SIP telephone audio redirection output method based on SIP voice server

Abstract

The invention discloses an SIP telephone audio redirection output method and system based on an SIP voice server, and the method comprises the steps: S1, enabling an SIP sound amplification soft terminal in the SIP voice server to be connected to the SIP voice server in a registration mode, and carrying out the qualification authentication and parameter configuration, so as to obtain the audio channel data of a specified SIP telephone on the SIP voice service; s2, the SIP voice service redirects appointed SIP telephone audio data to an appointed Domain Socket of the SIP loudspeaking soft terminal through an internal protocol; s3, the SIP sound amplification soft terminal obtains audio data of a plurality of SIP telephones, and efficient coding, decoding and sound mixing processing are carried out; and S4, the SIP sound amplification soft terminal sends the audio data to the FPGA module through DMA for signal conversion, and then the audio data is directionally output through a loudspeaker. The invention discloses an SIP (Session Initiation Protocol) telephone audio redirection output method and an SIP telephone audio redirection output system based on an SIP voice server, which realize efficient redirection and output of one or more SIP telephone audios, remarkably improve the quality of voice communication and user experience, and provide an efficient and reliable audio solution while reducing the cost.

Description

Technical Field

[0001] The present invention relates to the field of voice communication technology. More specifically, the present invention relates to a SIP phone audio redirection output method based on a SIP voice server, which realizes the redirection of SIP phone audio and analog audio output in a vehicle-mounted environment by combining a SIP loudspeaker soft terminal with a SIP voice server. Background Art

[0002] With the development of voice communication technology, SIP (Session Initiation Protocol) has become the mainstream voice communication protocol and is widely used in IP phone systems. SIP phones can achieve efficient voice communication through call control and routing through SIP voice servers. However, in specific scenarios, such as in-vehicle communication systems, it is usually necessary to direct the voice output of multiple SIP phones to the in-vehicle speakers for unified playback. The existing system still lacks an effective method to easily redirect the voice of multiple SIP phones to a single speaker.

[0003] The existing solutions on the market mainly rely on hardware devices and complex circuit designs to achieve audio output and switching, which increases the cost and complexity of the system. At the same time, most of the existing solutions rely on imported components and are difficult to meet the needs of localization. Therefore, a solution based on a combination of software and hardware based on domestic components is urgently needed to simplify the system, reduce costs, and meet the needs of specific application scenarios. Summary of the invention

[0004] An object of the present invention is to solve at least the above problems and / or disadvantages and to provide at least the advantages which will be described hereinafter.

[0005] In order to achieve these purposes and other advantages of the present invention, a SIP telephone audio redirection output method based on a SIP voice server is provided, comprising:

[0006] S1. The SIP voice amplification soft terminal in the SIP voice server connects to the SIP voice service through registration, performs qualification authentication and parameter configuration, and obtains the audio channel data of the specified SIP phone on the SIP voice service;

[0007] S2, SIP voice service redirects the specified SIP phone audio data to the specified Domain Socket of the SIP speaker soft terminal through the internal protocol;

[0008] S3, SIP amplification soft terminal obtains audio data from multiple SIP phones and performs encoding, decoding and mixing;

[0009] S4, SIP amplification soft terminal sends the audio data to the FPGA module through DMA for signal conversion, and then outputs it directionally through the speaker.

[0010] Preferably, in S1, the SIP voice server has at least two SIP telephone interfaces and one analog audio data interface, the SIP voice service, SIP loudspeaker soft terminal, FPGA module and speaker form a complete network for audio processing and output, and the SIP voice server obtains access qualifications by registering in the network;

[0011] The SIP voice amplification soft terminal is registered with the SIP voice service through the SIP protocol, uses a predetermined user name and password for identity authentication, identifies the availability status of the SIP voice amplification soft terminal, and notifies the SIP voice service communication interface and the amplification object;

[0012] SIP phones register with the SIP voice server through the SIP protocol to form a basic connection for voice communication;

[0013] During the initialization phase, the SIP voice server creates a dedicated SIP account for the speaker to identify and manage the speaker.

[0014] Preferably, before S2, an audio channel is also established, and the audio channel establishment process includes:

[0015] The initiator's SIP phone sends a SIP INVITE request. After receiving the request, the SIP voice server forwards it to the target SIP phone.

[0016] The SIP voice server forwards the response status of the target SIP phone to the initiator SIP phone;

[0017] The initiator's SIP phone sends an ACK message to confirm the call establishment, and the audio channel is established after the SIP voice server completes the connection confirmation.

[0018] Preferably, in S3, the codec adaptive module of the SIP loudspeaker soft terminal dynamically selects a codec to perform a decoding operation on the single audio stream based on the received SIP telephone audio format.

[0019] Preferably, in S3, the audio mixing processing module in the SIP loudspeaker soft terminal synthesizes the audio data of multiple SIP phones into a single audio stream based on the audio mixing method, and before the audio mixing, the audio stream data of multiple SIP phones are synchronously processed by the audio stream synchronization module in the SIP voice server;

[0020] The synchronization process is to align the various audio streams in time through timestamp and buffer management technology;

[0021] The mixing processing module mixes multiple audio signals based on a weighted addition algorithm, and the dynamic gain algorithm ensures that the volume is stable and not distorted.

[0022] Preferably, in S4, the SIP sound amplification soft terminal registers a SIP account and receives data through a Domain Socket interface, so that the speaker is in a sound amplification state;

[0023] The DMA transmission module of the SIP voice server transmits the mixed audio data to the FPGA module.

[0024] Preferably, the weighted addition is to assign a corresponding weight based on the priority and signal strength of each audio stream, and then add the weighted audio streams together;

[0025] After the weighted addition, the sum is gain-adjusted by the mixing processing module so that the output volume is within a predetermined dynamic range and does not exceed a threshold allowed by the system.

[0026] A SIP phone audio redirection output system is applied to a SIP phone audio redirection output method based on a SIP voice server, comprising: a SIP voice server, a speaker and a plurality of SIP phones, and the SIP voice server comprises: a SIP voice service, a SIP loudspeaker soft terminal and an FPGA module;

[0027] The SIP voice server is provided with a SIP voice service, an audio data acquisition submodule, a codec adaptation submodule, an audio stream mixing and synchronization submodule, a mixing processing submodule, and a DMA transmission submodule, and the modules are all configured to use a CPU;

[0028] The FPGA module is equipped with an audio data conversion and processing submodule and an analog audio output submodule connected to the speaker.

[0029] The present invention has at least the following beneficial effects:

[0030] First, the present invention is easily compatible with all domestic component platforms to achieve independent control and cost optimization of hardware;

[0031] Secondly, the present invention combines the SIP speaker soft terminal and hardware device to redirect the voice data of the in-vehicle SIP phone and output the audio to the in-vehicle speaker. That is, it can realize an effective method to redirect the voice of one or more SIP phones to a single speaker;

[0032] Thirdly, the SIP voice server of the present invention integrates the SIP voice amplification soft terminal function, reduces the need for additional hardware, and simplifies the system structure;

[0033] Fourthly, the present invention utilizes the high-performance decoding capability of FPGA to ensure the real-time and high-quality of audio conversion, and the entire audio processing process is carried out in a real-time environment. DMA+Domain Socket transmission is used to ensure low latency and high efficiency of audio transmission, and real-time processing technology can ensure the synchronization and smoothness of audio.

[0034] Through the above effects, the present invention realizes efficient redirection and output of audio in multiple SIP phones, significantly improves the quality of voice communication and user experience, and provides an efficient and reliable audio solution while reducing costs through innovative technical solutions.

[0035] Other advantages, objectives and features of the present invention will be embodied in part through the following description, and in part will be understood by those skilled in the art through study and practice of the present invention. BRIEF DESCRIPTION OF THE DRAWINGS

[0036] Figure 1 A flow chart of a SIP phone of the speaker object of the present invention making a call and starting amplification through a voice server;

[0037] Figure 2 A flowchart for converting audio data into analog audio data according to the present invention;

[0038] Figure 3 It is a functional block diagram of the SIP telephone audio redirection output system of the present invention;

[0039] Figure 4 The present invention is an audio stream mixing process flow chart;

[0040] Figure 5 It is a DMA audio data transmission flow chart of the present invention;

[0041] Figure 6 It is a schematic diagram of the structure of the FPGA audio processing module of the present invention;

[0042] Figure 7 Schematic diagram of a typical application scenario of the present invention. DETAILED DESCRIPTION

[0043] The present invention is further described in detail below in conjunction with the accompanying drawings so that those skilled in the art can implement the invention with reference to the description.

[0044] It should be understood that the terms such as “having”, “including” and “comprising” used herein do not exclude the existence or addition of one or more other elements or combinations thereof.

[0045] In the description of the present invention, it should be noted that, unless otherwise clearly stipulated and limited, the terms "installed", "provided with", "mounted / connected", "connected", etc. should be understood in a broad sense. For example, "connected" can be a fixed connection, a detachable connection, or an integral connection. It can be a mechanical connection or an electrical connection. It can be a direct connection or an indirect connection through an intermediate medium. It can be the internal connection of two components. For ordinary technicians in this field, the specific meanings of the above terms in the present invention can be understood according to specific circumstances.

[0046] The present invention provides a SIP phone audio redirection output method based on a voice server. The method connects to a SIP voice service through a SIP loudspeaker soft terminal, configures a loudspeaker object and a communication interface, and obtains the voice channel of a designated SIP phone in a car in a SIP voice server. The SIP voice service redirects the data of the current voice channel of the designated SIP phone in the car to a Domain Socket designated by the SIP loudspeaker soft terminal through an internal communication protocol, and the SIP loudspeaker soft terminal performs efficient encoding, decoding and mixing processing on the audio data. The decoded PCM audio data is sent to an FPGA through DMA, and the FPGA converts the PCM audio data into an analog audio signal for output by a speaker.

[0047] From the perspective of system architecture, the present invention mainly includes a SIP voice server, a loudspeaker, and a SIP phone to form a complete audio processing and output link, which is particularly suitable for a vehicle-mounted environment. In actual applications, the SIP voice server is used to achieve the following requirements: first, to provide the SIP voice service function of the SIP phone to realize various call services; second, to provide a speaker interface to play the audio data of one or more SIP phones through a loudspeaker.

[0048] The SIP voice server includes: SIP voice service, SIP amplification soft terminal, FPGA module. As the core component, the SIP voice server is equipped with at least two SIP phone interfaces and one analog audio data interface to provide core SIP voice services. The SIP amplification soft terminal is responsible for the acquisition, encoding, decoding and mixing of audio data. It obtains voice channel data through Domain Socket, decodes the audio data into PCM format, and transmits it to the FPGA module through DMA. The FPGA module converts digital audio data into analog signals to ensure high-quality audio output, and transmits it to the speaker through the audio interface.

[0049] The workflow of the present invention mainly includes several key steps: system initialization, audio channel establishment and redirection, audio data processing, audio data conversion and audio output. The detailed description of each step is as follows:

[0050] 1. System initialization

[0051] a) SIP voice service startup:

[0052] When the SIP voice service is started, the core SIP voice service functions are initialized and the SIP phone and SIP speaker soft terminal interface configurations are generated. This configuration ensures that the server can provide a basis for processing the audio streams of multiple SIP phones at the same time.

[0053] The SIP voice service creates a dedicated SIP account loudspeaker for the SIP speakerphone. This account is used to authorize speakerphone qualifications and manage speakerphone functions to ensure that it can be quickly activated when needed.

[0054] The SIP voice server is registered in the network and is ready to receive connection requests from the SIP phone and SIP speaker soft terminal in the car. The registration process ensures the accessibility of the SIP voice server in the network and the stability of communication.

[0055] b) SIP voice amplification soft terminal registration:

[0056] The SIP loudspeaker soft terminal is registered with the SIP voice service through the SIP protocol. The SIP loudspeaker soft terminal connects to the SIP voice service through the registration method (adding the custom field X-loudspeadker-socket:xxxx in the SIP header to notify the SIP voice service of the Domain Socket (an efficient inter-process communication method in Linux system) to receive audio data, and X-loudspeadker-objects:xxx,xxx,xxx,xxx, to notify the SIP voice service of the objects that need to be loudspeaked) to perform qualification authentication and parameter configuration to obtain the audio channel of the specified SIP phone on the SIP voice service. After the connection is established, the SIP voice service redirects the audio data of the specified SIP phone to the Domain Socket specified by the SIP loudspeaker soft terminal through the internal protocol.

[0057] Specifically, during the registration process, the SIP loudspeaker soft terminal and the SIP voice service use a dedicated SIP account loudspeaker for identity authentication, confirm the connection status, prevent unauthorized device access, and ensure the reliability of communication and the security of data transmission.

[0058] By adding the Domain socket pipe name "X-loudspeaker-socket:xxxxx" and the speaker object SIP number "X-loudspeaker-objects:xxx,xxx,xxx" in the SIP registration signaling Message Header according to the specification.

[0059] After receiving the registration success response 200 (OK), initialize the receiving channel.

[0060] After receiving the SIP registration signaling, the SIP voice service initializes the sending channel according to the value of "X-loudspeaker-socket:xxxxx" and sets several speaker objects according to the value of "X-loudspeaker-objects:xxx,xxx,xxx".

[0061] c)SIP phone registration:

[0062] The SIP phone in the car registers with the SIP voice server through the SIP protocol to form the basic connection for voice communication. After successful registration, the SIP phone can make and receive calls through the SIP voice server.

[0063] This connection lays the foundation for subsequent audio data transmission and processing, ensuring the smooth conduct of the call.

[0064] 2. Audio channel establishment and redirection

[0065] The SIP phone initiates a SIP INVITE request, and after receiving the request, the SIP voice server forwards it to the target SIP phone. This process involves signaling processing and routing selection.

[0066] The target SIP phone responds with a 180 ring tone message and a 200 OK confirmation message, and the SIP voice server forwards this information to the initiator. The ring tone message is used to notify the initiator that the call is in progress.

[0067] The initiator sends an ACK message to confirm the call establishment, and the voice server completes the connection confirmation to ensure the smooth progress of the call. At this point, the audio channel has been established and is ready for data transmission.

[0068] The SIP voice service determines the status of the SIP speaker soft terminal and encapsulates the voice data in the corresponding SIP phone audio channel into an internal communication protocol and sends it to the Domain Socket.

[0069] 3. Audio data processing

[0070] a) Audio data reception and decoding:

[0071] The SIP speaker terminal reads multiple SIP phone audio data from the Domain Socket.

[0072] The SIP speaker soft terminal dynamically selects the appropriate codec to decode the audio data according to the audio format of each SIP phone. Common formats supported include G.711, G.729, Opus, etc.

[0073] b) Audio synchronization and mixing:

[0074] Before audio mixing, multiple audio streams need to be synchronized. Timestamp and buffer management techniques are used to ensure that each audio stream is aligned in time.

[0075] Advanced mixing technology is used to mix multiple synchronized audio streams, and a dynamic gain control algorithm is used to ensure volume balance. The mixing algorithm is based on weighted addition, and the weights can be adjusted according to the priority and signal strength of the audio stream.

[0076] The mixing process ensures the clarity and synchronization of audio output, avoiding audio distortion and delay. With mixing, multiple audio streams can be played in the same speaker, providing a consistent audio experience.

[0077] The mixed audio data is transmitted to the FPGA module through the DMA interface to ensure the stability and low latency of data transmission. DMA communication provides a reliable data transmission channel.

[0078] 4. Audio data conversion and audio output

[0079] a) Data reception:

[0080] Receive digital audio data from DMA and use FIFO buffer to store the data.

[0081] b) Audio conversion and output:

[0082] After receiving the audio data, the FPGA module converts it into an analog signal. The high-performance processing capability of the FPGA ensures the real-time performance of audio decoding.

[0083] The analog signal is connected to the speaker through the audio interface to achieve high-quality audio output, ensuring the clarity of voice communication in the car. The speaker converts the analog signal into audible sound, providing a high-quality audio experience.

[0084] 5. End of call and stop of amplification

[0085] a) End of call:

[0086] The SIP phone sends a BYE message to end the call, and the SIP voice server processes and confirms it to ensure the normal end of the call. The BYE message is used to notify the end of the call.

[0087] b) Stop amplifying

[0088] The SIP voice service receives a call end request and stops forwarding voice data.

[0089] c) Status refresh and management:

[0090] The SIP voice service refreshes the state of the speaker object after the call ends and ensures that the audio output of other SIP phones in call is not affected.

[0091] The present invention realizes efficient redirection and output of audio in multiple SIP phones through the above solution, significantly improving the quality of voice communication and user experience. Through innovative technical solutions, the present invention provides an efficient and reliable audio solution while reducing costs.

[0092] Example:

[0093] This embodiment provides a system for designing efficient multi-channel audio acquisition, codec adaptation, efficient mixing and synchronization, hardware interface transmission and analog audio output. The core idea is to obtain the audio data of one or more SIP phones through the SIP voice service and redirect it to the SIP speaker soft terminal. The SIP speaker soft terminal performs efficient codec and mixing processing, and transmits the audio data to the FPGA through the DMA interface. The FPGA converts the audio data and finally outputs the analog audio signal through the DAC (digital-to-analog converter). The voice data redirection of the in-vehicle SIP phone is realized, and the audio is output to the in-vehicle speakers.

[0094] Specifically, if Figure 3 As shown, the system of the present invention mainly includes: SIP voice service module, audio data acquisition module, codec adaptation module, audio stream mixing and synchronization module, mixing processing module, DMA transmission module, FPGA processing and conversion module, analog audio output module, etc., and the functions of each module are as follows:

[0095] a. SIP voice service module 1: responsible for providing core SIP call control and routing, obtaining audio data from multiple SIP phones and encapsulating it as an internal protocol and redirecting it to the audio data acquisition module.

[0096] b. Audio data collection module 2: responsible for collecting audio data from multiple SIP phones. The system calls receive this data and inputs it into the next processing chain.

[0097] c. Codec Adaptive Module 3: This module dynamically selects the appropriate codec to decode the audio data according to the audio format of each SIP phone. Common formats supported include G.711, G.729, Opus, etc.

[0098] d. Audio stream synchronization module 4: Before audio mixing, multiple audio streams need to be synchronized. Timestamp and buffer management technology are used to ensure that each audio stream is aligned in time.

[0099] e.CPU mixing processing module 5: This module is responsible for mixing multiple synchronized audio streams and uses a dynamic gain control algorithm to ensure volume balance. The mixing algorithm is based on weighted addition, and the weight can be adjusted according to the priority and signal strength of the audio stream.

[0100] f.DMA transfer module 6: The mixed audio data is transferred to the FPGA via DMA. The direct memory access function of DMA allows the audio data to be efficiently transferred between the CPU and FPGA without increasing the CPU load.

[0101] g. FPGA processing and conversion module 7: FPGA receives digital audio data from DMA, performs digital to analog conversion (DAC), and performs additional processing such as audio enhancement and noise suppression as needed.

[0102] h. Analog audio output module 8: Outputs the analog audio signal converted by FPGA to speakers, headphones or other audio devices. This solution combines CPU and FPGA to perform efficient data transmission through DMA (Direct Memory Access). In the process of transmitting audio data from CPU to FPGA, DMA can achieve high-speed memory migration without occupying CPU resources and reducing CPU load. FPGA is responsible for further processing of audio data, digital to analog conversion (DAC), etc.

[0103] Figure 4 This article describes the audio stream mixing process, mainly from audio stream decoding, synchronization, weighted addition mixing, dynamic gain adjustment to the entire process of final output. In the specific implementation, the SIP audio data collection and encoding and decoding process mainly includes:

[0104] a. Audio data reception: Multiple SIP phones send audio data to the SIP server via RTP, and the server passes the audio data to the audio processing system.

[0105] b. Codec Adaptation: Based on the audio format information contained in the audio data packet header, the codec adaptation module automatically selects the appropriate codec for decoding.

[0106] Common codecs include:

[0107] ●G.711: No compression required, high sound quality, but occupies a large bandwidth.

[0108] G.729: High compression rate, suitable for bandwidth-constrained environments.

[0109] ●Opus: highly efficient compression, excellent sound quality, and wide adaptability.

[0110] c. Data decoding: The codec adaptation module decodes the received compressed audio data into original PCM format audio data according to the audio format of each SIP phone. After decoding, the PCM data is passed to the audio stream synchronization module. This solution enables the system to automatically identify and adapt to a variety of common audio codec formats (such as G.711, G.729, Opus, etc.), and can process audio streams of different formats without human intervention, thereby improving the compatibility and flexibility of the system. In actual applications, multiple SIP phones may use different codecs, and the system's adaptive function ensures compatibility with these different formats.

[0111] Figure 5 It shows how to transfer audio data between the CPU and FPGA through the DMA interface. In the specific implementation, the audio stream synchronization and mixing process includes:

[0112] a. Audio stream synchronization: The audio streams of multiple SIP phones may be inconsistent in time due to network delay or jitter. The audio stream synchronization module synchronizes these audio streams through the timestamp mechanism and buffer management technology. Each audio stream is assigned a timestamp. The audio stream synchronization module aligns different audio streams according to the timestamp to ensure that they have the same time reference when entering the mixing process, and passes the synchronized audio streams with timestamp alignment to the mixing processing module. This step uses timestamp and buffer management technology to enable the system to accurately synchronize multiple audio streams and ensure that the audio streams between different SIP phones are aligned in time. This is crucial to avoid audio misalignment or delay problems, and achieves low latency and high sound quality in audio transmission.

[0113] b. Mixing process: The synchronized multiple audio streams will be passed to the CPU mixing processing module. During the mixing process, the system uses weighted addition to mix multiple audio streams into one audio stream. The core of the mixing algorithm is the dynamic gain regulator, which is used to ensure that the volume of each audio stream is within a reasonable range to avoid audio distortion or overload. The specific steps of the mixing algorithm are as follows:

[0114] ●Weighted addition: Assign weights according to the priority and signal strength of each audio stream, and then add the weighted audio streams together. If there are n input signals, the discrete time sequence is represented by g1(t), g1(t), ... gn(t), and the corresponding weights are h1, h2, ... hn, then the calculation formula for the mixing result signal y(t) is as follows:

[0115] y(t)=∑[h_n*g_n(t)]

[0116] Dynamic gain control: After weighted addition, the mixer will adjust the gain of the sum to ensure that the output volume does not exceed the threshold allowed by the system while maintaining sufficient dynamic range.

[0117] This solution uses a mixing algorithm based on weighted average and dynamic gain adjustment. The algorithm can process multiple audio streams in real time and perform weighted processing based on the priority and signal strength of each audio stream to ensure volume balance and signal strength consistency. In addition, the system also has dynamic gain adjustment capabilities to avoid overload and distortion of audio signals, reflecting the solution's efficient processing capabilities.

[0118] Figure 6 It shows how to transfer audio data between the CPU and FPGA through the DMA interface. In the specific implementation, the audio data after mixing is transferred to the FPGA through DMA, and the digital-to-analog conversion (DAC) is performed in the FPGA, and then the analog audio output is performed. It mainly includes:

[0119] a.DMA transfer: The mixed single-channel PCM audio stream is transferred from the CPU to the FPGA through the DMA (Direct Memory Access) interface. The advantage of DMA is that it can complete data transmission without occupying CPU computing resources, greatly improving the efficiency of data transmission. The DMA controller will be responsible for high-speed data transfer and ensure that audio data will not be lost during the transmission process.

[0120] b. Use FIFO buffer to store data

[0121] c.FPGA data processing

[0122] d.DAC conversion:

[0123] Digital to Analog Conversion (DAC): After receiving the mixed PCM audio data, the FPGA first needs to convert it from a digital signal to an analog audio signal. This process is completed by the DAC module inside the FPGA.

[0124] e.Analog audio output

[0125] Analog audio output module: After being processed by FPGA, the final analog audio signal will be played through the output interface speaker. This module ensures the smooth output of the audio signal and avoids distortion.

[0126] Embodiment 2:

[0127] like Figure 7 The present invention is applied to a vehicle command communication system, and a usage scenario in which multiple SIP phones are used to amplify the sound through the vehicle's speakers is demonstrated.

[0128] Specifically, in the use of vehicle command communication systems, multiple SIP user calls need to be played through a single speaker, and the above processing method is perfectly compatible with this usage scenario. The call audio data is transmitted through the network, and the audio format and network conditions may vary. The system's codec adaptation function can automatically process a variety of codec formats to adapt to different network bandwidths and conditions. Through the synchronization and mixing of audio streams, the system can ensure that there is no audio imbalance or echo problems in multi-party calls. The hardware acceleration and audio processing functions provided by the FPGA module enable the system to maintain stable and efficient operation in large-scale VoIP communication scenarios.

[0129] The above solution is only an illustration of a preferred embodiment, but is not limited thereto. When implementing the present invention, appropriate replacement and / or modification can be performed according to user needs.

[0130] The number of devices and processing scales described here are used to simplify the description of the present invention. Applications, modifications and variations of the present invention will be obvious to those skilled in the art.

[0131] Although the embodiments of the present invention have been disclosed above, they are not limited to the applications listed in the specification and the embodiments. It can be fully applied to various fields suitable for the present invention. For those familiar with the art, additional modifications can be easily realized. Therefore, without departing from the general concept defined by the claims and equivalent scope, the present invention is not limited to the specific details and the illustrations shown and described here.

Claims

1. A SIP telephone audio redirection output method based on a SIP voice server, characterized in that: include: S1. The SIP voice amplification soft terminal in the SIP voice server connects to the SIP voice service through registration, performs qualification authentication and parameter configuration, and obtains the audio channel data of the specified SIP phone on the SIP voice service; S2, SIP voice service redirects the specified SIP phone audio data to the specified Domain Socket of the SIP speaker soft terminal through the internal protocol; S3, SIP amplification soft terminal obtains audio data from multiple SIP phones and performs encoding, decoding and mixing; S4, SIP amplification soft terminal sends the audio data to the FPGA module through DMA for signal conversion, and then outputs it directionally through the speaker.

2. The SIP telephone audio redirection output method based on the SIP voice server as claimed in claim 1, characterized in that: In S1, the SIP voice server has at least two SIP phone interfaces and one analog audio data interface. The SIP voice service, SIP loudspeaker soft terminal, FPGA module and speaker form a complete network for audio processing and output. The SIP voice server obtains access qualifications by registering in the network. The SIP voice amplification soft terminal is registered with the SIP voice service through the SIP protocol, uses a predetermined user name and password for identity authentication, identifies the availability status of the SIP voice amplification soft terminal, and notifies the SIP voice service communication interface and the amplification object; SIP phones register with the SIP voice server through the SIP protocol to form a basic connection for voice communication; During the initialization phase, the SIP voice server creates a dedicated SIP account for the speaker to identify and manage the speaker.

3. The SIP telephone audio redirection output method based on the SIP voice server as claimed in claim 1, characterized in that: Before S2, the audio channel is also established. The audio channel establishment process includes: The initiator's SIP phone sends a SIP INVITE request. After receiving the request, the SIP voice server forwards it to the target SIP phone. The SIP voice server forwards the response status of the target SIP phone to the initiator SIP phone; The initiator's SIP phone sends an ACK message to confirm the call establishment, and the audio channel is established after the SIP voice server completes the connection confirmation.

4. The SIP telephone audio redirection output method based on the SIP voice server as claimed in claim 1, characterized in that: In S3, the codec adaptive module of the SIP sound amplification soft terminal dynamically selects a codec to perform a decoding operation on a single audio stream based on the received SIP telephone audio format.

5. The SIP telephone audio redirection output method based on the SIP voice server as claimed in claim 1, characterized in that: In S3, the audio mixing processing module in the SIP voice amplification soft terminal synthesizes the audio data of multiple SIP phones into a single audio stream based on the audio mixing method, and before the audio mixing, the audio stream data of multiple SIP phones are synchronously processed by the audio stream synchronization module in the SIP voice server; The synchronization process is to align the various audio streams in time through timestamp and buffer management technology; The mixing processing module mixes multiple audio signals based on a weighted addition algorithm, and the dynamic gain algorithm ensures that the volume is stable and not distorted.

6. The SIP telephone audio redirection output method based on the SIP voice server as claimed in claim 1, characterized in that: In S4, the SIP sound amplification soft terminal registers a SIP account and receives data through the Domain Socket interface, so that the speaker is in a sound amplification state; The DMA transmission module of the SIP voice server transmits the mixed audio data to the FPGA module.

7. The SIP telephone audio redirection output method based on the SIP voice server as claimed in claim 3, characterized in that: The weighted addition method is to assign a corresponding weight based on the priority and signal strength of each audio stream, and then add the weighted audio streams together; After the weighted addition, the sum is gain-adjusted by the mixing processing module so that the output volume is within a predetermined dynamic range and does not exceed a threshold allowed by the system.

8. A SIP telephone audio redirection output system, applied to the SIP telephone audio redirection output method based on a SIP voice server as described in any one of claims 1 to 7, characterized in that: include: SIP voice server, speaker and multiple SIP phones, and the SIP voice server includes: SIP voice service, SIP loudspeaker soft terminal, FPGA module; The SIP voice server is provided with a SIP voice service, an audio data acquisition submodule, a codec adaptation submodule, an audio stream mixing and synchronization submodule, a mixing processing submodule, and a DMA transmission submodule, and the modules are all configured to use a CPU; The FPGA module is provided with an audio data conversion and processing submodule and an analog audio output submodule connected to a speaker.

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