JP2019115044A - Voice service system and voice service method - Google Patents

Abstract

A voice service system and a voice service method are provided.
A voice service system comprises: a first server for forwarding an incoming call to another server after a called-side device receives an incoming call from the calling-side device; and an incoming call transferred from the first server , And a second server that receives a called party's telephone number called by the incoming call and transfers the incoming call to another server, an incoming call transferred from the second server, and a called party telephone called by the incoming call And a third server that receives the number and provides voice service for the call.
[Selected figure] Figure 4

Description

  The present invention relates to voice services, and more particularly to a voice service system and a voice service method.

  With the integration of voice and data networks, data services can be used to transport voice services, enabling telecommunications carriers and network operators to meet the diverse needs of network consumers and gain more. Various network value added services can be deployed to obtain Network voice services, such as voice messaging and personal call centers, are network value added services with great development potential. Because the network voice service provider does not own its telecommunications network, and rents the telecommunications carrier's network to provide services, it provides consumers with limited telecommunications network resources and competitive prices. Providing network voice service is very important.

  Therefore, there is a need for a voice service system and voice service method that can save telecommunications network resources and provide voice services.

  In one aspect of the present invention, a first server that transfers the incoming call to another server after the called-side apparatus receives an incoming call from the calling-side apparatus, and an incoming call transferred from the first server; And a second server that receives a called party telephone number called by the incoming call and transfers the incoming call to another server, an incoming call transferred from the second server, and a called party telephone number called by the incoming call And a third server that provides voice service to the call.

  In another aspect of the present invention, the first server transmits the telephone number of the called device to the second server after the called device receives an incoming call of the calling device; Providing a voice service method comprising the steps of: 2) transferring the telephone number to a third server; and 3) providing voice service based on the telephone number.

FIG. 1 is a system diagram showing a voice service system according to an embodiment of the present invention. FIG. 7 is a system diagram showing a voice service system according to another embodiment of the present invention. FIG. 1 is a diagram showing a SIP voice service system according to an embodiment of the present invention. 5 is a flowchart of a voice service method of the voice service system of FIG. 2;

  FIG. 1 is a system diagram showing a voice service system 1 according to an embodiment of the present invention. The voice service system 1 may include a first server 100, a second server 120 and a third server 140, wherein the first server 100 may be located in the carrier network 10, and the second server 120 may be an internet telephone service provider ( The ITSP may be located in the Internet Telephony Service Provider (ITP) network 12 and the third server 140 may be located in the Internet Protocol Voice Service Provider (IPVS) network 14. The second server 120 and the third server 130 may be connected by an Internet Protocol (IP: Internet Protocol) network 16 or may not be connected via the IP network 16. , The second server 120, the IP network 16, and the third server 140 may be sequentially connected by a wired method or a wireless method. The user equipments A and B can subscribe to the telephone service and network service of the telecommunications carrier network 10, in which case the entity numbers Na and Nb in the telecommunications carrier network 10 for the user equipments A and B are added. Each is assigned. Also, user equipments A and B can subscribe to the network telephone service of the ITSP network 12, for example, user equipment B can subscribe to the network telephone service on the ITSP network 12, in this case with the telecommunications company Based on the protocol with the ITSP network operator, the corresponding virtual number Vb is assigned to the user apparatus B in the ITSP network 12, and all incoming calls to the entity number Nb of the user apparatus B are the carrier network 10 to the virtual number Vb in the ITSP network 12. Since the ITSP network is based on IP (Internet Protocol), the user apparatus B can receive an incoming call from outside the network via the IP network according to the communication protocol of the network telephone such as SIP. In the present invention, the correspondence between the above-described entity number Nb and the virtual number Vb is referred to as a virtual pair.

  The configuration of FIG. 1 may be applied to the provision of an automatic voice value-added service, for example, the function of call forwarding on busy or call forwarding on no answering of the entity number Nb. It is assumed that another entity number Nc is set, and a virtual number Vc to be a virtual pair with the entity number Nc is located in the IPVS network 14. As a result, when there is an incoming call of the outside line to the entity number Nb, and the user apparatus B is in a call or in a non-answer state, the incoming call is a virtual number Vc located in the IPVS network 14 via the entity number Nc. At this time, this missed call may be provided with a value-added network voice service application, such as Intelligent Voice Assistant or Interactive Voice Response.

  In the network voice service application configured as shown in FIG. 1, one virtual pair (entity number Nc: virtual number Vc) is used for user device B (the user device B uses entity number Nb), and each virtual pair is constant. The system resource may be occupied, and this system resource may be regarded as the cost of the value added service. Providing services to 10,000 users requires occupying the resources of 10,000 virtual pairs, and can not effectively increase the economic benefits. As will be described later, the present invention provides a method that can provide the above-mentioned network voice service efficiently and at low cost.

  FIG. 2 is a system diagram showing a voice service system 2 according to another embodiment of the present invention. The network arrangement and connection of the voice service system 2 are similar to the voice service system 1 of FIG. 1, and the detailed description thereof will be omitted here. The voice service system 2 can provide network voice services to subscribers such as user equipments B1 and B2 by means of the telecommunications carrier network 10, the ITSP network 12, the IP network 16 and the IPVS network 14. The carrier network 10 can provide telephone services including landline telephone service and mobile telephone service to the user devices A1, A2, B1 and B2. The IPVS Network 14 can be used for phone answering, voice messaging, call screening, phone survey, telephone call center and other secretarial services. It can provide network voice services including: The user devices B1 and B2 can also subscribe to the network telephony service provided by the ITSP network 12 to convert the incoming call of the first server 100 into a network telephony. The ITSP network 12 can provide a Voice over Internet Protocol (VoIP) service, and the Voice over Internet Protocol is a Session Initiation Protocol (SIP), H. H.323, Media Gateway Control Protocol (MGCP), H.323. 248, Real-time Transport Protocol (RTP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), Skype Protocol (Skype Protocol), It may be another VoIP service. The first server 100 may be a gateway server, may be located in the carrier network 100, or may be located in the ITSP network 12. The third server 140 may be an application server capable of providing an interactive voice response (IVR) service.

  In the network voice service system 2 of FIG. 2, the user devices B1 and B2 both use the value-added network voice service, the user device B1 uses the entity number Nb1, and the user device B2 uses the entity number Nb2. In this case, another set of numbered resources is required for designating call forwarding to the third server 140 in the IPVS network 14, but as described above, each set of numbered resources or virtual pairs is In order to occupy a certain system resource, the designated transfer number (virtual pair) used by the user device B1 and the designated transfer number (virtual pair) used by the user device B2 are identical. Assuming Vc is used, system resources and system cost can be saved. However, when all incoming calls are transferred to Vc, the third server 140 in the IPVS network 14 can not determine whether the incoming call is an incoming call to the user apparatus B1 or an incoming call to the user apparatus B2. . Thus, the present invention provides a way for application servers in the IPVS network 14 to correctly distinguish the called party (Callee). The following describes the specific method.

  When a call forwarded call arrives at the IPVS network 14, the normal incoming call information carries only the number of the calling party (Caller) according to the network telephony protocol, for example, the SIP communication protocol, and the call is It is not possible to distinguish between designated transfer and transfer from a certain number. On the other hand, in the present embodiment, when the telecommunication carrier network 10 and the ITSP network 12 forward incoming calls, the entity numbers Nb1 or Nb2 (also referred to as a forwarding party) for setting forwarding By recording and notifying the IPVS network 14, the IPVS network 14 can correctly distinguish the incoming called side device.

  As described above, in one embodiment, the diversion header field in the SIP communication protocol is used to distinguish the incoming call. When an incoming call is Forwarded, the carrier network 10 and the ITSP network 12 may notify the Callee server of Forwarding party information via the diversion header field. . According to this method, it is possible to reduce the cost of value-added network voice service and efficiently increase the economic benefit.

  The first server 100 sends the second server 120 the telephone number of the callee to the second server 120 after the callee receives the incoming call from the callee when the callee calls the callee. In particular, the first server 100 may transmit the telephone number of the called apparatus to the third server 140 via the second server 120 based on the virtual telephone number corresponding to the network voice service. For example, the first server 100 may transmit the incoming of the entity number Nc by the ITSP network 12 to the corresponding virtual number Vc of the network voice service in the ITSP network 14. When the second server 120 receives the telephone number of the called device transferred from the first server 100, the second server 120 may transfer the telephone number of the called device to the third server 140 by the IP network 16, for example, The second server 120 may transfer the incoming of the entity number Nc to the corresponding virtual number Vc of the network voice service in the ITSP network based on the virtual pair Nc: Vc. The third server 140 may provide the network voice service based on the telephone number of the called device after receiving the telephone number of the called device transferred from the second server 120. For example, in the practical example where the above-mentioned user device B1 or B2 provides voice automatic value added service using virtual pair Nc: Vc, user device B1 uses the telephone response service of IPVS network 14 and user device A1 does not. If the telephone number Nb1 of the user device B1 is called and the user device B1 is in a call or in a non-answer state, the first server 100 adds the entity number Nb1 of the user device B1 to the incoming information of the user device B1; The incoming information is transferred by the second server 120 to the virtual number Vc of the third server 140 via the entity number Nc. The first server 100 may transmit call termination information including the telephone number of the called party apparatus to the second server 120 using a wired method, for example, a leased line. The first server 100 may transmit incoming information including the telephone number of the called device to the second server 120 by VoIP, and the second server 120 may transmit information fields to be transmitted to the third server 140, for example. The telephone number of the called device may be written in the diversion header field of the SIP information, so that the third server 140 can provide voice service based on the telephone number in the field. The second server 120 may write the program name of the application program for providing the voice service in the field of the information to be transmitted to the third server 140, for example, the diversion reason field of the SIP information. The server 140 transmits the telephone number of the called device to the application program based on the program name of the application program in the field, and the application program identifies the called device based on the telephone number of the called device, and the called device is called. A corresponding service may be provided based on the calling party's telephone number.

  FIG. 3 is a diagram illustrating a Session Initiation Protocol (SIP) voice service system 3 according to an embodiment of the present invention. The SIP voice service system 3 includes a Public Switching Telephone Network (PSTN) 30, a telecommunications carrier gateway server 31, an ITSP network 32, and SIP trunking 33. The PSTN 30 is connected to the carrier gateway server 31, and the carrier gateway server 31 is connected to the ITSP network 32 by activating the SIP trunking 33. The telecommunications carrier gateway server 31 uses common channel signal No. 7 protocol / service integrated digital network user side protocol (SS7 / ISUP: Signaling System number 7 / Integrated Services digital network User Part) to SIP block 310, SS7 / ISUP block 312, and SIP block 314. The block 310 of SS7 / ISUP to SIP is connected to the SS7 / ISUP block 312 and the SIP block 314. The ITSP server 320 is connected to the IVR server 322 and includes a SIP block 3200. The SS7 / ISUP to SIP block 310, the SS7 / ISUP block 312, the SIP block 314 and the SIP block 3200 may be implemented in software or hardware manner.

  The telecommunications carrier gateway server 31 and the ITSP network 32 exchange SIP information by SIP trunking 33. The SS7 / ISUP protocol is a communication protocol used by the PSTN 30. When the user device A calls the user device B, the telephone information is transmitted to the ITSP server 320 via the carrier gateway server 31 based on the corresponding virtual telephone number of the network voice service. Since the telephone number of the voice user apparatus B is included in the telephone information, even when a plurality of user apparatuses share the same virtual telephone number, the user apparatus B can be identified by the telephone number of the user apparatus B. it can. In the telecommunication carrier gateway server 31, the telephone information is received by the SS7 / ISUP block 312, converted into SIP information by the SS7 / ISUP to SIP block 310, and the SIP block 314 to the SIP block 3200 by the SIP trunking 33. It is transmitted. The SIP information may include the telephone number of the user device B and the software name of the application software for providing the voice service. In the ITSP server 320, the SIP block 3200 receives the SIP information and transfers the SIP information to the IVR server 322. The IVR server 322 identifies the user apparatus B by the telephone number of the user apparatus B, and voices by the software name of the application software. By identifying the application software for providing the service, the application software can provide the network voice service subscribed to the user device B, for example, playing back a pre-recorded voice message by the user device B Can.

  FIG. 4 is a flowchart of the voice service method of the voice service system of FIG. As shown in FIG. 4, the method includes steps S400 to S404. Adjustment of any reasonable technical change or step is within the scope of the present invention. The following describes steps S400 to S404.

  Step S400: The first server 100 transfers an incoming call to the second server 120 after the call-in device B receives an incoming call from the call-in device A, and the call-in to the second server 120 according to a specific field Transmit the device B's telephone number.

  Step S402: The second server 120 transfers the incoming call to the third server 140, and transfers the telephone number to the third server 140 according to a specific field.

  Step S404: The third server identifies the user based on the telephone number in the specific field and provides the related voice service.

  Since the above has described step S400 to step S404 in detail, the detailed description thereof is omitted here. By steps S400 to S404, the same virtual telephone number can be repeatedly assigned to a plurality of user devices, and user devices can be identified using the telephone numbers of a plurality of user devices, and corresponding voice services can be provided. , System resources and system costs of virtual telephone numbers can be saved.

  The voice service system and voice service method of FIG. 2, FIG. 3 and FIG. 4 identify the user equipment by using the telephone number of the user equipment, and provide the voice service subscribed to the user equipment by using the virtual telephone number. System resources and system cost can be saved, and voice services can be provided.

  Although the preferred embodiments of the present invention have been described above, all changes and modifications that are equivalent and uniform based on the claims of the present invention fall within the scope of the present invention.

  Although the preferred embodiments of the present invention have been described above, all changes and modifications that are equivalent and uniform based on the claims of the present invention fall within the scope of the present invention.

1, 2 voice service system 10 telecommunication carrier network 100 first server 12, 32 ITSP network 120 second server 14 IPVS network 140 third server 16 IP network 3 SIP voice service system 30 PSTN
31 Telecommunications carrier gateway server 310 SS7 / ISUP to SIP block 312 SS7 / ISUP block 314 SIP block 320 ITSP server 3200 SIP block 322 IVR server 33 SIP trunking A, B, A1, A2, B1, B2 User equipment Na , Nb, Nb1, Nb2, Nc Entity No. Vb, Vc Virtual No. S400 to S404 Steps

Claims (11)

A first server that forwards the incoming call to another server after the called machine receives an incoming call of the calling machine;
A second server receiving an incoming call transferred from the first server and a called party's telephone number called by the incoming call, and forwarding the incoming call to another server;
A third server receiving an incoming call transferred from the second server and a called party's telephone number called by the incoming call, and providing a voice service for the incoming call.   The voice service system according to claim 1, wherein the second server and the third server are connected by an Internet Protocol (IP) network. The first server transfers the incoming call to the second server by voice over IP.
The voice service system according to claim 1, wherein the second server writes specific information necessary for the voice service in a field of information to be transmitted to the third server. The voice service uses a third telephone number other than the calling party's telephone number and the called party's telephone number,
The second server writes the third telephone number in the field of information to be transmitted to the third server;
The voice service system according to claim 3, wherein the third server provides the voice service based on the third telephone number in the field. The second server dynamically writes the third telephone number into the field of information to be transmitted to the third server based on each incoming call;
The voice service system according to claim 4, wherein the third server identifies a user who has subscribed to the voice service based on the third telephone number in the field.   The voice service system according to claim 3, wherein the voice over IP is a session initiation protocol (SIP).   The voice service system according to claim 3, wherein the field storing the specific information is a diversion header field of a session initiation protocol. The first server transmitting the telephone number of the called device to the second server after the called device receives an incoming call of the calling device;
Transferring the telephone number to the third server by the second server;
The third server providing a voice service based on the telephone number.   When the first server transmits the telephone number of the called side device to the second server, the first server transmits the second server via the second server based on a virtual telephone number corresponding to the voice service. The voice service method according to claim 8, wherein the telephone number is transmitted to a third server. Further comprising the second server writing the telephone number in the field of information to be transmitted to the third server;
When the first server transmits the telephone number of the called device to the second server, the first server transmits the telephone number of the called device to the second server by voice over IP,
The voice service method according to claim 8 or 9, wherein when the third server provides voice service based on the telephone number, the third server provides the voice service based on the telephone number in the field. .   The voice service method according to claim 8 or 9, further comprising the step of the second server writing an application program for providing the voice service in a field of information to be transmitted to the third server.