A method, device and equipment for voiceband data transmission based on dynamic protocol link
By dynamically selecting anti-packet loss protocols in the Session Border Controller (SBC), and combining redundant coding and retransmission techniques, the parameters are dynamically adjusted according to network conditions. This solves the problems of packet loss and latency in voice with data in IP networks, and achieves high reliability and low latency transmission under harsh network conditions.
Patent Information
- Application Number
- CN202511695831.X
- Authority / Receiving Office
- CN · China
- Patent Type
- Patents(China)
- Current Assignee / Owner
- Filing Date
- 2025-11-19
- Publication Date
- 2026-02-24
- Estimated Expiration
- 2045-11-19
AI Technical Summary
Existing single-technology solutions are insufficient to provide adequate reliability and low latency under controllable bandwidth overhead when dealing with packet loss and latency issues in voice data services. Redundant coding/forward error correction technology has limited reliability in the face of sudden packet loss, while the latency uncertainty of retransmission technology makes it difficult to meet real-time requirements.
By dynamically selecting anti-packet loss protocols in the Session Border Controller (SBC), and combining redundancy coding and retransmission protocols, the redundancy parameters and retransmission counts are dynamically adjusted according to network conditions to achieve dynamic processing of voiceband data streams. This includes segmenting data blocks, applying redundancy coding, and acknowledgment retransmission mechanisms, ensuring data integrity and low latency under adverse network conditions.
It effectively resists high packet loss while controlling latency, improves the transmission quality of voice data, and realizes reliable transmission of voice data in IP networks.
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Figure CN121173426B_ABST
Abstract
Description
Technical Field
[0001] This invention relates to the field of information processing technology, and in particular to a method, apparatus and device for transmitting voice data based on a dynamic protocol link. Background Technology
[0002] In the field of telephone communication based on IP networks (such as the Internet), especially when carrying Voice Band Data (VBD) services (such as alarms, faxes, and modem signals), the inherent packet loss, latency, and jitter characteristics of the network pose a severe challenge to transmission quality. To combat packet loss, two main solutions have emerged. The first is forward error correction technology, represented by redundancy coding. This technology adds redundant information to data packets in advance at the sending end or directly sends multiple copies, such as redundancy coding (RED), so that the receiving end can directly recover from partial packet loss using redundant data without feedback or retransmission, thus not introducing additional latency. However, this method essentially sacrifices bandwidth for reliability; in the event of continuous bursts of packet loss, all redundant copies may be lost simultaneously, leading to recovery failure and limited reliability. The second is reliable transmission protocols based on retransmission, such as Quick UDP Internet Connection (QUIC) or Simple Packet Relay Transport (SPRT) protocols built on top of User Datagram Protocol (UDP). This type of technology uses an acknowledgment mechanism at the receiving end to trigger retransmission only when data packets are lost, thus ensuring reliability while saving bandwidth compared to blind redundancy. However, its core problem is that retransmission inevitably introduces at least one round-trip delay. When packet loss is severe or the network latency itself is high, the accumulated delay from multiple retransmissions will increase sharply, far exceeding the threshold that VBD services can tolerate, leading to communication interruption.
[0003] In summary, existing single-technology solutions face a fundamental contradiction when dealing with the stringent requirements of VBD services: Redundant coding / forward error correction (RED / FEC) technologies struggle to provide sufficient reliability against sudden packet loss within controllable bandwidth overhead; while SPRT-type retransmission technologies, although highly reliable, suffer from inherent latency uncertainties that fail to meet the real-time requirements of services. Although attempts have been made to simply combine the two, without meticulous design and parameter optimization, it is often impossible to find the optimal balance among the three interdependent factors of "bandwidth overhead," "transmission latency," and "packet loss resilience." This can result in either excessive redundancy wasting bandwidth and exacerbating network congestion, or aggressive retransmission strategies leading to uncontrolled latency. Summary of the Invention
[0004] This invention provides a method, apparatus, and device for transmitting data via voiceband based on a dynamic protocol link, which solves the problem of balancing packet loss resistance and low latency in communication networks.
[0005] To solve the above-mentioned technical problems, the technical solution of the present invention is as follows:
[0006] This invention provides a method for voice band data transmission based on a dynamic protocol link, applied to a Session Border Controller (SBC), comprising:
[0007] The system receives a voice data stream sent by a first client front-end device (CPE) through a first Internet Protocol link. The first client front-end device (CPE) communicates with a first voice data terminal device (CTP), and the voice data stream is sent by the first CTP to the first client front-end device (CPE).
[0008] When the transmission quality of the first Internet Protocol link is detected to be lower than a preset threshold, at least one target anti-packet loss protocol is dynamically selected for processing of the voiceband data stream to obtain the target voiceband data stream;
[0009] The target voiceband data stream is sent to the second client front-end device (CPE) via the second Internet Protocol link, and then sent to the second voiceband data terminal device via the second client front-end device (CPE).
[0010] Optionally, at least one target packet loss mitigation protocol is dynamically selected from the voiceband data stream for processing to obtain a target voiceband data stream, including:
[0011] The voice data stream is segmented to obtain multiple voice data blocks;
[0012] The multiple voiceband data blocks are processed using a first target anti-packet loss protocol to obtain a composite message sequence;
[0013] The composite message sequence is processed using a second target anti-packet loss protocol to obtain the target voiceband data stream.
[0014] Optionally, the voiceband data stream is segmented to obtain multiple voiceband data blocks, including:
[0015] According to a preset segmentation period, the voice data stream is segmented to obtain multiple voice data blocks, which are arranged in chronological order and assigned sequence numbers.
[0016] Optionally, the plurality of voiceband data blocks are processed using a first target anti-packet loss protocol to obtain a composite message sequence, including:
[0017] Obtain the transmission quality data of the first Internet Protocol link;
[0018] Based on the transmission quality data, determine the target redundancy parameters of the first target packet loss resistance protocol;
[0019] Based on the target redundancy parameters, the multiple voiceband data blocks are copied to obtain multiple redundant voiceband data blocks;
[0020] The multiple voiceband data blocks and the multiple redundant voiceband data blocks are encapsulated using a first target anti-packet loss protocol to obtain a composite message sequence.
[0021] Optionally, obtain transmission quality data of the first Internet Protocol link, including:
[0022] The voice data stream is extracted and processed to obtain multiple voice data packets;
[0023] The integrity of the multiple voice packet data is checked. If there are missing voice packet data, message data is generated. The message data includes the sequence number and time information of the missing voice packet data.
[0024] The message data is statistically processed to obtain the transmission quality data of the first Internet Protocol link.
[0025] Optionally, based on the transmission quality data, the target redundancy parameters of the first target packet loss resistance protocol are determined, including:
[0026] Based on the transmission quality data, the average packet loss rate and burst packet loss index are determined;
[0027] The transmission success rate is determined based on the average packet loss rate.
[0028] Obtain preset redundancy parameters;
[0029] When the burst packet loss index is greater than the first preset value, the preset redundancy parameter is reduced to obtain the target redundancy parameter;
[0030] When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset redundancy parameters are incrementally processed to obtain the target redundancy parameters.
[0031] Optionally, the composite message sequence is processed using a second target anti-packet loss protocol to obtain a target voiceband data stream, including:
[0032] Get the preset retransmission parameters;
[0033] When the burst packet loss index is greater than the first preset value, the preset retransmission parameter is incrementally processed to obtain the target retransmission parameter;
[0034] When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset retransmission parameters are reduced to obtain the target retransmission parameters.
[0035] Based on the target retransmission parameters, the composite message sequence is encapsulated using the second target anti-packet loss protocol to obtain the target voiceband data stream.
[0036] This invention also provides a voiceband data transmission device based on a dynamic protocol link, applied to a session boundary controller (SBC), comprising:
[0037] The transceiver module is used to receive the voice data stream sent by the first client front-end device (CPE) through the first Internet Protocol link. The first client front-end device (CPE) communicates with the first voice data terminal device (CTP), and the voice data stream is sent by the first voice data terminal device to the first client front-end device (CPE).
[0038] The processing module is used to dynamically select at least one target anti-packet loss protocol to process the voiceband data stream when the transmission quality of the first Internet Protocol link is lower than a preset threshold, so as to obtain the target voiceband data stream.
[0039] The transceiver module is also used to send the target voiceband data stream to the second client front-end device (CPE) via the second Internet Protocol link, and to send the target voiceband data stream to the second voiceband data terminal device via the second client front-end device (CPE).
[0040] This invention also provides a computing device, including: a processor and a memory storing a computer program, wherein the computer program, when run by the processor, executes the above-described method.
[0041] This invention also provides a computer-readable storage medium storing instructions that, when executed on a computer, cause the computer to perform the above-described method.
[0042] The technical solution of the present invention has at least the following effects:
[0043] The above-described solution of the present invention receives a voice data stream sent by a first client front-end device (CPE) via a first Internet Protocol link. The first client front-end device (CPE) communicates with a first voice data terminal device (VDT), and the voice data stream is sent by the first VDT to the first client front-end device (CPE). When the transmission quality of the first Internet Protocol link is detected to be lower than a preset threshold, at least one target anti-packet loss protocol is dynamically selected to process the voice data stream to obtain a target voice data stream. The target voice data stream is then sent to a second client front-end device (CPE) via a second Internet Protocol link, and the second client front-end device (CPE) then sends the target voice data stream to the second voice data terminal device. This effectively resists high packet loss while controlling latency, thus improving the voice data quality. Attached Figure Description
[0044] Figure 1 This is a flowchart of a voiceband data transmission method based on a dynamic protocol link provided in an embodiment of the present invention;
[0045] Figure 2 This is a communication network structure diagram of the voice-band data transmission method based on dynamic protocol links provided in an embodiment of the present invention;
[0046] Figure 3 This is a structural diagram of a voice-band data transmission device based on a dynamic protocol link provided in an embodiment of the present invention;
[0047] Figure 4 This is a schematic diagram of the structure of the computing device provided in an embodiment of the present invention. Detailed Implementation
[0048] Exemplary embodiments of the invention will now be described in more detail with reference to the accompanying drawings. While exemplary embodiments of the invention are shown in the drawings, it should be understood that the invention may be implemented in various forms and should not be limited to the embodiments set forth herein. Rather, these embodiments are provided so that this invention will be thorough and complete, and will fully convey the scope of the invention to those skilled in the art.
[0049] like Figure 1 As shown, an embodiment of the present invention proposes a voiceband data transmission method based on a dynamic protocol link, applied to a session boundary controller (SBC), comprising:
[0050] Step 11: Receive the voice data stream sent by the first client front-end device (CPE) through the first Internet Protocol link. The first client front-end device (CPE) communicates with the first voice data terminal device (CTP). The voice data stream is sent by the first voice data terminal device to the first client front-end device (CPE).
[0051] Step 12: When the transmission quality of the first Internet Protocol link is detected to be lower than a preset threshold, at least one target anti-packet loss protocol is dynamically selected for processing of the voiceband data stream to obtain the target voiceband data stream;
[0052] Step 13: Send the target voiceband data stream to the second client front-end device (CPE) via the second Internet Protocol link, and then send the target voiceband data stream to the second voiceband data terminal device via the second client front-end device (CPE).
[0053] In step 11 of this embodiment, the voice data generated by the voice data terminal device (such as a fire alarm panel, fax machine, or SCADA MODEM), for example, analog voice data, is first sent to the first client front-end device (CPE) connected to it. The CPE device is responsible for converting the analog signal into IP data packets and transmitting them to the Session Border Controller (SBC) via a first Internet Protocol link, such as an Internet connection. This first Internet Protocol link is the core connection between the CPE and the SBC.
[0054] As the central node, the SBC continuously listens for and receives data streams from the first CPE device. The data stream is typically encapsulated in RTP (Real-Time Transport Protocol) or a similar format, but given the sensitivity of voiceband data to packet loss, the SBC will initially monitor link transmission quality parameters (such as packet loss rate, latency, and jitter). The first CPE device can employ multi-path technology, such as QUIC Multipath, to enhance reliability.
[0055] In step 12, the SBC incorporates a quality monitoring module to evaluate the transmission quality of the first Internet Protocol link in real time. Preset thresholds are set based on the sensitivity of the voiceband data, such as a packet loss rate exceeding 0.5%, latency exceeding 100ms, or jitter exceeding 20ms. When quality is detected to be below the threshold (e.g., an increase in packet loss rate or a sudden packet loss event), the SBC triggers a dynamic protocol selection mechanism.
[0056] SBC dynamically selects the most suitable packet loss mitigation scheme from multiple protocols. The selection criteria include current network conditions (e.g., packet loss mode: random or bursty packet loss), bandwidth cost, and latency constraints; protocol options include:
[0057] (1) Combining RED / FEC with SPRT; for example, using a combination of "3-fold RED + SPRT with a maximum retransmission count of 2". RED (Redundancy Coding) combats random packet loss by sending data copies in multiple packets, while SPRT (Reliable Transport Protocol) combats burst packet loss based on an acknowledgment retransmission mechanism. This combination balances latency and bandwidth, achieving a session success rate of 98.51% when the packet loss rate is below 1%.
[0058] (2) Variant schemes; Based on network statistics, SBC may choose other combinations, such as "2-layer RED + 3-layer SPRT" to reduce bandwidth overhead (increase bandwidth by 50%, but increase latency by 3 RTT), or "0-layer RED + 2-layer SPRT + 3 retransmissions for the last time" to minimize initial latency. In addition, when network conditions are good, standard reliable protocols such as TCP or QUIC may be used directly.
[0059] The dynamic decision-making process includes: SBC automatically adjusting the number of RED layers and SPRT retransmission count by analyzing real-time packet loss rate, packet loss burstiness (such as the duration of continuous packet loss), and historical performance data through algorithms. For example, if random packet loss is detected as the main issue, RED redundancy is increased first; if burst packet loss is the main issue, the number of SPRT retransmissions is increased. This ensures that the voiceband data stream maintains its integrity even under harsh network conditions.
[0060] Data processing includes: The SBC processes the received raw voiceband data stream using the selected anti-packet loss protocol. For example, in SPRT mode, the data stream is segmented into packets and sequence numbers are added, awaiting acknowledgment; in RED mode, redundant packets are generated. The processed output is the target voiceband data stream, whose format can be changed to a reliable transmission-based encapsulation (such as a QUIC frame or proprietary protocol data unit), thereby enhancing anti-packet loss capabilities.
[0061] In step 13, the SBC utilizes intelligent routing technology to forward the target voiceband data stream to the second CPE device via a second Internet Protocol link. The second link includes... Figure 2 Link 2 shown (connects SBC to client front-end device B).
[0062] After receiving the target voiceband data stream, the second CPE performs reverse processing: decapsulating the packet loss prevention protocol, recovering the original voiceband data signal (such as analog audio or data), and sending it to the second voiceband data terminal device (such as the receiving fax machine or alarm controller). The second CPE may also employ multi-link technology to ensure reception reliability, thereby completing the complete voiceband data communication from the first terminal to the second terminal.
[0063] The technical solution described in this embodiment effectively solves the packet loss and latency problems of IP network voice data transmission through SBC's dynamic protocol selection and multi-link management. Its core advantage lies in adaptively adjusting the anti-packet loss strategy according to real-time network conditions, combined with RED / SPRT hybrid technology, to improve reliability while ensuring low latency.
[0064] In an optional embodiment of the present invention, step 12, which involves dynamically selecting at least one target anti-packet loss protocol to process the voiceband data stream to obtain a target voiceband data stream, may include:
[0065] Step 121: The voice data stream is segmented to obtain multiple voice data blocks;
[0066] Step 122: Select a first target anti-packet loss protocol to process the multiple voiceband data blocks to obtain a composite message sequence;
[0067] Step 123: Select the second target anti-packet loss protocol to process the composite message sequence to obtain the target voiceband data stream.
[0068] In step 121 of this embodiment, basic data units are prepared for subsequent packet loss mitigation. The original voice data stream, such as a continuous modulated signal from a fax machine or alarm panel, is an uninterrupted bit stream that must be segmented into fixed-size data blocks suitable for network transmission. For example, a common packetization period is 20 milliseconds, meaning that every 20ms, the analog signal is sampled, encoded, and a fixed-size digital voice data block is generated. Segmentation ensures that the data can be encapsulated into independent network packets (such as RTP packets). Each data block carries timestamp information so that it can be reconstructed at the receiving end in the correct time and order.
[0069] In step 122, the first target anti-packet loss protocol refers to Redundancy Coding (RED) technology. For each newly generated Voice Band Data (VBD) block (which can be called a "master data block," denoted as D),... n The sender does not only send D n Instead of itself, it is combined with one or more previous data blocks, such as D. n-1 and D n-2 These components are combined according to a certain redundancy strategy to form a composite message. For example, in the "triple RED" scheme, the currently sent message Packetn contains not only the current main data block D, but also other components. n It also redundantly includes the first two data blocks D. n-1 and D n-2 A copy.
[0070] After this processing step, the original data block sequence is transformed into a composite message sequence. Each message contains redundant information about the current and historical data. Thus, during transmission, even if the packet... n If lost, only the subsequent packets n+1 or Packet n+2 If the message arrives successfully, the receiving end can extract the redundant D from these subsequent messages. n The data can be recovered without waiting for retransmission and without introducing additional latency.
[0071] In step 123, the second target packet loss mitigation protocol refers to the SPRT protocol based on acknowledgments. In this step, each packet in the composite message sequence output in step 122 needs to be sent through the transmission channel. The SBC maintains a timer for each packet and waits for an acknowledgment (ACK) signal from the second CPE device. If no ACK is received for a packet within one round-trip time (RTT), the sender determines that the packet has been lost.
[0072] At this point, the SBC will retransmit the lost packets, but there is a strict upper limit to the number of retransmissions (e.g., 2 times) to avoid infinitely increasing latency due to unlimited retransmissions, thus ensuring the real-time requirements of the voiceband data. After processing by the SPRT protocol, the final generated target voiceband data stream ensures that even in the event of continuous packet loss that the RED layer cannot compensate for, data can still be reliably delivered to the receiving end through controlled retransmissions.
[0073] In an optional embodiment of the present invention, step 121, which involves segmenting the voice data stream to obtain multiple voice data blocks, may include:
[0074] Step 1211: According to the preset segmentation period, the voice data stream is segmented to obtain multiple voice data blocks, which are arranged in chronological order and assigned sequence numbers.
[0075] In step 1211 of this embodiment, the length of the segmentation period directly affects the transmission latency, bandwidth efficiency, and packet loss resistance. A shorter segmentation period results in lower packetization latency and smaller packets, leading to less data loss in the event of packet loss. However, the disadvantage is that the overhead of protocol headers (such as RTP / UDP / IP headers) relative to the payload increases, increasing bandwidth consumption and the processing burden on network devices. Conversely, a longer segmentation period reduces the proportion of protocol header overhead and improves bandwidth utilization. However, the disadvantage is increased packetization latency, and the loss of a single packet leads to a longer data interruption, resulting in a more severe impact on services. In this embodiment, a preset segmentation period of 20ms is used to ensure processing simplicity and timing stability.
[0076] Voice-on data terminals (such as fax machines and alarm panels) generate continuous analog or unencapsulated digital signal streams. The Customer Pre-Employer (CPE), acting as a signal converter, buffers and slices the input continuous data stream within a precisely preset segmentation period. For each period, the CPE packages all received signal data into a single voice-on data block. This data block is the basic unit for subsequent packet loss mitigation. The segmented data blocks are generated in chronological order, and the receiving end needs to reconstruct these data blocks in exactly the same order to correctly reconstruct the original signal. To ensure the receiving end can identify and correct the order of data blocks in unreliable IP networks, the transmitting end (CPE) assigns a unique and incremental sequence number to each generated data block.
[0077] The receiving end reassembles data blocks into a continuous stream based on sequence numbers. Even if the order of arrival of messages is out of order due to different network paths, they can still be sorted by sequence numbers. If the receiving end finds that the sequence numbers are not consecutive (for example, it receives messages with sequence numbers 1, 2, and 4, but is missing 3), it can determine that packet loss has occurred. The receiving end informs the sending end which data blocks have been successfully received and which have been lost by confirming the next expected sequence number.
[0078] In an optional embodiment of the present invention, step 122, processing the plurality of voiceband data blocks using a first target anti-packet loss protocol to obtain a composite message sequence, may include:
[0079] Step 1221: Obtain the transmission quality data of the first Internet Protocol link;
[0080] Step 1222: Determine the target redundancy parameters of the first target anti-packet loss protocol based on the transmission quality data;
[0081] Step 1223: Based on the target redundancy parameters, the multiple voiceband data blocks are copied to obtain multiple redundant voiceband data blocks;
[0082] Step 1224: Encapsulate the plurality of voiceband data blocks and the plurality of redundant voiceband data blocks using the first target anti-packet loss protocol to obtain a composite message sequence.
[0083] In step 1221 of this embodiment, the SBC continuously monitors the performance metrics of the first Internet Protocol link (i.e., the uplink from the CPE to the SBC). The transmission quality data mainly includes:
[0084] Real-time packet loss rate: The packet loss rate directly determines the dynamic packet loss mitigation measures required.
[0085] Packet loss mode: Packet loss mode is used to distinguish between random packet loss and sudden packet loss. It can be determined by analyzing the number and distribution of consecutively lost packets.
[0086] In step 1222, the target redundancy parameter refers to the number of redundancy coding layers. When network quality is good (e.g., packet loss rate is below 0.1%, and random packet loss is predominant), a lower number of redundancy layers can be selected, such as single-layer RED or double-layer RED. This can provide basic protection while maximizing bandwidth savings. When network quality deteriorates (e.g., packet loss rate rises to 0.5% to 1%, but random packet loss is still predominant), a higher number of redundancy layers, such as triple-layer RED, is required, which can effectively combat moderate random packet loss. When burst packet loss is detected, RED has limited effectiveness against burst packet loss. In this case, the decision system can appropriately reduce the number of RED layers and delegate the main task of combating packet loss to subsequent SPRT layers. For example, using double-layer RED instead of triple-layer RED can save bandwidth to cope with the additional traffic that may be caused by retransmission. This process dynamically balances bandwidth, latency, and packet loss rate based on the characteristics of network packet loss.
[0087] In step 1223, specific redundant data is generated based on the strategy calculated in the previous step; this process involves data copying and caching. The sending end maintains a first-in-first-out (FIFO) buffer of size n (the target redundancy layer number) to store the n most recently generated call data blocks and their sequence numbers. For example, when the target redundancy parameter n=3, the buffer will always retain the current data block (D). k ) and the previous two data blocks (D k-1 and D k-2 For each newly generated data block, not only is the data block itself used, but the previous n-1 data blocks are also read from the buffer as "redundant callband data blocks" for later use.
[0088] In step 1224, for the message to be sent (i.e., the original data block D) k Its payload is not a single data block, but a composite structure. This structure includes:
[0089] (1) Master data block: the current original data block D k .
[0090] (2) Redundant data blocks: one or more previously cached data blocks, such as D k-1 and D k-2 (For triple RED).
[0091] This composite payload can be encapsulated into a transport protocol. For example, it can be encapsulated into an RTP message (using the RTP redundant payload format), and then wrapped with UDP and IP headers to form a RED message.
[0092] Each sent message contains redundant information about the current data and historical data. This generates a continuous sequence of composite messages. When the receiving end receives any message, it can extract the main data block and the redundant data block. If a message (such as one carrying D...)... k-1 If the message is lost, the receiver can retrieve it from subsequent messages (such as those carrying D). k The message, its redundant part contains D k-1 The error is restored in the process, thus achieving the effect of forward error correction (FEC) with almost no increase in latency.
[0093] In an optional embodiment of the present invention, step 1221, obtaining the transmission quality data of the first Internet Protocol link, may include:
[0094] Step 12211: Extract and process the voice data stream to obtain multiple voice data packets;
[0095] Step 12212: Perform integrity verification on the multiple voice packet data. If there are missing voice packet data, generate message data. The message data includes the sequence number and time information of the missing voice packet data.
[0096] Step 12213: Perform statistical processing on the message data to obtain the transmission quality data of the first Internet Protocol link.
[0097] In step 12211 of this embodiment, the SBC identifies and separates independent transmission units carrying valid voiceband data from the received network bitstream. First, the SBC receives a series of IP datagrams from the first Internet Protocol link (i.e., the link from the CPE to the SBC). The SBC decapsulates these IP datagrams layer by layer according to the established communication protocol stack. Specifically, this process includes: first, stripping the IP header and UDP header, and then parsing out the RTP message or other proprietary reliable protocol-defined message structure carrying the actual voiceband data payload. Each successfully parsed, complete application layer message is considered a voiceband data packet. This extraction process allows the system to access the key information carried within each data packet for analyzing transmission quality, namely the RTP sequence number and RTP timestamp.
[0098] In step 12212, each time the SBC successfully extracts and verifies a voice packet, it checks whether its RTP sequence number is consecutive to the expected next sequence number. If the received packet sequence number jumps (for example, the expected sequence number is 100, but the next correctly received sequence number is 102), the system determines that the packet with sequence number 101 has been lost. Once packet loss is confirmed, the system immediately generates a structured packet loss record, namely "message data". This record contains two core pieces of information:
[0099] (1) If missing, include the sequence number of the packet: that is, the RTP sequence number that was detected as lost (e.g., 101), which is the unique identifier for locating the packet loss.
[0100] (2) Time information: This includes the timestamp when the packet loss was discovered, and can also be combined with the timestamp of the received message to calculate the original sending time of the lost message.
[0101] In step 12213, SBC performs centralized statistical analysis on all packet loss records within a sliding statistical time window (e.g., the past 10 seconds, 30 seconds, or 1 minute).
[0102] SBC analyzes the distribution patterns of sequence numbers and times in packet loss records: if packet loss events occur sparsely and independently, they are judged as random packet loss; if multiple consecutive data packets with the same sequence number (e.g., data corresponding to 50ms or longer) are lost in a concentrated manner, they are judged as burst packet loss.
[0103] After the above statistical processing, the final transmission quality data is a comprehensive evaluation report, which mainly includes: real-time average packet loss rate and packet loss mode identifier.
[0104] In an optional embodiment of the present invention, step 1222, determining the target redundancy parameters of the first target packet loss resistance protocol based on the transmission quality data, may include:
[0105] Step 12221: Determine the average packet loss rate and burst packet loss index based on the transmission quality data;
[0106] Step 12222: Determine the transmission success rate based on the average packet loss rate;
[0107] Step 12223: Obtain the preset redundancy parameters;
[0108] Step 12224: When the burst packet loss index is greater than the first preset value, the preset redundancy parameter is reduced to obtain the target redundancy parameter.
[0109] Step 12225: When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset redundancy parameters are incrementally processed to obtain the target redundancy parameters.
[0110] In step 12221 of this embodiment, the average packet loss rate and burst packet loss index are determined based on the statistical results of the transmission quality data, wherein the formula for calculating the average packet loss rate is:
[0111] ;
[0112] in, This represents the average packet loss rate. To count the total number of data packets lost within the window; This is to count the total number of data packets expected to be received within the window.
[0113] Average packet loss rate reflects the proportion of data packets lost within a statistical time window. It is a macro-level indicator of network quality deterioration.
[0114] The burst packet loss index is a derivative index used to quantify the burstiness of packet loss. This index can be set as the maximum number of consecutive packet losses per unit of time.
[0115] In step 12222, the transmission success rate refers to the probability that the entire message will not be lost in a single transmission attempt. The calculation process includes:
[0116] (1) Probability of single message loss:
[0117] ;
[0118] in, This represents the probability of a single message being lost. The average packet loss rate is N; the number of retransmissions is N.
[0119] (2) Probability of no packet loss (transmission success rate):
[0120] ;
[0121] in, Transmission success rate; Total number of messages , For message duration, Packaging cycle;
[0122] In step 12223, the preset redundancy parameter is a default value pre-configured by the system, based on historical experience or assumptions about general network conditions. For example, the preset redundancy parameter can be set to 3 based on past experience, i.e., triple RED. This preset value represents the standard operating mode when no deep dynamic analysis is performed.
[0123] In step 12224, when a severe burst of packet loss is detected, the investment in the RED strategy can be reduced, because the RED effect is not significant at this time, and SPRT retransmission should be relied upon more. The first preset value is an empirical threshold used to define what level of packet loss burstiness is considered "severe". For example, the system may be set to trigger this condition when more than 3 consecutive packet losses are detected (i.e., the first preset value is 3); at this time, the redundancy parameter is reduced. For example, if the preset redundancy parameter is 3 (i.e., 3-layer RED), it can be reduced to 2 or 1.
[0124] RED mitigates packet loss by redundantly carrying historical data in different packets. However, during periods of continuous burst packet loss, the current packet and its redundant copy (carrying earlier historical data) are likely to be lost simultaneously in the network, causing the RED mechanism to completely fail. Continuing with high redundancy in this situation only consumes bandwidth and increases network load without achieving the desired packet loss mitigation effect. Therefore, reducing redundancy saves bandwidth, allowing the subsequent SPRT layer to perform necessary retransmissions.
[0125] In step 12225, when the network quality is good and stable (mainly random packet loss), RED protection is enhanced to further improve reliability in a near-zero latency manner, thereby reducing reliance on high-latency retransmission mechanisms. A transmission success rate greater than the second preset value indicates that the basic network quality is very good. For example, the second preset value can be set to 99% (corresponding to a packet loss rate of 1%), indicating that the network is in an excellent state. A burst packet loss index less than the third preset value indicates that there is no serious burst packet loss in the network, and packet loss is mainly random and sporadic. The third preset value can be set to a natural number less than the first preset value (e.g., 2 or 1). When this condition is met, the preset 3-layer RED can be upgraded to 4-layer RED.
[0126] In an optional embodiment of the present invention, step 123, processing the composite message sequence using a second target anti-packet loss protocol to obtain the target voiceband data stream, may include:
[0127] Step 1231: Obtain the preset retransmission parameters;
[0128] Step 1232: When the burst packet loss index is greater than the first preset value, the preset retransmission parameters are incrementally processed to obtain the target retransmission parameters.
[0129] Step 1233: When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset retransmission parameters are reduced to obtain the target retransmission parameters.
[0130] Step 1234: Based on the target retransmission parameters, encapsulate the composite message sequence using the second target anti-packet loss protocol to obtain the target voiceband data stream.
[0131] In step 1231 of this embodiment, the preset retransmission parameter refers to the maximum number of retransmissions, which is a predefined safety value of the system. This parameter can be set to 2 times as the preset, which is a balance between reliability and latency, because 2 retransmissions will only increase the latency by a maximum of 2 RTT, while significantly improving the ability to resist packet loss.
[0132] In step 1232, when the burst packet loss index exceeds a first preset value, it indicates that the network has entered a severe state dominated by burst packet loss. At this point, the maximum number of retransmissions needs to be increased beyond the preset number of retransmissions; for example, the maximum number of retransmissions can be increased from 2 (preset) to 3 or 4. This is because burst packet loss means that packets are lost consecutively, and more retransmission attempts may be needed for a single copy to successfully traverse the network. Increasing the retransmission limit improves the final delivery probability in this extreme situation. Although retransmission increases latency, in burst packet loss scenarios, prioritizing the establishment and maintenance of communication is crucial.
[0133] This step is triggered in conjunction with step 12224 (reducing RED redundancy). When the system detects that a sudden packet loss has caused the RED to fail almost completely, it immediately performs two operations: front-end (RED) throttling and back-end (SPRT) boosting, avoiding wasting bandwidth on invalid REDs and reserving network resources for more efficient retransmission mechanisms.
[0134] In step 1233, when network quality is good and stable, the number of retransmissions is reduced to minimize communication latency. This step is triggered in conjunction with step 12225 (increasing RED redundancy). At this time, the system performs two operations: front-end (RED) enhancement and back-end (SPRT) simplification. Because the enhanced RED can already solve most random packet loss problems, the probability of triggering SPRT retransmission is greatly reduced. Setting the maximum number of retransmissions to 1 means that the latency increase in the worst case is only 1 RTT. This minimizes the total communication latency, further improves the success rate of time-sensitive services such as fax handshakes, reduces unnecessary retransmissions, and saves network bandwidth and processing resources.
[0135] In steps 1234, a dynamically determined retransmission strategy is applied to encapsulate the data stream processed by RED into a highly reliable data stream. This involves using the composite message sequence output in step 122 as input and encapsulating it using the second target packet loss mitigation protocol (SPRT). This process includes:
[0136] (1) Add a sequence number: assign a unique sequence number at the reliable transport layer to each composite message.
[0137] (2) Start timer: Start a retransmission timer for each sent message. The timeout is dynamically calculated based on the measured RTT.
[0138] (3) Execute retransmission logic: Manage retransmission behavior according to the target retransmission parameters (new maximum number of retransmissions) determined in step 1232 or 1233. When packet loss occurs, the protocol will retransmit, but the number of retransmissions will not exceed this dynamically set upper limit.
[0139] After this processing step, the final output is a series of data packets carried over a reliable connection. These packets not only contain RED redundancy but are also protected by controlled, latency-managed retransmission. This data stream is then sent to a second Internet Protocol link and transmitted to the target CPE.
[0140] A specific embodiment of the voice data transmission method based on dynamic protocol links provided in this invention is as follows:
[0141] Step 1: Voiceband data acquisition and initial encapsulation;
[0142] First, assume that the fire alarm panel is triggered, generating a continuous analog alarm sound signal.
[0143] CPE device A samples and encodes the analog signal with a preset segmentation period of 20ms, generating a series of voice data blocks. Each data block is assigned a sequence number that increases sequentially in time.
[0144] CPE device A encapsulates these data blocks into RTP packets, ready to be sent over the Internet.
[0145] Step 2: First Internet Protocol link transmission and quality monitoring;
[0146] CPE device A sends a data stream to SBC via the Internet link (link 1). SBC receives data packets from link 1, decapsulates them, and extracts the voice packet data (RTP message).
[0147] Step 3: Dynamic transmission quality assessment;
[0148] The SBC checks the sequence numbers of received RTP packets. If the sequence numbers are consecutive, no packet loss is considered. If a sequence number jump is detected (e.g., receiving Seq100 and then immediately receiving Seq102), Seq101 is determined to be lost, and a packet loss record is generated containing {lost sequence number: 101, discovery time: T1}. The SBC performs statistical analysis on all packet loss records within a sliding time window (e.g., 30 seconds). It calculates the average packet loss rate (e.g., 0.8%) and the burst packet loss index (e.g., the longest consecutive packet loss within the statistical window is 3, i.e., 60ms of data loss).
[0149] Step 4: Dynamically select and execute the first target anti-packet loss protocol (RED);
[0150] The default redundancy parameter is 3 (i.e., triple RED).
[0151] Scenario A - Sudden Packet Loss: If the sudden packet loss index (3) is greater than the first preset value (2), the SBC determines that the network is dominated by sudden packet loss. At this time, the preset redundancy parameter is reduced and the target redundancy parameter is set to 2 (i.e., double RED).
[0152] Scenario B - Random Packet Loss: If the transmission success rate (99.2%) corresponding to the average packet loss rate (0.8%) is greater than the second preset value (99%), and the burst packet loss index (1) is less than the third preset value (2), the SBC determines that the network quality is good and that random packet loss is the main issue. At this time, the preset redundancy parameter is incrementally processed, and the target redundancy parameter is set to 4 (i.e., 4-fold RED).
[0153] Perform RED encapsulation: The SBC copies and encapsulates the data block according to the target redundancy parameter (2). For the current data block D k The composite message it generates not only contains D k It also redundantly contains the previous data block D k-1 A copy. This yields the composite message sequence.
[0154] Step 5: Dynamically select and execute the second target packet loss mitigation protocol (SPRT);
[0155] The default retransmission parameter is 2 (i.e., a maximum of 2 retransmissions).
[0156] Scenario A - Sudden Packet Loss: Since the sudden packet loss index is greater than the threshold, SBC performs incremental processing on the preset retransmission parameters and sets the target retransmission parameter to 3 (i.e., a maximum of 3 retransmissions).
[0157] Scenario B—Random Packet Loss: Due to good network quality, SBC reduces the preset retransmission parameters, setting the target retransmission parameter to 1 (i.e., a maximum of 1 retransmission). In cases of good network quality, a method of "0 RED + n SPRT + last retransmission m times" can also be used (where n and m are natural numbers, such as "0 RED + 2 SPRT + last retransmission 3 times").
[0158] SPRT encapsulation is performed: The SBC uses the SPRT protocol to reliably encapsulate the composite message sequence; it assigns a transport layer sequence number to each message, starts a retransmission timer, and allows up to 3 retransmission attempts for lost messages according to the target retransmission parameter (3). Finally, a highly reliable target voiceband data stream is obtained.
[0159] Step 6: Intelligent routing and transmission via the second Internet Protocol link;
[0160] Intelligent routing decision: SBC identifies that the data stream originates from a fire alarm (ALARM service).
[0161] Send to target CPE: Data is called and transmitted to the second CPE (CPE device B) via the Internet network.
[0162] Step 7: Data reception and restoration;
[0163] After receiving the data stream from the SBC via the IP link, CPE device B acknowledges the received data stream to ensure its integrity and order. If packet loss occurs, it requests retransmission (within the retransmission limit set by the SBC). It decapsulates the composite message and uses redundant data to recover the original data blocks lost during transmission. The recovered, ordered voiceband data blocks are decoded to restore a continuous analog alarm signal, which is then sent to voiceband data terminal B (alarm receiver), successfully triggering the alarm.
[0164] The technical solution described in this embodiment combines the RED and SPRT protocols to effectively improve the transmission quality of audio and video services in the Internet IP network environment.
[0165] First, this technical solution utilizes RED technology to carry multiple copies of data within the same message, combating random packet loss without incurring latency costs. Then, it introduces the SPRT protocol, performing a limited number of retransmissions only upon confirmation of packet loss, thus managing sudden, continuous packet loss with a controllable increase in latency. This hybrid mechanism overcomes the unreliability of pure redundancy schemes under sudden packet loss while avoiding the excessive latency problem of pure retransmission schemes.
[0166] Secondly, even under severe network conditions with a packet loss rate as high as 10%, this technical solution can still guarantee a 300-second call session with a complete packet loss probability of over 98.5%; at the same time, it strictly controls the transmission latency to within twice the round-trip time (RTT), effectively balancing the contradictions between bandwidth, packet loss, and latency.
[0167] Finally, the technical solution proposes a variety of parameter combination variations, which can dynamically adjust redundancy and retransmission strategies according to the actual network packet loss characteristics, further enhancing the adaptability and practicality of the technical solution in different network environments.
[0168] like Figure 3 As shown, this embodiment of the invention also provides a voiceband data transmission device 30 based on a dynamic protocol link, applied to a session boundary controller (SBC), comprising:
[0169] Transceiver module 31 is used to receive voice data stream sent by the first client front-end device CPE through the first Internet Protocol link. The first client front-end device CPE communicates with the first voice data terminal device. The voice data stream is sent by the first voice data terminal device to the first client front-end device CPE.
[0170] Processing module 32 is used to dynamically select at least one target anti-packet loss protocol to process the voiceband data stream when the transmission quality of the first Internet Protocol link is lower than a preset threshold, so as to obtain a target voiceband data stream.
[0171] The transceiver module 31 is also used to send the target voiceband data stream to the second client front-end device (CPE) via the second Internet Protocol link, and to send the target voiceband data stream to the second voiceband data terminal device via the second client front-end device (CPE).
[0172] Optionally, processing module 32 is specifically used for:
[0173] The voice data stream is segmented to obtain multiple voice data blocks;
[0174] The multiple voiceband data blocks are processed using a first target anti-packet loss protocol to obtain a composite message sequence;
[0175] The composite message sequence is processed using a second target anti-packet loss protocol to obtain the target voiceband data stream.
[0176] Optionally, processing module 32 is specifically used for voice tape data:
[0177] According to a preset segmentation period, the voice data stream is segmented to obtain multiple voice data blocks, which are arranged in chronological order and assigned sequence numbers.
[0178] Optionally, processing module 32 is specifically used for the voiceband data target anti-packet loss protocol:
[0179] Obtain the transmission quality data of the first Internet Protocol link;
[0180] Based on the transmission quality data, determine the target redundancy parameters of the first target packet loss resistance protocol;
[0181] Based on the target redundancy parameters, the multiple voiceband data blocks are copied to obtain multiple redundant voiceband data blocks;
[0182] The multiple voiceband data blocks and the multiple redundant voiceband data blocks are encapsulated using a first target anti-packet loss protocol to obtain a composite message sequence.
[0183] Optionally, processing module 32 is specifically used for:
[0184] The voice data stream is extracted and processed to obtain multiple voice data packets;
[0185] The integrity of the multiple voice packet data is checked. If there are missing voice packet data, message data is generated. The message data includes the sequence number and time information of the missing voice packet data.
[0186] The message data is statistically processed to obtain the transmission quality data of the first Internet Protocol link.
[0187] Optionally, processing module 32 is specifically used for the target packet loss resistance protocol:
[0188] Based on the transmission quality data, the average packet loss rate and burst packet loss index are determined;
[0189] The transmission success rate is determined based on the average packet loss rate.
[0190] Obtain preset redundancy parameters;
[0191] When the burst packet loss index is greater than the first preset value, the preset redundancy parameter is reduced to obtain the target redundancy parameter;
[0192] When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset redundancy parameters are incrementally processed to obtain the target redundancy parameters.
[0193] Optionally, processing module 32 is specifically used for target anti-packet loss protocol call data:
[0194] Get the preset retransmission parameters;
[0195] When the burst packet loss index is greater than the first preset value, the preset retransmission parameter is incrementally processed to obtain the target retransmission parameter;
[0196] When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset retransmission parameters are reduced to obtain the target retransmission parameters.
[0197] Based on the target retransmission parameters, the composite message sequence is encapsulated using the second target anti-packet loss protocol to obtain the target voiceband data stream.
[0198] It should be noted that this device is a device corresponding to the above method. All implementation methods in the above method embodiments are applicable to this embodiment and can achieve the same technical effect.
[0199] like Figure 4 As shown, this embodiment of the invention also provides a computing device 40, including a processor 41, a memory 42, and a program or instructions stored in the memory 42 and executable on the processor 41. When the program or instructions are executed by the processor 41, they implement the various processes of the above-described embodiment of the voiceband data transmission method based on dynamic protocol links and achieve the same technical effects. To avoid repetition, they will not be described again here. It should be noted that the computing device in this embodiment of the invention includes the above-described mobile electronic devices and non-mobile electronic devices.
[0200] Those skilled in the art will recognize that the units and algorithm steps of the various examples described in conjunction with the embodiments disclosed herein can be implemented in electronic hardware, or a combination of computer software and electronic hardware. Whether these functions are implemented in hardware or software depends on the specific application and design constraints of the technical solution. Those skilled in the art can use different methods to implement the described functions for each specific application, but such implementations should not be considered beyond the scope of this invention.
[0201] Those skilled in the art will understand that, for the sake of convenience and brevity, the specific working processes of the systems, devices, and units described above can be referred to the corresponding processes in the foregoing method embodiments, and will not be repeated here.
[0202] In the embodiments provided by this invention, it should be understood that the disclosed apparatus and methods can be implemented in other ways. For example, the apparatus embodiments described above are merely illustrative; for instance, the division of units is only a logical functional division, and in actual implementation, there may be other division methods. For example, multiple units or components may be combined or integrated into another system, or some features may be ignored or not executed. Furthermore, the coupling or direct coupling or communication connection shown or discussed may be through some interfaces; the indirect coupling or communication connection between apparatuses or units may be electrical, mechanical, or other forms.
[0203] The units described as separate components may or may not be physically separate. The components shown as units may or may not be physical units; that is, they may be located in one place or distributed across multiple network units. Some or all of the units can be selected to achieve the purpose of this embodiment according to actual needs.
[0204] In addition, the functional units in the various embodiments of the present invention can be integrated into one processing unit, or each unit can exist physically separately, or two or more units can be integrated into one unit.
[0205] If the aforementioned functions are implemented as software functional units and sold or used as independent products, they can be stored in a computer-readable storage medium. Based on this understanding, the technical solution of this invention, or the part that contributes to the prior art, or a portion of the 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 to cause a computer device (which may be a personal computer, server, or network device, etc.) to execute all or part of the steps of the methods of the various embodiments of this invention. The aforementioned storage medium includes various media capable of storing program code, such as USB flash drives, portable hard drives, ROM, RAM, magnetic disks, or optical disks.
[0206] Furthermore, it should be noted that in the apparatus and method of the present invention, it is obvious that the components or steps can be decomposed and / or recombined. These decompositions and / or recombinations should be considered equivalent solutions of the present invention. Moreover, the steps performing the above series of processes can naturally be executed in the order described, but are not necessarily required to be executed in chronological order; some steps can be executed in parallel or independently of each other. Those skilled in the art will understand that all or any step or component of the method and apparatus of the present invention can be implemented in any computing device (including processors, storage media, etc.) or network of computing devices, in hardware, firmware, software, or a combination thereof. This is something that those skilled in the art can achieve by using their basic programming skills after reading the description of the present invention.
[0207] Therefore, the object of the present invention can also be achieved by running a program or a set of programs on any computing device. The computing device can be a known general-purpose device. Therefore, the object of the present invention can also be achieved simply by providing a program product containing program code for implementing the method or apparatus. That is, such a program product also constitutes the present invention, and the storage medium storing such a program product also constitutes the present invention. Obviously, the storage medium can be any known storage medium or any storage medium developed in the future. It should also be noted that in the apparatus and method of the present invention, it is obvious that the components or steps can be decomposed and / or recombined. These decompositions and / or recombinations should be considered equivalent to the present invention. Furthermore, the steps for performing the above series of processes can naturally be performed in the order described, but are not necessarily required to be performed in chronological order. Some steps can be performed in parallel or independently of each other.
[0208] The above are preferred embodiments of the present invention. It should be noted that, for those skilled in the art, several improvements and modifications can be made without departing from the principle of the present invention, and these improvements and modifications should also be considered within the scope of protection of the present invention.
Claims
1. A method for transmitting voiceband data based on a dynamic protocol link, characterized in that, Applied to Session Border Controllers (SBCs), including: The system receives a voice data stream sent by a first client front-end device (CPE) through a first Internet Protocol link. The first client front-end device (CPE) communicates with a first voice data terminal device (CTP), and the voice data stream is sent by the first CTP to the first client front-end device (CPE). When the transmission quality of the first Internet Protocol link is detected to be lower than a preset threshold, at least one anti-packet loss target protocol is dynamically selected for processing of the voiceband data stream to obtain the target voiceband data stream; The target voiceband data stream is sent to the second client front-end device (CPE) via the second Internet Protocol link, and then sent to the second voiceband terminal device via the second client front-end device (CPE). Specifically, the process of dynamically selecting at least one anti-packet loss target protocol to process the voiceband data stream to obtain the target voiceband data stream includes: The voice data stream is segmented to obtain multiple voice data blocks; The multiple voiceband data blocks are processed using a first anti-packet loss target protocol to obtain a composite message sequence; The composite message sequence is processed using a second anti-packet loss target protocol to obtain the target voiceband data stream; Specifically, the multiple voiceband data blocks are processed using a first anti-packet loss target protocol to obtain a composite message sequence, including: Obtain the transmission quality data of the first Internet Protocol link; Based on the transmission quality data, determine the target redundancy parameters of the first anti-packet loss target protocol; Based on the target redundancy parameters, the multiple voiceband data blocks are copied to obtain multiple redundant voiceband data blocks; The plurality of voiceband data blocks and the plurality of redundant voiceband data blocks are encapsulated using a first packet loss resistance protocol to obtain a composite message sequence; The determination of the target redundancy parameters of the first anti-packet loss target protocol based on the transmission quality data includes: Based on the transmission quality data, the average packet loss rate and burst packet loss index are determined; The transmission success rate is determined based on the average packet loss rate. Obtain preset redundancy parameters; When the burst packet loss index is greater than the first preset value, the preset redundancy parameter is reduced to obtain the target redundancy parameter; When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset redundancy parameters are incrementally processed to obtain the target redundancy parameters.
2. The voice data transmission method based on a dynamic protocol link according to claim 1, characterized in that, The voiceband data stream is segmented to obtain multiple voiceband data blocks, including: According to a preset segmentation period, the voice data stream is segmented to obtain multiple voice data blocks, which are arranged in chronological order and assigned sequence numbers.
3. The voice data transmission method based on a dynamic protocol link according to claim 1, characterized in that, Obtain transmission quality data for the first Internet Protocol link, including: The voice data stream is extracted and processed to obtain multiple voice data packets; The integrity of the multiple voice packet data is checked. If there are missing voice packet data, message data is generated. The message data includes the sequence number and time information of the missing voice packet data. The message data is statistically processed to obtain the transmission quality data of the first Internet Protocol link.
4. The voice-band data transmission method based on a dynamic protocol link according to claim 1, characterized in that, The composite message sequence is processed using a second anti-packet loss target protocol to obtain the target voiceband data stream, including: Get the preset retransmission parameters; When the burst packet loss index is greater than the first preset value, the preset retransmission parameter is incrementally processed to obtain the target retransmission parameter; When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset retransmission parameters are reduced to obtain the target retransmission parameters. Based on the target retransmission parameters, the composite message sequence is encapsulated using the second anti-packet loss target protocol to obtain the target voiceband data stream.
5. A voiceband data transmission device based on a dynamic protocol link, characterized in that, Applied to Session Border Controllers (SBCs), including: The transceiver module is used to receive the voice data stream sent by the first client front-end device (CPE) through the first Internet Protocol link. The first client front-end device (CPE) communicates with the first voice data terminal device (CTP), and the voice data stream is sent by the first voice data terminal device to the first client front-end device (CPE). The processing module is used to dynamically select at least one anti-packet loss target protocol to process the voiceband data stream when the transmission quality of the first Internet Protocol link is lower than a preset threshold, so as to obtain the target voiceband data stream. The transceiver module is also used to send the target voiceband data stream to the second client front-end device (CPE) via the second Internet Protocol link, and to send the target voiceband data stream to the second voiceband terminal device via the second client front-end device (CPE). Specifically, the process of dynamically selecting at least one anti-packet loss target protocol to process the voiceband data stream to obtain the target voiceband data stream includes: The voice data stream is segmented to obtain multiple voice data blocks; The multiple voiceband data blocks are processed using a first anti-packet loss target protocol to obtain a composite message sequence; The composite message sequence is processed using a second anti-packet loss target protocol to obtain the target voiceband data stream; Specifically, the multiple voiceband data blocks are processed using a first anti-packet loss target protocol to obtain a composite message sequence, including: Obtain the transmission quality data of the first Internet Protocol link; Based on the transmission quality data, determine the target redundancy parameters of the first anti-packet loss target protocol; Based on the target redundancy parameters, the multiple voiceband data blocks are copied to obtain multiple redundant voiceband data blocks; The plurality of voiceband data blocks and the plurality of redundant voiceband data blocks are encapsulated using a first packet loss resistance protocol to obtain a composite message sequence; The determination of the target redundancy parameters of the first anti-packet loss target protocol based on the transmission quality data includes: Based on the transmission quality data, the average packet loss rate and burst packet loss index are determined; The transmission success rate is determined based on the average packet loss rate. Obtain preset redundancy parameters; When the burst packet loss index is greater than the first preset value, the preset redundancy parameter is reduced to obtain the target redundancy parameter; When the transmission success rate is greater than the second preset value and the burst packet loss index is less than the third preset value, the preset redundancy parameters are incrementally processed to obtain the target redundancy parameters.
6. A computing device, characterized in that, include: A processor, a memory storing a computer program, wherein the computer program, when executed by the processor, performs the method as described in any one of claims 1 to 4.
7. A computer-readable storage medium, characterized in that, The system stores instructions that, when executed on a computer, cause the computer to perform the method as described in any one of claims 1 to 4.
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