Voice scheduling exchange system with dual-network integration

Abstract

The invention, which relates to the technical field of communication, discloses a dual-network-integrated voice scheduling exchange system comprising the following steps: collecting network state information in real time, receiving and analyzing a voice scheduling request, and carrying out division processing on both communication parties; calculating a communication path through an intelligent routing algorithm according to the network state information and the division processing; performing network quality evaluation on the communication path calculated by the intelligent routing algorithm, and selecting an optimal communication path; selecting a voice compression parameter to compress the voice signal, and generating a compressed voice data packet; sending the compressed voice data packet to a receiver through the optimal communication path; and the receiver receives the compressed voice data packet, decompresses the compressed voice data packet to obtain an original voice signal, and outputs clear voice information. According to the invention, through integrating the information collection module, the intelligent routing module, the network evaluation module, the voice compression module, the voice transmission module, the terminal receiving module and the like, the overall safety of the dual-network integrated voice scheduling exchange system and the voice scheduling efficiency and quality are effectively improved.

Description

Technical Field

[0001] The invention relates to the field of communication technology, and in particular to a dual-network integrated voice dispatching switching system. Background Art

[0002] With the rapid development of information technology, especially the advancement of mobile communications and Internet technologies, the demand for voice communications is growing, and higher requirements are placed on the reliability, stability and efficiency of voice dispatching and switching systems. In recent years, dual-network fusion technology has gradually become a research hotspot, which aims to integrate different types of network resources, optimize resource allocation, and improve service quality. In addition, with the application of emerging technologies such as cloud computing and edge computing, the development of intelligent routing algorithms has also greatly promoted the advancement of dual-network fusion technology, making it possible to achieve efficient and stable voice dispatching and switching in complex network environments.

[0003] Although dual-network fusion technology has made some progress, it still faces some challenges in the actual deployment process. First, when the network status changes frequently, how to accurately and quickly collect and parse network status information, and how to dynamically adjust the routing strategy based on this information, is a major problem faced by the existing technology. Secondly, due to the performance differences between different networks, the existing voice scheduling and switching systems often fail to fully take this into account, resulting in a significant decrease in voice quality when network conditions are poor. Finally, for the compression and decompression of voice data packets in a cross-network environment, the existing technology lacks an effective mechanism to ensure the balance between transmission efficiency and decoding quality, which directly affects the user experience. Therefore, it is particularly important to develop a dual-network fusion voice scheduling and switching system that can overcome the above-mentioned defects. Summary of the invention

[0004] In view of the above existing problems, the present invention is proposed.

[0005] Therefore, the present invention provides a dual-network integrated voice dispatching and switching system to solve the problems of inaccurate network status perception, non-optimal routing selection and unstable voice quality in the voice dispatching and switching system under the dual-network integrated environment.

[0006] In order to solve the above technical problems, the present invention provides the following technical solutions:

[0007] In a first aspect, an embodiment of the present invention provides a dual-network integrated voice scheduling switching system, which includes:

[0008] Information collection module: collects network status information in real time, receives and analyzes voice dispatch requests, and divides and processes the communicating parties;

[0009] Intelligent routing module: calculates the communication path through intelligent routing algorithm according to network status information and division processing;

[0010] Network evaluation module: It evaluates the network quality of the communication path calculated by the intelligent routing algorithm and selects the optimal communication path;

[0011] Voice compression module: selects voice compression parameters to compress voice signals and generate compressed voice data packets;

[0012] Voice transmission module: sends compressed voice data packets to the receiver through the optimal communication path;

[0013] Terminal receiving module: The receiver receives the compressed voice data packet and decompresses it to obtain the original voice signal, and outputs clear voice information.

[0014] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the network status information includes broadband utilization, packet loss rate and delay;

[0015] The division process includes extracting the identity information and priority of both communicating parties;

[0016] The speech compression parameters include compression algorithm and compression ratio.

[0017] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the real-time collection of network status information, receiving and parsing of voice dispatching requests, and division processing of the communicating parties are performed, the specific operation steps are as follows:

[0018] Combine intelligent routing with program-controlled switching systems and IP soft switching systems to build a dual-network integrated voice dispatching system. Place monitoring probes on the switches to collect network status information in real time and transmit it to the data processing unit for real-time processing.

[0019] Encode standard format data and parse the encoded data to obtain the three main network influencing factors: real-time bandwidth utilization, packet loss rate, and delay;

[0020] Receive and parse the scheduling request from the user terminal, collect the identification information of the scheduling request initiator and the receiver, and the scheduling priority information;

[0021] By querying user data, verify whether the identities of the communicating parties are authenticated users. If so, network resources are allocated; otherwise, communication is rejected.

[0022] The corresponding network resources are allocated to the scheduling request based on the network status information and the identification information and priority information of the communicating parties.

[0023] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the real-time collection of network status information and the real-time transmission to the data processing unit for real-time processing, the specific operation steps are as follows:

[0024] The monitoring probes deployed on the switches monitor network traffic in real time and collect key network status information;

[0025] The UDP protocol is used as the transmission protocol to transmit the collected network status information to the data processing unit in real time;

[0026] The data processing unit uses the stream processing framework to process the received network status data in real time, extract key indicators, and perform data cleaning and formatting.

[0027] As a preferred solution of the dual-network integrated voice dispatch switching system of the present invention, wherein: the receiving and parsing of the dispatch request from the user terminal, collecting the identification information of the dispatch request initiator and the receiver and the dispatch priority information, the specific operation steps are as follows:

[0028] The voice dispatch exchange system receives a dispatch request initiated by the initiator user terminal through the network, and uses a message parser to parse the data field in the request;

[0029] The identification information of the scheduling request initiator and the receiver, as well as the priority of the scheduling request, are extracted from the parsed result.

[0030] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the communication path is calculated by an intelligent routing algorithm according to the network status information and the division processing, and the specific operation steps are as follows:

[0031] Based on the network status information collected by the information collection module and the identification information and priority of the communicating parties, the quality of the communication path is scored. The expression is as follows:

[0032]

[0033] Among them, S path (P) is the communication path scoring function that quantifies the quality of the communication path P, i is the index variable, B i is the bandwidth utilization of the i-th node in the communication path, L i is the packet loss rate of the ith node, D i is the percentage representation of the delay of the i-th node, n is the number of nodes in the communication path, κ B is the bandwidth utilization adjustment factor, κ L is the packet loss rate adjustment factor, κ D is the delay adjustment factor;

[0034] Define two communication levels, high priority and standard priority:

[0035] For high priority communication, set κ B =0.25,κ L =1.5,κ D =2;

[0036] For standard priority communication, set κ B =0.5,κ L =1,κ D =1.5;

[0037] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the network quality evaluation is performed on the communication path calculated by the intelligent routing algorithm to select the optimal communication path, and the specific operation steps are as follows:

[0038] Based on the network status information collected by the information module and the communication path calculated by the intelligent routing module, a network quality evaluation function is defined, and the expression is as follows:

[0039]

[0040] Among them, Q(P) is the comprehensive consideration of the communication path scoring function S path (P) and the network status information of each node in the communication path and the communication priority are used to quantitatively evaluate the network quality of the communication path. pri is the priority coefficient of the communication, α is a positive constant that adjusts the influence of the priority coefficient on the final evaluation score;

[0041] The communication path with the highest Q(P) value among all paths and exceeding the corresponding priority threshold is selected as the optimal communication path; the intelligent routing sends instructions to the selected program-controlled switching system and IP soft switching system to establish the communication path.

[0042] Through simulation tests, we observe the effect of communication path selection under different α values. According to the test results and the feedback from users on communication quality, we continuously adjust the α value until the optimal balance point is reached.

[0043] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the voice compression parameter is selected to compress the voice signal to generate a compressed voice data packet, and the specific operation steps are as follows:

[0044] Adopting adaptive compression algorithm based on perceptual quality to compress voice data;

[0045] According to the output of the network quality evaluation function Q(P), the adaptive compression parameter adjustment function is defined, and the expression is as follows:

[0046]

[0047] Among them, C(P) refers to dynamically adjusting the compression ratio by considering the output of the network quality evaluation function, and β is a threshold that distinguishes the quality of the network;

[0048] The compression ratio R is adjusted by the output of C(P), and the expression is defined as:

[0049] R=R min +C(P)·(R max -R min );

[0050] Among them, R max To pre-set the maximum compression ratio, R min To preset the minimum compression ratio.

[0051] As a preferred solution of the dual-network integrated voice dispatching switching system of the present invention, wherein: the compressed voice data packet is sent to the receiving party through the optimal communication path, and the specific operation steps are as follows:

[0052] Encapsulate the compressed voice data packets according to the standard RTP header format;

[0053] Add CRC-32 checksum to each compressed voice data packet, and set a timer to detect unconfirmed data packets and resend them;

[0054] Initialize the initial sending rate and congestion window size, record the unconfirmed data packets detected by the timer to obtain the packet loss rate, and record the time from the data packet group being sent to the receiving confirmation, calculate the average round-trip time (RTT), adjust the congestion window size according to the performance of the packet loss rate and RTT, and dynamically adjust the sending rate;

[0055] A forward error correction mechanism is used to append redundant data packets to each compressed voice data packet sent;

[0056] Confirm the IP address and port number information of the receiving terminal and send the data packet via UDP protocol;

[0057] An ICMP Ping request is sent every 10 seconds to detect the status of the communication path. If a problem is detected in the communication path, the intelligent routing module recalculates the optimal communication path.

[0058] As a preferred solution of the dual-network integrated voice dispatching and switching system of the present invention, wherein: the receiving party receives the compressed voice data packet and decompresses it to obtain the original voice signal, and outputs clear voice information, the specific steps are as follows:

[0059] The receiver receives the compressed voice data packet and uses the CRC-32 checksum to verify the integrity of the compressed voice data packet;

[0060] If an error is found, the information in the redundant data packet is used for recovery. If the lost data packet cannot be recovered, the voice transmission module is requested to retransmit and the error log is recorded.

[0061] After verification is completed, the header information of the compressed voice data packet is separated and the voice data packets are sorted according to the sequence number and time stamp;

[0062] The payload part of the voice data packet is retained, and the data is decompressed through the inverse process of the same compression algorithm of the voice compression module to restore the original voice information;

[0063] According to the adaptive compression algorithm based on perceptual quality, the voice signal is reconstructed and output through the receiving terminal audio device to complete the communication.

[0064] The beneficial effects of the present invention are as follows: by authenticating both communicating parties, it is ensured that only authenticated users can communicate, thereby improving the overall security of the system; by collecting network status information in real time, timely processing of scheduling requests is guaranteed; the intelligent routing algorithm takes into account factors such as broadband utilization, packet loss rate and delay, so that the selected communication path is more efficient and stable; by setting different priority adjustment factors, the best path is selected according to the urgency of the communication, thereby ensuring the smooth progress of important communications; through network quality evaluation, the selected communication path can minimize voice delay and packet loss, and improve the clarity of voice calls; the compression ratio is dynamically adjusted according to the network quality evaluation result, so that the system can adapt to various network environments and maintain good voice quality; through CRC-32 check code, timer detection and forward error correction mechanism, the integrity of the data packet and the stability of transmission are guaranteed. BRIEF DESCRIPTION OF THE DRAWINGS

[0065] In order to more clearly illustrate the technical solutions of the embodiments of the present invention, the accompanying drawings required for use in the description of the embodiments will be briefly introduced below. Obviously, the accompanying drawings described below are only some embodiments of the present invention. For ordinary technicians in this field, other accompanying drawings can be obtained based on these accompanying drawings without paying creative work.

[0066] Figure 1 This is a flow chart of the dual-network integrated voice dispatching switching system in Example 1.

[0067] Figure 2 This is a module diagram of the dual-network integrated voice scheduling switching system in Example 1. DETAILED DESCRIPTION

[0068] In order to make the above-mentioned objects, features and advantages of the present invention more obvious and easy to understand, the specific implementation methods of the present invention are described in detail below in conjunction with the accompanying drawings.

[0069] In the following description, many specific details are set forth to facilitate a full understanding of the present invention, but the present invention may also be implemented in other ways different from those described herein, and those skilled in the art may make similar generalizations without violating the connotation of the present invention. Therefore, the present invention is not limited to the specific embodiments disclosed below.

[0070] Secondly, the term "one embodiment" or "embodiment" as used herein refers to a specific feature, structure, or characteristic that may be included in at least one implementation of the present invention. The term "in one embodiment" that appears in different places in this specification does not necessarily refer to the same embodiment, nor does it refer to a separate or selective embodiment that is mutually exclusive with other embodiments.

[0071] Example 1, reference Figure 1 and Figure 2 , which is the first embodiment of the present invention, and which provides a dual-network integrated voice dispatching switching system, comprising the following steps:

[0072] S1. Collect network status information in real time, receive and parse voice dispatch requests, and divide and process the communicating parties.

[0073] Furthermore, the network status information refers to bandwidth utilization, packet loss rate and latency.

[0074] Division processing refers to extracting the identification information and priority of the communicating parties.

[0075] Combine intelligent routing with program-controlled switching systems and IP soft switching systems to build a dual-network integrated voice dispatching system. Deploy monitoring probes on switches to detect network status in real time, and collect the following raw network status data through the Simple Network Management Protocol (SNMP): the number of inbound and outbound bytes of the interface, the total number of packets sent and received, and the round-trip time (RTT) of data in the network;

[0076] The UDP protocol is used as the transmission protocol to transmit the collected original network status data to the data processing unit in real time;

[0077] The data processing unit pre-processes the original network status data through a specific algorithm hash table, identifies and removes duplicate data records, sets thresholds to filter out data points that are obviously beyond the normal range, converts all data fields into consistent integer data types, and uses the stream processing framework Apache Storm to process the pre-processed data in real time, extract key indicators, and ensure that the final network status information is more accurate and reliable, which is convenient for subsequent analysis and decision-making;

[0078] Preferably, the JSON (JavaScript Object Notation) format is used, which is more concise, easier to read and write than other formats and widely supports multiple programming languages. The standard format data is mapped to the JSON object, and the JSON parser is used to parse the encoded JSON format data to obtain the three main network influencing factors of real-time bandwidth utilization, packet loss rate and delay;

[0079] The voice dispatch system receives a dispatch request initiated by the initiator user terminal through the network, uses a message parser to parse the data field in the request, and extracts the identification information of the dispatch request initiator and the receiver, as well as the priority of the dispatch request from the parsed result;

[0080] By querying the user registration database, verify whether the unique user ID of the communicating parties is a registered and authenticated user. If so, network resources are allocated. Otherwise, communication is rejected and communication abnormality logs are recorded for later inspection;

[0081] The corresponding network resources are allocated to the scheduling request based on the network status information, identification information such as the unique user ID of the communicating parties, user level and priority information such as the communication urgency.

[0082] It should be noted that the use of the UDP protocol provides low-overhead services without the need to establish connections, which is suitable for real-time transmission requirements in voice dispatching, can ensure the accuracy and timeliness of data, and improve the overall efficiency of the dual-network integrated voice dispatching switching system. Other protocols such as the TCP protocol are far inferior to the UDP protocol in terms of transmission speed and cannot meet real-time requirements. It verifies user identities and ensures that only authenticated users can use services, thereby enhancing the security of the dual-network integrated voice dispatching switching system and preventing unauthorized access and malicious attacks.

[0083] S2. Calculate the communication path through an intelligent routing algorithm based on network status information and partition processing.

[0084] Furthermore, based on the network status information collected by the information collection module and the identification information and priority of the communicating parties, the quality of the communication path is scored, and the expression is as follows:

[0085]

[0086] Among them, S path (P) is the communication path scoring function that quantifies the quality of the communication path P. The threshold is calculated to be (2.6, 12.9). This interval ensures that different levels of communication quality can be properly evaluated and can distinguish the network quality levels required for different communication needs;

[0087] where i indexes the variable, B i is the bandwidth utilization of the i-th node in the communication path, L i is the packet loss rate of the ith node, D i is the percentage representation of the delay of the i-th node, n is the number of nodes in the communication path, κ B is the bandwidth utilization adjustment factor, κ L is the packet loss rate adjustment factor, κ D is the delay adjustment factor, which is used to adjust the impact of bandwidth utilization, packet loss rate and delay on path scoring. By dynamically adjusting the factor, the communication path selection strategy is automatically adjusted according to different communication requirements, reflecting the S path (P) Flexibility and adaptability to different application scenarios,

[0088] When S path When the (P) value is [8.0, 12.9), it means that the communication path quality is very high, which means that the communication path is very reliable and has excellent performance. It is suitable for communication in critical and emergency situations. Even if the network quality is not ideal, a program-controlled switching system can be selected to ensure the communication quality. When the score is [4.0, 8.0), it means that the communication path quality is good and can meet all communication needs. When the score is (2.6, 4.0), it means that the path quality is poor, the bandwidth utilization rate is high, the packet loss rate is large, and the delay is high. It is not suitable for high-quality and high-importance communications;

[0089] Furthermore, two communication levels are defined: communication between high-level users, communication in emergency situations is high priority, and communication between ordinary users, general business communication is standard priority:

[0090] For high priority communication, set κ B =0.25,κ L =1.5,κ D =2;

[0091] For standard priority communication, set κ B =0.5,κ L =1,κ D =1.5;

[0092] Integrate communication priority directly into the communication path scoring function instead of through external weights, which simplifies the path selection process while also ensuring the network resources required for high-priority communications;

[0093] When S path When the (P) score is low, the program-controlled switching system is selected. Thanks to its stable, reliable and secure characteristics, it can maintain good communication quality when the communication path quality is poor. path(P) When the score is high, choose an IP soft switch system with flexibility, scalability and low cost;

[0094] It should be noted that this intelligent dynamic routing algorithm can dynamically adjust the path selection strategy according to real-time network status information and communication needs to ensure the quality and reliability of communication services. Through a sophisticated scoring mechanism and flexible communication system selection, it not only improves communication efficiency, but also significantly improves user experience, especially when dealing with emergency and critical communication tasks.

[0095] S3. Select the optimal communication path by evaluating the network quality of the communication path calculated by the intelligent routing algorithm.

[0096] Furthermore, based on the network status information collected by the information module and the optimal communication path calculated by the intelligent routing module, a network quality evaluation function is defined, and the expression is as follows:

[0097]

[0098] Among them, Q(P) is the comprehensive consideration of the communication path scoring function S path (P), the network status information of each node in the communication path and the communication priority, a function to quantitatively evaluate the network quality of the communication path, P pri is the priority coefficient of the communication. For standard priority communication, P pri =0, for high priority communication P pri =1, α is a positive constant that adjusts the influence of the priority coefficient on the final evaluation score, and its value range is [3,7]. The smaller α is, the smaller the influence of the priority coefficient on the final evaluation score is, and the larger α is, the greater the influence of the priority coefficient on the final evaluation score is;

[0099] The standard priority Q(P) threshold is set to 6, the high priority Q(P) threshold is set to 17, and the communication path with the highest Q(P) value and exceeding the corresponding priority threshold is selected as the optimal communication path. The intelligent routing sends instructions to the selected program-controlled switching system and IP soft switching system to establish the communication path.

[0100] When the output of standard priority Q(P) is less than 2.6 and the output of high priority Q(P) is less than 6, it is determined that the quality of all path networks does not meet the most basic communication requirements, and the intelligent routing module recalculates;

[0101] The simulation is conducted under different conditions of broadband utilization, packet loss rate, delay and different communication priorities. First, α is set to 3. In each operating scenario, the optimal communication path is selected, and the Q(P) value of the selected path and the communication quality including communication voice clarity and delay are recorded. The path selection effect and communication quality under different α values ​​are analyzed. If the communication quality is better when the α value is smaller, the α value is gradually reduced, otherwise the α value is gradually increased. Through this iterative optimization, the α value with the best communication quality is found.

[0102] It should be noted that, through the comprehensive evaluation function Q(P), it is ensured that the path with the highest network quality is selected, thereby improving the stability, clarity and response speed of communication. By dynamically adjusting the value of α, the optimal communication path is selected according to different communication requirements and network conditions, thereby enhancing the flexibility and adaptability of the dual-network integrated voice scheduling and switching system. By setting thresholds and priorities, the communication path selection logic is simplified, unnecessary calculations are reduced, and the working efficiency of the dual-network integrated voice scheduling and switching system is improved, ensuring that good communication conditions can be obtained under various communication requirements.

[0103] S4. Select a voice compression parameter to compress the voice signal and generate a compressed voice data packet.

[0104] Furthermore, an adaptive compression algorithm based on perceptual quality is used to compress the speech data. This perceptual quality adaptive compression algorithm is a method for efficiently compressing speech data by combining perceptual linear prediction (PLP) and improved codebook excited linear prediction (CELP).

[0105] This adaptive compression algorithm based on perceptual quality combines the advantages of PLP and CELP. PLP provides a feature extraction method that is more in line with human hearing characteristics, making the compressed speech more natural and more suitable for human hearing. CELP is an efficient coding method that can effectively reduce the bit rate of speech data while maintaining high speech quality.

[0106] According to the output of the network quality evaluation function Q(P), the adaptive compression parameter adjustment function is defined, and the expression is as follows:

[0107]

[0108] Among them, C(P) refers to dynamically adjusting the compression ratio by considering the output of the network quality evaluation function, β is a positive constant that distinguishes the quality of the network, and its value range is [10,40]. The larger the value, the better the network quality.

[0109] By calculating the extreme value of C(P) above, we get the value range to be (0,1).

[0110] The compression ratio R is adjusted by the output of C(P), and the expression is defined as:

[0111] R=R min +C(P)·(R max -R min );

[0112] Among them, R max To pre-set the maximum compression ratio, R max = 10, that is, when the network quality is very poor, the compression ratio is increased to adapt to the network limitations and ensure smooth communication. min The minimum compression ratio R is set in advance min =1, that is, when the network conditions are very good, reducing the compression ratio can improve the communication sound quality and enhance the user experience;

[0113] The PLP technology is used to extract key features from the original speech signal, and the improved CELP method is used to encode the extracted features to generate a compressed data stream. The compression ratio is dynamically adjusted according to the output of Q(P) and changes in network status.

[0114] It should be noted that by combining PLP and improved CELP technology, the compressed speech is more natural and closer to the original timbre, which improves the user experience. By adaptively adjusting the compression parameters C(P), the compression ratio can be automatically adjusted according to changes in network status, thereby enhancing the adaptability of communication, flexibly responding to changes in network status, and ensuring smooth communication and voice quality.

[0115] S5. Send the compressed voice data packet to the receiver via the optimal communication path.

[0116] Furthermore, the compressed voice data packets are encapsulated according to the standard RTP header format;

[0117] The polynomial used by the CRC-32 standard is selected as the generating polynomial. The transmitted data is regarded as a binary number. 32 zeros are appended to the end of the data to match the 32-bit CRC value. The generated polynomial is used to perform binary division on the extended data. The remainder is the CRC-32 value. The CRC-32 value is appended to the end of the compressed voice data packet. At the same time, a timer is set to detect data packets that have not been confirmed by the receiver. If they are not confirmed for more than 0.5 seconds, the data packets are considered lost and resent.

[0118] Define the initial sending rate as 100kbps, set the congestion window to 10 packets, record the unconfirmed packets detected by the timer to get the packet loss rate, record the time from when the packet group is sent to when it is confirmed, and calculate the average round-trip time (RTT). If the packet loss rate exceeds 5% and the RTT exceeds 200 milliseconds, reduce the congestion window, otherwise increase the congestion window.

[0119] The forward error correction mechanism is used to add redundant data packets to each compressed voice data packet sent. The redundant data packets contain enough information for the receiving end to reconstruct the complete voice data even if some data packets are lost, thus ensuring smooth communication.

[0120] Confirm the IP address and port number information of the receiving terminal and send the data packet via UDP protocol;

[0121] An ICMP Ping request is sent every 10 seconds to detect the status of the communication path. If the response time exceeds 200ms, the path network status is considered poor, and the intelligent routing module and network evaluation module are informed to recalculate and evaluate the optimal communication path.

[0122] UDP has the advantage of being more suitable for voice scheduling where high transmission rates are required, but it cannot guarantee the integrity of data packets. Therefore, by adding methods such as setting a timer to detect packet loss rate and RTT to adjust the congestion window and forward error correction mechanism, the high transmission rate of UDP can be guaranteed while ensuring the integrity of data packets.

[0123] It should be noted that the integrity and accuracy of the data packet are ensured through the CRC-32 checksum and fast timeout retransmission mechanism, which improves the reliability of communication. The forward error correction mechanism reduces the number of retransmissions and improves communication efficiency. By regularly monitoring the path status and feeding back to the intelligent routing module and network evaluation module, it is ensured that the system can dynamically adjust the optimal communication path according to the latest network conditions and enhance network path management.

[0124] S6. The receiving party receives the compressed voice data packet and decompresses it to obtain the original voice signal.

[0125] Furthermore, the receiving party receives the compressed voice data packet containing the CRC-32 value, removes the CRC-32 value therein, recalculates the CRC-32 value in the same way as the CRC-32 value is generated, and compares the received CRC-32 value with the recalculated CRC-32 value. If the two are the same, the compressed voice data packet is intact. If the two are different, it indicates that an error occurred in the transmission of the compressed voice data packet.

[0126] If an error is found, the information in the redundant data packet is used for recovery. If the lost data packet cannot be recovered, the voice transmission module is requested to retransmit and the error log is recorded.

[0127] After the verification is completed, the header information of the compressed voice data packet is separated, a data list is created to store the compressed voice data packet, and the sequence number and timestamp are extracted. If the compressed voice data packet is sent in the sequence number order, the compressed voice data packet can be sorted according to the sequence number. If the sequence number is incomplete and is not sent in the sequence number order, it is further sorted according to the timestamp. If some compressed voice data packets arrive too late due to high latency, it can be decided whether to discard these compressed voice data packets based on the timestamp to avoid obvious delays and echoes during playback;

[0128] Based on the sorted compressed speech data stream, the LPC coefficient (speech feature extracted by the PLP method during the compression process) and the codebook index of the excitation signal are decoded from the compressed data, and the corresponding excitation signal sample is searched from the fixed codebook and the adaptive codebook according to the codebook index. The decoded LPC system is used to construct a synthesis filter, and the spectrum of the speech signal is generated by processing the excitation signal through the synthesis filter;

[0129] The inverse discrete cosine transform (IDCT) performs an inverse transform on the generated spectrum and converts it back into a time domain signal to obtain a reconstructed speech signal, which is output through the receiving terminal audio device to complete the communication.

[0130] It should be noted that the CRC-32 verification and error recovery mechanism ensures that the received data packets are complete and correct, thereby improving the voice quality. The retransmission request and error log records improve the stability and reliability of the system. The sorting and reorganization mechanism ensures that the voice signals are played in the correct order to avoid voice distortion caused by disorder.

[0131] In summary, the present invention performs identity authentication on both communicating parties to ensure that only authenticated users can communicate, thereby improving the overall security of the system, and ensures timely processing of scheduling requests by collecting network status information in real time; the intelligent routing algorithm takes into account factors such as broadband utilization, packet loss rate and delay, making the selected communication path more efficient and stable, and by setting different priority adjustment factors, the best path is selected according to the urgency of the communication, thereby ensuring the smooth progress of important communications; through network quality evaluation, the selected communication path can minimize voice delay and packet loss, and improve the clarity of voice calls; the compression ratio is dynamically adjusted according to the network quality evaluation result, so that the system can adapt to various network environments and maintain good voice quality; through CRC-32 check code, timer detection and forward error correction mechanism, the integrity of the data packet and the stability of transmission are guaranteed.

[0132] Example 2, referring to Table 1, is the second example of the present invention. To further verify the advancement of the present invention, experimental simulation data of a dual-network integrated voice dispatching switching system is provided.

[0133] First, two typical working environments were selected in the experiment: office environment and complex network environment. The former adopted a stable enterprise-level network, and the latter simulated unstable network conditions with high packet loss and delay. A dual-network integrated voice dispatching and switching system (hereinafter referred to as "this system") and a voice dispatching and switching system under a traditional single network architecture were deployed, and multiple tests were carried out to ensure the reliability of the data.

[0134] During the experiment, monitoring probes were deployed to monitor network traffic in real time and collect key network status information such as broadband utilization, packet loss rate, and latency. The collected network status information was transmitted to the processing unit in real time using the UDP protocol for real-time processing, extraction of key indicators, and data cleaning and formatting.

[0135] Furthermore, a simulation tool is used to simulate the user terminal initiating a scheduling request, collect the identification information of the scheduling request initiator and receiver and the scheduling priority information, and authenticate each request. Only when the scheduling request initiator and receiver are authenticated users can a valid scheduling request be initiated;

[0136] Furthermore, after receiving the scheduling request, the communication path is calculated through the intelligent routing algorithm according to the current network status and the identification information and priority of the communicating parties, and the quality score S corresponding to each communication path is given. path (P), further quantify the network quality of these paths, consider the quality score of the communication path, the network status information of each node in the communication path, and the priority of the two communicating parties, and select the communication path with the highest Q(P) output value and exceeding the corresponding priority threshold among all communication paths as the optimal communication path;

[0137] Furthermore, according to the quantitative evaluation value of the communication path network quality, an adaptive compression parameter adjustment function C(P) is generated to dynamically adjust the compression ratio R, and an adaptive compression algorithm based on perceptual quality is used to compress the voice data. After confirming the IP address and port number information of the receiving terminal, the compressed data packet is sent through the UDP protocol;

[0138] Finally, after receiving the compressed data, the receiver verifies the integrity and decompresses the compressed data packet to obtain the original voice signal.

[0139] The details are shown in Table 1 below:

[0140] Table 1 Performance evaluation data experimental record of dual network fusion voice dispatching switching system

[0141] Experimental subjects Broadband utilization Packet loss rate Delay (ms) MOS score Office Environment 0.85 0.02 30 4.3 Complex environment 0.75 0.05 60 3.8 Prior art A 0.8 0.03 35 3.9 Prior Art B 0.7 0.07 70 3.2

[0142] By comparing the bandwidth utilization, packet loss rate, delay and MOS scores of different experimental subjects in the experiment, it is obvious that;

[0143] In an office environment, the broadband utilization rate of the present invention reaches 0.85, while the existing technologies A and B are 0.8 and 0.7 respectively. This shows that the present system can more efficiently utilize bandwidth resources under stable network conditions. In a complex network environment, although the broadband utilization rate drops to 0.75, it is still higher than 0.7 of the existing technology B, showing the good adaptability of the present system in an unstable network environment.

[0144] The packet loss rate in the office environment is 0.02, which is much lower than 0.03 of the prior art A and 0.07 of the prior art B, which shows that the system can significantly reduce the data loss problem during data transmission;

[0145] The delay in the office environment is 30ms, while the existing technologies A and B are 35ms and more than 35ms respectively, which shows that the system can provide high-quality voice services while maintaining low delay. In a complex network environment, although the delay increases to 60ms, it is still lower than the 70ms of the existing technology B, which reflects the advantages of the system in dealing with complex network environments;

[0146] MOS score is a standard for measuring the quality of voice calls. The higher the value, the better the call quality. It can be clearly seen from the table that this system has significant advantages in ensuring call quality, whether in an office environment or a complex environment.

[0147] In summary, the present invention ensures that high-quality voice service experience can be provided to users even under poor network conditions through the effective application of a series of innovative intelligent routing algorithms and the dynamic adjustment of adaptive compression parameters. It is not only innovative in theory, but also demonstrates communication effects that are superior to existing technologies in practice, providing strong technical support for the field of communication technology.

[0148] It should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention rather than to limit it. Although the present invention has been described in detail with reference to the preferred embodiments, those skilled in the art should understand that the technical solutions of the present invention may be modified or replaced by equivalents without departing from the spirit and scope of the technical solutions of the present invention, which should all be included in the scope of the claims of the present invention.

Claims

1. A dual-network integrated voice dispatching switching system, characterized in that: include, Information collection module: collects network status information in real time, receives and analyzes voice dispatch requests, and divides and processes the communicating parties; Intelligent routing module: calculates the communication path through intelligent routing algorithm according to network status information and division processing; Network evaluation module: It evaluates the network quality of the communication path calculated by the intelligent routing algorithm and selects the optimal communication path; Voice compression module: selects voice compression parameters to compress voice signals and generate compressed voice data packets; Voice transmission module: sends compressed voice data packets to the receiver through the optimal communication path; Terminal receiving module: The receiver receives the compressed voice data packet and decompresses it to obtain the original voice signal, and outputs clear voice information.

2. The dual-network integrated voice dispatching switching system according to claim 1, characterized in that: The network status information includes bandwidth utilization, packet loss rate and delay; The division process includes extracting the identity information and priority of both communicating parties; The speech compression parameters include compression algorithm and compression ratio.

3. The dual-network integrated voice dispatching switching system as claimed in claim 2, characterized in that: The real-time collection of network status information, receiving and parsing of voice dispatch requests, and division and processing of the communicating parties are as follows: Combine intelligent routing with program-controlled switching systems and IP soft switching systems to build a dual-network integrated voice dispatching system. Place monitoring probes on the switches to collect network status information in real time and transmit it to the data processing unit for real-time processing. Encode standard format data and parse the encoded data to obtain the three main network influencing factors: real-time bandwidth utilization, packet loss rate, and delay; Receive and parse the scheduling request from the user terminal, collect the identification information of the scheduling request initiator and the receiver, and the scheduling priority information; By querying user data, verify whether the identities of the communicating parties are authenticated users. If so, network resources are allocated; otherwise, communication is rejected. The corresponding network resources are allocated to the scheduling request based on the network status information and the identification information and priority information of the communicating parties.

4. The dual-network integrated voice dispatching switching system as claimed in claim 3, characterized in that: The real-time collection of network status information and the transmission to the data processing unit for real-time processing are as follows: The monitoring probes deployed on the switches monitor network traffic in real time and collect key network status information; The UDP protocol is used as the transmission protocol to transmit the collected network status information to the data processing unit in real time; The data processing unit uses the stream processing framework to process the received network status data in real time, extract key indicators, and perform data cleaning and formatting.

5. The dual-network integrated voice dispatching switching system as claimed in claim 4, characterized in that: The receiving and parsing of the scheduling request from the user terminal, collecting the identification information of the scheduling request initiator and the receiver and the scheduling priority information, the specific operation steps are as follows: The voice dispatch exchange system receives a dispatch request initiated by the initiator user terminal through the network, and uses a message parser to parse the data field in the request; The identification information of the scheduling request initiator and the receiver, as well as the priority of the scheduling request, are extracted from the parsed result.

6. The dual-network integrated voice dispatching switching system as claimed in claim 5, characterized in that: The communication path is calculated by intelligent routing algorithm according to the network status information and division processing. The specific operation steps are as follows: Based on the network status information collected by the information collection module and the identification information and priority of the communicating parties, the quality of the communication path is scored. The expression is as follows: Among them, S path (P) is the communication path scoring function that quantifies the quality of the communication path P, i is the index variable, B i is the bandwidth utilization of the i-th node in the communication path, L i is the packet loss rate of the ith node, D i is the percentage representation of the delay of the i-th node, n is the number of nodes in the communication path, κ B is the bandwidth utilization adjustment factor, κ L is the packet loss rate adjustment factor, κ D is the delay adjustment factor; Define two communication levels, high priority and standard priority: For high priority communication, set κ B =0.25,κ L =1.5,κ D =2; For standard priority communication, set κ B =0.5,κ L =1,κ D =1.

5.

7. The dual-network integrated voice dispatching switching system as claimed in claim 6, characterized in that: The network quality evaluation is performed on the communication path calculated by the intelligent routing algorithm to select the optimal communication path. The specific operation steps are as follows: Based on the network status information collected by the information module and the communication path calculated by the intelligent routing module, a network quality evaluation function is defined, and the expression is as follows: Among them, Q(P) is the comprehensive consideration of the communication path scoring function S path (P) and the network status information of each node in the communication path and the communication priority are used to quantitatively evaluate the network quality of the communication path. pri is the priority coefficient of the communication, α is a positive constant that adjusts the influence of the priority coefficient on the final evaluation score; The communication path with the highest Q(P) value among all paths and exceeding the corresponding priority threshold is selected as the optimal communication path; the intelligent routing sends instructions to the selected program-controlled switching system and IP soft switching system to establish the communication path. Through simulation tests, we observe the effect of communication path selection under different α values. According to the test results and the feedback from users on communication quality, we continuously adjust the α value until the optimal balance point is reached.

8. The dual-network integrated voice dispatching switching system as claimed in claim 7, characterized in that: The voice compression parameters are selected to compress the voice signal to generate a compressed voice data packet. The specific operation steps are as follows: Adopting adaptive compression algorithm based on perceptual quality to compress voice data; According to the output of the network quality evaluation function Q(P), the adaptive compression parameter adjustment function is defined, and the expression is as follows: Among them, C(P) refers to dynamically adjusting the compression ratio by considering the output of the network quality evaluation function, and β is a threshold that distinguishes the quality of the network; The compression ratio R is adjusted by the output of C(P), and the expression is defined as: R=R min +C(P)·(R max -R min ); Among them, R max To pre-set the maximum compression ratio, R min To preset the minimum compression ratio.

9. The dual-network integrated voice dispatching switching system as claimed in claim 8, characterized in that: The compressed voice data packet is sent to the receiving party through the optimal communication path, and the specific operation steps are as follows: Encapsulate the compressed voice data packets according to the standard RTP header format; Add CRC-32 checksum to each compressed voice data packet, and set a timer to detect unconfirmed data packets and resend them; Initialize the initial sending rate and congestion window size, record the unconfirmed data packets detected by the timer to obtain the packet loss rate, and record the time from the data packet group being sent to the receiving confirmation, calculate the average round-trip time (RTT), adjust the congestion window size according to the performance of the packet loss rate and RTT, and dynamically adjust the sending rate; A forward error correction mechanism is used to append redundant data packets to each compressed voice data packet sent; Confirm the IP address and port number information of the receiving terminal and send the data packet through the UDP protocol; An ICMP Ping request is sent every 10 seconds to detect the status of the communication path. If a problem is detected in the communication path, the intelligent routing module recalculates the optimal communication path.

10. The dual-network integrated voice dispatching switching system according to claim 9, characterized in that: The receiving party receives the compressed voice data packet and decompresses it to obtain the original voice signal, and outputs clear voice information. The specific steps are as follows: The receiver receives the compressed voice data packet and uses the CRC-32 checksum to verify the integrity of the compressed voice data packet; If an error is found, the information in the redundant data packet is used for recovery. If the lost data packet cannot be recovered, the voice transmission module is requested to retransmit and the error log is recorded. After verification is completed, the header information of the compressed voice data packet is separated and the voice data packets are sorted according to the sequence number and time stamp; The payload part of the voice data packet is retained, and the data is decompressed through the inverse process of the same compression algorithm of the voice compression module to restore the original voice information; According to the adaptive compression algorithm based on perceptual quality, the voice signal is reconstructed and output through the receiving terminal audio device to complete the communication.

Images