Communication system, relay device, exchange device, communication method, relay method, exchange method, program for relay device, and program for exchange device

Abstract

To provide a variety of voice services in a form of connecting a WebRTC system to a PBX system.SOLUTION: The present invention is a communication system 100 including a gateway 3 and a PBX 4. The gateway 3 includes: a WebRTC UA unit 31 that processes interaction with a WebRTC terminal 1 that is currently in a call; and a SIP UA unit 32 that, upon receiving a transfer request to a SIP terminal 5b from the WebRTC terminal 1, transmits to the PBX 4 first information, which is information relating to the transfer request and has a SIP INFO message. A transfer unit 42 of the PBX 4 executes first call-making processing using SIP for the SIP terminal 5b, and executes hold processing using SIP for a SIP terminal 5a that is currently in a call. The transfer unit 42 also executes response processing relating to the first information for the gateway 3 using second information having the SIP INFO message. The transfer unit 42 also executes second call-making processing using SIP for the gateway 3.SELECTED DRAWING: Figure 1

Description

[Technical Field]

[0001] The present invention relates to a communication system, a relay device, an exchange device, a communication method, a relay method, an exchange method, a program for a relay device, and a program for an exchange device. [Background technology]

[0002] Currently, the majority of audio conferences that are widely used over the Internet are systems that use WebRTC (Web Real-Time Communication). An example of an invention that uses a WebRTC system is described in Patent Document 1. The invention in Patent Document 1 makes it possible to use images and audio in a web conference system that uses an Internet line or the like. [Prior art documents] [Patent documents]

[0003] [Patent Document 1] Japanese Patent Publication No. 2022-164121 Summary of the Invention [Problem to be solved by the invention]

[0004] Connecting a WebRTC system to a traditional PBX (Private Branch Exchange) system requires a gateway. Even when connecting a WebRTC system to a PBX system, simple connections such as calls with the other party require only protocol conversion at the gateway. However, for various PBX voice services such as call holding and transfer, the sequence is too complex, making it difficult to provide voice services.

[0005] In view of the above circumstances, an object of the present invention is to provide a variety of voice services by connecting a WebRTC system to a PBX system. [Means for solving the problem]

[0006] The present invention, which solves the above problem, is a communication system comprising a relay device and an exchange device, wherein the relay device comprises a first relay processing unit that processes interactions with a browser terminal that is currently using a call, and a second relay processing unit that, when receiving a transfer request to a destination terminal from the browser terminal, transmits first information relating to the transfer request, which has an INFO message of SIP (Session Initiation Protocol), to the exchange device, and wherein the exchange device, when receiving the first information, performs a first call processing using SIP to the destination terminal and performs a hold processing using SIP to the source terminal that is currently using a call, a second exchange processing unit that, after the first call processing has been performed and after the hold processing has been performed, performs a response processing related to the first information to the relay device using second information having an INFO message of SIP, and a third exchange processing unit that, after the response processing has been performed, performs a second call processing using SIP to the relay device.

[0007] The present invention also provides a communication method in a communication system including a relay device and an exchange device, the method comprising: a first relay step in which the relay device processes communication with a browser terminal that is currently in a call; and a second relay step in which, upon receiving a transfer request to a transfer destination terminal from the browser terminal, the relay device transmits first information, which is information regarding the transfer request and has an SIP INFO message, to the exchange device; and a first exchange step in which, upon receiving the first information, the exchange device performs a first call process using SIP with the transfer destination terminal and a hold process using SIP with the transfer source terminal that is currently in a call; a second exchange step in which, after the first call process has been performed and after the hold process has been performed, the relay device performs a response process regarding the first information using second information having an SIP INFO message; and a third exchange step in which, after the response process has been performed, the relay device performs a second call process using SIP with the relay device.

[0008] The present invention also provides a relay device comprising a first relay processing unit that processes interactions with a browser terminal that is currently on a call, and a second relay processing unit that, when receiving a transfer request to a destination terminal from the browser terminal, transmits first information, which is information regarding the transfer request and has a SIP INFO message, to an exchange device.

[0009] The present invention also provides a program for a relay device that causes a computer to function as a relay device, and that causes the computer to function as a first relay processing unit that processes interactions with a browser terminal that is currently on a call, and a second relay processing unit that, when receiving a transfer request to a destination terminal from the browser terminal, sends first information, which is information regarding the transfer request and has a SIP INFO message, to an exchange device.

[0010] The present invention also provides a relay method in which a relay device executes a first relay step in which the relay device processes an exchange with a browser terminal that is currently engaged in a call, and a second relay step in which, upon receiving a transfer request to a destination terminal from the browser terminal, the relay device transmits first information, which is information relating to the transfer request and has a SIP INFO message, to an exchange device.

[0011] The present invention also provides an exchange device comprising: a first exchange processing unit that, upon receiving a transfer request to a destination terminal from a browser terminal that is currently using a call, executes a first call processing using SIP to the destination terminal and executes a hold processing using SIP to the source terminal that is currently using a call, when the relay device receives the transfer request from the browser terminal that is currently using a call, the first information being information related to the transfer request, the first information having an SIP INFO message; a second exchange processing unit that, after the first call processing has been executed and after the hold processing has been executed, executes a response processing related to the first information to the relay device using second information having an SIP INFO message; and a third exchange processing unit that, after the response processing has been executed, executes a second call processing using SIP to the relay device.

[0012] The present invention also provides a program for an exchange device that causes a computer to function as an exchange device, the program causing the computer to function as a first exchange processing unit that, when it receives first information having an SIP INFO message, which is information regarding a transfer request to a transfer destination terminal, from a relay device that has received the transfer request from a browser terminal that is currently using a call, executes a first call processing using SIP to the transfer destination terminal and executes a hold processing using SIP to the transfer source terminal that is currently using a call; a second exchange processing unit that, after the first call processing has been executed and after the hold processing has been executed, executes a response processing related to the first information to the relay device using second information having an SIP INFO message; and a third exchange processing unit that, after the response processing has been executed, executes a second call processing using SIP to the relay device.

[0013] The present invention also provides an exchange method in which, when an exchange device receives, from a relay device that has received a transfer request to a transfer destination terminal from a browser terminal that is currently using a call, first information that is information regarding the transfer request and has an SIP INFO message, the exchange device executes a first call processing using SIP to the transfer destination terminal and executes a hold processing using SIP to the transfer source terminal that is currently using a call; a second exchange step, after the first call processing has been executed and after the hold processing has been executed, of executing a response processing regarding the first information to the relay device using second information that has an SIP INFO message; and a third exchange step, after the response processing has been executed, of executing a second call processing using SIP to the relay device. [Effects of the Invention]

[0014] According to the present invention, a variety of voice services can be provided by connecting a WebRTC system to a PBX system. [Brief explanation of the drawings]

[0015] [Figure 1] FIG. 1 is a functional configuration diagram of a communication system according to an embodiment of the present invention. [Figure 2A] FIG. 10 is a sequence diagram of hold and transfer processing. [Figure 2B] FIG. 10 is a sequence diagram of hold and transfer processing. DETAILED DESCRIPTION OF THE INVENTION

[0016] Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings. Each drawing is merely a schematic illustration to allow a sufficient understanding of the present invention. Therefore, the present invention is not limited to the illustrated examples. In each drawing, common or similar components are designated by the same reference numerals, and redundant explanations thereof will be omitted.

[0017] [Explanation regarding the premise of the present invention] This invention relates to a gateway for connecting WebRTC terminals to a conventional PBX voice system. By deploying the gateway of this invention, all of the complex sequences of the PBX system are absorbed by the gateway, and WebRTC terminals can use PBX services with only simple call control.

[0018] [composition] 1 is a functional configuration diagram of a communication system according to this embodiment. The communication system 100 is a system that provides a voice service realized between a WebRTC terminal 1 and a SIP (Session Initiation Protocol) terminal 5. The communication system 100 includes a gateway 3 and a PBX 4. The communication system 100 also includes the WebRTC terminal 1, a signaling server 2, and SIP terminals 5a(5) and 5b(5).

[0019] (WebRTC device 1) The WebRTC terminal 1 is a telephone terminal that can be used from a browser. The WebRTC terminal 1 may be, for example, a browser phone or a PC with a telephone app installed and a device capable of audio input and output, but is not limited to these. The WebRTC terminal 1 has a browser function and a WebRTC function. The browser function is a function that executes applications consisting of various data such as HTML (HyperText Markup Language), JavaScript (registered trademark), still image data, text data, and audio data. The WebRTC function is a function that transmits and receives signaling signals (signaling information), media (e.g., audio), etc. to and from other devices in real time according to a standard (protocol) that realizes high-speed data communication via a browser or the like. In addition, the URI (Uniform Resource Identifier) ​​assigned to the WebRTC terminal 1 corresponds to a telephone number.

[0020] (Signaling Server 2) The signaling server 2 is a server for mutual communication between two or more devices. The signaling server 2 can identify the communication partner at the time of communication by obtaining information about the communication partner, such as IP address and codec, before communication begins. Specifically, the signaling server 2 transmits a signaling (call control) message received from one device to the other device (call allocation). The signaling server 2 has a WebSocket function. The WebSocket function is a function for sending and receiving signaling (call control) messages to and from other devices in accordance with a standard (protocol) for two-way communication between a browser and a web server.

[0021] (SIP terminal 5) The SIP terminal 5 is a telephone terminal that uses the SIP protocol, and is a SIP phone. The SIP phone is located in a SIP network. The SIP terminal 5 has a SIP UA (SIP User Agent). The SIP UA has the function of transmitting and receiving messages related to signaling (call control), media (e.g., voice), etc. to and from other devices in accordance with SIP. The URI assigned to the SIP terminal 5 corresponds to a telephone number.

[0022] (PBX4) The PBX 4 is a switching device that controls connections between extensions and between external lines and extensions to realize voice communications. The PBX 4 includes a proxy unit 41 and a transfer unit .

[0023] The proxy unit 41 manages SIP communications. Managing SIP communications is one of the functions of a SIP server. In other words, the proxy unit 41 can function as a SIP server. The proxy unit 41 can manage communications between SIP UAs. Specifically, when the proxy unit 41 receives a request message or a response message from a SIP UA, it can send (transfer) the received message to the SIP UA of the other party based on the connection destination information for the other party.

[0024] The forwarding unit 42 processes messages related to forwarding. Specifically, when the forwarding unit 42 receives a request message related to forwarding from a SIP UA, it can transmit the received message to the SIP UA of the other party involved in the forwarding, or receive a response message, based on the connection destination information of the other party involved in the forwarding (forwarding destination). The forwarding unit 42 can also process messages related to holding for the other party involved in the forwarding.

[0025] (Gateway 3) The gateway 3 is a relay device for mutual communication between two or more devices. The gateway 3 relays messages related to signaling (call control) between devices using different protocols, and can transmit a message related to signaling (call control) received from one device to another device. The gateway 3 also relays media (e.g., audio) transmitted based on a message related to signaling (call control) between devices using different protocols, and can transmit media received from one device to another device. The gateway 3 includes a WebRTC UA unit 31, a SIP UA unit 32, and a conversion unit 33.

[0026] The WebRTC UA unit 31 transmits and receives messages related to signaling (call control) and media (e.g., audio) to and from other devices in accordance with a standard (protocol) that realizes high-speed data communication via a browser, etc. The WebRTC UA unit 31 exchanges data with the WebRTC terminal 1.

[0027] The SIP UA unit 32 is a functional unit equivalent to a SIP UA. The SIP UA unit 32 allows the gateway 3 to function as a device that can operate on a SIP network. In other words, the gateway 3 can have a configuration in which a SIP terminal is embedded. The SIP UA unit 32 can send and receive messages related to signaling (call control), media (e.g., voice), etc. to and from other devices in accordance with SIP.

[0028] The conversion unit 33 converts a message related to signaling (call control) received by the WebRTC UA unit 31 into a message that conforms to the standard (protocol) used by the SIP UA unit 32. The conversion unit 33 also converts a message related to signaling (call control) received by the SIP UA unit 32 into a message that conforms to the standard (protocol) used by the WebRTC UA unit 31. It should be noted that media (e.g., audio) is exchanged between the WebRTC UA unit 31 and the SIP UA unit 32, and does not go through the conversion unit 33 (no conversion takes place).

[0029] The gateway 3 includes hardware such as an input unit, an output unit, a control unit, and a storage unit. For example, if the control unit is configured with a CPU (Central Processing Unit), information processing by a computer including the control unit is realized by program execution processing by the CPU. Furthermore, a storage unit included in the computer stores various programs for realizing the functions of the computer in response to instructions from the CPU. This realizes collaboration between software and hardware. The programs can be provided by recording them on a recording medium or via a network. The gateway 3 may be communicably connected to a console (not shown) used by a user. The console can display the processing results of the gateway 3 and input requests to the gateway 3. The above description of the gateway 3 also applies to the PBX 4, the WebRTC terminal 1, the signaling server 2, and the SIP terminal 5.

[0030] [process] The processing of this embodiment will be described. When making a call from the WebRTC terminal 1 to the SIP terminal 5, a call connection (session) can be established and the call can be started by performing simple processing such as protocol conversion by the gateway 3. In this embodiment, a case where a call between the WebRTC terminal 1 and the SIP terminal 5a is transferred to the SIP terminal 5b will be described. Hold will also be described. Figures 2A and 2B are sequence diagrams relating to the hold and transfer processing.

[0031] For ease of explanation, the URI of the WebRTC terminal 1 is assumed to be 9000. The URIs of the SIP terminals 5a and 5b are assumed to be 8000 and 8001, respectively. When making a call from the WebRTC terminal 1, the URI of the gateway 3 is 9000. The WebRTC terminal 1 and the SIP terminal 5a are in a call, and sessions 61 and 62 are established via the gateway 3. Session 61 is formed between the WebRTC terminal 1 and the gateway 3, and data is exchanged according to SRTP (Secure Real-time Transport Protocol). SRTP is a protocol that uses encryption specifications in RTP (Real-time Transport Protocol). Session 62 is formed between the gateway 3 and the SIP terminal 5a, and data is exchanged according to RTP. RTP is a protocol for playing back data such as audio data and video data in real time.

[0032] Furthermore, during a call, SDP (Session Description Protocol) is exchanged between devices, and P2P (Peer to Peer) communication is carried out in accordance with RTP. SDP is the body of the SIP message format, and is set on both the request and response sides. The SIP message format has a start line, header, and body. Here, the contents of the start line and header are specified by SIP, but the contents of the body are not specified by SIP and can be written freely. The start line describes the request (SIP method), etc. The header describes the request's destination, origin, path taken by the request, etc.

[0033] For example, the SDP prepared for a device can describe the IP address, port number, and audio codec (standard) of the device. In this embodiment, the SDP provided in the WebRTC terminal 1 is SDP <1> SDP <1> The IP address written in is 192.0.2.1, the port number is 16384, and the audio codec is G.711. In addition, the SDP provided in Gateway 3 is SDP <2> ,SDP <3> SDP <2> is the SDP prepared in the WebRTC UA unit 31. <3> is the SDP prepared in the SIP UA unit 32. <2> The IP address written in the SDP is 192.0.2.3, the port number is 16384, and the audio codec is G.711. <3> The IP address written in is 192.0.2.4, the port number is 5060, and the audio codec is G.711. The SDP prepared in the SIP terminal 5a is <4> SDP <4> The IP address written in is 192.0.2.6, the port number is 5060, and the audio codec is G.711. The SDP prepared in the SIP terminal 5b is SDP <5> SDP <5> The IP address written in is 192.0.2.7, the port number is 5060, and the audio codec is G.711. Also, the SDP provided in PBX4 is SDP <6> SDP <6> The IP address written in is 192.0.2.5, the port number is 5060, and the audio codec is G.711.

[0034] As shown in FIG. 2A, first, the WebRTC terminal 1 transmits a message called INFO(PBXFwd8000+8001) to the signaling server 2 (step S1). INFO(PBXFwd8000+8001) is a request indicating a transfer command. This transfer command requests the PBX 4 to transfer from the SIP terminal 5a to the SIP terminal 5b. INFO is a SIP method indicating information notification within a session. Next, the signaling server 2 transmits a message called INFO(PBXFwd8000+8001) to the gateway 3 (step S2). The messages in steps S1 and S2 are based on the WebRTC standard (protocol).

[0035] Next, the gateway 3 sends a message called SIP:INFO(PBXFwd8001) to the PBX 4 (step S3). SIP:INFO(PBXFwd8001) is a request indicating a transfer command. This transfer command is a transfer command based on the transfer command from the WebRTC terminal 1. Next, the PBX 4 sends a message called SIP:200OK to the gateway 3 (step S4). SIP:200OK is a response indicating a successful response corresponding to SIP:INFO(PBXFwd8001). The messages in steps S3 and S4 are based on the SIP standard (protocol). In other steps, messages marked with "SIP:" also indicate that they are based on the SIP standard (protocol).

[0036] Next, PBX4 sends a SIP:INVITE(8001,SDP <3> ) to the SIP terminal 5b (step S5). <3> ) is a request indicating a session initiation request to the SIP terminal 5b, which is the transfer destination. INVITE is a SIP method indicating a session initiation request. In step S5, the message sent is SDP <3> is included (added) in order to exchange the SIP UA capability of the SIP UA provided in the gateway 3 (the SIP UA unit 32 thereof) with the SIP UA capability of the SIP terminal 5b.

[0037] Next, the PBX 4 transmits a message SIP:INVITE(8000, C=0) to the SIP terminal 5a (step S6). Next, the SIP terminal 5a transmits a message SIP:200OK to the PBX 4 (step S7). Next, the PBX 4 transmits a message SIP:ACK to the SIP terminal 5a (step S8). Steps S6 to S8 are pre-processing steps in the process of placing the call on hold at the transfer source. The SIP:INVITE (8000, C=0) in step S6 is a request indicating a session initiation request to the SIP terminal 5a that is the transfer source. The reason why C=0 is included (added) in the message sent in step S6 is to stop the media and put the destination on hold by sending an SDP including C=0 to the destination (the SIP terminal 5a that is the transfer source). The SDP that is sent also serves to enable the PBX 4 to send a hold tone. The SIP:200OK in step S7 is a response indicating a successful response corresponding to the SIP:INVITE(8000,C=0). The SIP:ACK in step S8 is a request (SIP method) indicating confirmation of session establishment corresponding to the SIP:INVITE(8000,C=0).

[0038] Next, PBX4 sends the SIP:INVITE(8000,SDP <6> ) to the SIP terminal 5a (step S9). Next, the SIP terminal 5a sends a message saying SIP:200OK(SDP <4> ) to the SIP terminal 5a (step S10). Next, the PBX 4 sends a message SIP:ACK to the SIP terminal 5a (step S11). Steps S9 to S11 are a process of holding the call to the transfer source. Step S9: SIP: INVITE(8000,SDP <6> ) is a request indicating a session initiation request to the SIP terminal 5a that is the transfer source. <6> is included (added) in order to exchange the PBX capability of the PBX 4 with the SIP UA capability of the SIP terminal 5a. Step S10 SIP: 200 OK (SDP <4> ) is SIP:INVITE(8000,SDP <6> ) is a response indicating a successful response corresponding to the SDP <4> is included (added) in order to exchange the PBX capability of the PBX 4 with the SIP UA capability of the SIP terminal 5a. The SIP:ACK in step S11 is SIP:INVITE(8000,SDP <6> ) is a request (SIP method) indicating confirmation of session establishment corresponding to the In steps S9 to S11, the PBX 4 transmits a hold music instruction 63 to the SIP terminal 5a, and the hold music is played in the SIP terminal 5a. <6> or SDP <4> It is processed using the audio codec G.711 described in.

[0039] Next, the SIP terminal 5b sends SIP:200OK(SDP <5> ) message is sent to the PBX 4 (step S12). <5> ) is SIP:INVITE(8001,SDP <3> ) is a response indicating a successful response. In step S12, the message sent is SDP <5> is included (added) in order to exchange the SIP UA capability of the SIP UA provided in the gateway 3 (the SIP UA unit 32 thereof) with the SIP UA capability of the SIP terminal 5b. Step S12 allows the call to arrive at the SIP terminal 5b.

[0040] Moving to FIG. 2B, next, the PBX 4 sends a message called SIP:INFO (transfer response) to the gateway 3 (step S13). The SIP:INFO (transfer response) is a request indicating a transfer response. This transfer response is a transfer response to the transfer command from the WebRTC terminal 1. Next, the gateway 3 sends a message called SIP:200OK to the PBX 4 (step S14). The SIP:200OK is a response indicating a successful response corresponding to the SIP:INFO (transfer response).

[0041] Next, the PBX 4 sends a message SIP:INVITE(9000,C=0) to the gateway 3 (step S15). Next, the gateway 3 sends a message SIP:200OK to the PBX 4 (step S16). Next, the PBX 4 sends a message SIP:ACK to the gateway 3 (step S17). Steps S15 to S17 are pre-processing for the transfer process to the gateway 3 that has requested the transfer command. The SIP:INVITE (9000, C=0) in step S15 is a request indicating a session start request to the gateway 3. The reason why C=0 is included (added) in the message sent in step S15 is to stop the media by sending an SDP including C=0 to the gateway 3 and to put the transfer destination SIP terminal 5b on hold (as a transfer preparation stage). The SDP sent also plays a role in sending a second call to the transfer destination SIP terminal 5b. The SIP:200OK in step S16 is a response indicating a successful response corresponding to the SIP:INVITE(9000,C=0). The SIP:ACK in step S17 is a request (SIP method) indicating confirmation of session establishment corresponding to the SIP:INVITE(9000,C=0).

[0042] Next, PBX4 sends the SIP:INVITE(9000,SDP <5> ) to the gateway 3 (step S18). Next, the gateway 3 sends a SIP:200OK (SDP <3> ) to the PBX 4 (step S19). Next, the PBX 4 sends a message SIP:ACK to the gateway 3 (step S20). Steps S18 to S20 are a transfer process to the gateway 3 that has requested the transfer command.

[0043] Step S18: SIP: INVITE(9000, SDP <5> ) is a request indicating a session start request to the gateway 3. The message sent in step S18 includes an SDP <5> is included (added) in order to exchange the SIP UA capability of the SIP UA provided in the gateway 3 (the SIP UA unit 32 thereof) with the SIP UA capability of the SIP terminal 5b. Step S19 SIP: 200 OK (SDP <3> ) is SIP:INVITE(9000,SDP <5> ) is a response indicating a successful response corresponding to the SDP <3> is included (added) in order to exchange the SIP UA capability of the SIP UA provided in the gateway 3 (the SIP UA unit 32 thereof) with the SIP UA capability of the SIP terminal 5b. The SIP:ACK in step S20 is the SIP:INVITE(9000,SDP <5> ) is a request (SIP method) indicating confirmation of session establishment corresponding to the

[0044] Next, the PBX 4 transmits a message called SIP:ACK to the SIP terminal 5b (step S21). The SIP:ACK in step S21 is a response to the SIP:INVITE(8001, SDP) in step S5. <3> ) is a request (SIP method) indicating confirmation of session establishment corresponding to the Through steps S5, S12, and S18 to S21, sessions 64 and 65 are established via the gateway 3. That is, the second call is connected, and a call between the WebRTC terminal 1 and the SIP terminal 5b begins. Session 64 is formed between the WebRTC terminal 1 and the gateway 3, and communication is performed in accordance with SRTP (Secure Real-time Transport Protocol). Session 65 is formed between the gateway 3 and the SIP terminal 5b, and communication is performed in accordance with RTP.

[0045] In step S2, the message sent from the signaling server 2 to the gateway 3 is processed by the WebRTC UA unit 31 of the gateway 3. In steps S3, S4, and S13 to S20, the messages sent and received by the gateway 3 are processed by the SIP UA unit 32 of the gateway 3. In the processing in the gateway 3, the conversion unit 33 appropriately converts between WebRTC messages and SIP messages.

[0046] [effect] According to this embodiment, a variety of voice services can be provided by connecting a WebRTC system to a PBX system. Specifically, by providing the gateway 3 with a SIP UA unit 32, a configuration similar to that of a SIP terminal supporting the services of the PBX 4 placed in the gateway 3 can be achieved. Because the SIP UA unit 32 can support complex SIP sequences, the WebRTC terminal 1 can communicate with the gateway 3 with only basic call control. As a result, the WebRTC terminal 1 can support various services of the PBX 4, such as hold and transfer. When a call is put on hold or forwarded using the SIP service, the SIP UA unit 32 performs call control with the PBX 4 and SIP terminal 5, so the connection with the WebRTC terminal 1 can be continued without affecting the WebRTC UA unit 31. The gateway 3 can implement hold or forwarding for the WebRTC terminal 1 simply by switching the source of the audio data sent to the WebRTC terminal 1 to hold music or forwarding destination audio. According to the configuration of the gateway 3, the PBX 4 can connect to the WebRTC system even if it is an existing PBX system, and can provide complex services such as hold and transfer to the WebRTC terminal 1 as in the past.

[0047] [Notes on structure and processing] The WebRTC terminal 1 is an example of a "browser terminal." The gateway 3 is an example of a "relay device." The PBX 4 is an example of an "exchange device." The SIP terminal 5a is an example of a "transfer source terminal." The SIP terminal 5b is an example of a "transfer destination terminal." The WebRTC UA unit 31 is an example of a "first relay processing unit." The SIP UA unit 32 is an example of a "second relay processing unit." The transfer unit 42 is an example of a "first exchange processing unit," a "second exchange processing unit," or a "third exchange processing unit." The message SIP:INFO(PBXFwd8001) in step S3 of FIG. 2A is an example of "first information that is information related to a transfer request and has an INFO message of SIP (Session Initiation Protocol)." SIP: INVITE (8001, SDP) in step S5 of FIG. 2A <3> ) is an example of "first call processing using SIP." The process using the message SIP:INVITE(8000,C=0) in step S6 of FIG. 2A and the process using the message SIP:INVITE(8000,SDP) in step S9 of FIG. 2A <6> ) is an example of "hold processing using SIP." The message SIP:INFO (transfer response) in step S13 of FIG. 2A is an example of "second information having an INFO message of SIP." The process using the message SIP:INFO (transfer response) in step S13 of FIG. 2A is an example of a "response process related to the first information." The process by the message SIP:INVITE(9000,C=0) in step S15 of FIG. 2A and the process by the message SIP:INVITE(9000,SDP) in step S18 of FIG. 2A <5> ) is an example of "second call processing using SIP." SDP <1> , SDP <2> , SDP <3> , SDP <4> , SDP <5> , SDP <6> are examples of the "first SDP," "second SDP," "third SDP," "fourth SDP," "fifth SDP," and "sixth SDP," respectively.

[0048] [Variations] (a) The voice services that can be provided by the present invention are not limited to call holding and call forwarding. For example, the present invention can also be applied to voice services such as voice warp, answering machine, and call waiting. (b) The audio codec for processing the hold music is not limited to G.711, and may be, for example, G.722, G.723.1, G726, or G729.

[0049] (c): The above-mentioned invention-specific features can be combined as appropriate. (d): Means that can be realized in software can be realized in hardware, and means that can be realized in hardware can be realized in software. [Explanation of symbols]

[0050] 100 Call Center System 1 WebRTC device (browser device) 2. Signaling Server 3 Gateway (relay device) 4 PBX (exchange device) 5, 5a, 5b SIP terminal (transfer source terminal, transfer destination terminal) 31 WebRTC UA unit (first relay processing unit) 32 SIP UA unit (second relay processing unit) 33 Conversion unit 41 Proxy Section 42 Transfer unit (first exchange processing unit, second exchange processing unit, third exchange processing unit)

Claims

1. A relay device and an exchange device are provided, The relay device a first relay processing unit that processes communication with a browser terminal that is currently in a call; a second relay processing unit configured to, when receiving a transfer request to a transfer destination terminal from the browser terminal, transmit first information to the exchange device, the first information being information related to the transfer request and including an INFO message of SIP (Session Initiation Protocol); The exchange device a first exchange processing unit that, upon receiving the first information, executes a first call process using SIP for the transfer destination terminal and executes a hold process using SIP for the transfer source terminal that is currently in a call; a second exchange processing unit that, after the first call processing has been executed and after the hold processing has been executed, executes a response processing regarding the first information to the relay device by using second information having an INFO message of SIP; a third exchange processing unit that executes a second call process using SIP to the relay device after the response process has been executed.

2. the browser terminal prepares a first SDP (Session Description Protocol) for establishing a call connection with the relay device, the relay device prepares a second SDP for establishing a call connection with the browser terminal and a third SDP different from the second SDP, The transfer source terminal prepares a fourth SDP, The destination terminal has a fifth SDP prepared, The switching device prepares a sixth SDP; the first exchange processing unit adds the third SDP to a message of the first call processing and adds the sixth SDP to a message of the hold processing; The communication system according to claim 1 , wherein the third exchange processing unit adds the fifth SDP to a message of the second call processing.

3. A communication method in a communication system including a relay device and an exchange device, The relay device a first relay step of processing communication with a browser terminal that is currently in a call; a second relay step of transmitting, to the exchange device, first information relating to the transfer request and having an INFO message of SIP when a transfer request to the transfer destination terminal is received from the browser terminal; The exchange device a first switching step of, upon receiving the first information, executing a first call process using SIP for the transfer destination terminal and executing a hold process using SIP for the transfer source terminal that is currently in a call; a second exchange step of executing a response process regarding the first information to the relay device using second information having an INFO message of SIP after the first call process has been executed and after the hold process has been executed; a third exchange step of executing a second call process using SIP to the relay device after the response process is executed.

4. a first relay processing unit that processes communication with a browser terminal that is currently in a call; A relay device comprising: a second relay processing unit that, when receiving a transfer request to a transfer destination terminal from the browser terminal, transmits first information, which is information regarding the transfer request and has a SIP INFO message, to an exchange device.

5. The relay device according to claim 4 , further comprising: a second SDP for establishing a call connection with the browser terminal; and a third SDP different from the second SDP.

6. A relay device program that causes a computer to function as a relay device, The computer a first relay processing unit that processes communication with a browser terminal that is currently in a call; A program for a relay device that functions as a second relay processing unit that, when a transfer request to a destination terminal is received from the browser terminal, sends first information to the exchange device, which is information regarding the transfer request and has a SIP INFO message.

7. The relay device a first relay step of processing communication with a browser terminal that is currently in a call; a second relay step of transmitting, to an exchange device, first information relating to the transfer request and having an SIP INFO message when a transfer request to a transfer destination terminal is received from the browser terminal.

8. a first exchange processing unit that, when receiving first information having an SIP INFO message, which is information relating to a transfer request to a transfer destination terminal from a relay device that has received the transfer request from a browser terminal that is currently in a call, executes a first call process using SIP for the transfer destination terminal and executes a hold process using SIP for the transfer source terminal that is currently in a call; a second exchange processing unit that, after the first call processing has been executed and after the hold processing has been executed, executes a response processing regarding the first information to the relay device by using second information having an INFO message of SIP; a third exchange processing unit that executes a second call process using SIP to the relay device after the response process has been executed.

9. the first exchange processing unit adds a third SDP to the message of the first call processing and adds a sixth SDP to the message of the hold processing; the third exchange processing unit adds a fifth SDP to the message of the second call processing; the third SDP is an SDP prepared in the relay device and is different from the second SDP for establishing a call connection with the browser terminal, The fifth SDP is an SDP prepared in the destination terminal, 9. The switching device according to claim 8, wherein the sixth SDP is an SDP prepared in the switching device.

10. A program for an exchange device that causes a computer to function as an exchange device, The computer a first exchange processing unit that, upon receiving first information having an SIP INFO message, which is information relating to a transfer request to a transfer destination terminal, from a relay device that has received the transfer request from a browser terminal that is currently in a call, executes a first call process using SIP for the transfer destination terminal and executes a hold process using SIP for the transfer source terminal that is currently in a call; a second exchange processing unit that executes a response process regarding the first information to the relay device by using second information having an INFO message of SIP after the first call process has been executed and after the hold process has been executed; a program for an exchange device that causes the relay device to function as a third exchange processing unit that executes second call processing using SIP after the response processing is executed;

11. The exchange device a first switching step of, when receiving first information, which is information regarding the transfer request to the transfer destination terminal and has an SIP INFO message, from a relay device that has received the transfer request to the transfer destination terminal from a browser terminal that is currently in a call, executing a first call process using SIP to the transfer destination terminal and executing a hold process using SIP to the transfer source terminal that is currently in a call; a second exchange step of executing a response process regarding the first information to the relay device using second information having an INFO message of SIP after the first call process has been executed and after the hold process has been executed; a third switching step of executing a second call process using SIP to the relay device after the response process has been executed.

Images