A safe and controllable campus voice call management system

Abstract

The application provides a safe and controllable campus voice call management and control system, and belongs to the technical field of campus communication safety management and control. The application builds a whole-process management and control architecture integrating a SIP relay docking module, a two-way call routing module, a terminal access authentication module, a voice coding adaptation module, a call black and white list module, a call duration management and control module and a traceability audit module. The system is developed based on a standard SIP protocol, does not need to transform original campus voice hardware, guarantees link stability through a relay keep-alive mechanism, customizes a school-family exclusive two-way call routing, and realizes pre-interception of illegal calls in combination with terminal authentication and number black and white list. The application can be widely applied to classroom phones, dormitory IP phones, SIP soft phones and other scenes of primary and secondary schools, colleges and universities and boarding schools, effectively manages and controls call rights, compresses communication costs, standardizes usage behavior, improves campus communication safety and operation and maintenance efficiency, and has the advantages of simple deployment, strong compatibility and high management and control precision.

Description

Technical Field

[0001] This invention belongs to the field of campus communication security management technology, specifically a secure and controllable campus voice call management system. Background Technology

[0002] Currently, most primary, secondary, and higher education institutions in China have deployed network voice communication systems based on the standard SIP protocol. These systems include classroom public telephones, dormitory IP phones, access control voice terminals, and SIP softphones for teachers and students, primarily used for communication between home and school and internal campus communications. Existing traditional campus SIP voice systems only implement basic account login and authentication functions and lack customized control designs for the campus scenario, revealing several key shortcomings in actual operation: Call permissions are completely out of control: The existing system does not have number blocking or routing restrictions, and authorized terminals can freely dial overseas numbers, pay-to-call hotlines, and various nuisance calls. This not only causes a significant overspending of campus communication costs, but also poses a security risk of students being exposed to harmful information. At the same time, there are no dedicated home-school routing restrictions, which can easily lead to problems such as students making unauthorized calls to outside numbers and disorderly inter-school calls between terminals within the school.

[0003] Unrestricted call duration: Once a call is established, users can hang up at will. Some users occupy the communication trunk line for a long time, resulting in channel resource shortages, making normal home-school calls impossible, and significantly increasing maintenance costs.

[0004] Poor stability of trunk links: Traditional SIP trunks lack a timed keep-alive mechanism, and are prone to registration failure and link disconnection after long periods of inactivity, requiring manual periodic restart and maintenance, resulting in a large workload for operation and maintenance.

[0005] Insufficient codec compatibility: Some campus terminals do not match the voice codec of the operator's network, resulting in voice stuttering, noise, one-way silence and other faults, which affect the call experience.

[0006] Lack of traceability and auditing capabilities: Traditional systems simply record call numbers without complete signaling logs or call recordings. Once issues such as illegal calls or abnormal charges occur, it is impossible to trace the source of the behavior, making it difficult to meet the requirements of campus management and telecommunications compliance auditing.

[0007] Currently, there is no integrated SIP voice management solution specifically designed for campus scenarios that combines relay keep-alive, routing restrictions, terminal authentication, number blacklists and whitelists, time limits, and full-link traceability.

[0008] Therefore, developing a voice call management and control system that is suitable for campus use, secure, controllable, and easy to deploy has become an urgent need in the field of campus communication operation and maintenance. Summary of the Invention

[0009] To address the shortcomings of existing technologies, this invention provides a secure and controllable campus voice call management system. It achieves pre-blocking of call requests, dedicated restrictions on home and school routes, isolation of unauthorized terminals, automatic call duration limits, stable operation of trunk links, and full data traceability and auditing. Without modifying existing hardware, it standardizes campus voice usage, controls communication costs, and improves communication security and operational efficiency. It solves the problems of disordered calls, unlimited call duration, unstable links, and lack of traceability and auditing in existing campus SIP voice systems.

[0010] To achieve the above objectives, the present invention provides the following technical solution: A secure and controllable campus voice call management and control system is built on the standard SIP communication protocol, including a SIP trunk docking module, a two-way call routing module, a terminal access authentication module, a voice coding adaptation module, a call permission management and control module, a call duration management and control module, and a source tracing and auditing module. The SIP trunking module is used to connect to the operator's telecommunications network to achieve signaling interoperability and link keep-alive. The bidirectional call routing module is used to define a dedicated home-school call route; The terminal access authentication module is used to verify the legitimacy of the terminal and block unauthorized devices from accessing the network. The voice encoding adaptation module is used to unify the voice encoding format and ensure call quality. The call access control module blocks calls from unauthorized numbers using blacklists and whitelists. The call duration control module keeps real-time time and automatically disconnects the call when the limit is exceeded; The source tracing and auditing module is used to store call logs and recording data and supports query auditing.

[0011] Furthermore, the SIP trunking module is configured with the operator-assigned IP address and communication port, and has a built-in timed keep-alive mechanism with a keep-alive sending cycle of 10~30s to continuously maintain the registration status of the softswitch and the telecom core network and prevent the trunk link from going offline.

[0012] Furthermore, the bidirectional call routing module only allows bidirectional call routing between student terminals and parents' mobile phones, blocking unauthorized routes such as calls between unrelated terminals within the school and ordinary external calls, thus limiting the scope of call flow at the link level.

[0013] Furthermore, the terminal access authentication module pre-stores a list of legitimate terminal registrations, allowing only registered IP phones and SIP softphones to access the system; it directly intercepts access requests from anonymous terminals and unregistered illegal terminals, refusing to establish a communication link.

[0014] Furthermore, the voice encoding adaptation module adopts the G.711A / U standard voice encoding format by default, while also being compatible with mainstream encodings such as G.729 and G.723.1, to achieve seamless encoding adaptation between campus terminals and operator networks and ensure the clarity of voice transmission.

[0015] Furthermore, the call access control module has a built-in dynamic number blacklist and whitelist. The blacklist includes overseas numbers, toll-free voice service numbers, high-frequency harassment numbers, and illegal government numbers. The system directly blocks call requests from blacklisted numbers, while the whitelist only retains the registered parents' mobile phone numbers.

[0016] Furthermore, the call duration control module collects SIP call signaling in real time and accumulates call duration. The system presets a maximum single-call duration threshold of 3 to 15 minutes. When the actual duration reaches the threshold, it automatically sends a SIP hang-up signaling to both ends of the call to forcibly terminate the call.

[0017] Furthermore, the source tracing and auditing module automatically generates call detail records and signaling interaction logs, and supports selectively enabling full-process voice recording; all stored data is stored locally and backed up in the cloud with dual backups, with a minimum data retention period of ≥90 days, and supports conditional retrieval, source tracing, and compliance auditing.

[0018] Furthermore, a campus voice call control method based on the system described in any one of claims 1-8 includes the following steps: Step S1. Link Initialization: The SIP trunk connection module completes the operator network registration and starts timed keep-alive; Step S2. Terminal Authentication: The terminal initiating the call accesses the system, and the terminal access authentication module verifies the registration status. Illegal terminals are directly blocked. Step S3. Routing and Number Verification: The bidirectional call routing module determines the legality of the call route, and the call access control module verifies whether the called number is in the blacklist. If either is invalid, the call is blocked. Step S4. Call Establishment: After verification, the voice coding adaptation module performs unified coding to establish the call link; Step S5. Duration Monitoring: The call duration control module keeps track of the time in real time and automatically disconnects the call if the time expires. Step S6. Data Retention: After the call ends, the source tracing and auditing module automatically saves logs, call details and recording data.

[0019] Furthermore, the system is compatible with existing traditional IP phones, SIP softphones, dormitory access control voice terminals, and classroom public telephones on campus. It does not require replacement of existing hardware equipment; only the management and control program needs to be deployed on the existing SIP softswitch server. It is suitable for various campus communication scenarios in boarding primary and secondary schools, full-time colleges and universities, and vocational schools.

[0020] This invention provides a secure and controllable campus voice call management system. It has the following beneficial effects: 1. This invention provides a secure and controllable campus voice call management system. Through dual blocking of routing restrictions and number blacklists and whitelists, it completely eliminates calls to illegal numbers such as voice information, overseas numbers, and harassing numbers, avoids the risk of exposure to harmful information, regulates student usage behavior, and automatically controls the duration of single calls to prevent long-term line occupation, improves channel utilization, and significantly reduces communication costs incurred by excessively long calls, thereby reducing campus operating costs.

[0021] 2. This invention provides a secure and controllable campus voice call management system with a built-in timed keep-alive mechanism, ensuring that the trunk link is online for a long time without the need for manual restart and maintenance, reducing the workload of operation and maintenance by more than 70%. It adopts a multi-encoding compatible design to solve the problems of noise and one-way communication in traditional systems, and the voice transmission is clear and stable.

[0022] 3. This invention provides a secure and controllable campus voice call management system. Logs and recordings within the system are stored long-term, and violations can be accurately traced. It fully meets the compliance audit standards of campus management and the telecommunications industry. Moreover, it does not require modification of the original voice hardware, and can be launched simply by loading the program on the existing SIP server. It is compatible with various campus voice terminals and has a low threshold for promotion. Attached Figure Description

[0023] Figure 1 This is a diagram illustrating the overall architecture of the secure and controllable campus voice call management system of the present invention. Figure 2 This is a flowchart illustrating the overall process of intercepting campus call requests according to the present invention. Figure 3 This is a timing diagram for the automatic call duration control of the present invention; Figure 4 This is a diagram of the data storage and query architecture for call tracing and auditing according to the present invention. Detailed Implementation

[0024] The present invention will now be described in further detail with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative of the invention and not intended to limit it. Furthermore, it should be noted that, for ease of description, the accompanying drawings show only the parts relevant to the present invention, and not all of the structures.

[0025] like Figures 1-4As shown, this invention provides a secure and controllable campus voice call management system, developed based on the standard SIP session initiation protocol. It includes seven functional modules: SIP trunk connection module, bidirectional call routing module, terminal access authentication module, voice coding adaptation module, call permission management module, call duration management module, and source tracing and auditing module. These modules work together to achieve full-process management from the link layer to the access layer, call layer, call layer, and auditing layer. The system does not require replacing existing IP phones, softphones, or other hardware on campus; it can be deployed simply by loading the management program into the existing SIP softswitch server.

[0026] The SIP trunking module is used to connect to the operator's telecommunications network to achieve signaling interoperability and link keep-alive; the two-way call routing module is used to limit the dedicated home-school call route; the terminal access authentication module is used to verify the legitimacy of the terminal and block the access of unauthorized devices; the voice coding adaptation module is used to unify the voice coding format and ensure call quality; the call permission control module blocks calls from illegal numbers through blacklists and whitelists; the call duration control module keeps real-time time and automatically hangs up when the limit is exceeded; the source tracing and auditing module is used to store call logs and recording data and supports query auditing.

[0027] The functions of each module are as follows: SIP trunk connection module: Responsible for connecting the campus softswitch platform with the telecom operator's core network, configuring the operator's dedicated IP, communication port, registration account and password; built-in timed keep-alive mechanism, continuously sending SIP keep-alive signaling at 10-30 second intervals to maintain trunk registration status, prevent link drop-off, and ensure long-term stability of internal and external network voice links.

[0028] Two-way call routing module: Customize campus-specific communication routing rules, only opening one two-way call route from student terminal to parent's mobile phone, blocking all unauthorized routes such as calls between terminals within the school, ordinary external calls, and cross-regional calls, thus blocking unauthorized call flow from the bottom layer of the link.

[0029] Terminal access authentication module: A whitelist of legitimate terminals is pre-established, which includes the device number, IP address, and SIP account of all IP phones, SIP softphones, and access control voice terminals in the entire school; when a terminal initiates an access request, it automatically compares with the whitelist. Unregistered anonymous terminals and illegal terminals will have their access requests directly blocked and will be prohibited from accessing the voice system.

[0030] Voice encoding adaptation module: It adopts the general G.711A / U encoding format of China Telecom by default, and is also compatible with mainstream voice encoding such as G.729 and G.723.1. When the terminal accesses, it automatically negotiates the encoding format to solve the problems of noise, stuttering and one-way communication caused by encoding incompatibility, and ensures the clarity of voice transmission.

[0031] Call access control module: Configures dynamically editable blacklists and whitelists: the blacklist includes overseas numbers, toll-free numbers, high-frequency harassment numbers, and numbers used by government agencies in violation of regulations; the whitelist only retains registered parents' mobile phone numbers. The system completes number verification during the SIPINVITE call signaling phase, and calls matching the blacklist are directly blocked without establishing a call link.

[0032] Call duration control module: After a call is established, SIP media stream signaling is captured in real time and the call duration is accumulated; administrators can preset a maximum call threshold of 3 to 15 minutes according to the campus scenario. When the actual duration reaches the threshold, the system automatically sends SIPBYE hang-up signaling to the calling and called terminals to forcibly terminate the call and release channel resources.

[0033] Source tracing and auditing module: Collects SIP signaling interaction logs, caller and called numbers, call start and end times, and routing information throughout the entire process; supports enabling full-process voice recording as needed. All data adopts a dual backup mode of local storage and cloud storage, with a data retention period of no less than 90 days; administrators can query records through account retrieval, time period retrieval, number retrieval, etc., to complete source tracing and compliance auditing of violations.

[0034] The overall system workflow is as follows: 1) Link initialization: After the system is powered on, the SIP trunk connection module registers with the operator's core network and starts a timed keep-alive mechanism to keep the link online; 2) Terminal authentication: When a campus terminal initiates an access request, the terminal access authentication module verifies the device registration information, and unauthorized terminals are directly blocked; 3) Pre-call verification: When a legitimate terminal initiates a call, the bidirectional routing module verifies the legitimacy of the route, and the call authorization module verifies the called number. If either of these two steps is invalid, the call is directly intercepted and an interception log is generated. 4) Call Establishment: After all verifications pass, the voice encoding module completes encoding negotiation and establishes a two-way voice call link; 5) Call duration monitoring: The call duration control module keeps track of the time in real time, and automatically sends a hang-up signal to terminate the call when the preset threshold is reached; 6) Data Retention: After the call ends, the source tracing and auditing module automatically saves the call details, signaling logs, and recording data for subsequent query and auditing.

[0035] The technical solution of the present invention will be further described below with reference to the accompanying drawings and specific embodiments.

[0036] The system of this invention is deployed on a general-purpose Linux server and a standard SIP softswitch platform. The communication protocols and module programs used are all general technologies in the field of communication. The test environment is divided into three typical campus scenarios: boarding primary and secondary schools, full-time colleges and universities, and vocational colleges. Three sets of implementation examples and one set of comparative examples are set up. The test indicators include trunk online rate, illegal call interception rate, duration control accuracy, voice call quality, operation and maintenance workload, and cost saving ratio.

[0037] Example 1: Voice Control System for Boarding Primary and Secondary Schools ① Deployment Scenario For fully boarding primary and secondary schools, the deployment of shared IP phones in classrooms and voice terminals in dormitories is driven by the core requirements of managing home-school communication, strictly limiting call duration, and blocking toll-free numbers.

[0038] ②System parameter configuration 1) SIP trunk: The operator's IP and port are configured according to the actual situation, and the keep-alive cycle is 15 seconds; 2) Routing rules: Only bidirectional routing between student terminal and parent's mobile phone number is allowed; 3) Terminal authentication: Record the registration information of all 186 voice terminals in the school and block all unregistered devices; 4) Voice encoding: Default is G.711A encoding, compatible with G.729 encoding; 5) Blacklist and whitelist: The blacklist contains more than 2,300 voice, overseas, and harassing numbers, while the whitelist only retains the registered parents' numbers; 6) Duration threshold: The maximum single-call duration is 5 minutes; the call will automatically disconnect after the timeout. 7) Source tracing audit: Enable full recording, with data backed up both locally and in the cloud, and retained for 90 days.

[0039] ③ Running effect After 30 days of continuous operation and testing: 100% trunk online rate; 100% illegal call interception rate; 100% accuracy rate in duration control; no noise or one-way communication in voice messages; monthly communication tariffs decreased by 42% year-on-year; and no manual maintenance of trunk lines was required.

[0040] Example 2: Voice Control System for Full-Time Higher Education Institutions ① Deployment Scenario For comprehensive universities, the requirements for deploying office IP phones and student dormitory SIP softphones are to appropriately relax call duration, accommodate a small number of compliant external lines within the campus, and improve auditing functions.

[0041] ②System parameter configuration 1) SIP relay keep-alive cycle: 20s; 2) Routing rules: Retain the two-way home-school routing and add non-compliant routes for faculty and staff; 3) Terminal authentication: Register all 420 fixed terminals and authorized softphone accounts of teachers and students across the school; 4) Voice encoding: G.711U as the primary encoding, fully compatible with mainstream encodings; 5) Duration threshold: Maximum single-call duration is 10 minutes; 6) Source tracing audit: Important calls are recorded, while ordinary calls are only logged, and data is retained for 120 days.

[0042] ③ Running effect 30-day continuous testing: trunk online rate 99.9%; illegal call interception rate 99.8%; duration control accuracy 100%; channel utilization increased by 35%, and tariffs decreased by 28% year-on-year.

[0043] Example 3: Vocational School Voice Control System ① Deployment Scenario For vocational schools that are both day schools and mixed boarding schools, the number of terminals is large and the usage frequency is high, so it is important to strengthen the stability of relays and the ability to block numbers.

[0044] ②System parameter configuration 1) The SIP relay keep-alive cycle is 10 seconds, which shortens the keep-alive interval in high-frequency usage scenarios; 2) Routing: Strictly implement two-way routing between home and school; 3) Duration threshold: Maximum single-call duration is 8 minutes; 4) Blacklist and whitelist: Expand the database of harassing numbers and update it automatically on a regular basis; 5) The remaining configurations are the same as in Example 1.

[0045] ③ Running effect 30 consecutive days of testing: no dropped connections on the trunk line, 100% online rate; zero unauthorized calls, and no abnormalities in duration control.

[0046] Comparison: Traditional uncontrolled campus SIP voice system Using a conventional campus SIP voice platform, only account login authentication is retained, with no routing restrictions, no blacklists or whitelists, no duration control, no relay keep-alive, and no complete audit logs. The hardware terminal is completely consistent with that of Example 1.

[0047] Performance testing: A 30-day comparative test was conducted simultaneously on the voice control systems in Examples 1-3 and the comparative example. The results of various test indicators are shown in the table below: SIP trunk online rate 100% 99.9% 100% 82.3% illegal call blocking rate 100% 99.8% 100% 0% Accuracy of call duration control 100% 100% 100% Uncontrolled Voice call anomaly (noise / one-way communication) incidence rate 0.5% 0.6% 0.4% 18.7% Monthly communication tariff reduction 42% 28% 36% 0% Frequency of manual relay maintenance (times / month) 0 1 0 12 The comparison results above show that 1. This invention features a timed keep-alive mechanism, achieving a relay online rate of nearly 100%. Traditional systems lack keep-alive design, resulting in frequent disconnections and requiring extensive manual restarts and maintenance every month. 2. This invention uses a dual approach of routing and blacklists / whitelists to block virtually all unauthorized calls, whereas traditional systems have no blocking capabilities and unauthorized calls can be made freely. 3. This invention allows for precise time-limited call termination, completely solving the problem of excessively long busy lines, whereas traditional systems rely entirely on manual control; 4. Multi-encoding adaptive design results in a significantly lower speech anomaly rate compared to traditional systems; 5. This invention significantly reduces communication costs, greatly reduces manual maintenance workload, and has outstanding overall cost advantages; In summary, the performance indicators of this invention are superior to those of traditional campus SIP voice systems, and it is fully adaptable to various campus communication management and control needs.

[0048] The system of this invention is developed based on a general SIP softswitch platform. It does not require replacing existing IP phones, softphones, or other hardware on campus. Deployment can be completed simply by upgrading the software or loading modules, requiring minimal engineering work.

[0049] The system of this invention is compatible with all mainstream SIP terminals and server equipment on the market, and is adapted to campus voice systems of different brands and eras, with a wide range of applications.

[0050] The call duration threshold, keep-alive period, blacklist / whitelist, and data retention duration of this invention can all be flexibly adjusted according to the school type and management requirements, demonstrating strong customization capabilities.

[0051] The system of this invention features automatic relay keep-alive and fully automated process control, reducing manual intervention and resulting in low long-term operation and maintenance costs.

[0052] The system of this invention has a complete log and audio recording traceability system, which meets the compliance audit requirements of education management departments and telecommunications operators, and is ready for large-scale promotion.

[0053] This invention system can be fully implemented in primary and secondary schools, universities, and vocational colleges across the country, with a broad market potential and extremely high industrial applicability and commercial value.

[0054] The following points should be noted in this article: 1. The accompanying drawings of the embodiments disclosed herein only relate to the structures involved in the embodiments disclosed herein; other structures can be referred to in general design.

[0055] 2. Where there is no conflict, the embodiments of this disclosure and the features in the embodiments can be combined with each other to obtain new embodiments.

[0056] Although embodiments of the present invention have been shown and described, it will be understood by those skilled in the art that various changes, modifications, substitutions, and variations can be made to these embodiments without departing from the principles and spirit of the invention, the scope of which is defined by the appended claims and their equivalents. All other embodiments obtained by those skilled in the art based on the embodiments of the present invention without inventive effort are within the scope of protection of the present invention.

Claims

1. A secure and controllable campus voice call management system, characterized in that, Built on the standard SIP communication protocol, it includes a SIP trunking module, a two-way call routing module, a terminal access authentication module, a voice coding adaptation module, a call permission control module, a call duration control module, and a source tracing and auditing module. The SIP trunking module is used to connect to the operator's telecommunications network to achieve signaling interoperability and link keep-alive. The bidirectional call routing module is used to define a dedicated home-school call route; The terminal access authentication module is used to verify the legitimacy of the terminal and block unauthorized devices from accessing the network. The voice encoding adaptation module is used to unify the voice encoding format and ensure call quality. The call access control module blocks calls from unauthorized numbers using blacklists and whitelists. The call duration control module keeps real-time time and automatically disconnects the call when the limit is exceeded; The source tracing and auditing module is used to store call logs and recording data and supports query auditing.

2. The secure and controllable campus voice call management system according to claim 1, characterized in that, The SIP trunking module is configured with the operator-assigned IP address and communication port, and has a built-in timed keep-alive mechanism with a keep-alive sending cycle of 10~30s to continuously maintain the registration status of the softswitch and the telecom core network and prevent the trunk link from going offline.

3. The secure and controllable campus voice call management system according to claim 1, characterized in that, The bidirectional call routing module only allows bidirectional call routing between student terminals and parents' mobile phones, blocking unauthorized routes such as calls between unrelated terminals within the school and ordinary external calls, thus limiting the scope of call flow at the link level.

4. The secure and controllable campus voice call management system according to claim 1, characterized in that, The terminal access authentication module pre-stores a list of legally registered terminals and only allows registered IP phones and SIP softphones to access the system; it directly intercepts access requests from anonymous terminals and unregistered illegal terminals and refuses to establish a communication link.

5. The secure and controllable campus voice call management system according to claim 1, characterized in that, The voice encoding adaptation module uses the G.711A / U standard voice encoding format by default, while also being compatible with mainstream encodings such as G.729 and G.723.1, enabling seamless encoding adaptation between campus terminals and operator networks and ensuring the clarity of voice transmission.

6. The secure and controllable campus voice call management system according to claim 1, characterized in that, The call access control module has a built-in dynamic number blacklist and whitelist. The blacklist includes overseas numbers, toll-free numbers, high-frequency harassment numbers, and illegal government numbers. The system directly blocks call requests from blacklisted numbers, while the whitelist only retains the registered parents' mobile phone numbers.

7. The secure and controllable campus voice call management system according to claim 1, characterized in that, The call duration control module collects SIP call signaling in real time and accumulates call duration. The system presets a maximum single-call duration threshold of 3 to 15 minutes. When the actual duration reaches the threshold, it automatically sends a SIP hang-up signaling to both ends of the call to forcibly terminate the call.

8. The secure and controllable campus voice call management system according to claim 1, characterized in that, The source tracing and auditing module automatically generates call detail records and signaling interaction logs, and supports selectively enabling full-process voice recording. All data is stored locally and backed up twice in the cloud, with a minimum data retention period of ≥90 days. It supports conditional search, traceability, and compliance auditing.

9. A campus voice call control method based on the system described in any one of claims 1-8, characterized in that, Includes the following steps: Step S1. Link Initialization: The SIP trunk connection module completes the operator network registration and starts timed keep-alive; Step S2. Terminal Authentication: The terminal initiating the call accesses the system, and the terminal access authentication module verifies the registration status. Illegal terminals are directly blocked. Step S3. Routing and Number Verification: The bidirectional call routing module determines the legality of the call route, and the call access control module verifies whether the called number is in the blacklist. If either is invalid, the call is blocked. Step S4. Call Establishment: After verification, the voice coding adaptation module performs unified coding to establish the call link; Step S5. Duration Monitoring: The call duration control module keeps track of the time in real time and automatically disconnects the call if the time expires. Step S6. Data Retention: After the call ends, the source tracing and auditing module automatically saves logs, call details and recording data.

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