maintaining the call connection on the VoIP between the telephone terminal 41 and the IP telephone terminal 31a. That is, the voice from the telephone terminal 41 is interrupted. It is transferred to the analog telephone terminal 11a that has been made, while the voice packet from the IP telephone terminal 31a is discarded. Thereby, a telephone call with the analog telephone terminal 11a with the telephone terminal 41 becomes possible. That is, the subscriber line accommodation unit 25 performs voice analog signal relay processing on the telephone terminal 41 (step S19). In response to this, the office line accommodating unit 24 performs voice analog signal relay processing on the analog telephone terminal 11a (step S20).
[0026]
After the telephone call between the telephone terminal 41 and the analog telephone terminal 11a, if the operator of the telephone terminal 41 determines to end the telephone call, the hooking operation is performed again. The subscriber line accommodation unit 25 receives this hooking signal (step S21). The hooking signal is transferred to the state management unit 26, and the state management unit 26 performs a recovery process for the suspended VoIP session (step S22). That is, the exchange of voice packets over VoIP between the telephone terminal 41 and the IP telephone terminal 31a is resumed. The network control unit 27 stops sending the holding tone RTP packet (step S23). Thereby, the telephone call between the operators of the telephone terminal 41 and the IP telephone terminal 31a can be resumed.
[0027]
With the above sequence, a call waiting service in an IP telephone call that enables an incoming call via PSTN 10 is realized.
FIG. 4 shows a processing procedure at the time of call waiting service of the VoIP relay device 20 in the same manner as FIG. 3, but further describes the operation when an interrupt incoming call occurs from the IP telephone terminal. The sequence of this figure will be described with reference to the components shown in FIG. In the VoIP relay device 20, the subscriber line accommodation unit 25, the station line accommodation unit 24, the state management unit executed by the MPU 21, and the network control unit 27 operate in cooperation.
[0028]
First, an IP telephone terminal, for example, an IP telephone terminal 31a, makes an IP telephone call via the telephone terminal 41 via the subscriber line accommodating part 25 and the network control part. That is, the state management unit (MPU) 21 sets up a VoIP session between the telephone terminal 41 and the IP telephone terminal 31a (step S31). Thereby, the subscriber line accommodation unit 25 performs a voice analog signal relay process for a call from the telephone terminal 41 (step S32). The network control unit 27 performs voice packet relay processing for the call from the IP telephone terminal 31a (step S33). The audio signal between them is processed by the DSP 23.
[0029]
At this time, if an interrupt incoming call occurs from the IP telephone terminal 31b via the IP network 30, the network control unit 27 receives the incoming IP network (step S34). In response to the incoming call notification from the network control unit 27, the state management unit (MPU) 21 issues an IIT (incoming call display tone) transmission instruction in the same manner as a normal telephone exchange procedure (step S35). The IIT is sent to the subscriber line accommodating unit 25 and reproduced by the telephone terminal 41, so that the operator of the telephone terminal 41 can recognize that the incoming call has arrived.
[0030]
When the operator of the telephone terminal 41 that has recognized the incoming call is accepting the incoming call, the operator performs a hooking operation on the handset. Alternatively, an operation button for accepting an interrupt incoming call is operated. In response to this, the subscriber line accommodation unit 25 receives a hooking signal (step S36).
This hooking signal is notified to the state management unit 21. In response to this, the state management unit 21 interrupts the VoIP session so far and issues a hold sound transmission instruction (step S37). In response to this, the network control unit 27 sends the hold tone RTP packet generated by the DSP 23 to the IP telephone terminal 31a (step S38). Next, the state management unit 21 sets up a VoIP session with the telephone terminal 41 and the IP terminal 31b (step S39). As a result, the voice packet from the telephone terminal 41 is transferred to the interrupted IP terminal 31b, while the voice packet from the IP telephone terminal 31a is discarded. That is, the subscriber line accommodation unit 25 performs voice analog signal relay processing on the telephone terminal 41, and the network control unit 27 performs voice packet relay processing on the IP terminal 31b (steps S40 and S40 '). As a result, the telephone terminal 41 can make a call with the IP terminal 31b.
[0031]
After the telephone call between the telephone terminal 41 and the IP terminal 31b, when the operator of the telephone terminal 41 determines to end the telephone call, the hooking operation is performed again. The subscriber line accommodation unit 25 receives this hooking signal (step S41). The hooking signal is transferred to the state management unit 21. The state management unit 21 terminates the VoIP session between the telephone terminal 41 and the IP terminal 31b, and performs recovery processing for the suspended session between the telephone terminal 41 and the IP terminal 31a. (Step S42). That is, the exchange of voice packets over VoIP between the telephone terminal 41 and the IP telephone terminal 31a is resumed. The network control unit 27 stops sending the holding tone RTP packet (step S43). Thereby, the telephone call between the operators of the telephone terminal 41 and the IP telephone terminal 31a can be resumed.
[0032]
With the above sequence, a call waiting service in an IP telephone call that enables an incoming call via the IP network 30 is realized.
In the above embodiment, a mode has been described in which one interrupt telephone terminal interrupts the connection of two opposing telephone terminals, that is, a caller and a callee. However, the telephone communication method and apparatus according to the present invention provides a conference. The present invention can also be applied to a form in which two or more telephone terminals are simultaneously connected like a telephone system.
[0033]
【The invention's effect】
As described above, according to the telephone communication method and apparatus according to the present invention, the call waiting service can be realized regardless of the type of protocol implemented in the telephone terminal connected to the packet data network. For example, as a protocol for realizing VoIP, ITU-TH. There are various protocols such as H.323 or SIP, but the spread can be expanded by realizing the call waiting service without depending on the difference between the protocols.
[Brief description of the drawings]
FIG. 1 is a block diagram showing a configuration of an entire system including a VoIP relay device according to an embodiment of the present invention.
FIG. 2 is a block diagram showing a software configuration of the VoIP relay device shown in FIG. 1;
FIG. 3 is a sequence diagram showing a processing procedure when an interrupt is received from a public telephone line network in the VoIP relay device shown in FIG. 1;
4 is a sequence diagram showing a processing procedure when an interrupt is received from an IP network in the VoIP relay device shown in FIG. 1. FIG.
[Explanation of symbols]
10 Public telephone network (PSTN)
11a, 11b, 11c Analog telephone terminal 20 VoIP relay device 21 MPU
22 storage unit 23 DSP
24 office line accommodation unit 25 subscriber line accommodation unit 27 network control unit 30 IP network 31a to 31c IP telephone terminal 41 telephone terminal 50 SIP proxy server

Claims (7)

A telephone communication method for transmitting and receiving telephone call signals in the form of voice packets via a packet data network,
A first connection step of connecting between a plurality of telephone terminals and relaying a telephone call signal; receiving an interruption signal from the interruption source telephone terminal to any one of the plurality of telephone terminals; An interrupt signal transfer step of transferring the interrupt signal to the interrupt destination telephone terminal;
In response to a hooking signal from the interrupt destination telephone terminal, the connection set in the first relay step is suspended, and the telephone call signal is connected between the interrupt source telephone terminal and the interrupt destination telephone terminal. A second connection step for relaying,
A holding sound transmission step of transmitting a holding sound signal in the form of a voice packet to a telephone terminal other than the interrupt destination telephone terminal among the plurality of telephone terminals. The first and second connection steps include SIP protocol or ITU-T recommendation H.264. The telephone communication method according to claim 1, wherein the telephone communication method is performed according to a protocol based on H.323. 3. The telephone communication method according to claim 2, wherein the music on hold transmission step transmits the music on hold packet as a packet based on an RTP protocol. A telephone communication device that is connected to a packet data network and transmits / receives a telephone call signal in the form of voice packets via the packet data network,
A first connection means for connecting a plurality of telephone terminals and relaying a telephone call signal;
An interrupt signal transfer means for receiving an interrupt signal for an interrupt destination telephone terminal of any of the plurality of telephone terminals from an interrupt source telephone terminal, and transferring the interrupt signal to the interrupt destination telephone terminal;
In response to a hooking signal from the interrupt destination telephone terminal, the connection set by the first relay means is suspended, and a telephone call signal is connected between the interrupt source telephone terminal and the interrupt destination telephone terminal. A second connection means for relaying,
A telephone communication apparatus comprising: a hold sound transmitting means for transmitting a hold sound signal in the form of a voice packet to a telephone terminal other than the interrupt destination telephone terminal among the plurality of telephone terminals. 5. The telephone communication according to claim 4, wherein said first and second connection means include means for mutually encoding or decoding between an analog signal of said telephone communication signal and a digital signal of a voice packet. apparatus. 6. The telephone communication apparatus according to claim 4, further comprising office line accommodation means for accommodating office lines of a public telephone line network. The telephone communication device according to claim 4, comprising at least one of the plurality of telephone terminals.

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