Communication method and device of VOIP equipment, equipment and storage medium

Abstract

The invention provides a communication method and device of VOIP equipment, equipment and a storage medium. The method comprises the steps of collecting and encoding voice data, detecting encoded data features and a network environment state, generating a feature vector set and a channel stability index, and sending the feature vector set and the channel stability index to an access point; receiving network resources and scheduling information allocated by the access point, and processing the encoded voice data packet to form a VOIP data packet to be transmitted; sending the VOIP data packet to other VOIP equipment through a wireless channel, coordinating transmission resources with other equipment, sharing the wireless channel, and adjusting transmission parameters according to the interference condition; and monitoring the transmission performance of the sent data packet, performing call quality evaluation, and recording the transmission performance and a quality evaluation result. According to the invention, through dynamic adjustment of a transmission strategy, cooperative transmission and comprehensive quality evaluation, the performance and user experience of VOIP communication in a high-density wireless network environment are significantly improved.

Description

Technical Field

[0001] The present invention relates to the field of communication technologies, and in particular, to a communication method, apparatus, device, and storage medium for a VOIP device. Background Art

[0002] With the development of network technologies, voice communication based on the IP network (i.e., VOIP) has been widely applied to various communication scenarios. However, in a high-density wireless network environment, VOIP communication faces many challenges. Traditional VOIP communication methods usually adopt fixed transmission strategies and cannot dynamically adjust transmission parameters according to changes in the network environment, resulting in difficulty in ensuring the quality of voice communication when the network is congested or the channel quality fluctuates. Especially in the scenario where multiple devices simultaneously conduct VOIP communication, the competition for transmission resources is intense, easily causing delays, jitters, and packet losses of voice data packets, seriously affecting the user's call experience.

[0003] In the prior art, most VOIP devices only focus on their own transmission requirements, lack a cooperation mechanism with other devices, and cannot effectively share and coordinate network resources. At the same time, the monitoring and evaluation of call quality by traditional VOIP communication methods are relatively simple, cannot comprehensively reflect the actual call experience, and are difficult to provide an effective basis for optimizing transmission strategies. These problems are particularly prominent in a high-density wireless network environment, severely restricting the capacity and quality of VOIP communication systems. Summary of the Invention

[0004] The main objective of the present invention is to solve the technical problems that existing VOIP devices cannot dynamically adapt to network changes, lack a cooperation mechanism between devices, and have an incomplete call quality evaluation in a high-density wireless network environment; In a first aspect of the present invention, a communication method for a VOIP device is provided. The communication method for the VOIP device includes: Collect and encode the voice data of the VOIP device, detect the characteristics of the encoded data and the network environment status, generate a feature vector set and a channel stability index, and send the feature vector set and the channel stability index to an access point; Receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, process the encoded voice data packets to form VOIP data packets to be transmitted; Send the VOIP data packets to other VOIP devices in the network through a wireless channel, and at the same time, coordinate transmission resources with other VOIP devices in the network, share the wireless channel, and adjust transmission parameters according to the interference situation during the transmission process; Monitor the transmission performance of the sent VOIP data packets, conduct call quality evaluation, and record the quality evaluation results of the transmission performance and call quality evaluation.

[0005] Optionally, in the first implementation manner of the first aspect of the present invention, the collecting and encoding the voice data of the VOIP device, detecting the characteristics of the encoded data and the network environment state, and generating a feature vector set and a channel stability index include: Collect the voice data of the VOIP device and perform encoding processing using a codec to obtain encoded voice data, and analyze the encoded voice data to detect the codec type, the packet generation period, and the packet size, and obtain encoding feature parameters; Analyze the voice activity state, background noise level, and call mode in the encoded voice data to generate a load prediction vector, and measure the modulation and coding scheme level, maximum transmit power, and battery state supported by the VOIP device to generate a terminal capability vector; Periodically measure the received signal strength indication, signal-to-noise ratio, and current modulation and coding scheme of the wireless channel, calculate the channel fluctuation situation, and obtain a channel stability index; Combine the encoding feature parameters, the load prediction vector, the terminal capability vector, and the channel stability index into a feature vector set.

[0006] Optionally, in the second implementation manner of the first aspect of the present invention, the receiving the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, and processing the encoded voice data packet to form a VOIP data packet to be transmitted includes: Receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, and analyze the encoded voice data packet according to the channel stability index and the network resources to calculate a packet aggregation degree adjustment coefficient; Perform dynamic aggregation processing on multiple encoded voice data packets according to the aggregation degree adjustment coefficient to form an aggregated data packet, and in the aggregation process, perform real-time analysis on the encoded voice data packet to identify key voice data containing burst sounds or significantly changed volume segments, and assign a priority mark to the key voice data; Select a transmission modulation and coding scheme for the aggregated data packet according to the scheduling information and the channel stability index, and combine the aggregated data packet, the priority mark, and the transmission modulation and coding scheme to form a VOIP data packet to be transmitted.

[0007] Optionally, in the third implementation manner of the first aspect of the present invention, the analyzing the encoded voice data packet according to the channel stability index and the network resources to calculate a packet aggregation degree adjustment coefficient includes: Perform feature analysis on the encoded voice data packets, extract features such as packet size, frame interval, and payload distribution, and generate voice data feature values; Process the channel stability index, combine historical data of received signal strength indication and signal-to-noise ratio, and calculate the channel reliability value of the current transmission window; Perform resource analysis on the network resources, detect available bandwidth, time slot allocation, and current network congestion status, and obtain resource utilization parameters; According to the voice data feature values, the channel reliability value, and the resource utilization parameters, query a preset aggregation parameter table, select the corresponding basic aggregation value, and correct it in combination with the current voice activity state to obtain a packet aggregation degree adjustment coefficient.

[0008] Optionally, in the fourth implementation manner of the first aspect of the present invention, the sending the VOIP data packet to other VOIP devices in the network through a wireless channel, coordinating transmission resources with other VOIP devices in the network, sharing the wireless channel, and adjusting transmission parameters according to interference conditions during the transmission process include: Receive a cooperative group mapping table from the access point, and identify the affiliated cooperative group and cooperative group transmission rules according to the network resources and scheduling information allocated by the access point through the cooperative group mapping table; According to the cooperative group transmission rules determined by the cooperative group mapping table, listen for trigger frames sent by the access point, and send the VOIP data packet to other VOIP devices in the network through the wireless channel according to the indication in the trigger frame; According to the cooperative group transmission rules, monitor and analyze the interference situation in the network, generate an interference map, and based on the interference map, adjust the transmission parameters of the VOIP data packet, including adjusting transmission power, modulation and coding scheme, and transmission timing, to adapt to the interference situation of the wireless channel.

[0009] Optionally, in the fifth implementation manner of the first aspect of the present invention, the receiving a cooperative group mapping table from the access point and identifying the affiliated cooperative group and cooperative group transmission rules according to the network resources and scheduling information allocated by the access point through the cooperative group mapping table include: Receive the cooperative group mapping table sent by the access point through a management frame, and perform parsing processing on the network resources allocated by the access point to identify the resource configuration data structure of the resource unit of the network resources; Perform timing analysis on the scheduling information provided by the access point, parse the trigger period, target wake-up time, contention window parameters, and transmission opportunity limit of the trigger frame, determine the transmission timing and duration constraints of the VOIP device, and construct a time slot access rule table; Perform a byte-by-byte comparison operation between the physical address of the VOIP device and the member address list in the collaboration group mapping table, and determine the collaboration group identifier of the collaboration group to which the VOIP device belongs through an address matching algorithm; Retrieve the corresponding transmission parameters from the collaboration group mapping table according to the collaboration group identifier, and construct a collaboration group transmission rule in combination with the resource configuration data structure and the time slot access rule table.

[0010] Optionally, in the sixth implementation manner of the first aspect of the present invention, monitoring the transmission performance of the sent VOIP data packet and performing call quality evaluation, and recording the quality evaluation results of the transmission performance and call quality evaluation include: Perform real-time tracking on the end-to-end delay, network jitter, and packet loss rate of the sent VOIP data packet, analyze the transmission control parameters, including the retransmission rate, sequence number interval, and round-trip time, evaluate the efficiency of the transmission process, and obtain the transmission performance; Analyze the decoded voice data, evaluate the voice clarity, voice continuity, and mean opinion score, perform call quality evaluation, and obtain the quality evaluation results of the call quality evaluation; Record the transmission performance and the quality evaluation results into the transmission performance reference database under different network environments.

[0011] The second aspect of the present invention provides a communication device for a VOIP device, and the communication device for the VOIP device includes: A data processing module, configured to collect and encode the voice data of the VOIP device, detect the characteristics of the encoded data and the network environment state, generate a feature vector set and a channel stability index, and send the feature vector set and the channel stability index to an access point; A data packet generation module, configured to receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, process the encoded voice data packets, and form VOIP data packets to be transmitted; A collaborative transmission module, configured to send the VOIP data packets to other VOIP devices in the network through a wireless channel, coordinate transmission resources with other VOIP devices in the network, share the wireless channel, and adjust transmission parameters according to the interference situation during the transmission process; A quality evaluation module, configured to monitor the transmission performance of the sent VOIP data packets and perform call quality evaluation, and record the quality evaluation results of the transmission performance and call quality evaluation.

[0012] In a third aspect of the present invention, a communication device for a VOIP device is provided, including: a memory and at least one processor, wherein instructions are stored in the memory, and the memory and the at least one processor are interconnected by a line; the at least one processor invokes the instructions in the memory to cause the communication device of the VOIP device to execute the steps of the above-mentioned communication method of the VOIP device.

[0013] In a fourth aspect of the present invention, a computer-readable storage medium is provided, in which instructions are stored. When it runs on a computer, it causes the computer to execute the steps of the above-mentioned communication method of the VOIP device.

[0014] The above-mentioned communication method, device, equipment and storage medium of the VOIP device collect and encode voice data, detect the characteristics of the encoded data and the network environment status, generate a feature vector set and a channel stability index and send them to the access point; receive the network resources and scheduling information allocated by the access point, process the encoded voice data packets to form VOIP data packets to be transmitted; send the VOIP data packets to other VOIP devices through a wireless channel, coordinate the transmission resources with other devices, share the wireless channel, and adjust the transmission parameters according to the interference situation; monitor the transmission performance of the sent data packets, conduct call quality evaluation, and record the transmission performance and quality evaluation results. The present invention significantly improves the performance and user experience of VOIP communication in a high-density wireless network environment through dynamic adjustment of the transmission strategy, cooperative transmission and comprehensive quality evaluation.

[0015] Other features and advantages of the present invention will be described in the subsequent specification, and part of them will become obvious from the specification or be understood by implementing the present invention. The objectives and other advantages of the present invention are achieved and obtained by the structures specifically pointed out in the specification, claims and drawings.

[0016] To make the above-mentioned objectives, features and advantages of the present invention more obvious and understandable, the following specific preferred embodiments are given and described in detail in conjunction with the accompanying drawings. Brief Description of the Drawings

[0017] Figure 1 It is a schematic diagram of the first embodiment of the communication method of the VOIP device in the embodiment of the present invention; Figure 2 It is a schematic diagram of an embodiment of the communication device of the VOIP device in the embodiment of the present invention; Figure 3 It is a schematic diagram of an embodiment of the communication equipment of the VOIP device in the embodiment of the present invention. Detailed Description of the Embodiment

[0018] To make the objectives, technical solutions, and advantages of the embodiments of the present invention clearer, the technical solutions of the present invention will be clearly and completely described below with reference to the accompanying drawings. Apparently, the described embodiments are some, but not all, of the embodiments of the present invention. All other embodiments obtained by those of ordinary skill in the art based on the embodiments of the present invention without creative efforts shall fall within the protection scope of the present invention.

[0019] As used in the embodiments of the present invention, the terms "including" and "having" and any variations thereof are intended to cover non-exclusive inclusion. For example, a process, method, system, product, or device that includes a series of steps or units is not limited to the listed steps or units, but optionally further includes other steps or units not listed, or optionally further includes other steps or units inherent to these processes, methods, products, or devices.

[0020] To facilitate the understanding of this embodiment, first, a communication method of a VOIP device disclosed in the embodiments of the present invention will be introduced in detail. As Figure 1 shown, the method includes the following steps: 101. Collect and encode the voice data of the VOIP device, detect the characteristics of the encoded data and the network environment status, generate a feature vector set and a channel stability index, and send the feature vector set and the channel stability index to the access point; In an embodiment of the present invention, the step of collecting and encoding the voice data of the VOIP device, detecting the characteristics of the encoded data and the network environment status, and generating a feature vector set and a channel stability index includes: Collect the voice data of the VOIP device and perform encoding processing using a codec to obtain the encoded voice data, and analyze the encoded voice data to detect the codec type, packet generation period, and packet size, and obtain encoding feature parameters; analyze the voice activity status, background noise level, and call mode in the encoded voice data to generate a load prediction vector, and measure the modulation and coding scheme level, maximum transmit power, and battery status supported by the VOIP device to generate a terminal capability vector; periodically measure the received signal strength indication, signal-to-noise ratio, and current modulation and coding scheme of the wireless channel, calculate the channel fluctuation condition, and obtain a channel stability index; combine the encoding feature parameters, the load prediction vector, the terminal capability vector, and the channel stability index into a feature vector set.

[0021] Specifically, the VOIP device collects the user's voice signal through a built-in microphone array. During the collection process, a sampling rate of 16 kHz and a quantization precision of 16 bits are adopted to ensure the high fidelity of the original voice signal. The collected original voice data is preprocessed through echo cancellation, noise suppression, and automatic gain control, etc., and then sent to the codec for encoding. The codec selects the corresponding encoding algorithm according to the current configuration, such as G.711, G.729, or a low-bitrate encoder, and compresses and encodes the preprocessed voice data to generate encoded voice data. The encoding process follows the standard process of the corresponding codec. For example, G.711 uses non-linear quantization pulse code modulation technology, generating 8000 samples per second, with each sample represented by 8 bits, forming a constant bitrate stream of 64 kbps; while G.729 uses conjugate structure algebraic code excited linear prediction technology to compress the voice signal to 8 kbps. After encoding, the VOIP device deeply analyzes the encoded voice data to detect the type of codec currently in use, the packet generation period, and the packet size. The codec type is determined by checking the encoding header identification bit; the packet generation period is obtained by calculating the time interval between two consecutive packets, usually 10 ms, 20 ms, or 30 ms; the packet size is determined by calculating the total number of bytes of the encoded voice frame plus the protocol header. These parameters together constitute the encoding feature parameters, which are used to describe the basic transmission characteristics of the voice data.

[0022] Specifically, while obtaining the encoding feature parameters, the VOIP device also conducts a deeper analysis of the encoded voice data to evaluate the trend of communication load changes. Specifically, the device calculates the short-time energy and zero-crossing rate, and combines with the voice activity detection algorithm to judge the activity state of the current voice data, separating the voice segment and the silent segment. The background noise level is obtained by analyzing the energy level and spectral characteristics of the silent segment, and is quantified into three levels: low, medium, and high. The call mode is identified by analyzing the time distribution pattern of voice activity, whether it is a one-way call, a two-way alternating call, or a two-way overlapping call. Based on these analysis results, the VOIP device constructs a triple data structure, including the voice activity state, the noise level, and the call mode, as the load prediction vector, which is used to predict the trend of voice data generation in the next period of time. At the same time, the device evaluates its own hardware capabilities, including the supported modulation and coding scheme levels (from MCS0 to MCS11), the maximum transmit power (in dBm), and the current battery state (percentage). The modulation and coding scheme level determines the maximum transmission rate of the device under different channel conditions, the maximum transmit power affects the signal transmission distance and anti-interference ability, and the battery state is related to the device's energy efficiency strategy. These hardware parameters form the terminal capability vector, providing device capability constraints for the formulation of the transmission strategy.

[0023] Specifically, in order to comprehensively evaluate the network environment status, the VOIP device periodically measures the wireless channel, and the measurement period is usually set to 100ms. In each measurement period, the device obtains the received signal strength indication in dBm by listening to the beacon frame and other control frames of the access point, and calculates the signal-to-noise ratio in dB to represent the signal quality; and records the modulation and coding scheme currently used. The device maintains a sliding window of historical measurement values with a length of 10, and calculates the channel fluctuation based on these historical data. The channel fluctuation calculation adopts a method combining standard deviation and coefficient of variation. The standard deviation reflects the absolute fluctuation amplitude, and the coefficient of variation reflects the relative fluctuation degree. Specifically, first calculate the standard deviation of the received signal strength and signal-to-noise ratio in the sliding window, and then calculate the respective coefficients of variation, that is, the standard deviation divided by the average value. Finally, the channel stability index is calculated by weighted combination of the coefficient of variation and the mapping value of the modulation and coding scheme. The closer the channel stability index value is to 1, the more stable the channel is, and the closer it is to 0, the greater the channel fluctuation is. This index directly affects the selection of subsequent transmission strategies. Stable channels are suitable for more efficient transmission parameters, while channels with large fluctuations require more conservative transmission strategies.

[0024] Specifically, after completing the acquisition and calculation of the above parameters, the VOIP device combines the coding feature parameters, load prediction vector, terminal capability vector and channel stability index into a unified feature vector set. The feature vector set adopts a structured data format and contains four main parts: the coding feature part records the codec type, data packet generation cycle and data packet size; the load prediction part contains the voice activity status, background noise level and call mode; the terminal capability part describes the supported modulation and coding scheme level, maximum transmission power and battery status; the channel status part contains the channel stability index and its calculation basis. As a comprehensive description of the current status and network environment of the VOIP device, the feature vector set will be sent to the access point through the enhanced buffer status reporting mechanism to provide a decision basis for the access point to allocate network resources and formulate scheduling strategies. This mechanism extends multiple information fields on the basis of the standard buffer status report to ensure that the key information in the feature vector set can be fully transmitted, and at the same time optimizes the triggering conditions of the report, actively sends reports when the channel status changes significantly or the voice characteristics change significantly, and reduces unnecessary signaling overhead. The transmission of the feature vector set adopts a compression coding method to quantize the key parameters into fields with a fixed bit width, ensuring transmission efficiency while ensuring accurate transmission of information.

[0025] 102. The receiving access point allocates network resources and scheduling information according to the feature vector set and the channel stability index, and processes the encoded voice data packet to form a VOIP data packet to be transmitted; In an embodiment of the present invention, processing the encoded voice data packets according to the network resources and scheduling information allocated by the access point based on the feature vector set and the channel stability index to form VOIP data packets to be transmitted includes: receiving the network resources and scheduling information allocated by the access point based on the feature vector set and the channel stability index, and analyzing the encoded voice data packets according to the channel stability index and the network resources to calculate a packet aggregation degree adjustment coefficient; dynamically aggregating a plurality of encoded voice data packets according to the aggregation degree adjustment coefficient to form an aggregated packet, and in the aggregation process, performing real-time analysis on the encoded voice data packets to identify key voice data containing explosive sounds or significantly changed volume segments, and assigning a priority mark to the key voice data; selecting a transmission modulation and coding scheme for the aggregated packet according to the scheduling information and the channel stability index, and combining the aggregated packet, the priority mark, and the transmission modulation and coding scheme to form a VOIP data packet to be transmitted.

[0026] Specifically, after completing voice data acquisition and encoding and sending the feature vector set and the channel stability index to the access point, the VOIP device needs to receive the network resources and scheduling information allocated by the access point and process the encoded voice data packets to finally form VOIP data packets to be transmitted. The access point, based on the received feature vector set and channel stability index, allocates appropriate network resources and transmission time for the VOIP device through a resource allocation algorithm. The VOIP device receives the resource allocation information frame broadcast by the access point through the wireless channel, and this information frame includes two parts: resource unit allocation information and scheduling information. The resource unit allocation information clearly specifies the number, bandwidth, and location of the frequency domain resource units allocated to the VOIP device. These resource units are the basic resource allocation units in the orthogonal frequency division multiple access technology. Each resource unit consists of several subcarriers, and different devices can simultaneously use different resource units for transmission, avoiding the strict queuing waiting in the traditional time division access. The scheduling information includes transmission time slot arrangement, transmission duration limit, and priority indication, clearly specifying when the VOIP device starts transmission and the maximum allowable transmission time. The media access control layer of the VOIP device receives and parses this information, determines the available network resources and transmission opportunities, and then passes this information to the upper layer for packet processing.

[0027] Specifically, after receiving network resources and scheduling information, the VOIP device first needs to calculate the packet aggregation degree adjustment coefficient according to the channel stability index and network resource conditions. Packet aggregation is a technology that combines multiple small data packets into a larger one for transmission, which can reduce protocol header overhead and the number of transmissions, thereby improving bandwidth utilization. The VOIP device extracts the encoded voice data packets to be sent from the buffer. These packets are usually small in size (for example, a G.729 encoded voice frame of 20 bytes is generated every 20 ms). If each packet is sent individually, the protocol header (such as RTP+UDP+IP+MAC headers, about 60 bytes in total) overhead will be much larger than the payload. Therefore, it is crucial to determine the appropriate aggregation degree according to the current network conditions. Specifically, the VOIP device first analyzes the allocated network resources, calculates the available maximum transmission unit (MTU), and also considers the value of the channel stability index. For a situation with a relatively high channel stability (the index is close to 1), the device tends to adopt a higher aggregation degree because a stable channel means a lower transmission error rate and a higher transmission success rate for large data packets; while for a situation with large channel fluctuations (the index is close to 0), the aggregation degree is reduced to avoid a high retransmission rate of large data packets in an unstable channel. In addition, the VOIP device also needs to consider the time limit in the scheduling information to ensure that the aggregated packets can be transmitted within the allocated time slots. Based on the above analysis, the VOIP device obtains the packet aggregation degree adjustment coefficient through a table lookup or calculation method, and this coefficient will be multiplied by the base aggregation degree to obtain the final number of aggregated packets.

[0028] Specifically, after the packet aggregation degree is determined, the VOIP device starts to perform dynamic aggregation processing on multiple encoded voice data packets to form aggregated data packets. During the aggregation process, the VOIP device first extracts a specified number of voice data packets from the send buffer and then encapsulates them according to the specifications of the Real-time Transport Protocol (RTP). Each original voice data packet is encapsulated into an RTP packet, which contains information such as a sequence number, a timestamp, and a synchronization source identifier. These information are crucial for the receiving end to correctly decode and play the voice. Subsequently, multiple RTP packets are further encapsulated into an RTP compound packet to form the first-level aggregation. The compound packet after the first-level aggregation is further encapsulated through the UDP and IP layers, adding the corresponding transport layer and network layer header information. Finally, a MAC header is added at the link layer to form a complete aggregated data packet. During the aggregation process, the VOIP device simultaneously performs real-time analysis on the encoded voice data packets to identify key segments containing important voice content. By calculating the short-time voice energy, the spectral change rate, and the voice feature parameters, the device can accurately identify plosive sounds (such as / p / , / t / , / k / , etc.) and voice segments with obvious volume changes. These key voice segments usually carry important semantic information, and ensuring their transmission quality has a significant impact on the overall call quality. Therefore, the VOIP device assigns a high-priority mark to the data packets containing these key voice data. The mark uses 2-bit binary encoding, representing four priority levels from 0 (ordinary) to 3 (highest). The priority mark is inserted into the type-of-service field of the data packet header for reference in resource scheduling during network transmission.

[0029] Specifically, finally, the VOIP device needs to select an appropriate transmission modulation and coding scheme for the aggregated data packet according to the scheduling information and the channel stability index. The modulation and coding scheme (MCS) determines the encoding and modulation methods of data at the physical layer, directly affecting the transmission rate and anti-interference ability. The VOIP device supports multiple MCS levels, ranging from low-order modulation and low coding rate (such as BPSK1 / 2, suitable for poor channels) to high-order modulation and high coding rate (such as 256QAM 5 / 6, suitable for good channels). When selecting the MCS, the VOIP device comprehensively considers the size of the aggregated data packet, the transmission time slot length in the scheduling information, and the channel stability index. For a higher channel stability situation, the device selects a higher-order MCS to improve the transmission efficiency; while for a situation with larger channel fluctuations, a lower-order MCS is selected to ensure the transmission reliability. The selection of the MCS also needs to ensure that the data transmission can be completed within the allocated transmission time slot. After the MCS is selected, the VOIP device combines the aggregated data packet, the priority tag, and the transmission modulation and coding scheme together to form a complete VOIP data packet structure to be transmitted. This structure includes three main parts: physical layer parameters, MAC layer control information, and payload data. The physical layer parameters specify physical transmission parameters such as the modulation and coding scheme, the guard interval length, and the spatial stream configuration; the MAC layer control information includes a frame control field, a duration field, an address field, and a sequence control field, etc.; the payload data is the actual content of the aggregated data packet, including the voice data with marked priority. The VOIP data packet to be transmitted formed after such combination contains all necessary transmission parameters and payloads, and is completely ready to be sent to other VOIP devices in the network through the wireless channel.

[0030] Furthermore, the analysis of the encoded voice data packet according to the channel stability index and the network resources to calculate the data packet aggregation degree adjustment coefficient includes: performing feature analysis on the encoded voice data packet, extracting the data packet size, frame interval, and payload distribution features, and generating a voice data feature value; processing the channel stability index, combining the historical data of the received signal strength indication and the signal-to-noise ratio, and calculating the channel reliability value of the current transmission window; performing resource analysis on the network resources, detecting the available bandwidth, time slot allocation situation, and the current network congestion state, and obtaining resource utilization parameters; according to the voice data feature value, the channel reliability value, and the resource utilization parameters, querying a preset aggregation parameter table, selecting the corresponding basic aggregation value, and correcting it in combination with the current voice activity state to obtain the data packet aggregation degree adjustment coefficient.

[0031] Specifically, during VOIP communication, packet aggregation technology directly affects transmission efficiency and call quality. To accurately calculate the optimal packet aggregation degree adjustment coefficient, the VOIP device needs to comprehensively analyze the encoded voice packets based on the channel stability index and network resources. First, the VOIP device performs feature analysis on the encoded voice packets, extracts samples of voice packets to be sent from the buffer, and usually selects the last 10 packets as the analysis objects. For each packet, the device extracts three core features: packet size, frame interval, and payload distribution characteristics. The packet size refers to the total number of bytes of the voice packet, including the RTP header (12 bytes), UDP header (8 bytes), IP header (20 bytes), and voice payload; the frame interval refers to the generation time interval between two consecutive voice packets, and typical values are 10ms, 20ms, or 30ms, depending on the codec settings; the payload distribution characteristics describe the proportion and distribution of voice data in the entire packet. The VOIP device calculates the average value and standard deviation of these features through statistical analysis to form a feature vector, which is the voice data feature value. The voice data feature value reflects the statistical characteristics of the current voice stream. Different voice stream patterns (such as continuous voice, intermittent voice, or silent periods) will generate different feature values, and these feature values directly affect the selection of the optimal aggregation strategy.

[0032] Specifically, the VOIP device needs to further process the channel stability index to obtain a more accurate evaluation of the current transmission window channel reliability. The channel stability index itself already contains an overall evaluation of channel fluctuations, but for a specific upcoming transmission window, it is also necessary to combine the latest received signal strength indication (RSSI) and signal-to-noise ratio (SNR) historical data for a refined evaluation. The device obtains RSSI and SNR sampling values within the last 100ms from the physical layer interface, usually sampling once every 10ms to form a historical data sequence of length 10. For these historical data, the device first calculates the average value and change trend, and then combines the previously calculated channel stability index to calculate the channel reliability value of the current transmission window through a weighted average method. In the calculation formula, the latest RSSI and SNR samples are given higher weights to reflect the current channel state, and the channel stability index participates in the calculation as a basic evaluation value. The obtained channel reliability value is a normalized value between 0 and 1. The closer the value is to 1, the more reliable the channel and the higher the transmission success rate; the closer the value is to 0, the worse the channel quality and the higher the probability of transmission errors. The channel reliability value directly affects the conservativeness of the aggregation strategy. Under low reliability conditions, the system will select a smaller aggregation degree to reduce retransmission overhead.

[0033] Specifically, the VOIP device conducts a detailed analysis of the network resources allocated to the access point to evaluate the currently available transmission resources. The device obtains resource allocation information from the media access control layer, including the number of allocated resource units, their locations, and durations. Based on this information, the device first calculates the available bandwidth, which is the bit rate that can be used within the current transmission window, measured in kbps; then analyzes the time slot allocation, including the start time, end time of the time slot, and the overlap of time slots with other devices; finally, evaluates the current network congestion status by monitoring indicators such as channel utilization, contention window size, and collision rate to determine the network busyness. These analysis results are organized into a triple <bandwidth, slot_duration, congestion_level>, which constitutes the resource utilization parameter. The resource utilization parameter directly reflects the transmission capacity and time constraints of the current network environment. In the case of high bandwidth, long time slots, and low congestion, the system can select a larger aggregation degree to improve transmission efficiency; while in the case of resource constraints, a more cautious selection of the aggregation strategy is required to avoid exceeding resource limits or causing network congestion.

[0034] Specifically, based on the above analysis results, the VOIP device finally calculates the packet aggregation degree adjustment coefficient. This calculation process first queries a preset aggregation parameter table based on the voice data eigenvalue, channel reliability value, and resource utilization parameter. The aggregation parameter table is a three-dimensional lookup table, where each dimension corresponds to different intervals of voice characteristics, channel reliability, and resource utilization respectively, and the table stores the empirically optimal basic aggregation values under the corresponding conditions. The table lookup operation is performed through trilinear interpolation, that is, first finding the two closest index points on each dimension, and then calculating the interpolation result based on the distance weights. This method can more accurately reflect the continuous change of parameters than the simple rounding method. After obtaining the basic aggregation value from the table lookup operation, the VOIP device also needs to make corrections in combination with the currently detected voice activity status in real time. The voice activity detector monitors the energy level and characteristic parameters of the voice signal in real time to determine whether it is a voice active period or a silent period. During the voice active period, especially when detecting semantically important voice segments (such as plosives or the beginning of vowels), the system will appropriately reduce the aggregation degree to reduce the transmission delay of these important voice segments; while during the silent period or the smooth vowel continuation segment, the system will maintain or increase the aggregation degree to optimize the transmission efficiency. Finally, the corrected value is the packet aggregation degree adjustment coefficient, which is usually between 0.5 and 2.0, representing the adjustment ratio relative to the standard aggregation degree. For example, a coefficient of 1.5 means that the aggregation degree under the current conditions should be 1.5 times the standard value. This adjustment coefficient will be directly applied to the subsequent packet aggregation processing process, multiplied by the default basic aggregation quantity of the device (usually determined by the codec type), to obtain the final number of aggregated packets. Through this dynamic aggregation degree adjustment mechanism based on multi-dimensional feature analysis, the VOIP device can adaptively select the optimal aggregation strategy according to the current voice characteristics, channel conditions, and network resources, and balance the requirements of transmission efficiency and call quality.

[0035] 103. Send the VOIP packets to other VOIP devices in the network through the wireless channel, coordinate the transmission resources with other VOIP devices in the network, share the wireless channel, and adjust the transmission parameters according to the interference situation during the transmission process; In one embodiment of the present invention, sending the VOIP data packet to other VOIP devices in the network through a wireless channel, coordinating transmission resources with other VOIP devices in the network, sharing the wireless channel, and adjusting transmission parameters according to interference conditions during the transmission process includes: receiving a cooperative group mapping table from the access point, and identifying the belonging cooperative group and cooperative group transmission rules according to the network resources and scheduling information allocated by the access point through the cooperative group mapping table; according to the cooperative group transmission rules determined by the cooperative group mapping table, listening for trigger frames sent by the access point, and sending the VOIP data packet to other VOIP devices in the network through the wireless channel according to the indication in the trigger frame; monitoring and analyzing the interference conditions in the network according to the cooperative group transmission rules, generating an interference map, and adjusting the transmission parameters of the VOIP data packet based on the interference map, including adjusting the transmission power, modulation and coding scheme, and transmission timing, to adapt to the interference conditions of the wireless channel.

[0036] Specifically, in a VOIP communication system, the efficient transmission of data packets requires coordinated cooperation between devices. Especially in a high-density wireless environment, to ensure that VOIP data packets can be successfully sent to other VOIP devices in the network, the device first needs to receive a cooperative group mapping table and identify the belonging cooperative group. The VOIP device obtains the cooperative group mapping table by listening for management frames periodically broadcast by the access point. This mapping table is embedded in the beacon frame or management frame in the form of a dedicated information element and is updated every 100 ms. The cooperative group mapping table is a structured data that contains detailed information of multiple cooperative groups. Each cooperative group entry contains fields such as a cooperative group identifier, a list of member addresses, a set of transmission parameters, and group priority. After receiving the cooperative group mapping table, the VOIP device first decodes and verifies it to ensure data integrity; then extracts the cooperative group information in the mapping table and performs a comparative analysis with the network resources and scheduling information allocated by the current access point. Specifically, the device compares its own MAC address with the list of member addresses of each cooperative group, using a byte-by-byte comparison method to identify the cooperative group to which it belongs. At the same time, the device also extracts the set of transmission parameters of this cooperative group, including the transmission period, time slot arrangement, frequency domain resource allocation, and cooperative transmission strategy, etc. These parameters together constitute the cooperative group transmission rules, which stipulate how group members coordinate resource usage, how to avoid interference, and how to handle transmission conflicts, etc. The cooperative group transmission rules and the network resources and scheduling information allocated by the access point jointly determine the specific transmission behavior of the VOIP device.

[0037] Specifically, after identifying the affiliated cooperation group and obtaining the cooperation group transmission rules, the VOIP device needs to listen for trigger frames sent by the access point according to these rules and perform data transmission according to the indications in the trigger frames. The trigger frame is a control frame used by the access point to coordinate the uplink transmission of multiple devices and contains information such as resource unit allocation, transmission sequence, and transmission parameters. The VOIP device continuously listens to the control channel through the physical layer receiver. When detecting the trigger frame preamble, it immediately switches to the high-priority decoding mode to demodulate and decode the trigger frame. The content of the decoded trigger frame first undergoes frame check to confirm integrity and validity; then the device extracts the target device list from it and checks whether it is included in the transmission sequence. If the VOIP device identifies itself in the transmission sequence of the current trigger frame, it further parses the resource unit information allocated to itself, including the frequency-domain position, bandwidth, and time-domain duration of the resource unit. Subsequently, the device performs the final encapsulation of the VOIP data packet to be sent according to the transmission protocol specified in the cooperation group transmission rules, including adding a physical layer preamble, training sequence, and frame check sequence, etc. After the encapsulation is completed, the device starts the transmitter at the precise moment specified in the trigger frame and sends the VOIP data packet through the allocated resource unit to the wireless channel. This trigger-frame-based synchronous transmission mechanism ensures that multiple VOIP devices can orthogonally use wireless resources in terms of time and frequency, minimizing transmission conflicts and resource waste. At the same time, since all devices follow the cooperation group transmission rules, multiple VOIP communication links sharing the same wireless channel can coexist harmoniously, improving the overall network capacity.

[0038] Specifically, during the data transmission process, the interference situation in the wireless environment will change dynamically. The VOIP device needs to continuously monitor and analyze the interference situation and adjust the transmission parameters accordingly. According to the interference monitoring protocol in the cooperation group transmission rules, the VOIP device will periodically switch to the receiving mode during non-transmission periods to perform an energy scan of the entire frequency band, measure the signal energy levels and interference characteristics of each frequency band. Specifically, the device samples and performs energy detection on each subcarrier within a 20MHz bandwidth, records the noise plus interference power spectral density of each subcarrier, and at the same time identifies the type characteristics of the interference source, such as narrowband interference, broadband interference, or impulse interference, etc. These measurement results are statistically processed in the time and frequency dimensions to form a two-dimensional interference distribution matrix, that is, an interference map. The interference map takes frequency as the horizontal axis and time as the vertical axis, and records the interference levels at different time-frequency resource points. Each element in the interference map is an interference intensity value, usually in dBm, indicating the interference plus noise power level at that time-frequency resource point. In addition, the interference map also contains interference type markers and periodic analysis results to help the device identify periodic interference sources and predictable interference patterns.

[0039] Specifically, based on the generated interference map, the VOIP device dynamically adjusts the transmission parameters of subsequent VOIP data packets to adapt to the changing wireless channel interference situation. The adjustment of transmission parameters mainly includes three aspects: transmission power adjustment, modulation and coding scheme adjustment, and transmission timing adjustment. In terms of transmission power, the device adjusts the transmission power of each resource unit according to the interference level of each resource unit in the interference map. For resource units with high interference, the transmission power is appropriately increased to ensure sufficient signal-to-noise ratio; for resource units with low interference, the transmission power is appropriately reduced to reduce interference to other devices and save energy consumption. It should be noted that the power adjustment needs to be carried out within the maximum power limit that complies with regulatory requirements, and the power amplifier efficiency curve and battery status of the device itself need to be considered. In terms of the modulation and coding scheme, the device selects an appropriate modulation order and coding rate according to the channel quality and interference level of each resource unit. For resource units affected by strong interference, lower-order modulation and lower coding rate (such as QPSK 1 / 2) are selected to enhance the anti-interference ability; for resource units with less interference, higher-order modulation and higher coding rate (such as 64-QAM 3 / 4) are selected to improve the spectrum efficiency. In terms of transmission timing, the device adjusts the sending timing of data packets according to the interference time pattern identified in the interference map, and tries to avoid the interference peak period as much as possible. For interference sources with obvious periodicity, the device can predict their interference patterns and arrange transmissions during the interference gaps; for random interference, the time window with a lower interference probability is found through statistical analysis. The parameter adjustments in these three aspects cooperate with each other to form a comprehensive transmission strategy, enabling VOIP data packets to be transmitted more reliably in a complex interference environment, ensuring the quality of voice communication while improving the utilization efficiency of spectrum resources.

[0040] Further, the receiving the cooperative group mapping table from the access point and identifying the belonging cooperative group and cooperative group transmission rules according to the network resources and scheduling information allocated by the access point through the cooperative group mapping table include: receiving the cooperative group mapping table sent by the access point through the management frame, and parsing and processing the network resources allocated by the access point to identify the resource configuration data structure of the resource units of the network resources; performing a timing analysis on the scheduling information provided by the access point, parsing the trigger period, target wake-up time, contention window parameters, and transmission opportunity limit of the trigger frame, determining the transmission timing and duration constraint of the VOIP device, and constructing a time slot access rule table; performing a byte-by-byte comparison operation on the physical address of the VOIP device and the member address list in the cooperative group mapping table, and determining the cooperative group identifier of the cooperative group to which the VOIP device belongs through an address matching algorithm; retrieving the corresponding transmission parameters from the cooperative group mapping table according to the cooperative group identifier, and combining the resource configuration data structure and the time slot access rule table to construct the cooperative group transmission rules.

[0041] Specifically, in a high-density wireless network environment, VOIP devices need to coordinate with other devices to use limited wireless resources, and the cooperative group mechanism provides an efficient resource coordination solution. First, the VOIP device needs to receive the cooperative group mapping table from the access point and identify the cooperative group to which it belongs. The access point distributes the cooperative group mapping table to all devices in the network through specific management frames, which include beacon frames, association response frames, or independent action frames. The MAC layer receiver of the VOIP device continuously monitors the control channel. When the MAC header of the management frame is detected, the frame content is demodulated and decoded. The decoded management frame undergoes integrity verification to confirm that the frame is not damaged and the source is indeed the currently associated access point. After passing the verification, the VOIP device extracts the cooperative group mapping table information element from the management frame. This element uses a specific information element identifier (usually 221, indicating vendor-specific extension) and contains the complete configuration information of the cooperative group. The cooperative group mapping table adopts a structured format, including the mapping table version number, the number of cooperative groups, global parameters, and detailed information of each cooperative group. The global parameters include the cooperative group update period, default permission settings, and global resource limits, etc.; while the detailed information of each cooperative group includes the cooperative group identifier (8-byte unique identifier), the member address list (supporting up to 20 MAC addresses at most), the transmission parameter set (including multiple transmission-related configuration items), and the group priority (0-7, the larger the value, the higher the priority). The VOIP device stores the received cooperative group mapping table in a dedicated buffer and records the reception timestamp for subsequent validity verification.

[0042] Specifically, upon receiving the collaboration group mapping table, the VOIP device needs to parse and process the network resources allocated by the access point. Network resource information is usually included in the resource allocation frame or trigger frame together with the scheduling information. The VOIP device obtains these frames through the physical layer receiver and performs decoding processing. The decoded resource allocation information first undergoes format verification and integrity check, and then detailed parsing. The first step of resource parsing is to identify the basic parameters of the resource unit, including the number of resource units, bandwidth configuration, and location distribution. For a 20 MHz channel, the system supports multiple resource unit combinations, such as 9 26-tone resource units, 4 52-tone resource units, or more complex hybrid configurations. The VOIP device determines the type and number of resource units allocated to itself according to the resource allocation information, such as one 52-tone resource unit or two 26-tone resource units. Next, the device analyzes the frequency domain position of each resource unit, i.e., the starting subcarrier index and ending subcarrier index in the entire channel bandwidth, which determines the specific frequency range used during transmission. Then, the device extracts the time domain configuration of the resource unit, including the symbol start position, number of symbols, and guard interval type. These parameters jointly determine the transmission time length and time positioning. Finally, the device also needs to parse the physical layer parameters related to the resource unit, such as the maximum allowable transmit power, spatial stream configuration, and precoding matrix information. All this information is organized into a structured resource configuration data structure, which contains all necessary resource parameters for reference in subsequent transmission processes.

[0043] Specifically, in addition to network resources, the VOIP device also needs to perform timing analysis on the scheduling information provided by the access point to determine when to transmit and for how long. The scheduling information is usually contained in the trigger frame, Target Wake Time (TWT) information element, or resource allocation frame. The VOIP device first needs to extract the key parameters related to timing. For the trigger frame, the device parses the trigger period field, which specifies the transmission interval of the trigger frame, with typical values of 5 ms, 10 ms, or 20 ms. Then it extracts the target wake time, which precisely specifies when the VOIP device should switch from the dormant state to the active state to prepare for receiving or sending data. Next, it analyzes the contention window parameters, including the minimum contention window size, maximum contention window size, and contention window adjustment rule, which affect the device's channel access opportunity during contention-based access. Finally, it extracts the transmission opportunity limit, i.e., the maximum continuous transmission time that the device can have once it obtains the channel access right, usually in microseconds. Based on these timing parameters, the VOIP device constructs a time slot access rule table, which is a time series data structure. Each entry contains the start time, end time, access mode (such as scheduled access or contention access), and priority information of the time slot. The time slot access rule table clearly stipulates when the VOIP device can access the channel and the duration constraints of the access, ensuring that the device's transmission behavior conforms to the global scheduling arrangement of the access point.

[0044] Specifically, after obtaining the cooperation group mapping table, resource configuration data structure, and time slot access rule table, the VOIP device needs to determine the cooperation group it belongs to. The device first obtains its own MAC address (a 48-bit globally unique identifier) from the internal storage or hardware register, and then compares this address with the member address lists of each cooperation group in the cooperation group mapping table. The comparison process uses a byte-by-byte comparison algorithm, starting from the most significant byte and comparing each byte of the MAC address in turn until a complete match or non-match is confirmed. To improve efficiency, an optimized comparison algorithm, such as batch byte comparison or hardware-accelerated comparison, is usually adopted in actual implementation. When the VOIP device finds an entry in the member list of a cooperation group that exactly matches its own MAC address, it confirms that it belongs to this cooperation group and records the corresponding cooperation group identifier. In some cases, a device may belong to multiple cooperation groups at the same time. In this case, the device will select the primary cooperation group according to the cooperation group priority, or switch between different cooperation groups according to the specific application scenario. After determining the cooperation group it belongs to, the VOIP device extracts the transmission parameter set of this cooperation group from the cooperation group mapping table. These parameters include resource sharing strategies, transmission sequence arrangements, cooperation transmission protocols, and conflict handling mechanisms, etc. Finally, the VOIP device integrates the transmission parameters extracted from the cooperation group mapping table with the previously parsed resource configuration data structure and time slot access rule table to generate a complete cooperation group transmission rule.

[0045] 104. Monitor the transmission performance of the sent VOIP data packets, conduct call quality assessment, and record the quality assessment results of the transmission performance and call quality assessment.

[0046] In one embodiment of the present invention, the monitoring of the transmission performance of the sent VOIP data packets, conducting call quality assessment, and recording the quality assessment results of the transmission performance and call quality assessment include: real-time tracking of the end-to-end delay, network jitter, and packet loss rate of the sent VOIP data packets, and analyzing transmission control parameters, including retransmission rate, sequence number interval, and round-trip time, to evaluate the efficiency of the transmission process and obtain the transmission performance; analyzing the decoded voice data, evaluating voice clarity, voice continuity, and mean opinion score, conducting call quality assessment, and obtaining the quality assessment results of the call quality assessment; and recording the transmission performance and the quality assessment results into the transmission performance reference database under different network environments.

[0047] Specifically, during VOIP communication, continuous monitoring and assessment of communication quality are crucial for maintaining a high-quality call experience. The VOIP device first needs to monitor the transmission performance of the sent VOIP data packets. When the VOIP device sends each data packet, it adds a timestamp and sequence number information to the packet header, and at the same time maintains a transmission record table inside the device to record the sending time, sequence number, size, and importance flag of each packet. When the receiving end receives the data packet, it sends an acknowledgment message or status report back to the sending end, including the receiving timestamp, sequence number, and receiving quality metrics. Based on this information, the VOIP device calculates three core network performance metrics: end-to-end delay, network jitter, and packet loss rate. The end-to-end delay refers to the total time from the packet being sent to being received, which is calculated by comparing the sending timestamp and the receiving timestamp. Typically, VOIP communication requires the end-to-end delay not to exceed 150 milliseconds. Network jitter refers to the degree of change in the delay of consecutive data packets, which is calculated by the standard deviation of the delay difference between adjacent data packets, reflecting the stability of the network transmission time. Larger jitter will cause the voice playback to be uneven. The packet loss rate refers to the proportion of the sent data packets that are not successfully received, which is calculated by comparing the sent sequence number and the received acknowledgment sequence number. A packet loss rate exceeding 5% usually significantly affects the call quality. In addition to these basic metrics, the VOIP device also analyzes deeper transmission control parameters, including retransmission rate, sequence number interval, and round-trip time. The retransmission rate refers to the proportion of data packets that need to be resent. A high retransmission rate means poor network conditions. Analyzing the sequence number interval can detect packet out-of-order phenomena caused by network congestion or routing changes. The round-trip time measures the time interval from sending a data packet to receiving an acknowledgment, and is an important parameter for evaluating the network response speed. These parameters together constitute a comprehensive transmission performance evaluation system, enabling the VOIP device to accurately grasp the current network transmission status.

[0048] Specifically, while monitoring the network transmission performance, the VOIP device also needs to evaluate the actual voice call quality. The call quality evaluation starts from the perspective of the user experience and deeply analyzes the decoded voice data. First, the VOIP device obtains the original voice samples from the decoder output, usually PCM data with a sampling rate of 16 kHz and a quantization precision of 16 bits. For these data, the device first conducts a voice clarity evaluation. By calculating parameters such as the signal spectrum characteristics, energy distribution, and background noise level, the clarity level of the voice is quantified. The voice clarity reflects the discernibility of the voice content in the voice signal and is affected by factors such as codec algorithms, transmission packet loss, and network noise. Secondly, the device evaluates the voice continuity. By analyzing the time distribution of voice segments, the number and duration of voice interruptions, it judges whether the voice playback is smooth. Voice continuity has an important impact on the call experience. Frequent interruptions or pauses will seriously affect the call communication efficiency. Then, the device calculates the Mean Opinion Score (MOS) by synthesizing multiple indicators. This is a widely used standardized indicator for evaluating voice quality, with a score range from 1 (very poor) to 5 (excellent). The MOS calculation takes into account various factors, including voice clarity, background noise level, echo intensity, voice delay, and interruption frequency. In actual implementation, the VOIP device usually uses a simplified version of the E-model or PESQ algorithm to calculate the MOS value and evaluate the perceived voice quality without the need to refer to the original voice. Through these analyses, the device forms a complete call quality evaluation result, including specific indicators and the overall quality score, comprehensively reflecting the current call user experience level.

[0049] Specifically, after obtaining the transmission performance metrics and call quality assessment results, the VOIP device needs to record this information in a dedicated database to support long-term analysis and optimization. The VOIP device maintains a structured transmission performance reference database, which is organized according to different network environment types. The network environment type is the result of multi-dimensional classification, including dimensions such as network type (e.g., home WiFi, enterprise WiFi, public hotspot, etc.), channel congestion level (low, medium, high three levels), interference level (low, medium, high three levels), and mobility (stationary, low-speed movement, high-speed movement). For each call session, the VOIP device first identifies the corresponding network environment type based on the current network characteristics, and then adds the monitored transmission performance metrics and call quality assessment results to the data entry of this environment type. The database entry adopts a time series structure, recording the detailed metrics at each time point, while maintaining the statistical summary of the sliding window, including statistical features such as mean, standard deviation, maximum value, minimum value, and quantile. In addition, the database also records the correlation analysis results between various metrics, such as the correlation curve between packet loss rate and MOS, to help understand the impact mechanism of network performance on user experience. The VOIP device regularly maintains the database, including aging expired data, merging similar entries, and optimizing the storage structure, to ensure that the database size is controllable and has good query performance.

[0050] Specifically, the transmission performance reference database plays multiple roles in the VOIP communication system. Firstly, it is used for system adaptive optimization. The VOIP device can understand the best transmission parameter configurations in different network environments, such as aggregation degree, MCS selection, and transmission power, by querying historical data, so as to achieve a more accurate transmission strategy selection. Secondly, it is used for performance trend analysis. By comparing the current performance with historical records, it can timely detect the trend of network quality decline and take preventive measures. Thirdly, it is used for fault diagnosis and location. When the call quality significantly deteriorates, the abnormal metrics can be quickly identified and the root cause of the problem can be located by comparing historical data. Finally, the database also supports the user feedback mechanism, allowing users to subjectively evaluate the call quality. These evaluations are stored together with the objective measurement metrics to establish the mapping relationship between subjective and objective evaluations, and continuously improve the accuracy of the evaluation algorithm. By establishing and maintaining the transmission performance reference database, the VOIP device can continuously accumulate communication experience, continuously optimize communication strategies based on a large amount of historical data, adapt to various complex network environments, and provide more stable and reliable voice communication services. The entire database adopts a lightweight design, is stored in the device's local flash memory, usually occupies no more than 10MB of space, has low requirements for the device's hardware resources, and at the same time supports differential privacy policies to ensure the security of user communication data.

[0051] In this embodiment, by collecting and encoding voice data, detecting the characteristics of the encoded data and the network environment status, generating a feature vector set and a channel stability index, and sending them to the access point; receiving the network resources and scheduling information allocated by the access point, processing the encoded voice data packets to form VOIP data packets to be transmitted; sending the VOIP data packets to other VOIP devices through a wireless channel, coordinating transmission resources with other devices in the network, sharing the wireless channel, and adjusting transmission parameters according to the interference situation; monitoring the transmission performance of the sent data packets, performing call quality evaluation, and recording the transmission performance and quality evaluation results. Through dynamic adjustment of the transmission strategy, cooperative transmission, and comprehensive quality evaluation, the present invention significantly improves the performance and user experience of VOIP communication in a high-density wireless network environment.

[0052] The communication method of the VOIP device in the embodiment of the present invention has been described above. Next, the communication device of the VOIP device in the embodiment of the present invention will be described. For the communication device of the VOIP device, please refer to Figure 2 , an embodiment of the communication device of the VOIP device in the embodiment of the present invention includes: A data processing module 201, configured to collect and encode the voice data of the VOIP device, detect the characteristics of the encoded data and the network environment status, generate a feature vector set and a channel stability index, and send the feature vector set and the channel stability index to the access point; A data packet generation module 202, configured to receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, process the encoded voice data packets, and form VOIP data packets to be transmitted; A cooperative transmission module 203, configured to send the VOIP data packets to other VOIP devices in the network through a wireless channel, and at the same time coordinate transmission resources with other VOIP devices in the network, share the wireless channel, and adjust transmission parameters according to the interference situation during the transmission process; A quality evaluation module 204, configured to monitor the transmission performance of the sent VOIP data packets, perform call quality evaluation, and record the transmission performance and the quality evaluation results of the call quality evaluation.

[0053] In an embodiment of the present invention, the communication device of the VOIP device runs the communication method of the VOIP device. The communication device of the VOIP device collects and encodes voice data, detects the characteristics of the encoded data and the network environment status, generates a feature vector set and a channel stability index, and sends them to the access point; receives the network resources and scheduling information allocated by the access point, processes the encoded voice data packets, and forms VOIP data packets to be transmitted; sends the VOIP data packets to other VOIP devices through a wireless channel, coordinates the transmission resources with other devices, shares the wireless channel, and adjusts the transmission parameters according to the interference situation; monitors the transmission performance of the sent data packets, conducts call quality evaluation, and records the transmission performance and quality evaluation results. The present invention significantly improves the performance and user experience of VOIP communication in a high-density wireless network environment through dynamic adjustment of the transmission strategy, cooperative transmission, and comprehensive quality evaluation.

[0054] Above Figure 2 The communication device of the VOIP device in the embodiment of the present invention is described in detail from the perspective of modular functional entities. Next, the communication device of the VOIP device in the embodiment of the present invention is described in detail from the perspective of hardware processing.

[0055] Figure 3 FIG. 9 is a schematic structural diagram of a communication device of a VOIP device provided by an embodiment of the present invention. The communication device 300 of the VOIP device may vary greatly due to configuration or performance differences, and may include one or more processors (central processing units, CPU) 310 (for example, one or more processors) and a memory 320, and one or more storage media 330 for storing application programs 333 or data 332 (for example, one or more mass storage device terminals). Among them, the memory 320 and the storage media 330 may be transient storage or persistent storage. The program stored in the storage media 330 may include one or more modules (not shown in the figure), and each module may include a series of instruction operations on the communication device 300 of the VOIP device. Further, the processor 310 may be configured to communicate with the storage media 330 and execute a series of instruction operations in the storage media 330 on the communication device 300 of the VOIP device to implement the steps of the above-mentioned communication method of the VOIP device.

[0056] The communication device 300 of the VOIP device may further include one or more power supplies 340, one or more wired or wireless network interfaces 350, one or more input / output interfaces 360, and / or one or more operating systems 331, such as Windows Serve, Mac OS X, Unix, Linux, FreeBSD, etc. Those skilled in the art can understand that Figure 3The communication device structure of the VOIP device shown does not constitute a limitation on the communication device of the VOIP device provided by the present invention, and may include more or fewer components than shown, or combine certain components, or have different component arrangements.

[0057] The present invention also provides a computer-readable storage medium, which can be a non-volatile computer-readable storage medium or a volatile computer-readable storage medium. Instructions are stored in the computer-readable storage medium. When the instructions run on a computer, the computer is caused to execute the steps of the communication method of the VOIP device.

[0058] Those skilled in the art can clearly understand that for the convenience and brevity of description, the specific working processes of the above-described system, device, or unit can refer to the corresponding processes in the foregoing method embodiments and will not be elaborated herein.

[0059] If the integrated unit is implemented in the form of 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 solution of the present invention, in essence, or the part that contributes to the prior art, or all or part of this technical solution, can be embodied in the form of a software product. This computer software product is stored in a storage medium and includes several instructions for causing a computer device (which can be a personal computer, a server, or a network device, etc.) to execute all or part of the steps of the methods described in various embodiments of the present invention. The foregoing storage medium includes: various media such as USB flash drives, mobile hard disks, read-only memory (ROM), random access memory (RAM), magnetic disks, or optical discs that can store program codes.

[0060] As described above, the above embodiments are only used to illustrate the technical solutions of the present invention and are not intended to limit them; although the present invention has been described in detail with reference to the foregoing embodiments, those of ordinary skill in the art should understand that they can still modify the technical solutions described in the foregoing embodiments or perform equivalent replacements for some of the technical features; and these modifications or replacements do not cause the essence of the corresponding technical solutions to deviate from the spirit and scope of the technical solutions of various embodiments of the present invention.

Claims

1. A communication method for a VOIP device, characterized in that, The communication method of the VOIP device includes: Collect and encode the voice data of the VOIP device, detect the characteristics of the encoded data and the network environment status, generate a feature vector set and a channel stability index, and send the feature vector set and the channel stability index to the access point; Receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, process the encoded voice data packets, and form VOIP data packets to be transmitted; Send the VOIP data packets to other VOIP devices in the network through the wireless channel, and at the same time coordinate the transmission resources with other VOIP devices in the network, share the wireless channel, and adjust the transmission parameters according to the interference situation during the transmission process; Monitor the transmission performance of the sent VOIP data packets, conduct call quality evaluation, and record the quality evaluation results of the transmission performance and call quality evaluation.

2. The communication method of the VOIP device according to claim 1, wherein The collecting and encoding the voice data of the VOIP device, detecting the characteristics of the encoded data and the network environment status, and generating a feature vector set and a channel stability index includes: Collect the voice data of the VOIP device and perform encoding processing using a codec to obtain the encoded voice data, and analyze the encoded voice data to detect the codec type, packet generation period, and packet size, and obtain encoding feature parameters; Analyze the voice activity status, background noise level, and call mode in the encoded voice data to generate a load prediction vector, and measure the modulation and coding scheme level, maximum transmit power, and battery status supported by the VOIP device to generate a terminal capability vector; Periodically measure the received signal strength indication, signal-to-noise ratio, and current modulation and coding scheme of the wireless channel, calculate the channel fluctuation situation, and obtain the channel stability index; Combine the encoding feature parameters, the load prediction vector, the terminal capability vector, and the channel stability index into a feature vector set.

3. The communication method of the VOIP device according to claim 1, characterized in that, The receiving the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, and processing the encoded voice data packets to form VOIP data packets to be transmitted includes: Receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, and analyze the encoded voice data packets according to the channel stability index and the network resources to calculate the packet aggregation degree adjustment coefficient; Perform dynamic aggregation processing on multiple encoded voice data packets according to the aggregation degree adjustment coefficient to form an aggregated data packet, and during the aggregation process, perform real-time analysis on the encoded voice data packets to identify key voice data containing burst sounds or segments with obvious volume changes, and assign a priority mark to the key voice data; Select a transmission modulation and coding scheme for the aggregated data packet according to the scheduling information and the channel stability index, and combine the aggregated data packet, the priority mark, and the transmission modulation and coding scheme to form a VOIP data packet to be transmitted.

4. The communication method of the VOIP device according to claim 3, characterized in that, Analyzing the encoded voice data packets according to the channel stability index and the network resources, and calculating the packet aggregation degree adjustment coefficient includes: Performing feature analysis on the encoded voice data packets, extracting the packet size, frame interval, and payload distribution features, and generating voice data feature values; Processing the channel stability index, combining the historical data of the received signal strength indication and the signal-to-noise ratio, and calculating the channel reliability value of the current transmission window; Performing resource analysis on the network resources, detecting the available bandwidth, time slot allocation, and current network congestion status, and obtaining resource utilization parameters; According to the voice data feature values, the channel reliability value, and the resource utilization parameters, querying a preset aggregation parameter table, selecting the corresponding basic aggregation value, and correcting it in combination with the current voice activity status to obtain the packet aggregation degree adjustment coefficient.

5. The communication method of the VOIP device according to claim 1, characterized in that, Sending the VOIP data packets to other VOIP devices in the network through a wireless channel, coordinating transmission resources with other VOIP devices in the network, sharing the wireless channel, and adjusting transmission parameters according to the interference situation during the transmission process includes: Receiving a cooperative group mapping table from the access point, and identifying the affiliated cooperative group and the cooperative group transmission rules according to the network resources and scheduling information allocated by the access point through the cooperative group mapping table; According to the cooperative group transmission rules determined by the cooperative group mapping table, listening for the trigger frames sent by the access point, and sending the VOIP data packets to other VOIP devices in the network through the wireless channel according to the indication in the trigger frames; According to the cooperative group transmission rules, monitoring and analyzing the interference situation in the network, generating an interference map, and based on the interference map, adjusting the transmission parameters of the VOIP data packets, including adjusting the transmission power, modulation and coding scheme, and transmission timing, to adapt to the interference situation of the wireless channel.

6. The communication method of the VOIP device according to claim 5, characterized in that, Receiving a cooperative group mapping table from the access point, and identifying the affiliated cooperative group and the cooperative group transmission rules according to the network resources and scheduling information allocated by the access point through the cooperative group mapping table includes: Receiving the cooperative group mapping table sent by the access point through a management frame, and performing parsing processing on the network resources allocated by the access point to identify the resource configuration data structure of the resource units of the network resources; Performing timing analysis on the scheduling information provided by the access point, parsing the trigger period, target wake-up time, contention window parameters, and transmission opportunity limit of the trigger frame, determining the transmission timing and duration constraints of the VOIP device, and constructing a time slot access rule table; Performing a byte-by-byte comparison operation on the physical address of the VOIP device and the member address list in the cooperative group mapping table, and determining the cooperative group identifier of the cooperative group to which the VOIP device belongs through an address matching algorithm; Retrieving the corresponding transmission parameters from the cooperative group mapping table according to the cooperative group identifier, and combining the resource configuration data structure and the time slot access rule table to construct cooperative group transmission rules.

7. The communication method of the VOIP device according to claim 1, wherein Monitoring the transmission performance of the sent VOIP data packets, performing call quality evaluation, and recording the quality evaluation results of the transmission performance and call quality evaluation includes: Track the end-to-end delay, network jitter, and packet loss rate of the sent VOIP packets in real time, analyze the transmission control parameters, including the retransmission rate, sequence number interval, and round-trip time, evaluate the efficiency of the transmission process, and obtain the transmission performance; Analyze the decoded voice data, evaluate the voice clarity, voice continuity, and mean opinion score, conduct a call quality assessment, and obtain the quality assessment result of the call quality assessment; Record the transmission performance and the quality assessment result into the transmission performance reference database under different network environments.

8. A communication device of a VOIP device, characterized in that, The communication device of the VOIP device includes: A data processing module, configured to collect and encode the voice data of the VOIP device, detect the characteristics of the encoded data and the network environment state, generate a feature vector set and a channel stability index, and send the feature vector set and the channel stability index to the access point; A packet generation module, configured to receive the network resources and scheduling information allocated by the access point according to the feature vector set and the channel stability index, process the encoded voice packets, and form VOIP packets to be transmitted; A cooperative transmission module, configured to send the VOIP packets to other VOIP devices in the network through a wireless channel, coordinate transmission resources with other VOIP devices in the network, share the wireless channel, and adjust the transmission parameters according to the interference situation during the transmission process; A quality assessment module, configured to monitor the transmission performance of the sent VOIP packets, conduct a call quality assessment, and record the transmission performance and the quality assessment result of the call quality assessment.

9. A communication device for a VOIP device, characterized in that, The communication device of the VOIP device includes: a memory and at least one processor, and instructions are stored in the memory; The at least one processor calls the instructions in the memory to cause the communication device of the VOIP device to execute the steps of the communication method of the VOIP device according to any one of claims 1-7.

10. A computer-readable storage medium, on which instructions are stored, characterized in that, When the instructions are executed by the processor, the steps of the communication method of the VOIP device according to any one of claims 1-7 are implemented.

Images