Connection Establishment Method, System and Storage Medium of SIP Call Center System

Abstract

The present application discloses a method for establishing a connection of a SIP call center system, including: in response to the client receiving a call request from a user to the SIP call center system, adding the user to the RTC room through the client, and sending a call establishment request to the voice platform, carrying the room identifier of the RTC room; the voice platform initiating a call to the SIP call center system based on the call establishment request; the voice platform initiating a call to the RTC gateway and instructing the RTC gateway to join the room specified by the room identifier; the voice platform bridging the calls respectively initiated to the SIP call center system and the RTC gateway to establish a communication connection between the SIP call center system and the APP through the RTC gateway. By applying the present application, an enterprise APP can call the enterprise's SIP call center system and establish a connection with the call center, greatly reducing the call cost with the call center.

Description

Technical Field

[0001] This application relates to the technical field of call centers, and particularly to a method, a system, and a computer-readable storage medium for establishing a connection in a SIP call center system. Background Art

[0002] Currently, the call centers of large enterprises are mostly based on the SIP technical architecture. Users and call center agents mostly conduct voice interactions via PSTN phones. Users must select from a complex and lengthy Interactive Voice Response (IVR) menu and wait for an agent to answer. Moreover, PSTN call charges are relatively expensive. Taking the 1010 number as an example, both incoming and outgoing calls cost ￥0.11 per minute.

[0003] Network voice applications that emerged in previous years, such as Skype calls, QQ calls, WeChat voice, and enterprise WeChat calls, can basically only conduct voice or text - image interactions between the same type of apps. They cannot perform heterogeneous communication and are difficult to be deeply integrated with enterprise - self - built applications. Therefore, they have not become the mainstream communication method. Summary of the Invention

[0004] In view of the above - mentioned prior art, embodiments of this application disclose a method for establishing a connection in a SIP call center system, which can call the SIP call center system of an enterprise through an enterprise APP and establish a connection with the call center, greatly reducing the call cost between the user and the call center.

[0005] To achieve the above object, this application adopts the following technical solutions:

[0006] A method for establishing a connection in a Session Initiation Protocol (SIP) call center system, the method comprising:

[0007] In response to the client receiving a call request from a user, the client adds the user to a Real - Time Communication (RTC) room, and sends a call - establishment request to a voice platform through a signaling server and a network voice control server. The call - establishment request carries the room identifier of the RTC room;

[0008] The voice platform initiates a call to the SIP call center system based on the call - establishment request;

[0009] The voice platform initiates a call to an RTC gateway and instructs the RTC gateway to join the room specified by the room identifier;

[0010] The voice platform bridges the calls initiated to the SIP call center system and the RTC gateway respectively, to establish a communication connection between the SIP call center system, through the RTC gateway, and the user.

[0011] Preferably, the method further includes: when the client receives a call request from the user to the SIP call center system, further receiving the service category selected by the user, and further carrying the proxy group identifier corresponding to the service category when sending a call establishment request to the voice platform;

[0012] Carrying the proxy group identifier when the voice platform makes a call to the SIP call center system;

[0013] After establishing the communication connection between the SIP call center system and the user through the RTC gateway, connecting the agent specified by the proxy group identifier to the communication connection.

[0014] Preferably, the operations of the voice platform making calls to the SIP call center system and the RTC gateway and the operations of instructing the RTC gateway are all carried out under the control of the network voice control server.

[0015] Preferably, the method further includes: when the agent or the user hangs up the connection, disconnecting the connections between the voice platform and the SIP call center system and the RTC gateway respectively, and the RTC gateway and the user exiting the room specified by the identifier of the room.

[0016] Preferably, the method further includes: during the maintenance of the communication connection, the agent records or videos the call; after the communication connection is disconnected, the agent views the recording or video of the call.

[0017] A connection establishment system for a SIP call center system, the system includes: an application module, a voice platform, a signaling server, a network voice control server, an RTC gateway, and a SIP call center system;

[0018] The application module is used to, in response to receiving a call request sent by the user to the SIP call center system by itself, add the user to the RTC room, and send a call establishment request to the voice platform through the signaling server and the network voice control server, and the call establishment request carries the room identifier of the RTC room;

[0019] The voice platform is used to initiate a call to the SIP call center system based on the call establishment request; is also used to initiate a call to the RTC gateway and instruct the RTC gateway to join the room specified by the room identifier; is also used to bridge the calls made to the SIP call center system and the RTC gateway respectively, and establish a communication connection between the SIP call center system and the user through the RTC gateway;

[0020] The RTC gateway is used to join the RTC room specified by the room identifier;

[0021] The SIP call center system is used to establish the communication connection with the user through the RTC gateway.

[0022] Preferably, when the application module receives a call request from the user to the SIP call center system, it is further configured to receive the service category selected by the user, and further carry the proxy group identifier corresponding to the service category when sending a call establishment request to the voice platform;

[0023] The voice platform carries the proxy group identifier when initiating a call to the SIP call center system;

[0024] The SIP call center system is further configured to, after establishing the communication connection with the user through the RTC gateway, access the agent specified by the proxy group identifier to the communication connection.

[0025] Preferably, the network voice control server is further configured to control the operations of the voice platform to initiate calls to the SIP call center system and the RTC gateway respectively and instruct the operations of the RTC gateway.

[0026] Preferably, when the agent or the user hangs up the call connection, the connections between the voice platform and the SIP call center system and the RTC gateway are disconnected, and the user and the RTC gateway exit the room specified by the identifier of the room.

[0027] Preferably, during the maintenance of the communication connection, the agent in the SIP call system is configured to record or video the communication connection; after the communication connection is disconnected, the agent is further configured to view the recorded or videoed content.

[0028] In the above processing of the present application, in response to the client receiving a call request from the user to the SIP call center system, the user is added to the RTC room through the client, and a call establishment request is sent to the voice platform through the client via the signaling server and the network voice control server, carrying the room identifier of the joined room. The voice platform initiates calls to the SIP call center system and the RTC gateway respectively based on the call establishment request, and instructs the RTC gateway to join the room specified by the room identifier; the voice platform bridges the two initiated calls to establish a communication connection between the SIP call center system and the user through the RTC gateway. Through the above processing, an end-to-end communication connection between the SIP call center and the user can be established using RTC. Description of the Drawings

[0029] To more clearly illustrate the technical solutions in the embodiments of the present application, the following will briefly introduce the accompanying drawings required for the description of the embodiments. Obviously, the accompanying drawings in the following description are only some embodiments of the present application. For those of ordinary skill in the art, without creative efforts, other accompanying drawings can be obtained based on these drawings.

[0030] Figure 1 It is a schematic diagram of the basic process of the call establishment method of the SIP call center system in the present application;

[0031] Figure 2 It is a schematic diagram of the structure of the connection establishment system of the SIP call center system in the present application;

[0032] Figure 3 It is a composition diagram of an example of the connection establishment system of the SIP call center system in the present application;

[0033] Figure 4 Based on Figure 3 the system and data flow, it is a message timing diagram for transmission between each entity module. Specific Embodiments

[0034] The following will clearly and completely describe the technical solutions in the embodiments of the present application in combination with the accompanying drawings in the embodiments of the present application. Obviously, the described embodiments are only some embodiments of the present application, rather than all embodiments. Based on the embodiments in the present application, all other embodiments obtained by those of ordinary skill in the art without creative efforts belong to the scope of protection of the present application.

[0035] The terms "first", "second", "third", "fourth", etc. (if any) in the specification and claims of the present invention and the above accompanying drawings are used to distinguish similar objects and do not necessarily need to describe a specific order or sequence. It should be understood that such used data can be interchanged under appropriate circumstances so that the embodiments of the present invention described herein can be implemented in an order other than those illustrated or described herein. In addition, the terms "comprising" and "having" and any variations thereof are intended to cover non-exclusive inclusion. For example, a process, method, system, product, or device that includes a series of steps or units does not necessarily have to be limited to those steps or units clearly listed, but may include other steps or units not clearly listed or inherent to these processes, methods, products, or devices.

[0036] The following will detail the technical solutions of the present invention with specific embodiments. The following several specific embodiments can be combined with each other, and the same or similar concepts or processes may not be repeated in some embodiments.

[0037] With the rapid development of network real-time communication (RTC) technologies such as live streaming, video conferencing, and online education, as well as the deployment of RTC vendor content delivery network (CDN) nodes and the continuous stability and maturity of RTC SDKs, it has become possible to connect enterprise mobile applications (Apps) with SIP call centers.

[0038] Embodiments of the present disclosure describe a method that combines RTC technology with SIP call center technology to connect the interactive voice response (IVR) and agent voice calls of mobile Apps with call centers, enabling users to quickly access enterprise call center resources and obtain corresponding services in such a network environment.

[0039] Figure 1 It is a schematic diagram of the basic process of the call establishment method of the SIP call center system in this application. As Figure 1 shown, the method includes:

[0040] Step 101, in response to the client receiving a call request from the user to the SIP call center system, the user is added to the RTC room through the client, and a call establishment request is sent to the voice platform through the client via the signaling server and the network voice control server. The call establishment request carries the room identifier of the RTC room that has been joined.

[0041] Embodiments of this application provide an application (APP) that allows users to use the APP to call the SIP call center system without using the public switched telephone network (PSTN) for calling.

[0042] In this step, the user can operate the APP to select to call the SIP call center system. The enterprise APP receives the call request sent by the user to the SIP call center system, invokes the RTC SDK to join an RTC room, and obtains the room identifier of the joined room.

[0043] The client (which can be an APP, for example) sends a call request to the voice platform (such as freeswitch) and carries the obtained room identifier. When the APP sends a call request to the voice platform, it can initiate a call request to the voice platform through the signaling server and the network voice control server.

[0044] Further preferably, various different service information of the SIP call center system can be provided on the APP for the user to select the service category of the current call, such as order consultation, complaint and suggestion, etc. The APP receives the selected service category by the user and carries the proxy group identifier corresponding to the service category when sending a call request to the voice platform. The proxy group identifier is the identification information of the proxy group in the SIP call center system that provides a certain type of service.

[0045] Step 102: The voice platform initiates a call to the SIP call center system based on the call establishment request in Step 101.

[0046] After receiving the call request, the voice platform initiates a call to the SIP call center system. The specific internal processing of the SIP call center system can be carried out in the existing manner, such as playing voice navigation and other information.

[0047] In addition, preferably, if the user selects a certain service category through the APP in Step 101, the proxy group identifier corresponding to the service category can be obtained in the received call request in this step. When establishing a connection to the SIP call center system, this proxy group identifier is provided to the SIP call center system.

[0048] The above call operation of the voice platform for the SIP call center system can be completed under the control of the network voice control server.

[0049] Step 103: The voice platform initiates a call to the RTC gateway and instructs the RTC gateway to join the room specified by the room identifier.

[0050] After receiving the call request, the voice platform establishes a connection with the SIP call center system in Step 102, and in this step, it establishes a connection with the RTC gateway. Specifically, after receiving the call request, the voice platform obtains the room identifier from it, initiates a call to establish a connection with the RTC gateway, and carries the room identifier to instruct the RTC gateway to join the room specified by the room identifier. In this way, the RTC gateway and the APP join the same RTC gateway, and real-time communication between the RTC gateway and the user can be realized. At the same time, the voice platform also establishes a connection with the RTC gateway, that is, the voice platform establishes a connection with the APP through the RTC gateway. Among them, the signaling and data transmission between the voice platform and the RTC gateway can use the SIP protocol, and between the RTC gateway and the APP can be based on the RTC protocol. In this case, the RTC gateway can complete the function of protocol conversion and is responsible for the protocol conversion between RTC data packets and SIP data packets.

[0051] In addition, the above operation of the voice platform for the RTC gateway can be completed under the control of the network voice control server.

[0052] Step 104: The voice platform bridges the calls initiated to the SIP call center system and the RTC gateway to establish a communication connection between the SIP call center system, through the RTC gateway, and the APP.

[0053] Through the processing of the above steps 102 and 103, the voice platform initiates calls to the SIP call center system and the RTC gateway respectively. These two calls can be executed in any order. For example, steps 102 and 103 are executed in parallel, or step 102 is executed first and then step 103, or step 103 is executed first and then step 102.

[0054] After the voice platform initiates the above two calls, in this step, the two calls are bridged into one call, and the SIP call center system establishes a communication connection with the APP through the RTC gateway, such as a two-way real-time communication connection. In this way, the SIP call center system and the APP are connected, and an end-to-end connection between the SIP call center system and the user is established, enabling communication (such as two-way real-time communication) between the APP and the SIP call center system. The processing of bridging the two calls by the voice platform can also be completed under the control of the network voice control server.

[0055] In addition, preferably, if the voice platform sends the agent group identifier to the SIP call center in step 102, after establishing the connection between the SIP call center system and the APP, the SIP call center system can connect the agent specified by the agent group identifier to the two-way communication connection. Among them, according to the agent group identifier, the service type required by the user can be determined. The agent group of this service type includes several agents for providing corresponding services to the user. One of the agents can be selected to access the SIP connection between the APP and the SIP call center system. The agent phone of the SIP call center rings, the CRM pops up a screen, and a communication connection is established with the App user through FreeSwitch, RTC gateway, RTC cloud. Here, since the RTC gateway performs the conversion between SIP and RTC and does not involve the traffic process control of the service, the voice platform FreeSwitch is still required for connection establishment. The specific process of connecting the corresponding agent to the connection can be carried out in the existing manner. For example, when the user inputs a selected service through IVR interaction when calling the SIP call center system through PSTN, the SIP call center system will connect an agent of a certain service according to the user's selection. Then, the corresponding processing part in the existing PSTN call can be reused for the above-mentioned processing of connecting the agent. As can be seen from the above, the processing of the SIP call center system in the embodiments of the present application is the same as the existing one, and the existing SIP call center system can be directly used without adding new components.

[0056] After the communication connection (e.g., a two-way real-time communication connection) between the seats in the APP and the SIP call center system is established, the seat can record or video the corresponding communication and perform various existing configurations and operations (such as CRM, reports, etc.); if the SIP call center system or the user hangs up the connection, then the connection between the SIP call center system and the voice platform and the connection between the voice platform and the RTC gateway are torn down, and the RTC gateway and the APP exit the RTC room. After the call connection is disconnected, the seat can view the recording or video of the corresponding call.

[0057] Thus, the process of the connection establishment method for the SIP call center system in this application ends.

[0058] This application also provides a connection establishment system for a SIP call center system. Figure 2 It is a schematic diagram of the basic structure of the system. As Figure 2 shown, the system basically includes an application module, a voice platform, a signaling server, a network voice control server, an RTC gateway, and a SIP call center system.

[0059] Among them, the application module is used to, in response to receiving a call request sent by the user to the SIP call center system, add the user to the RTC room, and send a call establishment request to the voice platform through the signaling server and the network voice control server, and the call establishment request carries the room identifier of the RTC room that has been joined. The voice platform is used to initiate a call to the SIP call center system based on the call establishment request; it is also used to initiate a call to the RTC gateway and instruct the RTC gateway to join the room specified by the room identifier; it is also used to bridge the calls respectively initiated to the SIP call center system and the RTC gateway to establish a communication connection between the SIP call center system, the RTC gateway, and the user. The RTC gateway is used to join the RTC room specified by the room identifier. The SIP call center system is used to establish a communication connection with the user through the RTC gateway.

[0060] Optionally, when the application module receives the call request from the user to the SIP call center system, it can further be used to receive the service category selected by the user and further carry the proxy group identifier corresponding to the service category when sending the call establishment request to the voice platform. The voice platform carries the received proxy group identifier when initiating a call to the SIP call center system. The SIP call center system is further used to, after establishing a connection with the user through the RTC gateway, connect the seat specified by the received proxy group identifier to the communication connection. In this way, the user can directly select the required service through the application module (e.g., an APP) and directly establish a communication connection with the seat of the service.

[0061] Optionally, the network voice control server can also be used to control the voice platform to initiate calls to the SIP call center system and the RTC gateway respectively and to instruct the operation of the RTC gateway.

[0062] Optionally, when the agent or user hangs up the call connection, the connections between the voice platform and the SIP call center system and the RTC gateway are disconnected respectively, and the user and the RTC gateway exit the room specified by the identification of the room.

[0063] Optionally, during the communication connection, the agent in the SIP call system can be used to record or video the communication connection; after the communication connection is disconnected, the agent can also be used to view the recording or video of the corresponding communication connection.

[0064] Taking the example that the user initiates a voice call through the APP to the SIP call center system to establish a two-way real-time communication connection, a specific composition and process example of the system and method in this application are given below.

[0065] Figure 3 This is an example composition diagram of the connection establishment system of the SIP call center system of this application, specifically including an application module, a signaling server, a network voice control server, a voice platform (FreeSwitch), a SIP call center system, an RTC gateway, and an RTC cloud. Among them, HTTP / RPC, WS / TCP, kafka, and SIP in the figure are all data of the signaling layer, mainly used for call establishment, connection, resource preparation, and negotiation of relevant information such as codec types of the bitstream; RTP belongs to the data of the RTP layer, focusing on audio and video data processing. In this example, audio processing is used to illustrate. To conduct a real-time voice call, steps such as audio data collection, preprocessing, encoding, decoding, and playback are required.

[0066] As Figure 3 shown, the processing flow in the system specifically includes:

[0067] A) The App side initiates a voice call:

[0068] (1) The App calls the rtc sdk (android or ios) to join the RTC room;

[0069] (2) The App sends the userid, roomid, and agentgroupid to the signaling server;

[0070] (3) The signaling server sends a call request to the network voice control server, carrying userid / roomid / agentgroupid

[0071] B) Control operations of the network voice control server (network-voice)

[0072] (4) network-voice notifies its own FreeSwitch to initiate a call to the RTC-GW, and instructs the RTC-GW to join the RTC room specified by roomid, marking the call from its own FreeSwitch to the RTC-GW as call-leg-a;

[0073] (5) FreeSwitch initiates a call to the RTC gateway;

[0074] (6) The RTC-GW joins the RTC room specified by roomid;

[0075] (7) network-voice notifies Freeswitch to initiate a call to the SIP call center, carrying agentgroupid, and marking the call from FreeSwitch to the SIP call center as call-leg-b;

[0076] (8) network-voice notifies Freeswitch to bridge (bridge) call-leg-a and call-leg-b into one call;

[0077] (9) After the call reaches the call center sipserver, it is dispatched to the sip phone of the idle agent;

[0078] (10) The agent phone rings and is displayed on the CRM interface of the agent's computer;

[0079] (11) The agent answers the call through the softphone control bar embedded in the CRM;

[0080] (12) The agent conducts a voice call with the APP user through the sip phone via the sipserver and the bridged freeswitch, RTC-GW, and RTC cloud.

[0081] C) Call conversation and resource recovery

[0082] (13) The SIP call center performs call management such as recording and reporting;

[0083] (14) After the agent side or the app side hangs up the voice call, the relevant server disconnects the link and recovers the relevant resources, and the call ends.

[0084] Figure 4 For the system and data stream based on Figure 3 The message sequence diagram transmitted between each entity module. Specifically, it may include:

[0085] 1. The app initiates a voice call and joins the RTC room;

[0086] 2. The app sends a sip request to the signaling server, carrying the user ID, roomid, and agentgroupid;

[0087] 3. The signaling server sends a call request to network-voice, carrying the user ID, roomid, and the target agentgroupid;

[0088] 4. Network-voice sends a call request to RTC-GW, carrying the roomid;

[0089] 5. RTC-GW sends a request to the RTC cloud to join the room identified by the roomid;

[0090] 6. Network-voice sends a SIP call to the SIP call center, carrying the target agentgroupid and the user ID;

[0091] 7. The SIP call center plays the IVR;

[0092] 8. Network-voice bridges the call established with the SIP call center and the call established with the RTC gateway together;

[0093] 9. The app user in the room identified by the roomid hears the IVR and the queuing waiting tone;

[0094] 10. The agent at the SIP center answers;

[0095] 11. The app user communicates with the answering agent at the SIP center;

[0096] 12. The call center agent can listen to the recording afterwards.

[0097] Through the processing of the present application described above, the live broadcast of the currently popular RTC and the real-time voice capabilities of the audio and video conferencing App can be utilized, integrating the telephone / conference signaling function, and then connecting with the SIP telephone system. This system connects the enterprise self-developed App with the internal SIP telephone system:

[0098] (1) The App network voice module checks into the enterprise App, which can run on the Android or iOS system and integrate with internal and external enterprise services. Using this App, it is possible to directly call into the enterprise's SIP call center system;

[0099] (2) When the Wifi or mobile data signal is weak and it is impossible to make a normal call with the SIP call center system using the APP, it can be flexibly switched to a traditional PSTN phone call;

[0100] (3) No new components need to be added to the enterprise SIP phone system / call center system, and components such as the phone / CRM / reports / recording do not need to be changed;

[0101] (4) The App side uses voice coding of RTC such as Opus (sampling rate of 16k - 48k), which is converted into g.711 standard telephone voice coding (sampling rate of 8k) through the RTC voice gateway to maintain high-quality call quality.

[0102] The embodiment of the present application also provides a computer-readable storage medium. The computer-readable storage medium stores instructions, and when the instructions are executed by a processor, the steps in the connection establishment method of the above-mentioned SIP call center system can be executed. In practical applications, the computer-readable medium can be included in each device / device / system of the above embodiments, or can exist separately and not be assembled into the device / device / system. Among them, instructions are stored in the computer-readable storage medium, and when the stored instructions are executed by a processor, the steps in the connection establishment method of the above-mentioned SIP call center system can be executed.

[0103] According to the embodiments disclosed in the present application, the computer-readable storage medium can be a non-volatile computer-readable storage medium. For example, it can include but is not limited to: portable computer disks, hard disks, random access memories (RAM), read-only memories (ROM), erasable programmable read-only memories (EPROM or flash memories), portable compact disk read-only memories (CD-ROMs), optical storage devices, magnetic storage devices, or any suitable combination of the above, but is not used to limit the scope of protection of the present application. In the embodiments disclosed in the present application, the computer-readable storage medium can be any tangible medium that contains or stores a program, and this program can be used by or in combination with an instruction execution system, device, or device.

[0104] The flowcharts and block diagrams in the accompanying drawings of the present application illustrate the possible architectures, functions, and operations of systems, methods, and computer program products according to various embodiments disclosed in the present application. In this regard, each block in the flowchart or block diagram may represent a module, a program segment, or a portion of code that contains one or more executable instructions for implementing the specified logical function. It should also be noted that in some alternative implementations, the functions marked in the blocks may occur in a different order than that marked in the different drawings. For example, two consecutively represented blocks may actually be executed substantially in parallel, and they may sometimes be executed in the reverse order, depending on the functions involved. It should also be noted that each block in the block diagram or flowchart, as well as combinations of blocks in the block diagram or flowchart, can be implemented by a dedicated hardware-based system that performs the specified functions or operations, or can be implemented by a combination of dedicated hardware and computer instructions.

[0105] Those skilled in the art can understand that the features described in the various embodiments and / or claims of the present disclosure can be combined and / or combined in various ways, even if such combinations or combinations are not explicitly described in the present application. In particular, without departing from the spirit and teachings of the present application, the features described in the various embodiments and / or claims of the present application can be combined and / or combined in various ways, and all such combinations and / or combinations fall within the scope of the present disclosure.

[0106] Specific embodiments are used herein to illustrate the principles and implementation manners of the present invention. The description of the above embodiments is only used to help understand the method and its core idea of the present invention, and is not used to limit the present application. For those skilled in the art, changes can be made in the specific implementation manners and application scopes according to the ideas, spirits, and principles of the present invention. Any modifications, equivalent replacements, improvements, etc. made by them shall be included within the scope of protection of the present application.

Claims

1. A method for establishing a connection in a Session Initiation Protocol call center system, characterized in that, The method includes: In response to the client receiving a call request from a user, the user is added to a real-time communication room through the client, and a call establishment request is sent to the voice platform through the client via the signaling server and the network voice control server, where the call establishment request carries the room identifier of the real-time communication room; The voice platform initiates a call to the Session Initiation Protocol (SIP) call center system based on the call establishment request; The voice platform initiates a call to the RTC gateway and instructs the RTC gateway to join the room specified by the room identifier; The voice platform bridges the calls initiated to the SIP call center system and the RTC gateway respectively to establish a communication connection between the SIP call center system and the user via the RTC gateway.

2. The method according to claim 1, wherein The method further includes: when the client receives a call request from the user to the SIP call center system, the service category selected by the user is further received, and the proxy group identifier corresponding to the service category is further carried when sending the call establishment request to the voice platform; The proxy group identifier is carried when the voice platform initiates a call to the SIP call center system; After establishing the communication connection between the SIP call center system and the user via the RTC gateway, the agent specified by the proxy group identifier is connected to the communication connection.

3. The method according to claim 1 or 2, wherein The operations of the voice platform initiating calls to the SIP call center system and the RTC gateway and instructing the RTC gateway are all carried out under the control of the network voice control server.

4. The method according to claim 1 or 2, characterized in that, The method further includes: when the agent or the user hangs up the connection, the connections between the voice platform and the SIP call center system and the RTC gateway are disconnected respectively, and the RTC gateway and the user exit the room specified by the room identifier.

5. The method according to claim 4, characterized in that, The method further includes: during the period when the communication connection is maintained, the agent records or videos the call; after the communication connection is disconnected, the agent views the recording or video of the call.

6. A connection establishment system for a Session Initiation Protocol call center system, characterized in that, The connection establishment system includes: an application module, a voice platform, a signaling server, a network voice control server, an RTC gateway, and a SIP call center system; The application module is configured to, in response to receiving a call request sent by a user to the SIP call center system by itself, add the user to a real-time communication room, and send a call establishment request to the voice platform through the signaling server and the network voice control server, where the call establishment request carries the room identifier of the real-time communication room; The voice platform is configured to initiate a call to the SIP call center system based on the call establishment request; is further configured to initiate a call to the RTC gateway and instruct the RTC gateway to join the room specified by the room identifier; is further configured to bridge the calls initiated to the SIP call center system and the RTC gateway respectively to establish a communication connection between the SIP call center system and the user via the RTC gateway; The RTC gateway is used to join the real-time communication room specified by the room identifier; The Session Initiation Protocol call center system is used to establish the communication connection with the user through the RTC gateway.

7. The connection establishment system according to claim 6, wherein When receiving the call request from the user to the Session Initiation Protocol call center system, the application module is further used to receive the service category selected by the user, and further carry the proxy group identifier corresponding to the service category when sending a call establishment request to the voice platform; The voice platform carries the proxy group identifier when initiating a call to the Session Initiation Protocol call center system; The Session Initiation Protocol call center system is further used to connect the agent specified by the proxy group identifier to the communication connection after establishing the communication connection with the user through the RTC gateway.

8. The connection establishment system according to claim 6 or 7, wherein The network voice control server is further used to control the voice platform to initiate calls to the Session Initiation Protocol call center system and the RTC gateway respectively and instruct the operations of the RTC gateway.

9. The connection establishment system according to claim 6 or 7, characterized in that, When the agent or the user hangs up the call connection, disconnect the connections between the voice platform and the Session Initiation Protocol call center system and the RTC gateway respectively, and the user and the RTC gateway exit the room specified by the room identifier.

10. The connection establishment system according to claim 9, wherein, During the maintenance of the communication connection, the agent in the Session Initiation Protocol call center system is used to record or video the communication connection; after the communication connection is disconnected, the agent is further used to view the recorded or videoed content.

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