JP4808429B2 - Call system - Google Patents

Description

  The present invention relates to a telephone call system using a software telephone that operates on a computer, for example, an Internet protocol (hereinafter referred to as “IP”) implemented as software from a personal computer (hereinafter referred to as “PC”). A) Call center software phone for remote control of telephone functions.

In recent years, IP telephones using PCs have appeared, but operating systems that do not have a real-time control function (hereinafter referred to as “OS”) such as “MS Windows (registered trademark)” or “Linux (registered trademark)”. In a PC in which the IP phone is installed, an IP phone operating on such an OS may affect the sound quality of a call depending on the background processing of the OS or the processing load of other applications. Therefore, there is a problem that it is difficult to operate the IP phone stably.
On the other hand, in the case of an IP phone using a dedicated phone that is deployed independently of the PC, control of calling, disconnection, answering, etc. of the phone by the control of the PC application, or acquisition of caller information at the time of incoming call In order to do this, it is necessary to realize a unique linkage function (CTI). However, in such a CTI, there are many functional limitations, and there is a problem that it is difficult to control freely from the application side of the PC.

  All of the above IP telephones require a call center function. The call center function is a function in which a predetermined supervisor monitors a call between a telephone operator and a customer, intercepts the conversation as necessary, and supports the telephone operator. In the case of realizing such a call center function, when a network for connecting an IP telephone includes a switching HUB, a switching HUB with a monitor port is prepared in order to intercept the voice flowing on the network. It is necessary to connect a server device for interception. Therefore, an expensive system is required. As the number of terminals of the target telephone operators to be monitored increases, there is a problem that a device for voice intercept becomes larger and restrictions on the network occur.

Rather than intercepting voice over LAN equipment, the telephone operator and customer's IP communication path is connected via voice interception equipment, and the supervisor connects the call to equipment dedicated to voice interception. In order to enable voice interception at all times, it is necessary to connect all calls between the telephone operator and the customer via this voice interception device. The scale becomes large and expensive, and the merit of the IP phone that can distribute and transmit voice is lost.
Calls between telephone operators and customers usually do not go through the voice interception device, but only when performing the interception, it is not necessary to make the voice interception device large-scale by sequentially switching the connection to the voice interception device. There is a problem that it takes some time to switch the communication path, and the conversation is hindered because the call is interrupted.
In a call center, a conversation between a company and a customer may be recorded as voice data, and a function to record all calls may be required for the purpose of maintaining the contents of the negotiations. The mechanism for doing this becomes large-scale and network restrictions arise.

  Some telephone devices can control a plurality of calls at the same time, have a function of relaying the calls, and can be controlled from an application. When using such a telephone device, one call is connected to the IP phone via the PBX, the other call is connected to the call destination via the PBX, and each voice path is relayed. Allows outgoing and incoming call control. In this case, the outgoing call to the IP phone is a phone controller connected for remote use of the phone, and the other call is an actual outgoing call, but PBX recognizes that there were two calls. The For this reason, the PBX call log cannot distinguish between remote transmission and normal transmission. In order to grasp the actual outgoing call history, it is necessary to output the outgoing call history from the telephone that has controlled outgoing calls, and the monitoring function and log output function in the PBX may be useless.

  During a call between a customer and a telephone operator at a call center such as telephone banking, as a means of notifying a PIN number from the other party, verbal reading of the number is heard by the customer and the operator knows As a result, various inconveniences arise. For this reason, usually, a mechanism for inputting a personal identification number into the telephone system by pressing a push button of the telephone is required. In the conventional telephone system, in order to realize this function, a system in which a call is temporarily transferred to a separately prepared automatic answering device during a call, and after obtaining a personal identification number there, is transferred to the original telephone operator again. It was realized by building. In order to realize this mechanism, the entire telephone system including the automatic answering device needs to be completely integrated. In addition, prepare an appropriate number of lines according to the usage frequency. Furthermore, the original telephone operator enters another call while the automatic answering machine is answering, and based on the call from the automatic answering machine. Care must be taken so that it cannot be restored. Therefore, it was necessary to construct a large-scale and advanced telephone system. In an environment where session management of IP telephone calls based on terminal-led distributed control is used, it is difficult to replace these functions with IP telephones for centralized control system integration, and it is necessary to construct a more advanced system. However, there is a problem that the construction cost becomes higher by IP.

  As described above, when using an IP phone in a call center, the call quality maintenance function, CTI function, supervisory voice intercept function, etc. required by the call center realized by the conventional PBX system, There are many problems that need to be solved in order to realize them in an IP telephone environment, and not only can the advantages of IP telephones be utilized, but they are often expensive systems.

  The main object of the present invention is to provide an IP telephone mechanism that can solve the above-described problems.

According to the present invention, it is provided with a software telephone that operates on a computer and a hardware telephone that operates in a SIP environment under the control of the computer, and the software telephone controls the SIP of the hardware telephone. To perform first party call control including outgoing call, incoming call, hold, and transfer, and the hardware telephone performs voice IP packet transmission / reception processing independent of the CPU load of the computer. A call system is configured.
In addition, the software phone retrieves voice information of the call from the handset of the hardware telephone as an analog electric signal, and connects the voice data acquired by the microphone input of the computer to a hardware telephone that can be talked to by the caller using the SIP procedure. It is also possible to adopt a configuration that realizes a conversation interception function.
Further, the software telephone implements a call recording function that takes out voice information of a call from a handset of a hardware telephone as an analog electric signal, and stores the voice data acquired by the microphone input of the computer in a data file. In addition, it may be configured to acquire the call start and end timing information, the destination telephone number, the caller information of the caller, and the like by cooperating with the hardware telephone.
In addition, when the call is made to use the hardware telephone so that the connection control of the call to be performed with the hardware telephone can be distinguished from the SIP extension transmission and incoming call control, the software telephone has an extension number. A special number may be added before the call, and a special SIP server may interpret the special number to enable call management and history data identification.
Further, the software telephone connects the conversation between the operator and the customer by the hardware telephone to the microphone input of the computer from the pickup of the handset, and analyzes the voice, thereby the DTMF by the push button of the telephone transmitted by the customer. It is good also as a structure which makes it possible to determine and pass to the application which cooperates on PC.

In one embodiment of the present invention, when an incoming call is received by a software telephone on a computer, an IP telephone is called by adding voice session information = session description protocol (SDP) designated by the call destination, and a response to the call By responding to the caller with the SDP received in step 1, it means that the incoming call is answered directly by the software telephone on the computer, and the RTP voice path is connected to the IP telephone.
When making a call from a software phone, first, the IP phone is called, and the destination is called by specifying the SDP added to the response from the IP phone. When a response is returned from the transmission destination, the SDP of the transmission destination is specified in the response information. Therefore, this is notified to the IP telephone, and the voice session is switched (Re: INVITE), so Is connected to the P telephone. Thus, the telephone call is directly controlled by a software telephone on the computer, and the voice transmission / reception is performed by the IP telephone.
After receiving the incoming call information by the computer itself, the voice call is performed by the IP telephone. Rather than remote control to look at information received at the IP phone, the software phone on the computer directly makes and receives calls, and only the voice path is connected to the IP phone next to the computer. All information related to first party call control, such as dial-in information, caller ID, display information, destination dial number when making a call, is grasped by the software phone on the computer. Operations such as hold, transfer, and disconnection can be performed in close cooperation with a computer application.
The IP phone beside the computer performs the conversion of the voice to be transmitted to the other party to the RTP packet and the process of reproducing the RTP packet received from the other party into the voice, so that when the CPU load on the computer increases, If a failure occurs, the call quality is not affected.

  A software phone on a computer usually has sufficient processing capability for processing voice packets, and this can be used to provide voice information to the supervisor. A special eavesdropping function is required to intercept the conversation between the computer-side IP phone and the other party with IP, but a device (telephone pickup) for branching with an analog electrical signal is inserted into the handset of the computer-side IP phone. By connecting this electric signal to the microphone input of the computer, the conversation can be acquired by the computer, and the conversation can be recorded to provide a function for maintaining the contents of the call. In addition, during a call, the supervisor is using a SIP (Session Initiation Protocol) terminal as a separate session, and the content of this conversation is converted into an RTP packet so that the supervisor can hear the supervisor. Intercept customer conversations and provide operator assistance and supervision for education. Since the audio signal in this case is affected by the CPU load of the computer, there is a possibility that the sound may be temporarily interrupted, but it is more necessary to maintain the quality than the conversation between the operator and the customer. It is unlikely to be a problem. The connection between the operator and the supervisor can be performed in a state where the operator does not perceive the supervisor. During the call interception led by the supervisor, the supervisor can tell this to the operator by a text message.

  When the operator and the customer are talking and the supervisor is intercepting this call, if the "switch connection destination" is performed by operating the software phone on the computer, the call between the operator and the customer is put on hold, and the customer holds a hold sound. The operator and supervisor can make a call after listening to the message. When “connection destination switching” is performed again, the call is restored. In addition, when the “transfer execution” operation is performed, the call with the customer is transferred to the supervisor, and the operator is released from the call. The operations of “switching connection destination” and “execution of transfer” can be performed by remote operation from the supervisor side using the SIP message transfer function between the software phone of the operator and the supervisor.

The conversation between the operator and the customer is recorded by connecting to the microphone input of the computer from the pickup of the handset, and linked with the information related to the call (start time, end time, destination telephone number, caller ID, etc.) And transferred to a server that stores voice information. At this time, when the call is finished, the data is once saved in a local file as raw data, converted into a compressed WAV file, and then transferred to the server. Compared with the method of recording from voice recording to compression and storage for all calls recording, recording and compression processing is performed with each computer distributed, so the load on the server increases even when the number of terminals increases It is possible to cope with a large scale at a low cost.
As for call recording, there is a device that records on a computer, but since this method is a software phone in comparison with the computer's telephone control function realized by the CTI function, Can also be closely linked.

  Call connection control between a software telephone on a computer and an IP telephone normally placed next to the computer is indistinguishable from normal SIP extension call and incoming call control. When you make a call to use an IP phone, you do not specify the other party's extension number directly, but add a special number to distinguish it from normal extension calls before the extension number. The call log is also output in a distinguishable manner, and is also distinguished by the real-time status so that it can be distinguished by real-time aggregation display and history aggregation.

  With a software telephone on a computer, the conversation between the operator and the customer is recorded by being connected to the microphone input of the computer from the pickup of the receiver, so that it can have a function of analyzing voice. The audio signal received by the computer is only recorded to a normal audio file during a call, but when receiving an instruction from the application by the operator's operation, the audio signal is analyzed in real time, The DTMF by the push button of the telephone transmitted by the customer can be determined and returned to the instructing application. After completion of acquisition of DTMF, the CPU load of the computer is reduced by returning to the process of recording only a normal conversation. By this method, the acquisition of DTMF during a call with a customer can be distributed to the processing at each terminal, and the application on the terminal side can acquire information directly only in cooperation with a directly connected software phone. It becomes like this. Compared to the conventional method that required the entire system to be integrated, it can be realized by directly linking the applications in the distributed terminals and the software telephone, so that it can be easily realized and the necessary resources are not excessive. There is no need to consider shortages, and an efficient system can be constructed.

As described above, in the present invention, since the software phone itself as a terminal also has a switching function, the connection control of the voice when the terminal itself makes and receives a call is proxied to another terminal to perform such control. Make it.
As described above, in the present invention, the exchange function is distributed to the terminals, and each terminal independently changes its connection destination to another terminal.
In the conventionally known exchange server technology for connecting audio between computers peer-to-peer, the distribution of information “what terminal is connected to which terminal” is performed centrally in the center. Yes. Therefore, in such a conventional technique, basically, an IP telephone concept of “concentration of exchange control and control of voice connection distributed to terminals and performed between terminals” is realized.
Therefore, the present invention is different from such a conventional technique in that “the terminal itself has an exchange function”.

Hereinafter, an embodiment of the present invention will be described with reference to the drawings. In addition, in each figure, the same code | symbol was attached | subjected to the same part.
FIG. 1A is a functional explanatory diagram showing the entire configuration of an IP telephone system using a software telephone according to the present invention. In FIG. 1A, a public switched telephone network gateway (PSTNGW) [or Internet telephone service provider gateway (ITSPGW)] 200, a software telephone 300, and an IP telephone 400 are connected to the SIP server 100, respectively. The software phone 300 is connected to the application 500.
The software telephone 300 operates as a terminal of the SIP server 100 with respect to the SIP server 100, but proxies a call used by the SIP server 100 to an external SIP terminal, and the IP telephone 400 actually receives the call. To. Accordingly, the SIP server 100 recognizes the software telephone 300 as a terminal, but an actual call is made through the IP telephone 400 proxied from the software telephone 300.
Therefore, in this embodiment, the proxy function of PBX is distributed (in this embodiment, it is a desktop of a PC (personal computer), but if it is distributed, it does not need to be a PC or desktop, and is dedicated. Although the software telephone 300 is a terminal, the software telephone 300 is configured to control the external SIP terminal by transferring a call to another SIP terminal behind the terminal. Therefore, compared to a configuration in which proxy functions are concentrated on the PBX, the proxy functions are distributed in this way, so even if the load increases, the load is not concentrated but concentrated. Therefore, it is possible to suppress the influence caused by the increased load, and it is easy to maintain the call quality.

  The software telephone 300 includes a session unit 310, a main unit 320, an interface unit 330, and a GUI unit 340. The session unit 310 includes SIP-UA (SIP) as shown in FIG. (User agent) 311, 312, 313 are included. Two of them are a session for outgoing / incoming calls and a session for transfer like a conventional SIP terminal, and the other one is a session for remote connection for proxying a call to another SIP terminal. The operations of these sessions are controlled by the main unit 320, and the main unit 320 is configured to input and output data and commands through the interface unit 330 by using the application 500 and CTI (Computer Telephony Integration). Further, the interface unit 330 can receive a command from the user through the GUI 340.

A schematic diagram of the IP telephone system according to the present embodiment is shown in FIG. The software telephone 300 is connected to the SIP server 100 and the register server 110, and the SIP server 100 and the register server 110 are connected to the call control server 120. The media server 130 (IVR, ACD) is connected to the SIP server 100. A database DB1 of the setting table and status table is provided to be accessible from the call control server 120 and the media server 130, and a database DB2 for presence reference routing is accessible from the call control server 120, the media server 130, and the presence server 520, respectively. Is provided.
The software phone 300 realizes first party call control from an application by remote control of the SIP phone. Further, the software telephone 300 is connected to the application 500, and is called, disconnected, answered, held, reconnected, consult transfer, consulted transfer, connection destination switching, transfer stop, transfer execution, remote connection by CTI command / response event. Start / cancel, call extension status notification by event, incoming call, call start, idle, hold, held, and other DTMF acquisition.
The software phone 300 is connected to the remote phone management server 510, and login (phone number, ID, password), logout, ACD agent status (reception possible, post-processing) with the presence server 520 by the presence client (browser). , Leaving), skill registration (skill ID, skill level, desired level), reason code (going out, during a meeting, etc.), contact means (phone availability, email availability, short message availability, etc.).

FIG. 3 is an explanatory diagram of the connection configuration of the software telephone 300. The software telephone 300 has a dedicated extension number, but does not make a single call, and is used as a set with an IP telephone or other SIP terminal.
In this way, if an IP phone starting from 050 is used as the remote phone connection destination, even if the remote phone user is connected to the SIP server of the center from outside by Internet VPN etc., the voice signal (RTP) Since it is connected to the GW through the VoIP network, the voice call quality is not affected by fluctuations in the Internet load.

  FIG. 4 is a diagram for explaining an outline of the operation at the time of incoming call in this IP telephone system. In this example, the extension 5001 is assigned to the software telephone 300, and the call is proxied from the software telephone 300 to the SIP terminal to which the extension 201 is assigned. Further, in order to clarify the flow of operation, the order of occurrence of events such as incoming calls and incoming call notifications is indicated by (1) and (2) in the figure.

When an incoming call is received from the public switched telephone network (PSTN) to the public switched telephone network gateway (PSTNGW), (1) Incoming is notified to the PSTNGW.
In response, the incoming call notification (2) INVITE is made from the PSTNGW to the SIP server 100. In this case, the reception port of the gateway is specified by SDP (Session Description Protocol). The SIP server 100 performs an incoming call notification (3) INVITE from GW to the software telephone 300 to which the extension 5001 is assigned.
The software phone 300 notifies the application 500 of the incoming event (4) Delivered. Further, software telephone 300 transmits provisional response (5) Trying to SIP server 100. The SIP server 100 relays the provisional response (6) Trying from the software telephone 300 to the public switched telephone network gateway. In the software telephone 300, the extension 201 is called by designating the SDP notified by the incoming call from the public switched telephone network gateway (7) INVITE 201 with SDPGW.

The SIP server 100 calls the IP telephone 400 to which the extension 201 is assigned (8) INVITE with SDPGW. The IP telephone 400 to which the extension 201 is assigned returns a provisional response and a ringing notification (9) Trying / ringing to the SIP server 100. The SIP server 100 returns (10) Trying / ringing a provisional response from the IP telephone 400 to which the extension 201 is assigned and a ringing notification to the software telephone 300. The software telephone 300 transmits (11) ringing for the incoming call from the public switched telephone network gateway 200 to the SIP server 100.
The SIP server 100 relays (12) the ringing notification to the public switched telephone network gateway 200. At this time, the public switched telephone network gateway notifies the public switched telephone network that a call is being made.
After that, when the handset is lifted by the IP telephone 400 to which the extension 201 is assigned (13) and is in the OffHook state, the IP telephone 400 to which the extension 201 is assigned (at this time, the receiving port of the extension 201 is designated by SDP). Response notification (14) OKwithSDP201 is returned to the SIP server 100. At the same time, the voice is transmitted by the RTP (Real-time Transport Protocol) to the designated SDP (receiving port of the public switched telephone network gateway 200) from the software telephone 300. The software telephone 300 receives the response from the IP telephone 400 to which the extension 201 is assigned (15) OK with SDP 201, and the IP telephone 400 to which the extension 201 is assigned as a response to the incoming call from the public switched telephone network gateway 200 The designated reception port is designated and a response notification (16) OK with SDP 201 is made to the public switched telephone network gateway 200. Furthermore, the software phone 300 performs event transmission (17) Established to the application 500 that the call has been established.
The SIP server 100 relays the response notification (18) OKwithSDP201 from the software telephone 300 to the public switched telephone network gateway 200, and the public switched telephone network gateway 200 transmits the voice by RTP to the reception port specified by the response notification. To do. As a result, an RTP session is bidirectionally established (19) RTP Established between the public switched telephone network gateway 200 and the IP telephone 400 to which the extension 201 is assigned.

FIG. 5 shows a more detailed explanatory diagram when the software telephone 300 is received. In this figure, for simplification, IP telephone 400 in FIG. 4 is simply expressed as “extension”. Similarly, the public switched telephone network gateway 200 is abbreviated as GW, the public switched telephone network is abbreviated as PSTN, and the software telephone 300 is abbreviated as R.
As shown in FIG. 5A, when an incoming call is received from the public switched telephone network gateway 200 to the software telephone 300, the extension is called by the SDP at that time. Thereafter, as shown in FIG. 5B, when the extension responds, the SDP of the extension is designated and the response is returned from the software telephone 300 to the public switched telephone network gateway 200. Thereby, a telephone call is established. Unless the extension side handset is placed, the extension and the software telephone 300 remain connected.
In the case of disconnecting an established call and disconnecting from the extension side or disconnecting by operating the software telephone 300 and the extension is not on-hook, as shown in FIG. The SDP is switched to the software telephone 300 so that the extension and the software telephone 300 are connected.
When the extension goes on-hook, as shown in FIG. 5D, the extension is ringed down again when the next call is made or received. As described above, connection establishment / disconnection from the public switched telephone network gateway 200 to the IP telephone 400 is performed.

Next, with reference to FIG. 6, an outline of operation at the time of outgoing call will be described. In the calling operation in this case, the calling instruction is issued from the application 500 to the software telephone 300, and in response to this, the IP telephone 400 to which the extension 201 is assigned and the telephone connected to the public switched telephone network. Communication is established with
In FIG. 6, first, the application 500 transmits a call instruction (1) MakeCall to the software telephone 300. In response to this, the software telephone 300 notifies the SIP server 100 of a call to the IP telephone 400 to which the extension 201 is assigned (2) Invite. The SIP server 100 calls (3) Invite the IP telephone 400 to which the extension 201 is assigned. The IP telephone 400 to which the extension 201 is assigned returns a provisional response and ringing notification (4) Trying / Ringing to the SIP server 100. The SIP server 100 relays the received provisional response and ringing notification to the software telephone 300 (5) Trying / Ringing.

  After that, when the handset is lifted by the IP telephone 400 to which the extension 201 is assigned and (6) is in the OffHook state, the IP telephone 400 to which the extension 201 is assigned (at this time, the receiving port of the extension 201 is designated by SDP) Response notification (7) OKwithSDP201 is returned to the SIP server 100. Software telephone 300 receives a response from IP telephone 400 to which extension 201 is assigned (8) OKwithSDP201. Furthermore, the software telephone 300 adds the SDP designated from the IP telephone 400 to which the extension 201 is assigned, and transmits a call instruction (9) INVITEGWwithSDP 201 to the SIP server 100. The software telephone 300 notifies the application 500 of a call start (10) originated.

The SIP server 100 issues a call instruction (11) Invite to the public switched telephone network gateway 200. In response to this, the public switched telephone network gateway 200 returns a provisional response (12) Trying, while the public switched telephone network gateway 200 sends (13) outbound to the public switched telephone network (outside line).
The SIP server 100 returns (14) Trying the provisional response from the public switched telephone network gateway 200 to the remote phone. Upon receiving “calling” from the outside line, the public switched telephone network gateway 200 designates the receiving port of the public switched telephone network gateway 200 by SDP and transmits (15) Ringing to the SIP server 100.

The software telephone 300 receives the call with SDP (16) Ringing with SDPGW from the public switched telephone network gateway 200, and in response to this, notifies the application 500 of a call progress display (17) Call Progress (ringing). In addition, the software telephone 300 performs (18) Re: InvitewithSDPGW on the extension 201 using the SDP received from the public switched telephone network gateway 200.
The IP telephone 400 to which the extension 201 is assigned receives an RTP replacement instruction by Re: InvitewithSDPGW from the software telephone 300 and accepts (20) OK for this. As a result, the IP telephone 400 to which the extension 201 is assigned replaces the STP of the public switched telephone network gateway 200 designated by the software telephone 300 with the RTP, thereby the public switched telephone network gateway 200 and the IP to which the extension 201 is assigned. An RTP session is established (21) RTP Established with the telephone 400 in both directions. Thereafter, the approval (22) OK from the IP telephone 400 to which the extension 201 is assigned is relayed to the software telephone 300.

Thereafter, when the destination of the outside line responds, a response notification (23) OK is notified from the public switched telephone network gateway 200 to the SIP server 100.
The SIP server 100 relays (24) OK the response notification from the public switched telephone network gateway 200 to the software telephone 300. Then, the software telephone 300 notifies the application 500 of connection completion (25) Established.

FIG. 7 shows a more detailed explanatory diagram when the software telephone 300 is called. As in FIG. 5, in this figure, the IP telephone 400 in FIG. 6 is simply expressed as “extension” for the sake of simplicity. Similarly, the public switched telephone network gateway 200 is abbreviated as GW, the public switched telephone network is abbreviated as PSTN, and the software telephone 300 is abbreviated as R.
The example of FIG. 7 shows a case where the IP telephone 400 is connected to a telephone such as a customer via the public switched telephone network in response to a call request (call request from the application 500 in this example). As shown in FIG. 7A, when receiving a call request, the software telephone 300 calls an extension. Thereafter, when the extension responds, the public telephone network gateway 200 is called from the software telephone 300 by designating the extension SDP as shown in FIG. 7B. Further, the call originator, in this case, the application 500 is notified of the start of the call.
As shown in FIG. 7C, when the session progress with SDP is returned from the public switched telephone network gateway 200 to the software telephone 300, the software telephone 300 Re: Invitewith the reception port of the extension to this SDP, Make it possible to communicate in both directions. Further, software phone 300 informs application 500 that the call is in progress.
As shown in FIG. 7D, when the destination responds, the public switched telephone network gateway 200 notifies the software telephone 300 of OK, and the software telephone 300 confirms that the other party has responded to the application 500. To be notified.

  Next, additional functions of the software telephone 300 according to the present embodiment will be described with reference to FIG. As an additional function of the software telephone 300 according to the present embodiment, various functions required in a general call center are provided in addition to a normal call originating / holding / transfer incoming call response function. In the following example, the SIP telephone 400 is operated by an operator, and a call is made with a customer through a telephone whose connection is established through an external line via the public switched telephone network, and is different from the SIP telephone 400 and transferred by the software telephone 300. An example is shown in which a telephone call or text message transmission or the like is performed with a supervisor through a SIP telephone that can perform the above. With this configuration, when the operator is talking to the customer and the operator is unable to deal with the situation, the supervisor can ask for instructions or the supervisor can directly replace the customer. It is possible to make a call.

As shown in FIG. 8, the software telephone 300 includes a sound card 350, and includes a recording unit 321, a sound generation unit 322, a management information acquisition / message communication unit 323, and a presence registration / reference unit 324. It is provided as software or hardware.
The SIP-UAs 311 to 313 receive the hold sound, RBT, ringer sound, and the like generated by the sound generation unit 322, and can send the hold sound and the like to the call destination, the supervisor, and the SIP telephone 400, respectively.
On the other hand, the handset terminal of the SIP telephone 400 is configured to pick up a voice signal such as a speaker's voice and input it to the sound card 350 of the software telephone 300.

  The sound card 350 sends the received audio signal to the recording unit 321, and the recording unit 321 performs recording by recording the call. This call recording is triggered by the start and end of a call. In this embodiment, the call content is converted into a wav file format (the conversion format is not particularly limited and may be set as appropriate) with the start and end of the call as a trigger, and the management server together with call information such as call destination information and time stamp Send FTP and save. The recording unit 321 also detects DTMF (Dial Tone Multi Frequency). Thus, when a password is entered by pressing a push button of the telephone, the entered password can be detected by picking up DTMF from the handset of the SIP telephone 400.

In this embodiment, during a call between a customer (destination) connected through the public switched telephone network and an operator who makes a call using the SIP telephone 400, a password is input to the customer by a request from the application 500. Request and receive the entered PIN.
In this case, during a call, the call destination is switched from SIP of the software telephone 300 to RTP, and a voice file in which a voice prompting the user to input the password is played back and the customer is requested to input the password. The customer receives a DTMF input by pressing a push button on the telephone, detects a personal identification number, and transmits the detected number to the management server 510 in the application 500. Thereafter, the call is returned to the original SIP. In this embodiment, the DTMF is detected by the recording unit. However, the DTMF may be detected by an independent function unit or by another function unit.

In addition, the sound card 350 can transmit the audio signal from the SIP-UA 312 to the supervisor by sending the audio signal from the handset terminal of the SIP telephone 400 to the SIP-UA 312.
The management information acquisition and message communication unit 323 communicates with the management server 510 in the application 500 by SIP to perform management information acquisition and message communication. Also, the presence registration / reference unit 324 communicates with the presence server 520 in the application 500 via SIP to perform presence registration and reference. Thereby, a status registration function to the presence server, for example, agent login / logout, receivability, post-processing, leaving, reason display, skill setting, etc. can be performed.

Next, details of the transfer process in the example of FIG. 8 are shown in FIGS.
FIG. 9A shows that the software telephone 300 makes a call with the telephone (for the customer's call) through the public switched telephone network gateway 200 using the RTP of the extension 1 (for the operator call) by the SIP-UA 311 for making and receiving calls. Indicates the established state. When transfer is performed from this state, as shown in FIG. 9B, the public switched telephone network gateway 200 is placed on hold and a hold sound is transmitted to the extension 1. As shown in the figure, SDP-R is sent from software telephone 300 to extension 1, while hold-invite is sent from software telephone 300 to public switched telephone network gateway 200, and software telephone 300 and public switched telephone network gateway are sent. 200 is disconnected.
As shown in FIG. 9 (c), the software telephone 300 designates the SDP of the extension 1 by the transfer SIP-UA 312 and sends it to the extension 2. When Ringing returns, the RBT is sent to the extension 1. The public switched telephone network gateway 200 remains in the hold state shown in FIG.

As shown in FIG. 10A, when the extension 2 responds, the extension 2 is designated as SDP and OK is returned to the software telephone 300, the software telephone 300 transmits the extension 1 with the SDP from the extension 2. Re: Invite so that extension 1 and extension 2 can communicate bidirectionally. The public switched telephone network gateway 200 remains on hold.
As shown in FIG. 10B, the software telephone 300 sends the SDP of the public switched telephone network gateway 200 to the extension 1 and transmits Re: Invite to the public switched telephone network gateway 200. The public switched telephone network gateway 200 returns OK to the software telephone 300, and the software telephone 300 transmits a hold Invite to the extension 2, whereby the connection destination switching operation is performed. As a result, the extension line 2 is put on hold.
As shown in FIG. 10C, the SIP session between the software telephone 300 and the public switched telephone network gateway 200 and the software telephone are referred to by referring the public switched telephone network gateway 200 to the extension 2 by the transfer execution operation. The SIP session between 300 and extension 2 is released. On the other hand, “refer” is transmitted from the software telephone 300 to the public switched telephone network gateway 200, thereby enabling a two-way call between the public switched telephone network gateway 200 and the extension 2.
In this embodiment, character messages can be transmitted and received between the supervisor and the operator.

On the other hand, a call may be made from the supervisor to the operator who is talking, and the conversation between the operator and the customer (voice received by speaker input) may be monitored. In this case, the above-described monitoring is performed by starting and ending the call contents from the SIP-UA 311 and SIP-UA 313 communicating with the callee (customer) through the SIP-UA 312 communicating with the supervisor. Can be done.
A help function from the operator to the supervisor is also provided. The help function allows the operator to send a message, selects the message corresponding to a specific push button input from multiple fixed character messages, or sends a message to the supervisor. . The supervisor who has received the message can silently monitor the conversation between the operator and the customer (voice received by speaker input) by selecting and calling the SIP telephone 400 of the caller operator. In this example, a message is transmitted during monitoring, and “Monitoring” is displayed on the operator's screen to notify that the supervisor has entered support. When “connection destination switching” is performed in this state, the conversation with the customer is put on hold, and a call can be made with the supervisor. If “transfer execution” is performed as it is, the call with the customer can be transferred to the supervisor.

If the operator performs a “switch connection destination” operation while the supervisor is monitoring, the call between the operator and the customer is put on hold and switched to the call with the supervisor. By performing the “connection destination switching” operation again, the original state is restored. If the “transfer execution” operation is performed as it is, the customer's call is transferred to the supervisor. This transfer operation can also be executed as a pickup operation from the supervisor. In this case, the operator is notified in advance of the transfer to the supervisor by a text message.
Communication between the supervisor and the operator is established as follows. First, when an operator receives an incoming call from another terminal during a call, a SIP message indicating a new incoming event arrives at the software telephone 300. At this time, if it is a new incoming call from a general destination, it is already in a call, so a message indicating “BUSY” is returned to this event, the other party is busy, and the call is connected There is nothing.

In this embodiment, the incoming call information for a new incoming call can include information indicating that the other party is a supervisor and the call monitoring is requested, and the same SIP-UA as that for transfer is used. It is possible to respond to this incoming call in a new SIP session. At the time of transfer, one SIP session is held in a hold state and is transmitted by the transfer SIP-UA. In this case, one SIP session is left in a call, and the incoming call operation is performed by the transfer SIP-UA. Will do. For the call between the operator and the customer, an audio telephone pickup is attached to the telephone handset used for the call, so that the voice of the operator's transmission / reception is connected to the microphone input of the PC, and the PC of the remote phone is operating. It can be received by PC software as microphone input sound. The software telephone 300 transmits the voice information to the voice session connected by the transfer SIP-UA, so that the supervisor can hear the voice of the call monitor.
Therefore, if you want to monitor calls only with IP, a large-scale mechanism is required to receive voice information on the LAN from the outside, but by connecting the phone and PC with analog signals, The monitor sound can be taken into the PC and sent to the supervisor.

Next, input to the software telephone 300 will be described.
The software telephone 300 voluntarily registers with the SIP server, but otherwise performs an operation according to the state at that time by an external input. The input source is a SIP message (according to rfc3261), GUI, or CTI (considered to be the same as an event generated in the GUI).
In addition, the input from GUI and CTI in this embodiment is as follows.
-Start remote connection (use extension to set in config)
-Remote connection cancellation / outgoing (specify the destination telephone number. In this example, the outside telephone number is not distinguished for the destination telephone number)
・ Disconnect (call suspension)
・ Response (valid only when an incoming call is received during remote connection)
・ Hold, reconnect ・ Consult transfer (specify the transfer destination phone number. In this example, the transfer destination phone number is not distinguished between the outside line and the extension)
・ Transfer canceled (valid only during consult transfer)
・ Transfer execution (valid only during a consult transfer call)
-Connection destination switching (valid only during a consult transfer call)
・ Send text message (specify phone number of destination software phone 300)
-Call monitor call (specify the software phone number of the operator's destination from the supervisor)
・ Call monitor cancellation ・ DTMF signal acquisition (valid only when connected)

The contents of the state transition at the time of basic outgoing / incoming calls in software telephone 300 in response to the input from the input source are shown in the state transition diagram of FIG.
In FIG. 11, when the extension is on-hook in a state below the line y, the state returns to the empty state.
In addition, the contents of state transitions during hold, transfer, and call monitoring are shown in the state transition diagram of the figure. When the extension is on-hooked in the state on the left side of the line L in FIG. 12, the extension is set to the calling state and returned to the original state when responding.

1 is a function explanatory diagram showing the overall configuration of an IP telephone system according to the present invention. FIG. 1 is a schematic diagram of an IP telephone system according to an embodiment. Explanatory drawing of the connection structure of the software telephone 300. FIG. Operation | movement outline explanatory drawing at the time of the incoming call in an IP telephone system. FIG. 3 is a more detailed explanatory diagram when a software telephone 300 is received. Explanatory drawing of the operation | movement outline | summary at the time of transmission. FIG. 3 is a more detailed explanatory diagram at the time of outgoing call of software telephone 300. Explanatory drawing of the additional function of the software telephone 300 which concerns on this embodiment. Explanatory drawing of the detail of a transfer process. Explanatory drawing of the detail of a transfer process. The state transition figure at the time of the basic outgoing / incoming call in the software telephone 300. State transition at the time of hold, transfer and call monitoring in the software telephone 300.

Explanation of symbols

DESCRIPTION OF SYMBOLS 100 SIP server 110 Register server 120 Call control server 130 Media server 200 Public switched telephone network gateway 300 Software telephone 310 Session part 320 Main part 321 Recording part 322 Sound generation part 322 Management information acquisition and message communication part 324 Presence registration and reference part 330 Interface unit 340 GUI unit 350 Sound card 400 SIP telephone 500 Application 510 Remote phone management server 520 Presence server

Claims (4)

A software telephone that operates on a computer, and a hardware telephone that operates in a SIP environment under the control of the computer,
The software telephone performs first party call control including outgoing call, incoming call, hold, and transfer by controlling SIP of the hardware telephone,
The hardware telephone performs voice IP packet transmission / reception processing independently of the CPU load of the computer.
Call system. The software telephone receives voice data of a call from a handset of a hardware telephone as an analog electric signal through a microphone input of the computer ,
The software phone realizes the voice data acquired intercept function the conversation to connect callers to hardware phone capable calls SIP procedure,
The call system according to claim 1. The software telephone receives voice data of a call from a handset of a hardware telephone as an analog electric signal through a microphone input of the computer ,
The computer realizes a call recording function for storing the input voice data in a data file,
Acquires the timing information of the start and end of the call, the telephone number of the call destination and the caller information of the caller by cooperating with the hardware telephone,
The call system according to claim 1. The software telephone connects the conversation between the operator and the customer via the hardware telephone to the microphone input of the computer from the pickup of the receiver, and analyzes the voice to determine the DTMF by the push button of the telephone transmitted by the customer. And pass it to the application that works on the PC .
The call system according to claim 1.

Images