CN116055614A - Intelligent voice outbound system and method for realizing SIP relay through mobile phone - Google Patents
Intelligent voice outbound system and method for realizing SIP relay through mobile phone Download PDFInfo
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- CN116055614A CN116055614A CN202310054274.8A CN202310054274A CN116055614A CN 116055614 A CN116055614 A CN 116055614A CN 202310054274 A CN202310054274 A CN 202310054274A CN 116055614 A CN116055614 A CN 116055614A
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- 238000000034 method Methods 0.000 title claims abstract description 26
- 238000012545 processing Methods 0.000 claims abstract description 19
- 230000011664 signaling Effects 0.000 claims abstract description 16
- 230000006854 communication Effects 0.000 claims abstract description 13
- 238000004891 communication Methods 0.000 claims abstract description 13
- 230000005540 biological transmission Effects 0.000 claims abstract description 6
- 230000002618 waking effect Effects 0.000 claims abstract description 4
- 238000012790 confirmation Methods 0.000 claims description 8
- 230000007175 bidirectional communication Effects 0.000 claims description 4
- 238000006243 chemical reaction Methods 0.000 claims description 3
- 230000004044 response Effects 0.000 claims description 2
- 238000003672 processing method Methods 0.000 description 2
- 230000009286 beneficial effect Effects 0.000 description 1
- 238000010586 diagram Methods 0.000 description 1
- 230000000977 initiatory effect Effects 0.000 description 1
- 238000012986 modification Methods 0.000 description 1
- 230000004048 modification Effects 0.000 description 1
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Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
- H04M1/72409—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality by interfacing with external accessories
- H04M1/724092—Interfacing with an external cover providing additional functionalities
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/1046—Call controllers; Call servers
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
- H04M1/7243—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages
- H04M1/72433—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages for voice messaging, e.g. dictaphones
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72448—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
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- Y—GENERAL TAGGING OF NEW TECHNOLOGICAL DEVELOPMENTS; GENERAL TAGGING OF CROSS-SECTIONAL TECHNOLOGIES SPANNING OVER SEVERAL SECTIONS OF THE IPC; TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
- Y02—TECHNOLOGIES OR APPLICATIONS FOR MITIGATION OR ADAPTATION AGAINST CLIMATE CHANGE
- Y02D—CLIMATE CHANGE MITIGATION TECHNOLOGIES IN INFORMATION AND COMMUNICATION TECHNOLOGIES [ICT], I.E. INFORMATION AND COMMUNICATION TECHNOLOGIES AIMING AT THE REDUCTION OF THEIR OWN ENERGY USE
- Y02D30/00—Reducing energy consumption in communication networks
- Y02D30/70—Reducing energy consumption in communication networks in wireless communication networks
Abstract
The invention provides an intelligent voice outbound system and method for realizing SIP relay through a mobile phone, and belongs to the technical field of audio and video transmission. The system comprises an AI end, a voice processing end and a voice processing end, wherein the AI end is used for configuring client information and outbound information and processing voice streams of clients; the FreeWITCH terminal is used for establishing communication between the AI terminal and the mobile phone terminal and transmitting signaling and audio between the AI terminal and the mobile phone terminal; the mobile phone terminal is internally provided with an APP capable of interacting with a telephone module of the mobile phone, and is used for processing signaling sent by the AI terminal and obtaining client information; the mobile phone self-phone module is used for waking up the mobile phone self-phone module to dial a customer phone and playing the audio sent from the AI end and transmitted through the FreeWITCH end; for receiving a voice stream of a client: the invention replaces the method of directly calling the customer by FreeWITCH through the line, and solves the problems that the line is easy to be occupied and the customer cannot receive the feedback telephone.
Description
Technical Field
The invention relates to the technical field of audio and video transmission, in particular to an intelligent voice outbound system and method for realizing SIP relay through a mobile phone.
Background
FreeWITCH is used as an IP-telephone exchange and controlled by a route setting or program, and an enterprise can distribute a call dialed by a client to a telephone terminal where a designated seat representative is located through the FreeWITCH; also, the agent can use the phone terminal to place a call over FreeWITCH to place a fixed or mobile phone to the customer. However, the client displays a special number when the phone terminal calls out, and this number cannot be dynamically modified, possibly resulting in the client hanging up or refusing to receive; moreover, when the customer calls back, the line may be occupied, so that the enterprise cannot receive the incoming call of the customer in time.
Disclosure of Invention
Aiming at the problems, the invention provides an intelligent voice outbound system and method for realizing SIP relay through a mobile phone, and the adopted technical scheme is as follows:
in a first aspect, the present invention provides an intelligent voice outbound system for implementing SIP relay through a mobile phone, including:
the AI end is used for configuring client information and outbound information and processing the voice stream of the client;
the FreeWITCH terminal is used for establishing communication between the AI terminal and the mobile phone terminal, transmitting signaling and audio sent by the AI terminal to the mobile phone terminal, and transmitting voice stream of a client received by the mobile phone terminal to the AI terminal;
the mobile phone terminal is internally provided with an APP capable of interacting with a telephone module of the mobile phone, and the APP is used for processing signaling sent by the AI terminal and obtaining client information; the mobile phone self-phone module is used for waking up the mobile phone self-phone module to dial a customer phone and playing the audio sent from the AI end and transmitted through the FreeWITCH end; for receiving a voice stream of a client.
Further, the AI terminal includes:
the outbound configuration unit is used for configuring client information and outbound information, wherein the client information comprises a client mobile phone number;
a voice recognition unit for recognizing a voice stream of a client, converting the voice stream of the client into text;
a dialogue processing unit for generating a reply text according to the text converted from the voice stream of the client;
and the text conversion unit is used for converting the text in the outbound information configured at the AI end and/or the reply text generated by the dialogue processing unit into an audio file.
Further, the APP contact information built in the mobile phone terminal is registered in advance at the FreeSWITCH terminal.
Further, the FreeWITCH end and the APP built in the mobile phone end are communicated through an SIP protocol.
In a second aspect, the present invention provides an intelligent voice outbound method for implementing SIP relay through a mobile phone based on the outbound system, including the following steps:
s1, an AI end is used as a user agent client to initiate a call to a FreeWITCH end and transmit signaling and audio;
s2, the FreeWITCH end is used as a user agent server end to initiate a call and transmit signaling and audio through a media stream of a SIP protocol and a real-time transmission protocol to an APP built in the mobile phone end;
s3, the APP built in the mobile phone terminal is used as a user agent client terminal to receive the call of the FreeWITCH terminal and process the signaling sent by the AI terminal to acquire the client information;
s4, the built-in APP of the mobile phone wakes up the mobile phone self-phone module to dial the customer phone and plays the audio sent from the AI end and transmitted by the FreeWITCH end, and receives the voice stream of the customer;
s5, the APP built-in the mobile phone end transmits the received voice stream of the client to the AI end through the FreeWITCH end, response audio is generated after processing by the AI end and is transmitted to the APP built-in the mobile phone end through the FreeWITCH end, and a bidirectional communication link between the AI end and the APP built-in the mobile phone end is established.
Further, the phone module of the mobile phone sends a request to a communication operator through the traditional communication service and dials a client to perform two-way communication with the client.
Further, in step S1, the AI terminal initiates a session as a party of the SIP request, initiates a session request to the FreeSWITCH terminal, and sends an authentication message to the AI terminal after receiving the session request, and the AI terminal returns a confirmation character to confirm that the authentication request has been received, so that the AI terminal is connected to the FreeSWITCH terminal.
Further, in step S2, after the FreeSWITCH end processes the session request of the AI end, the FreeSWITCH end dials the network telephone to the APP built in the mobile phone end through the SIP protocol, when dialing the network telephone, the FreeSWITCH end initiates a session as a party of the SIP request, initiates a session request to the APP built in the mobile phone end, and sends an authentication message to the FreeSWITCH end after the APP built in the mobile phone end receives the session request, and the FreeSWITCH end returns a confirmation character to confirm that the authentication request has been received, so far, the FreeSWITCH end is connected with the APP built in the mobile phone end.
Further, in step S3, the APP built in the mobile phone terminal obtains the client phone number through session description protocol exchange in the SIP protocol, and stores the number in the memory.
Further, in step S4, after the phone module of the mobile phone dials the phone of the client, the client displays the number of the built-in phone card of the mobile phone.
The invention has the beneficial effects that: the invention replaces the method of directly calling the customer by FreeWITCH through the line, and solves the problems that the line is easy to be occupied and the customer cannot receive the feedback telephone.
Drawings
The accompanying drawings, which are included to provide a further understanding of the invention and are incorporated in and constitute a part of this specification, illustrate embodiments of the invention and together with the description serve to explain the invention and do not constitute a limitation on the invention. In the drawings:
fig. 1 is a schematic diagram of an intelligent voice outbound method for implementing SIP relay through a mobile phone according to an embodiment of the present invention.
Detailed Description
The technical solutions of the present application will be clearly and completely described below with reference to specific embodiments of the present application and corresponding drawings. It will be apparent that the described embodiments are only some, but not all, of the embodiments of the present application. All other embodiments, which can be made by one of ordinary skill in the art without undue burden from the present disclosure, are within the scope of the present disclosure.
As shown in fig. 1, the intelligent voice outbound method for implementing SIP relay by a mobile phone provided by the invention includes the following steps:
s1: the AI end is used as a user agent client end to initiate a call to the FreeWITCH end;
in this embodiment, before a call is initiated, contact information of an Application (APP) which is built in a mobile phone terminal and can interact with a phone module of the mobile phone itself is registered to a FreeSWITCH terminal, then an AI terminal is used as a party of an SIP request to initialize a session, a session request is initiated to the FreeSWITCH terminal, the FreeSWITCH terminal sends an authentication message to the AI terminal after receiving the session request, and the AI terminal sends a confirmation character confirmation message back to the FreeSWITCH terminal to confirm that an authentication requirement has been received, so far, the AI terminal is connected with the FreeSWITCH terminal.
S2: the FreeWITCH end is used as a user agent server end to transfer signaling and audio through a media stream of a SIP protocol and a real-time transmission protocol to an APP built in the mobile phone end;
in this embodiment, after the FreeSWITCH end processes the session initiation request of the AI end, the FreeSWITCH end dials the internet phone to the APP built in the mobile phone end through the SIP protocol, and transmits audio to the APP built in the mobile phone end through the real-time transmission protocol media stream. The voice call processing method comprises the steps that one part is audio generated by outbound information configured by an AI end and is set according to corresponding outbound tasks, and the other part is reply audio generated after voice streams of clients are processed, and the voice call processing method is used for realizing dialogue with the clients.
S3: an APP built in the mobile phone terminal is used as a user agent client to receive a call of the FreeWITCH terminal and process signaling sent by the AI terminal, so as to obtain client information;
in this embodiment, after receiving a session request from the FreeSWITCH end, the APP built in the mobile phone end sends an authentication message to the FreeSWITCH end, and the FreeSWITCH end returns a confirmation character confirmation message, so that the FreeSWITCH end is connected with the APP built in the mobile phone end, and complete communication from the AI end to the APP built in the mobile phone end is established, at this time, the APP built in the mobile phone end can obtain audio information from the FreeSWITCH end, and exchange and obtain a client telephone number based on a session description protocol in a SIP protocol, and store the number in a memory.
S4: an APP built in the mobile phone end dials a customer call through a telephone module of the mobile phone, transmits a voice stream of the AI end and receives the voice stream of the customer;
in this embodiment, the APP built in the mobile phone terminal interacts with the phone module of the mobile phone itself, wakes up the mobile phone call system to dial the acquired client number, establishes a bidirectional communication channel between the mobile phone terminal and the client terminal, and transmits the voice stream of the AI terminal and receives the voice stream of the client through the mobile phone terminal.
The mobile phone terminal sends a request to a communication operator through the traditional communication service and dials a number of a built-in phone card of the mobile phone terminal, so that the client terminal displays the number of the built-in phone card of the mobile phone terminal, and after the client dials back the displayed mobile phone number, the built-in APP of the mobile phone terminal interrupts the current connection, so that the user can answer the call back by himself, and the problems that the line is easy to be occupied and the user cannot receive the call back by himself are solved.
Corresponding to the foregoing embodiment of an intelligent voice outbound method for implementing SIP relay through a mobile phone, the present application further provides an embodiment of an intelligent voice outbound system for implementing SIP relay through a mobile phone, which mainly includes:
the AI end is used for configuring client information and outbound information and processing the voice stream of the client;
the FreeWITCH terminal is used for establishing communication between the AI terminal and the mobile phone terminal, transmitting signaling and audio sent by the AI terminal to the mobile phone terminal, and transmitting voice stream of a client received by the mobile phone terminal to the AI terminal;
the mobile phone terminal is internally provided with an APP capable of interacting with a telephone module of the mobile phone, and the APP is used for processing signaling sent by the AI terminal and obtaining client information; the mobile phone self-phone module is used for waking up the mobile phone self-phone module to dial a customer phone and playing the audio sent from the AI end and transmitted through the FreeWITCH end; for receiving a voice stream of a client.
The APP contact information built in the mobile phone terminal is registered in advance in the FreeWITCH terminal, and the FreeWITCH terminal and the APP built in the mobile phone terminal are communicated through an SIP protocol.
After the bi-directional communication link between the AI-terminal and the user-terminal is established, an intelligent dialogue between the AI-terminal and the client can be implemented, for example, the configuration can be performed in the AI-terminal:
the outbound configuration unit is used for configuring client information and outbound information, wherein the client information comprises a client mobile phone number;
a voice recognition unit for recognizing a voice stream of a client, converting the voice stream of the client into text;
a dialogue processing unit for generating a reply text according to the text converted from the voice stream of the client;
and the text conversion unit is used for converting the text in the outbound information configured at the AI end and/or the reply text generated by the dialogue processing unit into an audio file.
The specific manner in which the various modules perform the operations in relation to the systems of the above embodiments have been described in detail in relation to the embodiments of the method and will not be described in detail herein.
For system embodiments, reference is made to the description of method embodiments for the relevant points, since they essentially correspond to the method embodiments. The system embodiments described above are merely illustrative and the various modules may or may not be physically separate. In addition, each functional module in the present invention may be integrated in one processing unit, each module may exist alone physically, or two or more modules may be integrated in one unit. The integrated modules or units can be realized in a hardware form or a software functional unit form, so that part or all of the modules can be selected according to actual needs to realize the purpose of the scheme.
The foregoing list is only illustrative of specific embodiments of the invention. Obviously, the invention is not limited to the above embodiments, but many variations are possible. All modifications directly derived or suggested to one skilled in the art from the present disclosure should be considered as being within the scope of the present invention.
Claims (10)
1. An intelligent voice outbound system for implementing SIP relay by a mobile phone, comprising:
the AI end is used for configuring client information and outbound information and processing the voice stream of the client;
the FreeWITCH terminal is used for establishing communication between the AI terminal and the mobile phone terminal, transmitting signaling and audio sent by the AI terminal to the mobile phone terminal, and transmitting voice stream of a client received by the mobile phone terminal to the AI terminal;
the mobile phone terminal is internally provided with an APP capable of interacting with a telephone module of the mobile phone, and the APP is used for processing signaling sent by the AI terminal and obtaining client information; the mobile phone self-phone module is used for waking up the mobile phone self-phone module to dial a customer phone and playing the audio sent from the AI end and transmitted through the FreeWITCH end; for receiving a voice stream of a client.
2. The intelligent voice outbound system for implementing SIP relay through a mobile phone according to claim 1, wherein the AI terminal comprises:
the outbound configuration unit is used for configuring client information and outbound information, wherein the client information comprises a client mobile phone number;
a voice recognition unit for recognizing a voice stream of a client, converting the voice stream of the client into text;
a dialogue processing unit for generating a reply text according to the text converted from the voice stream of the client;
and the text conversion unit is used for converting the text in the outbound information configured at the AI end and/or the reply text generated by the dialogue processing unit into an audio file.
3. The intelligent voice outbound system for realizing the SIP relay through the mobile phone according to claim 1, wherein the built-in APP contact information of the mobile phone terminal is registered in advance at the FreeWITCH terminal.
4. The intelligent voice outbound system for realizing the SIP relay through the mobile phone according to claim 1, wherein the FreeWITCH terminal and the APP built-in the mobile phone terminal communicate through an SIP protocol.
5. An intelligent voice outbound method for realizing SIP relay through a mobile phone based on the outbound system as claimed in claim 1, comprising the steps of:
s1, an AI end is used as a user agent client to initiate a call to a FreeWITCH end and transmit signaling and audio;
s2, the FreeWITCH end is used as a user agent server end to initiate a call and transmit signaling and audio through a media stream of a SIP protocol and a real-time transmission protocol to an APP built in the mobile phone end;
s3, the APP built in the mobile phone terminal is used as a user agent client terminal to receive the call of the FreeWITCH terminal and process the signaling sent by the AI terminal to acquire the client information;
s4, the built-in APP of the mobile phone wakes up the mobile phone self-phone module to dial the customer phone and plays the audio sent from the AI end and transmitted by the FreeWITCH end, and receives the voice stream of the customer;
s5, the APP built-in the mobile phone end transmits the received voice stream of the client to the AI end through the FreeWITCH end, response audio is generated after processing by the AI end and is transmitted to the APP built-in the mobile phone end through the FreeWITCH end, and a bidirectional communication link between the AI end and the APP built-in the mobile phone end is established.
6. The method for implementing SIP relay intelligent voice outbound through mobile phone according to claim 5, wherein the mobile phone own phone module sends a request to a communication operator through a conventional communication service and dials a number for a client to perform two-way communication with the client.
7. The method for implementing SIP relay intelligent voice outbound through mobile phone according to claim 5, wherein in step S1, the AI terminal initiates a session as a party requesting SIP, initiates a session request to the FreeSWITCH terminal, and sends an authentication message to the AI terminal after receiving the session request, and the AI terminal returns a confirmation character to confirm that the authentication request has been received, so far, the AI terminal is connected to the FreeSWITCH terminal.
8. The method for implementing the intelligent voice outbound of the SIP relay through the mobile phone according to claim 5, wherein in step S2, after the FreeWITCH terminal processes the session request of the AI terminal, the FreeWITCH terminal dials the network telephone to the built-in APP of the mobile phone terminal through the SIP protocol, when the network telephone is dialed, the FreeWITCH terminal initiates the session as a party of the SIP request, initiates the session request to the built-in APP of the mobile phone terminal, the built-in APP of the mobile phone terminal sends an authentication message to the FreeWITCH terminal after receiving the session request, and the FreeWITCH terminal returns a confirmation character to confirm that the authentication request has been received, so far, the FreeWITCH terminal is communicated with the built-in APP of the mobile phone terminal.
9. The method for realizing the intelligent voice outbound of the SIP relay through the mobile phone according to claim 8, wherein in the step S3, the APP built in the mobile phone terminal obtains the telephone number of the client through the exchange based on the session description protocol in the SIP protocol, and stores the number in the memory.
10. The method for realizing the intelligent voice outbound of the SIP relay through the mobile phone according to claim 5, wherein in the step S4, after the mobile phone own phone module dials the client phone, the client displays the number of the built-in phone card of the mobile phone.
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