A method and apparatus for communicating over a VoIP and CT network

Abstract

The embodiment of the present specification provides a method and device for communication based on VoIP and CT network, wherein the method comprises: a calling terminal initiates a call request on an Internet side, wherein the call request contains a calling identification; an Internet server accesses a landing gateway; the Internet server or the landing gateway converts the calling identification into a calling number recognizable by a CT network; according to the call request, an Internet side signaling is generated, wherein the calling number is encapsulated in the Internet side signaling; the landing gateway performs authentication and verification on the Internet side signaling, and converts the Internet side signaling into a telecommunication side signaling after the verification; the telecommunication side signaling is sent to the CT network; the CT network receives the telecommunication side signaling, and connects a called terminal to realize communication with full-process traceability. The embodiment of the present specification can integrate traditional voice technology and emerging VoIP technology, and realize efficient and stable voice communication.

Description

Technical Field

[0001] This specification relates to the field of communications, and in particular to a method and apparatus for communicating with a VoIP and CT network. Background Technology

[0002] Existing traditional voice communication technologies mainly rely on cellular networks or operator core network authentication, which suffers from problems such as poor communication stability, poor indoor signal quality, extremely limited wireless bandwidth resources, high base station construction costs, and high communication expenses.

[0003] VoIP (Voice over Internet Protocol) is a technology that transmits voice over IP networks. It uses the Internet or a dedicated IP network to encode, compress, and group voice signals to achieve real-time voice communication.

[0004] The widespread adoption of VoIP has posed significant challenges to traditional voice technologies, primarily in terms of communication costs and the services they can provide. However, VoIP employs a connectionless technology, resulting in unreliable quality of service, which has severely hampered its development.

[0005] Therefore, there is an urgent need for a method based on VoIP and CT network communication that can integrate traditional voice technology with emerging VoIP technology to achieve efficient and stable voice communication. Summary of the Invention

[0006] The purpose of the embodiments in this specification is to provide a method and apparatus for VoIP and CT network communication to achieve efficient and stable voice communication.

[0007] To achieve the above objectives, in one aspect, embodiments of this specification provide a method for communication between VoIP and CT networks, including: The calling terminal initiates a call request on the Internet side, and the call request includes the calling identifier; The calling terminal connects to the local gateway via an internet server; The internet server or local gateway converts the caller ID into a caller number that can be recognized by the CT network; The Internet server or the landing gateway generates Internet-side signaling based on the call request, and the Internet-side signaling encapsulates the caller's number; The landing gateway authenticates and verifies the Internet-side signaling. After successful verification, the Internet-side signaling is converted into telecommunications-side signaling. The landing gateway sends the telecom-side signaling to the CT network; The CT network receives the signaling from the telecommunications side and connects the called terminal to achieve fully traceable communication.

[0008] Preferred options also include: The local gateway connects to the source tracing gateway; The calling terminal connects to the source tracing gateway; Throughout the entire communication process, the local gateway reports the metadata of the internet-side link in real time. Throughout the entire communication process, the calling terminal reports the metadata of the cellular side link in real time. Throughout the entire communication process, the traceability gateway records and associates metadata from both the internet-side link and the cellular-side link, forming a traceable traceability log.

[0009] Preferably, the metadata of the associated Internet sidelink and cellular sidelink, forming a traceable tracing log, further includes: Select an association identifier from the metadata, and associate the metadata of the Internet side link and the cellular side link through the association identifier; Based on the associated identifier, the metadata is arranged in chronological order to form a traceability log.

[0010] Preferably, the step of converting the caller ID into a caller number recognizable by the CT network further includes: Based on the mapping source priority from high to low, obtain the original number associated with the calling identifier; The original number is standardized and converted to obtain a calling number that can be recognized by the CT network.

[0011] Preferably, the step of standardizing the original number to obtain a caller ID recognizable by the CT network further includes:

[0012] The original number is cleaned to obtain the cleaned original number; Match the location of the cleaned original number, and complete the country code based on the location to obtain the completed original number; The format of the completed original number is validated, and the caller ID that can be recognized by the CT network is obtained after the validation passes.

[0013] Preferred options also include: The calling terminal continuously collects at least one of the following for the current call: signal strength, network latency, network jitter, packet loss rate, and network load traffic. At least one of the signal strength value, network latency, network jitter, packet loss rate, and network load traffic is preprocessed to obtain at least one type of data. The data at least one piece of data is converted to a uniform format to obtain at least one piece of data. A quality result is obtained by using a machine learning model to perform weighted calculations on at least one dataset in a uniform format; When the quality result is below the threshold, the network handover mechanism is triggered according to the handover strategy.

[0014] Preferably, the step of triggering a network handover mechanism according to a handover strategy when the quality result is below a threshold further includes: When the quality result is below a threshold, the calling terminal obtains the available WiFi and cellular networks in the vicinity. Sort WiFi networks and cellular networks according to their signal strength; Connect the corresponding networks in order of sorting to obtain the currently connected network; If the quality result of the currently connected network is not lower than the threshold, then switch to the currently connected network; If the quality result of the network currently being connected is lower than the threshold, then repeat the above steps of connecting the corresponding networks in order of sorting.

[0015] Preferred options also include: The local gateway connects to the backend management system; The local gateway pushes communication data to the backend management system in real time. The backend management system provides a visual representation and detects anomalies in the communication data.

[0016] Preferred options also include: The calling terminal accesses the Internet via WiFi.

[0017] On the other hand, embodiments of this specification provide a device based on VoIP and CT network communication, characterized in that the device includes: The calling module allows the calling terminal to initiate a call request on the internet side, and the call request includes the calling identifier. The access module allows the calling terminal to access the local gateway via an internet server. The conversion module, either an internet server or a local gateway, converts the caller ID into a caller number recognizable by the CT network. The generation module, the Internet server or the landing gateway, generates Internet-side signaling based on the call request, and the Internet-side signaling encapsulates the caller's number; The verification module, the landing gateway, performs authentication verification on the Internet-side signaling, and after the verification is successful, the Internet-side signaling is converted into telecommunications-side signaling; The sending module and the landing gateway send the telecom-side signaling to the CT network; The connection module receives the telecom-side signaling from the CT network and connects the called terminal to achieve fully traceable communication.

[0018] As can be seen from the technical solutions provided in the embodiments of this specification above, these embodiments achieve plug-and-play interoperability between VoIP and CT networks by mapping the caller ID to CT numbers on the Internet side, standardizing the encapsulation of the caller ID in the Internet-side signaling, and having the terminal gateway uniformly complete authentication and signaling conversion from the Internet side to the telecommunications side. This ensures correct caller ID display on the called side, significantly improves call connection rate, and reduces interception or failures caused by unidentifiable caller ID or signaling incompatibility. Centralized authentication and number mapping effectively suppress number spoofing and spam calls, enhances security, and meets regulatory compliance and billing traceability requirements. Simultaneously, it decentralizes complex protocol compatibility and policy control to the terminal gateway, requiring only the terminal to connect to the Internet server, reducing client-side and business-side modification costs, facilitating elastic scaling and unified operation and maintenance, and overall improving the system's reliability, manageability, controllability, and maintainability.

[0019] To make the above and other objects, features and advantages of this specification more apparent and understandable, preferred embodiments are described below in detail with reference to the accompanying drawings. Attached Figure Description

[0020] To more clearly illustrate the technical solutions in the embodiments or prior art of this specification, the drawings used in the description of the embodiments or prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of this specification. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0021] Figure 1 This specification illustrates a system architecture diagram provided in an embodiment. Figure 2 A flowchart illustrating the first method for VoIP-based communication with a CT network provided in the embodiments of this specification is shown. Figure 3 This document illustrates a flowchart of the process for converting a caller ID into a CT network-recognizable caller number, as provided in an embodiment of this specification. Figure 4 This document illustrates a flowchart of a process for standardizing and converting an original number to obtain a caller ID recognizable by a CT network, as provided in an embodiment of this specification. Figure 5 This document illustrates a second flowchart of a method for communicating with a CT network based on VoIP, as provided in an embodiment of this specification. Figure 6 This document illustrates a flowchart illustrating the process of forming a traceable tracing log by showing the metadata of associated Internet side links and cellular side links provided in the embodiments of this specification. Figure 7This document illustrates a third flowchart of a method for communicating with a CT network based on VoIP, as provided in an embodiment of this specification. Figure 8 This document illustrates a flowchart of a network handover mechanism triggered according to a handover strategy when the quality result is below a threshold, as provided in an embodiment of this specification. Figure 9 This document illustrates a fourth flowchart of a method for communicating with a CT network based on VoIP, as provided in an embodiment of this specification. Figure 10 A schematic diagram of the module structure of a device based on VoIP and CT network communication provided in an embodiment of this specification is shown. Figure 11 A schematic diagram of the structure of a computer device provided in an embodiment of this specification is shown.

[0022] Explanation of symbols in the attached drawings: 100. Call module; 200. Access module; 300. Conversion module; 400. Generation module; 500. Verification module; 600. Sending module; 700. Connection module; 1102. Computer equipment; 1104. Processor; 1106. Memory; 1108. Drive mechanism; 1110. Input / output module; 1112. Input device; 1114. Output device; 1116. Presentation device; 1118. Graphical user interface; 1120. Network interface; 1122. Communication link; 1124. Communication bus. Detailed Implementation

[0023] The technical solutions in the embodiments of this specification will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of this specification, and not all embodiments. Based on the embodiments in this specification, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the embodiments of this specification.

[0024] It should be noted that the user information (including but not limited to user device information, user personal information, etc.) and data (including but not limited to data used for analysis, data stored, data displayed, etc.) involved in this application are all information and data authorized by the user or fully authorized by all parties. Furthermore, the collection, storage, use, processing, transmission, provision, disclosure, and application of the relevant data all comply with the relevant laws, regulations, and standards of the relevant countries and regions, have taken necessary confidentiality measures, do not violate public order and good morals, and provide corresponding operation access points for users to choose to authorize or refuse.

[0025] Existing traditional voice communication technologies mainly rely on cellular networks or operator core network authentication, which suffers from problems such as poor communication stability, poor indoor signal quality, extremely limited wireless bandwidth resources, high base station construction costs, and high communication expenses.

[0026] VoIP (Voice over Internet Protocol) is a technology that transmits voice over IP networks. It uses the Internet or a dedicated IP network to encode, compress, and group voice signals to achieve real-time voice communication.

[0027] The widespread adoption of VoIP has posed significant challenges to traditional voice technologies, primarily in terms of communication costs and the services they can provide. However, VoIP employs a connectionless technology, resulting in unreliable quality of service, which has severely hampered its development.

[0028] To address the aforementioned issues, this specification provides a method for communication between VoIP and CT networks, which integrates traditional voice technology with emerging VoIP technology to achieve efficient and stable voice communication. Figure 1 This is a flowchart illustrating a method for VoIP and CT network communication based on an embodiment of this specification. This specification provides the operational steps of the method described in the embodiment or flowchart, but based on conventional or non-inventive methods, more or fewer operational steps may be included. The order of steps listed in the embodiment is merely one possible execution order among many and does not represent the only possible execution order. In actual system or device products, the methods shown in the embodiment or drawings can be executed sequentially or in parallel.

[0029] It should be noted that the terms "first," "second," etc., in the description, claims, and accompanying drawings of the embodiments in this specification are used to distinguish similar objects and are not necessarily used to describe a specific order or sequence. It should be understood that such data can be interchanged where appropriate so that the embodiments described herein can be implemented in orders other than those illustrated or described herein. Furthermore, the terms "comprising" and "having," and any variations thereof, are intended to cover non-exclusive inclusion; for example, a process, method, apparatus, product, or device that includes a series of steps or units is not necessarily limited to those steps or units explicitly listed, but may include other steps or units not explicitly listed or inherent to such processes, methods, products, or devices.

[0030] First, some terms that may be mentioned in the embodiments of this specification will be explained: VoIP (Voice over Internet Protocol) is a technology for transmitting voice over IP networks. It enables real-time voice communication by encoding, compressing and grouping voice signals and utilizing the Internet or a dedicated IP network.

[0031] CT network (Communication Technology Network) refers to traditional communication technology networks, mainly including the Public Switched Telephone Network (PSTN), mobile communication networks (2G / 3G / 4G / 5G cellular networks), etc., which carry traditional voice services and are based on circuit-switched or connection-oriented communication modes.

[0032] A landing gateway is an intermediate device used to connect IP networks and CT networks. It is responsible for converting VoIP IP voice data packets into a signal format compatible with CT networks (such as PSTN circuit-switched signals) to enable cross-network communication.

[0033] SIP (Session Initiation Protocol): A signaling protocol used to establish, modify, and terminate multimedia communication sessions (such as voice calls). Reference Figure 1 This is a system architecture diagram for an embodiment of this specification. The calling terminal can be a mobile phone, tablet, laptop, smartwatch, or other terminal device. It supports initiating call requests via WeChat mini-programs, standalone apps, or headless direct dialing mode. The calling terminal initiates calls via WiFi or other means and accesses the Internet through the inbound gateway. The landing gateway is a conversion device connecting the Internet (IP network) and the CT network, handling signaling conversion and voice codec conversion. The source tracing gateway is deployed on the landing gateway side for call source tracing. The backend management system is used for user management, call monitoring, balance management, and anomaly alarms.

[0034] Reference Figure 2 A method for communicating with a CT network based on VoIP may include the following steps: S101: The calling terminal initiates a call request on the Internet side, and the call request includes the calling identifier; S102: The calling terminal accesses the local gateway through the Internet server; S103: The Internet server or local gateway converts the caller ID into a caller number that can be recognized by the CT network; S104: The Internet server or the landing gateway generates Internet-side signaling based on the call request, and the Internet-side signaling encapsulates the calling number; S105: The landing gateway authenticates and verifies the Internet-side signaling. After the verification is successful, the Internet-side signaling is converted into telecommunications-side signaling. S106: The landing gateway sends the telecom-side signaling to the CT network; S107: The CT network receives the signaling from the telecommunications side and connects the called terminal to achieve fully traceable communication.

[0035] The calling terminal generates a call request containing the calling identifier (such as WeChat ID or phone card information). Of course, in addition to the calling identifier, the call request also contains other necessary information for communication, such as the called terminal's number, IP address, etc.

[0036] The calling terminal directly accesses the landing gateway through the Internet server, establishes a SIP session with the landing gateway, and negotiates voice codec formats (such as G.711, OPUS), transmission ports and other voice parameters. It does not need to be relayed through the cellular network core network and is compatible with multiple protocols such as S2b, IPSec, IKEv2, and SIP. The Internet server refers to the business backend or signaling server on the Internet side, such as the OTT (Over-The-Top) backend.

[0037] In particular, the calling terminal uses WiFi to access the Internet, extending to scenarios such as basements and mountainous areas, reducing core network relay links, lowering voice latency, improving call quality, and saving user data costs (WiFi is free or low-cost), making it especially suitable for long-duration call scenarios.

[0038] Reference Figure 3 The step of converting the caller ID into a caller number recognizable by the CT network further includes: S201: Obtain the original number associated with the calling identifier based on the mapping source priority from high to low; S202: Standardize and convert the original number to obtain a calling number that can be recognized by the CT network.

[0039] The caller ID can be a WeChat ID, and the original number associated with the caller ID can be a mobile phone number associated with the WeChat ID. The mobile phone number is stored in the core user binding information of the OTT backend (such as WeChat). From a technical perspective, this information is usually stored in encrypted form in the OTT backend user database (such as encrypted fields in MySQL / PostgreSQL, or encrypted key-value pairs in distributed KV storage such as Redis), and the association is controlled by user authorization (the user needs to actively bind the mobile phone number). The mobile phone number is retrieved from the user account system via internal API calls (such as HTTPS-based RESTful interfaces). The call process requires token authentication to ensure data security.

[0040] The caller ID can also be the terminal SIM card number (UE local number). In this case, the original number associated with the caller ID is the terminal SIM card number, which is the mobile phone number (MSISDN, Mobile Subscriber Integrated Services Digital Network Number) corresponding to the SIM card inserted in the calling terminal (mobile phone or tablet, etc.). This number is read through the operating system's Telephony interface (such as Android's TelephonyManager or iOS's CoreTelephony framework) and then uploaded to the backend server through the OTT backend SDK. If the terminal has multiple SIM cards, the terminal SIM card number needs to be determined based on user configuration (such as the default primary SIM card) or system policies (such as the SIM card used in the current data network).

[0041] When a calling terminal initiates a call request, the Call Control Module first queries the configuration center using the user ID to obtain the mapping source priority. If the user has not configured it, the default policy is triggered (usually "prioritize the WeChat account associated with the mobile phone number, and if there is no association, then use the terminal's SIM card number"). The parsing process needs to handle abnormal scenarios (such as when neither exists), which will trigger a call failure (because the CT network requires a valid calling number).

[0042] The goal of standardizing the original number is to convert it into the standardized E.164 number format required by the CT network—this is the only caller ID format that the CT network (PSTN, GSM, CDMA, etc.) can recognize, and it is also the basis for subsequent cross-network call routing.

[0043] The CT network operates based on a globally unified numbering scheme. Its calling numbers must conform to the ITU-T E.164 standard. The core specifications of this standard include: the number consists of "country code (CC) + domestic destination code (NDC) + subscriber number (SN)", with a total length not exceeding 15 digits (excluding the prefix "+"); it must carry the country code (e.g., "86" for China) and has no prefix "0" (e.g., a domestic mobile number 13800138000 needs to be converted to +8613800138000, but the CT network usually omits the "+" and stores it as 8613800138000).

[0044] In the embodiments described in this specification, reference is made to Figure 4 The process of standardizing and converting the original number to obtain a caller ID recognizable by the CT network further includes:

[0045] S301: Clean the original number to obtain the cleaned original number; S302: Match the location of the cleaned original number, and complete the country code according to the location to obtain the completed original number; S303: Perform format verification on the completed original number. If the verification passes, a caller ID that can be recognized by the CT network is obtained.

[0046] If the caller ID does not conform to the E.164 format, the landing gateway will directly reject the call (considering it an "illegal caller"), therefore "format conversion" is a mandatory requirement. The conversion process consists of three steps: "number cleaning → rule matching → format completion," and is usually implemented by the "Number Transformation Engine" in the OTT backend. 1) Number Cleansing The original phone number may have formatting issues (such as spaces, parentheses, or hyphens in the user input, or the country code being missing during terminal upload), so it needs to be "de-noiseed" first: Remove non-numeric characters: filter out symbols such as +, -, (,), and spaces (e.g., +86 138-0013-8000 → 8613800138000); Remove invalid prefixes: Remove common redundant prefixes in domestic mobile phone numbers (such as 013800138000→13800138000, because "0" is a domestic long-distance prefix, and the CT network needs to replace it with the country code).

[0047] 2) Country code matching and completion The original number may not include the country code (domestic users usually only save 13800138000 instead of +8613800138000), and it needs to be automatically completed through "Number Range Location Matching": The engine has a built-in global database of phone number segments (such as 130-139, 150-159, etc. for Chinese mobile phone numbers), and matches the country of origin by the first 3-4 digits of the original number. If a country of origin is matched (e.g., 138 matches "86" for China), then the country code is added before the number. If a match cannot be found (e.g., due to an abnormal number range), a "format error" alarm will be triggered, terminating the conversion process.

[0048] 3) E.164 format verification and correction After completing the country code, it is necessary to verify whether it conforms to the E.164 standard. Length check: The total number of digits (country code + number) must be ≤ 15 digits (e.g., a Chinese mobile number, after being completed, is 8613800138000, which is 13 digits and meets the requirement). Legality verification: Verify whether the number range is a valid number assigned by the operator through the location database (to prevent counterfeit numbers).

[0049] If the verification passes, the final "CT network identifiable caller ID" (e.g., 8613800138000) is generated; if the verification fails, "invalid caller ID" is returned and the call is terminated.

[0050] The converted caller ID number is not sent directly to the CT network, but requires protocol conversion and data delivery through the Access Gateway.

[0051] The landing gateway (usually a "softswitch gateway" or "IMS gateway") has a dual function: signaling protocol conversion. It converts Internet-side signaling (such as the SIP protocol, which is often used for VoIP calls) into CT network telecommunications signaling (such as the ISUP protocol, used for PSTN; MAP protocol, used for GSM). The calling number, as the "core field of Internet-side signaling", will be encapsulated in the signaling message and transmitted.

[0052] In addition, media stream conversion is required to convert the media stream from the Internet (such as OPUS-encoded speech) into encoding supported by the CT network (such as G.711).

[0053] Taking the most common scenario of "OTT VoIP (SIP) → CT network (ISUP)" as an example, the transmission process is as follows: 1) OTT side encapsulates the calling number The call control module at the OTT backend encapsulates the converted E.164 number (e.g., 8613800138000) into the From header field of the SIP signaling, in the following format: From:<sip:8613800138000@ott-domain.com> ;tag=xxx (where ott-domain.com is the domain name of the OTT platform). Meanwhile, to ensure the CT network recognizes the legitimacy of the calling number, "caller authentication information" (such as a token issued by the OTT backend) will be attached to the SIP signaling.

[0054] 2) Gateway parsing and protocol conversion After receiving the SIP signaling, the landing gateway first parses the From header field to extract the calling number, and then: Authentication and verification: Verify the authenticity of the calling number through the authentication interface of the OTT backend (to prevent spoofed numbers from making cross-network calls). Signaling conversion: Convert SIP signaling to ISUP signaling for the CT network, and encapsulate the calling party number into the "Calling Party Number" field of the ISUP signaling—this field is the core basis for CT network call routing and must strictly follow the E.164 format.

[0055] 3) Sending a call to the CT network The landing gateway sends the ISUP signaling, encapsulated with the caller's number, to the core network of the CT network (such as China Mobile's MSC or China Unicom's softswitch equipment) via a trunk line with the telecom operator (such as E1 / T1 digital trunk or IP trunk). Upon receiving the signal, the CT network routes the caller's information according to the country code and area code, ultimately connecting the called terminal.

[0056] In addition, the calling number must not only be able to be converted and transmitted, but also meet the CT network's requirement for "number authenticity," otherwise it will be judged as a "spam call" or a "fraudulent call." Therefore, two core technologies are required to ensure this: 1. Caller ID Authentication CT networks require that the calling number in cross-network calls be "traceable and verifiable" to prevent spoofing. Therefore, a "caller authentication mechanism" is established between the internet server, the terminal gateway, and the telecom operator. The internet server will perform "real-name authentication verification" on the mobile phone number associated with the user (interface with the operator to verify whether the mobile phone number belongs to the user).

[0057] When the gateway transmits a call to the telecom operator, it attaches a "digital signature from the Internet server". The operator verifies the signature using a preset public key to confirm that the caller ID number has not been tampered with.

[0058] 2. Compatibility and Adaptation: Supports multiple CT network standards. Different CT network standards (PSTN, GSM, CDMA, 5G SA) have slightly different requirements for the fields of the calling number (e.g., 5G SA network uses SIP signaling instead of ISUP), therefore the "number conversion engine" needs to have "multi-standard adaptation capability": Built-in "Network Standard - Signaling Field" mapping table (e.g., ISUP uses "Calling Party Number", 5GSIP uses the "P-Asserted-Identity" header field); The landing gateway dynamically adjusts the encapsulation position of the calling number according to the CT network standard it connects to, ensuring compatibility.

[0059] To achieve fully traceable communication, in the embodiments of this specification, refer to Figure 5 It also includes: S401: The grounding gateway connects to the traceability gateway; S402: The calling terminal connects to the traceability gateway; S403: The local gateway reports the metadata of the Internet-side link in real time throughout the entire communication process; S404: The calling terminal reports the metadata of the cellular side link in real time throughout the entire communication process; S405: The traceability gateway records and associates metadata of the Internet side link and the cellular side link throughout the entire communication process to form a traceable traceability log.

[0060] Among them, reference Figure 6 The metadata of the associated Internet sidelink and cellular sidelink, forming a traceable tracing log, further includes: S501: Select an association identifier from the metadata and associate the metadata of the Internet side link and the cellular side link through the association identifier; S502: Based on the associated identifier, the metadata is arranged in chronological order to form a traceability log.

[0061] By utilizing the dual-mode tracing module built into the intelligent fusion calling terminal, cross-domain correlation tracing of "cellular sidelink data - Internet sidelink data - user behavior characteristics" is achieved, specifically including: Data acquisition phase: Simultaneously capture metadata such as base station location, IMSI / IMEI identifier, and communication duration of the calling terminal's cellular mobile network, as well as metadata such as DNS resolution records, application access trajectory, and data transmission port of the Internet side access, to form a dual-link raw dataset; Association and fusion stage: Based on the association identifier, such as the unique hardware identifier of the terminal (e.g., UUID), an association engine is established to align the communication events on the cellular side with the application operations on the Internet side on the timeline, and generate a fusion traceability tag containing "communication scenario - network path - user operation", which is bound to the association identifier; Dynamic storage stage: Distributed encrypted ledger technology is used to store data in the database according to the sensitivity level (such as basic data like base station location being encrypted and desensitized, and user operation trajectory being encrypted), and cross-link joint queries are supported by calling terminal identifier, time interval, and data type. This specification's embodiments utilize hardware-level dual-mode collaboration to achieve full-scenario tracing of calling terminals in both mobile and internet environments, resolving the issue of gaps in cross-network behavior tracking. Employing dynamic anonymization and granular permission control (e.g., basic trajectory is traceable, sensitive operations require multi-level authorization), it maximizes user data privacy protection while maintaining tracing accuracy, complying with the hierarchical and categorized protection requirements of the Data Security Law.

[0062] In the embodiments described in this specification, reference is made to Figure 7 It also includes: S601: The calling terminal continuously collects at least one of the following parameters for the current call: signal strength, network latency, network jitter, packet loss rate, and network load traffic. S602: Preprocess at least one of the signal strength value, network latency, network jitter, packet loss rate, and network load traffic to obtain at least one type of data; S603: Convert the data format of the at least one data to obtain at least one data in a uniform format; S604: Use a machine learning model to perform weighted calculations on at least one data point in a uniform format to obtain a quality result; S605: When the quality result is lower than the threshold, the network handover mechanism is triggered according to the handover strategy.

[0063] Signal strength measurement: The calling terminal measures the power of the received WiFi signal through the Received Signal Strength Indicator (RSSI) sensor and obtains the signal strength value in real time.

[0064] Network latency and jitter are collected: The calling terminal measures latency by sending test data packets and recording the time difference between sending and receiving the packets. Simultaneously, jitter is evaluated by measuring the changes in latency multiple times. The expected interval is set as the "expected interval" between two consecutive packets (e.g., a 20 ms frame length corresponding to an RTP timestamp, or a transmission period T set by active probing). The actual interval is set as the "actual arrival interval" (time difference between arrivals) between two consecutive packets. Jitter is the fluctuation of the actual interval relative to the expected interval, typically characterized by exponential moving average (EWMA) or window statistics (standard deviation / percentile).

[0065] Packet loss rate: The calling terminal sends a series of test data packets, then counts the number of data packets that were not successfully received, and calculates the packet loss rate.

[0066] Collect network load traffic: Use traffic monitoring sensors to count the data traffic per unit time to determine the network load.

[0067] Data preprocessing: Denoising, normalization, and other preprocessing are performed on at least one of the received signal strength values, network latency, network jitter, packet loss rate, and network load traffic to convert different types of data into comparable formats for comprehensive evaluation.

[0068] Machine learning models such as the Relevance Vector Machine (RVM) algorithm are used to analyze the preprocessed data and calculate the network quality assessment results based on the weights of different parameters.

[0069] In the embodiments described in this specification, reference is made to Figure 8 The step of triggering a network handover mechanism according to a handover strategy when the quality result is below a threshold further includes: S701: When the quality result is below the threshold, the calling terminal obtains the available WiFi and cellular networks in the vicinity; S702: Sort WiFi networks and cellular networks according to network signal strength; S703: Connect the corresponding networks in order to obtain the currently connected network; S704: If the quality result corresponding to the network quality of the currently connected network is not lower than the threshold, then switch to the currently connected network; S705: If the quality result of the network currently being connected is lower than the threshold, then repeat the above steps of connecting the corresponding networks in order of sorting.

[0070] Depending on the specific communication application requirements, appropriate thresholds for evaluating network quality can be set. The obtained quality results are compared with the preset thresholds. If the quality falls below the threshold and communication needs cannot be met, a network switching mechanism is triggered: the calling terminal searches for available Wi-Fi and cellular networks in the vicinity and obtains relevant information such as Wi-Fi signal strength, cellular network signal strength, and frequency band. The switching strategy involves sorting available networks according to certain rules, for example, sorting Wi-Fi networks by signal strength from strongest to weakest, with cellular networks last. Connections are then attempted sequentially according to this order, prioritizing networks with strong signal strength and high network quality.

[0071] To achieve seamless handover, advance preparation is necessary: ​​before the current network quality deteriorates significantly, the calling terminal can perform pre-processing such as attaching and registering with the cellular network, ensuring rapid access when handover is required. Seamless handover technology utilizes collaborative technologies between multi-mode terminals and the internet side, such as dual connectivity (DC) technology, enabling the calling terminal to smoothly switch between Wi-Fi and cellular networks. During handover, mechanisms such as caching and retransmission ensure continuous data transmission, making network interruption virtually imperceptible to the user. At the application layer, optimized protocols and caching strategies minimize the impact on applications during handover.

[0072] In the embodiments described in this specification, reference is made to Figure 9 It also includes: S801: Grounding gateway connects to the backend management system; S802: The local gateway pushes communication data to the backend management system in real time; S803: The background management system visualizes and detects anomalies in the communication data.

[0073] The backend management system collects data such as terminal status (online / offline), call status (connected / hung up / busy), and user balance in real time, and provides a visual monitoring interface and anomaly detection alarms (such as insufficient balance or call interruption).

[0074] The embodiments in this manual achieve full-process management through a unified interface, improve operation and maintenance efficiency, support data visualization (such as call volume trend charts and terminal distribution heat maps), assist in operational decision-making, have short response time for abnormal alarms, and eliminate the need for manual inspection.

[0075] The embodiments in this specification are mainly applied to scenarios that require communication across IP networks and CT networks, including voice communication in areas with weak cellular signals (such as indoors or mountainous areas); and personal users making traditional phone calls via the Internet.

[0076] The following detailed description is provided through specific embodiments.

[0077] Specific Implementation Example 1: Traceable CT Network Calls on the WeChat Mini Program Front End Application scenario: In an indoor WiFi environment, users can call their family members' CT network mobile phone numbers (such as PSTN landlines or mobile phones) via WeChat mini-program. The caller's WeChat account is displayed as the mobile phone number bound to the user's WeChat account. The background monitors the call status in real time and supports call record tracing.

[0078] Implementation steps: Step S1: WeChat Mini Program access and call initiation; Execution subject: Front-end terminal (smartphone with WeChat mini program installed); Triggering conditions: The user opens the mini program, enters the called number (e.g., "010-12345678"), and selects the caller ID (default is "WeChat account bound mobile number 138xxxx8888"). Processing actions: The mini-program obtains the IP address (e.g., 192.168.1.100) and user identifier (WeChat OpenID: o6_bmjrPTlm6_2sgVt7hMZOPfL2M) of the terminal's WiFi access; it then generates a call request, including the caller ID (138xxxx8888), the called number (010-12345678), the IP address, and the timestamp. Processing result: The call request was sent to the local gateway via WiFi.

[0079] Step S2: SIP session establishment and gateway access; Executing entity: Landing gateway; Triggering condition: The landing gateway receives a call request; Processing steps: The landing gateway returns a SIP 100 Trying response, initiating session negotiation; both parties negotiate voice parameters via SDP protocol: codec format is G.711U, RTP transmission port is 5060; the landing gateway initiates a call to the CT network (performing circuit-switched signaling conversion for the called number 010-12345678); Processing result: SIP session established, RTP voice channel ready.

[0080] Step S3: End-to-end integration of WeChat and CT network communication tracing processing; Executing entity: Source tracing gateway; Triggering condition: After the SIP session is established, the calling terminal and the landing gateway synchronize session information with the tracing gateway; Processing action: The source tracing gateway records the mapping relationship of "calling terminal IP (192.168.1.100) - calling identifier (WeChat OpenID) - session ID (sip:138xxxx8888@192.168.1.200)"; During the call, a traceability log is generated every 30 seconds: {“Session ID”: “sip:xxx”, “Caller”: “138xxxx8888”, “Called”: “010-12345678”, “Duration”: “30s”, “Status”: “Calling”}; providing multiple traceability dimensions such as the caller’s terminal IP, caller number, and user identifier (WeChat ID).

[0081] Processing result: The traceability log is written to the database in real time, forming a traceable link.

[0082] Step S4: Intelligent mapping of calling number and call establishment; The core of this step is to achieve seamless integration between VoWiFi networks and traditional telecommunications networks through protocol conversion (SIP and ISUP), a rules engine (number mapping), and real-time media transmission (RTP / RTCP), ensuring correct display of caller ID and continuity of voice calls.

[0083] Executing entity: Landing gateway; Triggering condition: The CT network connects to the called terminal and requests the caller's number information; 1) The underlying triggering mechanism of the triggering conditions After the CT network (traditional telecommunications network) completes the paging and connection of the called terminal, it will send an ACM (Address Complete Message) or ANM (Answer Message) to the landing gateway through ISUP (ISDN User Part) signaling, which carries the signaling parameter of "requesting caller number information" (such as the Calling PartyNumber field in ISUP).

[0084] The landing gateway monitors the SIP-I (SIP with ISUP interworking) or BICC (Bearer Independent Call Control) protocol signaling streams in real time through the signaling parsing module. When it detects signaling containing the Calling PartyNumber Request flag, it triggers the calling number mapping process.

[0085] 2) Caller ID switching implementation 2.1) Storage and retrieval of user configuration data The gateway has a built-in user configuration database (such as Redis or MySQL) that stores user-preset mapping rules (such as "WeChat account bound to mobile phone number" or calling terminal card mobile phone number). The data format is key-value pairs: {WeChat user identifier: "wxid_xxxxxx", bound mobile phone number: "138xxxx8888", mapping switch: "enable"}. When mapping is triggered, the gateway queries the calling user's (WeChat client) configuration record through the configuration read interface (based on SQL or Redis protocol) to verify whether the mapping switch status is "enable".

[0086] 2.2) Number Conversion Rule Engine Rule Analysis: The engine has built-in number format conversion logic to map non-telecom network identifiers (such as WeChat IDs and VoIP accounts) to E.164 format telecom numbers (such as 138xxxx8888), ensuring compliance with CT network number specifications (Chinese mobile numbers are 11 digits, starting with 13 / 15 / 17 / 18 / 19). Validity Verification: Regular expressions are used to verify whether the mapped number is a valid mobile number. If invalid, default rules are triggered (such as using a virtual number assigned by the gateway).

[0087] 3) The underlying process of caller ID transmission via PSTN signaling 3.1) Signaling Protocol Conversion As a protocol conversion node between VoWiFi and PS TN / CT networks, the landing gateway needs to complete the SIP to ISUP protocol conversion: the gateway's signaling conversion module fills the mapped calling number (138xxxx8888) into the From header field of the SIP message (format: From: sip:138xxxx8888@gateway-domain.com>).

[0088] When converting to ISUP signaling, the number is written into the Calling Party Number information element (IE), which includes: Nature of Address Indicator: set to "national number"; Numbering Plan Indicator: set to "ISDN / Telephony Numbering Plan" (E.164); Presentation Indicator: set to "allowed" or "restricted" according to user configuration.

[0089] 3.2) Signaling Routing and Transmission The gateway sends ISUP signaling to the MSC (Mobile Switching Center) of the CT network via SS7 or the IP bearer network (SIGTRAN protocol stack). In the SIGTRAN protocol stack, M3UA (MTP3 User Adaptation Layer) is responsible for carrying MTP3 (Message Transfer Part 3) signaling on the IP network to ensure reliable transmission of ISUP messages (achieved through the coupling mechanism of the SCTP protocol).

[0090] 4) Underlying technologies for call establishment and voice transmission 4.1) Caller ID display triggered on the called terminal After receiving an ISUP signaling message containing the caller's number, the MSC of the CT network sends a SETUP message (GSM) or Paging Response (LTE) to the called terminal via the air interface (Um interface, GSM / LTE protocol), which carries the caller's number information. The protocol stack processing module of the called terminal parses the Calling Line Identification field in the message and displays the number "138xxxx8888" in the terminal's UI layer.

[0091] 4.2) Establishment and transmission of voice channel (RTP) Media Negotiation: The landing gateway and the CT network negotiate the voice encoding format (such as AMR-NB, G.711), RTP port (usually a dynamic port 10000-20000) and IP address via SDP (Session Description Protocol).

[0092] RTP Channel Establishment: After negotiation, the landing gateway establishes a bidirectional RTP stream between the calling party (VoWiFi side) and the called party (CT network side) via the UDP protocol. Voice data is transmitted in the form of RTP packets (each packet contains a 20ms voice frame, and the timestamp is generated by the sending end according to the sampling rate).

[0093] QoS Guarantee: The gateway enables DSCP (Differentiated Services Code Point) to mark RTP packets (such as EF marking to ensure low latency) and monitors packet loss rate and jitter in real time through RTCP (Real-time Transmission Control Protocol) to trigger retransmission or bitrate adjustment mechanisms.

[0094] 5) Technical verification of processing results At the signaling level: the landing gateway receives the ANM (Answer Message) signaling returned by the CT network, confirms that the called party has been answered, and the call status is switched to "active".

[0095] At the media level: Verify the continuity of the RTP stream (e.g., packet loss rate < 1%, jitter < 30ms) through RTCP's SR (Sender Report) and RR (Receiver Report) to ensure that the voice quality complies with the 3GPP TS 26.114 specification.

[0096] Step S5: Backend Management and Monitoring; Implementing entity: Backend management system; Triggering condition: After the call is established, the landing gateway pushes real-time data to the backend management system; Processing actions: The monitoring module displays the terminal's online status (WiFi), call status (connected), duration (1 minute 20 seconds), and the calling and called numbers; the balance management module deducts the call charges (based on duration, 0.1 yuan / minute), and the current balance is displayed as "98.5 yuan"; if the balance is less than 10 yuan, the alarm module triggers a pop-up reminder "Insufficient balance, please recharge"; Result: The backend management system updates data in real time, allowing administrators to view and intervene.

[0097] Step S6: Call End and Source Tracing Archiving Executing entities: calling terminal, landing gateway, or tracing gateway; Triggering condition: The user hangs up the call; Processing actions: The terminal sends a SIP BYE request, and the landing gateway releases session resources; the tracing gateway generates a final tracing report, which includes complete call records and IP-user mapping; the backend system updates the call records ("138xxxx8888→010-12345678, duration 1 minute 45 seconds, cost 0.3 yuan"). Processing result: The call ended, and all data was archived to the database for subsequent traceability queries.

[0098] Specific Implementation Example 2: Enterprise Call Center Application in Interface-Free Direct Dialing Mode (Extended Technical Solution) Application scenario: Enterprise customer service terminals (customized enterprise communication software interface, supporting VoIP direct dialing) dial customers' CT network numbers, and the caller ID displays the enterprise landline number (terminal phone card number). The background monitors the quality of customer service calls and supports call record tracing and auditing.

[0099] Implementation steps: Step S1: Direct dialing and call initiation via customer service terminal; Implementing entity: Customer service terminal; Triggering condition: The customer service terminal enters the called number (customer mobile number 139xxxx9999) and presses the dial button; Processing action: The customer service terminal obtains an IP address (10.0.0.5) through the enterprise WiFi, automatically reads the built-in phone card number (021-87654321) as the calling number; generates a call request, including the calling number (021-87654321), the called number (139xxxx9999), and the terminal identifier (SN: KF001). Processing result: The call request was sent to the local gateway via the enterprise intranet.

[0100] Step S2: SIP Session Interconnection with CT Network Executing entity: Landing gateway; Processing steps: The gateway establishes a SIP session with the client terminal and negotiates the encoding / decoding to OPUS (Optimized for Low Bandwidth Scenarios); it initiates a call to the mobile communication network (5G) and converts the calling party to circuit-switched format; Processing result: The called terminal (139xxxx9999) rang, displaying the incoming call "021-87654321".

[0101] Step S3: Enterprise-level traceability and access control Executing entity: Source tracing gateway; Processing actions: Associate terminal SN (KF001), customer service ID (007), and terminal IP (10.0.0.5); record voice metadata during the call (no voice call content, only timestamp and duration) to meet privacy compliance requirements; Processing result: The traceability information is only accessible to enterprise administrators.

[0102] Step S4: Back-end quality monitoring and billing Implementing entity: Backend management system; Actions taken: Monitor call quality (MOS value ≥ 3.5, network jitter ≤ 50ms); bill according to enterprise package (no charge during free time), and record customer service workload; Processing result: Call quality met standards, and customer service performance data was generated.

[0103] It enables customized interface design to support customer service operations, improves work efficiency, supports enterprise-level traceability for internal auditing, meets industry compliance requirements, displays the enterprise number to callers to enhance customer trust, ensures service level through quality monitoring, and automatically alerts when the MOS value is below standard.

[0104] This application provides users with access to relevant big data analysis (such as personal biometrics, identity data, consumption data, asset data, electronic terminal operation data, etc.), allowing users to choose to agree to or reject automated decision results; if the user chooses to reject, the process will proceed to the expert decision-making process.

[0105] Based on the method for VoIP-based communication with a CT network described above, this specification also provides a corresponding device for VoIP-based communication with a CT network. The device may include a system (including a distributed system), software (application), module, component, server, client, etc., using the method described in this specification, combined with necessary hardware implementation. Based on the same innovative concept, the devices in one or more embodiments provided in this specification are as described in the following embodiments. Since the implementation schemes and methods for solving the problem are similar, the implementation of specific devices in this specification can refer to the implementation of the aforementioned method, and repeated details will not be repeated. As used below, the terms "unit" or "module" can refer to a combination of software and / or hardware that implements a predetermined function. Although the device described in the following embodiments is preferably implemented in software, hardware implementation, or a combination of software and hardware, is also possible and contemplated.

[0106] Specifically, Figure 10 This is a schematic diagram of the module structure of one embodiment of a device based on VoIP and CT network communication provided in this specification. (Refer to...) Figure 10 As shown in the embodiments of this specification, a device based on VoIP and CT network communication includes: Call module 100: The calling terminal initiates a call request on the Internet side, and the call request includes the calling identifier; Access module 200: The calling terminal accesses the landing gateway through the Internet server. The conversion module 300, an internet server or a local gateway, converts the caller ID into a caller number that can be recognized by the CT network. The generation module 400 generates internet-side signaling based on the call request, and the internet-side signaling encapsulates the caller's number. The verification module 500 performs authentication verification on the Internet-side signaling at the landing gateway. After the verification is successful, the Internet-side signaling is converted into telecommunications-side signaling. The sending module 600 and the landing gateway send the telecom-side signaling to the CT network; The connection module 700 receives the telecommunications-side signaling from the CT network and connects the called terminal to achieve fully traceable communication.

[0107] Reference Figure 11As shown, based on the above-described method for VoIP and CT network communication, one embodiment of this specification also provides a computer device 1102, wherein the above method operates on the computer device 1102. The computer device 1102 may include one or more processors 1104, such as one or more central processing units (CPUs) or graphics processing units (GPUs), each processing unit implementing one or more hardware threads. The computer device 1102 may also include any memory 1106 for storing any kind of information such as code, settings, data, etc. In one specific embodiment, a computer program on the memory 1106 and executable on the processor 1104, when run by the processor 1104, can execute instructions according to the above method. Non-limitingly, for example, the memory 1106 may include any type of RAM, any type of ROM, flash memory, hard disk, optical disk, etc. More generally, any memory can use any technology to store information. Further, any memory can provide volatile or non-volatile retention of information. Further, any memory can represent a fixed or removable component of the computer device 1102. In one scenario, when processor 1104 executes associated instructions stored in any memory or combination of memories, computer device 1102 can perform any operation of the associated instructions. Computer device 1102 also includes one or more drive mechanisms 1108 for interacting with any memory, such as hard disk drive mechanisms, optical disk drive mechanisms, etc.

[0108] Computer device 1102 may further include an input / output module 1110 (I / O) for receiving various inputs (via input device 1112) and providing various outputs (via output device 1114). A specific output mechanism may include a presentation device 1116 and an associated graphical user interface 1118 (GUI). In other embodiments, the input / output module 1110 (I / O), input device 1112, and output device 1114 may be omitted, and the device may function solely as a computer device within a network. Computer device 1102 may also include one or more network interfaces 1120 for exchanging data with other devices via one or more communication links 1122. One or more communication buses 1124 couple the components described above together.

[0109] Communication link 1122 can be implemented in any way, such as via a local area network, a wide area network (e.g., the Internet), a point-to-point connection, or any combination thereof. Communication link 1122 may include any combination of hardwired links, wireless links, routers, gateway functions, name servers, etc., governed by any protocol or combination of protocols.

[0110] Corresponding to Figures 2-9 In addition to the methods described above, embodiments of this specification also provide a computer-readable storage medium storing a computer program that, when executed by a processor, performs the steps of the methods described above.

[0111] This specification also provides computer-readable instructions, wherein when a processor executes the instructions, the program therein causes the processor to perform the following... Figures 2 to 9 The method shown.

[0112] This specification also provides a computer program product, which, when executed by the processor of a computer device, performs the following... Figures 2 to 9 The method shown.

[0113] The computer program product described in this specification is a software product that mainly implements the methods described in this specification through a computer program.

[0114] It should be understood that in the various embodiments of this specification, the sequence number of each process does not imply the order of execution. The execution order of each process should be determined by its function and internal logic, and should not constitute any limitation on the implementation process of the embodiments of this specification.

[0115] It should also be understood that, in the embodiments of this specification, the term "and / or" is merely a description of the relationship between related objects, indicating that three relationships can exist. For example, A and / or B can represent: A existing alone, A and B existing simultaneously, and B existing alone. Furthermore, in the embodiments of this specification, the character " / " generally indicates that the preceding and following related objects have an "or" relationship.

[0116] Those skilled in the art will recognize that the units and algorithm steps of the various examples described in conjunction with the embodiments disclosed in this specification can be implemented in electronic hardware, computer software, or a combination of both. To clearly illustrate the interchangeability of hardware and software, the components and steps of each example have been generally described in terms of functionality in the foregoing description. Whether these functions are implemented in hardware or software depends on the specific application and design constraints of the technical solution. Those skilled in the art can use different methods to implement the described functions for each specific application, but such implementations should not be considered beyond the scope of the embodiments in this specification.

[0117] Those skilled in the art will clearly understand that, for the sake of convenience and brevity, the specific working processes of the systems, devices, and units described above can be referred to the corresponding processes in the foregoing method embodiments, and will not be repeated here.

[0118] In the several embodiments provided in this specification, it should be understood that the disclosed systems, apparatuses, and methods can be implemented in other ways. For example, the apparatus embodiments described above are merely illustrative; for instance, the division of units is only a logical functional division, and in actual implementation, there may be other division methods. For example, multiple units or components may be combined or integrated into another system, or some features may be ignored or not executed. Furthermore, the couplings or direct couplings or communication connections shown or discussed may be indirect couplings or communication connections through some interfaces, devices, or units, or they may be electrical, mechanical, or other forms of connection.

[0119] The units described as separate components may or may not be physically separate. The components shown as units may or may not be physical units; that is, they may be located in one place or distributed across multiple network units. Some or all of the units can be selected to achieve the purpose of the embodiments described in this specification, depending on actual needs.

[0120] Furthermore, the functional units in the various embodiments of this specification can be integrated into one processing unit, or each unit can exist physically separately, or two or more units can be integrated into one unit. The integrated unit can be implemented in hardware or as a software functional unit.

[0121] If the integrated unit is implemented as a software functional unit and sold or used as an independent product, it can be stored in a computer-readable storage medium. Based on this understanding, the technical solutions of the embodiments of this specification, in essence, or the parts that contribute to the prior art, or all or part of the technical solutions, can be embodied in the form of a software product. This computer software product is stored in a storage medium and includes several instructions to cause a computer device (which may be a personal computer, server, or network device, etc.) to execute all or part of the steps of the methods described in the various embodiments of this specification. The aforementioned storage medium includes various media capable of storing program code, such as USB flash drives, portable hard drives, read-only memory (ROM), random access memory (RAM), magnetic disks, or optical disks.

[0122] This specification uses specific embodiments to illustrate the principles and implementation methods of the embodiments. The above description of the embodiments is only for the purpose of helping to understand the methods and core ideas of the embodiments in this specification. At the same time, for those skilled in the art, there will be changes in the specific implementation methods and application scope based on the ideas of the embodiments in this specification. Therefore, the content of this specification should not be construed as a limitation on the embodiments in this specification.

Claims

1. A method for communication between VoIP and CT networks, characterized in that, include: The calling terminal initiates a call request on the Internet side, and the call request includes the calling identifier; The calling terminal connects to the local gateway via an internet server; The internet server or local gateway converts the caller ID into a caller number that can be recognized by the CT network; The Internet server or the landing gateway generates Internet-side signaling based on the call request, and the Internet-side signaling encapsulates the caller's number; The landing gateway authenticates and verifies the Internet-side signaling. After successful verification, the Internet-side signaling is converted into telecommunications-side signaling. The landing gateway sends the telecom-side signaling to the CT network; The CT network receives the signaling from the telecommunications side and connects the called terminal to achieve fully traceable communication.

2. The method according to claim 1, characterized in that, Also includes: The local gateway connects to the source tracing gateway; The calling terminal connects to the source tracing gateway; Throughout the entire communication process, the local gateway reports the metadata of the internet-side link in real time. Throughout the entire communication process, the calling terminal reports the metadata of the cellular side link in real time. Throughout the entire communication process, the traceability gateway records and associates metadata from both the internet-side link and the cellular-side link, forming a traceable traceability log.

3. The method according to claim 2, characterized in that, The metadata of the associated Internet sidelink and cellular sidelink, forming a traceable tracing log, further includes: Select an association identifier from the metadata, and associate the metadata of the Internet side link and the cellular side link through the association identifier; Based on the associated identifier, the metadata is arranged in chronological order to form a traceability log.

4. The method according to claim 1, characterized in that, The step of converting the caller ID into a caller number recognizable by the CT network further includes: Based on the mapping source priority from high to low, obtain the original number associated with the calling identifier; The original number is standardized and converted to obtain a calling number that can be recognized by the CT network.

5. The method according to claim 4, characterized in that, The step of standardizing the original number to obtain a caller ID recognizable by the CT network further includes: The original number is cleaned to obtain the cleaned original number; Match the location of the cleaned original number, and complete the country code based on the location to obtain the completed original number; The format of the completed original number is validated, and the caller ID that can be recognized by the CT network is obtained after the validation passes.

6. The method according to claim 1, characterized in that, Also includes: The calling terminal continuously collects at least one of the following for the current call: signal strength, network latency, network jitter, packet loss rate, and network load traffic. At least one of the signal strength value, network latency, network jitter, packet loss rate, and network load traffic is preprocessed to obtain at least one type of data. The data at least one piece of data is converted to a uniform format to obtain at least one piece of data. A quality result is obtained by using a machine learning model to perform weighted calculations on at least one dataset in a uniform format; When the quality result is below the threshold, the network handover mechanism is triggered according to the handover strategy.

7. The method according to claim 6, characterized in that, The mechanism for triggering network handover according to the handover strategy when the quality result is below the threshold further includes: When the quality result is below a threshold, the calling terminal obtains the available WiFi and cellular networks in the vicinity. Sort WiFi networks and cellular networks according to their signal strength; Connect the corresponding networks in order of sorting to obtain the currently connected network; If the quality result of the currently connected network is not lower than the threshold, then switch to the currently connected network; If the quality result of the network currently being connected is lower than the threshold, then repeat the above steps of connecting the corresponding networks in order of sorting.

8. The method according to claim 1, characterized in that, Also includes: The local gateway connects to the backend management system; The local gateway pushes communication data to the backend management system in real time. The backend management system provides a visual representation and detects anomalies in the communication data.

9. The method according to claim 1, characterized in that, Also includes: The calling terminal accesses the Internet via WiFi.

10. A device based on VoIP and CT network communication, characterized in that, The device includes: The calling module allows the calling terminal to initiate a call request on the internet side, and the call request includes the calling identifier. The access module allows the calling terminal to access the local gateway via an internet server. The conversion module, either an internet server or a local gateway, converts the caller ID into a caller number recognizable by the CT network. The generation module, the Internet server or the landing gateway, generates Internet-side signaling based on the call request, and the Internet-side signaling encapsulates the caller's number; The verification module, the landing gateway, performs authentication verification on the Internet-side signaling, and after the verification is successful, the Internet-side signaling is converted into telecommunications-side signaling; The sending module and the landing gateway send the telecom-side signaling to the CT network; The connection module receives the telecom-side signaling from the CT network and connects the called terminal to achieve fully traceable communication.

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