CN117692560A - Voice communication method and system based on ground station networking - Google Patents

Voice communication method and system based on ground station networking Download PDF

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Publication number
CN117692560A
CN117692560A CN202311612593.2A CN202311612593A CN117692560A CN 117692560 A CN117692560 A CN 117692560A CN 202311612593 A CN202311612593 A CN 202311612593A CN 117692560 A CN117692560 A CN 117692560A
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China
Prior art keywords
voice
cabin
client
voice processing
data
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CN202311612593.2A
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Chinese (zh)
Inventor
王维
付泱
屠熙
戴唐云
杨其林
田兴
周小林
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AVIC Chengdu Aircraft Design and Research Institute
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AVIC Chengdu Aircraft Design and Research Institute
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Priority to CN202311612593.2A priority Critical patent/CN117692560A/en
Publication of CN117692560A publication Critical patent/CN117692560A/en
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Abstract

The invention discloses a voice communication method and system based on ground station networking, and relates to the technical field of voice communication. One embodiment of the present invention includes: a cross-cabin point-to-point voice communication method, a single-cabin point-to-point voice communication method, a multi-cabin client-to-client conference communication method, a single-cabin client-to-client and a multi-radio communication method. The networking communication method adopts a modularized design, has strong general line, expansibility and economy, supports networking modes of various unmanned aerial vehicle ground stations, and can meet the application of system-level transformation of new stations and old stations.

Description

Voice communication method and system based on ground station networking
Technical Field
The invention relates to the technical field of voice communication, in particular to a voice communication method and system based on ground station networking.
Background
The ground station voice communication is an important component of the ground station of the unmanned aerial vehicle, and mainly realizes the voice communication between the ground station and the radio station, and between the ground station and the ground station, so that the mutual communication among users is satisfied. However, the existing communication device of the ground station generally only provides analog voice, and has the advantages of more voice devices, large space occupation and single function.
Although some unmanned aerial vehicle ground stations are digital communication, private equipment and communication protocols are adopted, related equipment is independently packaged, so that the voice communication system of the ground stations is poor in universality and integration, networking cannot be conveniently conducted in large unmanned aerial vehicle ground stations with digital radio stations, digital encryption cross cabin links and unmanned aerial vehicle autonomous network links, multiple sets of terminals such as headsets are required to be equipped for each voice link, and space waste of an operation desk is caused.
In addition, the existing ground station mainly realizes the conversation function, cannot support text communication, file transmission, voice recording and multi-party cross-cabin networking communication, lacks an open protocol interface, cannot be accessed into advanced artificial intelligent equipment such as voice recognition, voice control, voice synthesis and voice alarm, and cannot meet the application requirements of networking of large ground stations in the future.
At present, multimedia information such as transmission text, files, voice, video and the like on a VoIP (voice over Internet protocol) network based on packet forwarding is mature in the field of telecommunications, the technology is applied to large-scale ground station networking, the mutual networking of changeable large-scale ground stations can be realized, digital voice and video communication cooperative control combat can be performed with a plurality of control units transversely, data information sharing of a higher-level command control center is longitudinally performed, and the fine task of a complex battlefield environment is completed better. However, if the digital VoIP technology with high flow and low encryption capability in the field of the mobile telecommunication is mature, networking cross-linking cannot be developed with the existing mature unmanned aerial vehicle ground station simulation station, unmanned aerial vehicle inter-aircraft links with low flow and confidentiality requirements, space-ground autonomous network links and the like, and cross-cabin encryption leading links. The popularization of unmanned aerial vehicle voice systems of large-scale ground stations of the digitizing technology is affected.
Disclosure of Invention
The invention provides a voice communication method based on ground station networking, which adopts a modularized design, has strong general line, expansibility and economy, supports networking modes of various unmanned aerial vehicle ground stations, and can meet the application of system-level reconstruction of new stations and old stations. The hardware equipment can be deployed on the unmanned aerial vehicle ground station equipment rack, and related software forms can be installed on the unmanned aerial vehicle ground station seat control computer. Real-time digital communication of seats, ED137 digital radio stations and program control phones in the cabin can be realized conveniently through client side operation software, voice communication of unmanned aerial vehicle ground station simulation radio stations can be realized, interconnection networking between the cabin and each related node of the unmanned aerial vehicle can be realized through inter-cabin encryption leading links and inter-unmanned aerial vehicle and inter-air-ground autonomous network links, and a voice, video, file and text communication network between long-distance ground and air-ground large-scale cabins is constructed.
In view of this, according to an aspect of an embodiment of the present invention, there is provided a voice communication method based on ground station networking, a point-to-point voice communication method across cabins, including:
Step 2-1: the seat of the cabin A logs in a voice processing workstation of the cabin A through voice system client software;
step 2-2: the voice processing workstation of the cabin A acquires information of all seat 1 clients of the cabin, and multicast sharing is carried out on the information to other cabins by using a UDT protocol through a cross-cabin encryption link or an autonomous network link;
step 2-3: the voice processing workstation of the cabin B acquires and stores information of the seat 1 client side of the cabin A;
step 2-4: the voice system client software of the cabin A seat sends an SIP command with the cabin B client number to the cabin A voice processing workstation through an SIP protocol according to the user information of the cabin B client to realize sending of a cross-cabin Invite request to start calling;
step 2-5: after the cabin A voice processing workstation confirms that the user authentication is passed, the voice processing workstation inserts an own address in a Via header field of the request message, packages the Invite message according to a cross-cabin or autonomous network protocol, and distributes the Invite message to a cross-cabin encryption link receiving device according to a load requirement of a cross-cabin encryption or autonomous network link;
step 2-6: the cross-cabin encryption link guiding equipment encrypts the sub-packet data and sends the encrypted sub-packet data to the cabin B;
step 2-7: the cabin B cross-cabin encryption link receiving equipment decrypts and packetizes and sends the decryption and the packetization to a cabin B voice processing workstation according to an encryption protocol;
Step 2-8: the cabin B voice processing workstation receives the sub-packet data packet sent by the cross-cabin encryption link and then sends an Invite request To a cabin B seat client according To the To domain of the message;
step 2-9: the cabin B seat client sends response information with SIP cross-cabin expansion frames in call processing to the cabin B voice processing workstation;
step 2-10: the cabin B voice processing workstation receives SIP cabin-crossing response information sent by the equipment to the cabin B cabin-crossing encryption link; and sub-packaging;
step 2-11: the cabin B sends encrypted data to the cabin A after being encrypted by the cabin B cross-cabin encryption link guide receiving equipment;
step 2-12: the cabin A cross-cabin encryption link guide equipment performs decryption and then sends the decrypted decryption to a voice processing workstation of the cabin A;
step 2-13: the voice processing workstation of the cabin A sends SIP cross-cabin response information in the software call processing of the seat client of the cabin A;
step 2-14: the cabin B seat client side indicates the called user to ring, and after the user rings, ringing information is sent to the cabin B voice processing workstation;
step 2-15: the cabin B voice processing workstation executes the encryption forwarding steps from the step 2-9 to the step 2-12 to forward the ringing information of the called user to the cabin A seat client;
step 2-16: the cabin B seat client returns a response indicating that the connection is successful to the cabin B voice processing workstation;
Step 2-17: after the cabin B voice processing workstation divides the packets, forwarding the success indication to the cabin A client of the cabin A voice processing workstation through a cross-cabin encryption link;
step 2-18: the cabin A voice processing workstation forwards the success indication to the cabin A client;
step 2-19: after receiving the information, the cabin A client sends ACK information to the cabin A voice processing workstation for confirmation;
step 2-20: the cabin A voice processing workstation is used for transmitting the packetized data to the cabin B voice processing workstation through a cross-cabin encryption link;
step 2-21: the cabin B voice processing workstation forwards the ACK confirmation message to the cabin B client;
step 2-22: and the calling user and the called user establish communication connection and start talking. The client software of the seat compresses the voice data according to a contracted compression format;
step 2-23: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 2-24: each cabin voice processing workstation stores voice data according to RTP voice data;
step 2-25: and the voice processing workstations of each cabin carry out sub-packaging processing according to the cabin bit of the RTP extension header data receiving party and the client, and start cross-cabin encryption forwarding.
Step 2-26: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
step 2-27: after receiving the data, the client decodes the RTP audio data;
step 2-28: and the client sends the decoded data to the sound card for playing.
Optionally, the text and file transmission communication method comprises the following steps in addition to the steps 2-1 to 2-20:
step 2-29: after communication connection is established between the clients, cabin A client software sends relevant text information and file data to be subjected to lossless compression and then sends the compressed text information and file data to a cabin A voice processing workstation according to a UDT format so as to reduce bandwidth occupation;
step 2-30: the cabin A voice processing workstation stores text information and file data according to the UDT data;
step 2-31: and the cabin A voice processing workstation determines a client receiving the shelter according to the UDT extension header data to forward. The sub-packaging is executed under the condition of crossing the cabin, and the sub-packaging is sent to the cabin A guiding equipment for encryption transmission;
step 2-32: the cabin B voice processing workstation receives decryption data of the cabin B guiding equipment and then packs the decryption data;
Step 2-33: the cabin B voice processing workstation stores the packed data and forwards the data to a client of the cabin B;
step 2-34: after receiving the data, the client of the cabin B decompresses the UDT text information and the file data;
step 2-35: and displaying text information or storing files on the client side of the cabin B.
Optionally, the single-cabin point-to-point voice communication method further comprises:
step 1-1: logging in a voice processing workstation through voice system client software;
step 1-2: the voice processing workstation acquires information of all online clients of the cabin;
step 1-3: the seat 1 voice system client software initiates an Invite request to the voice processing workstation through a standard SIP protocol to initiate a call;
step 1-4: after the voice processing workstation confirms that the user authentication is passed, checking whether the Via header field in the request message contains the address of the user authentication; if yes, the loop back is indicated to occur, and a response indicating the error is returned; if not, the voice processing workstation inserts the self address in the Via header field of the request message and transmits an Invite request To the called seat 2 client indicated by the To field of the Invite message;
step 1-5: the seat 2 client sends response information in call processing to the voice processing workstation;
Step 1-6: the voice processing workstation sends response information in call processing to the seat 1 client;
step 1-7: the seat 2 client side indicates the called user to ring, and after the user rings, the ringing information is sent to the voice processing workstation;
step 1-8: the voice processing workstation forwards the ringing information of the called user to the seat 1 client;
step 1-9: the seat 2 client returns a response indicating that the connection is successful to the voice processing workstation;
step 1-10: the voice processing workstation forwards the success indication to the seat 1 client;
step 1-11: after receiving the information, the seat 1 client sends ACK information to the voice processing workstation for confirmation;
step 1-12: the voice processing workstation forwards the ACK confirmation message to the seat 2 client;
step 1-13: communication connection is established between the calling user and the called user, communication is started, and the client software of the seat compresses voice data according to a contracted compression format;
step 1-14: the seat client software sends the compressed and encoded data to a voice processing workstation according to an RTP format;
step 1-15: the voice processing workstation stores voice data according to the RTP voice data;
step 1-16: and the voice processing workstation judges the receiver client according to the RTP extension header data and forwards the receiver client.
Step 1-17: after receiving the data, the client decodes the RTP audio data;
step 1-18: the client sends the decoded data to the sound card for playing;
optionally, for text and file transmission communication methods, in addition to steps 1-1 to 1-12, the following steps are included:
step 1-19: the clients are connected in a communication way, and seat 1 client software sends relevant text information and file data to a voice processing workstation according to a UDT format after lossless compression;
step 1-20: the voice processing workstation stores text information and file data according to the UDT data;
step 1-16: the voice processing workstation determines a receiver client according to the UDT extension header data and forwards the receiver client;
step 1-17: after receiving the data, the client decompresses the UDT text information and the file data;
step 1-18: the client displays the text information or saves the file.
Optionally, the conference communication method between clients among multiple cabins includes:
step 3-1: the cabin A seat 1 establishes a conference room through voice system client software;
step 3-2: after receiving the conference room establishment message, the voice processing workstation of the cabin A uses a UDT protocol to multicast and share the conference room information to other cabins through a cross-cabin encryption link or an autonomous network link;
Step 3-3: other client software enters a conference room by using a Session Initiation Protocol (SIP) handshake command with a conference number; each cabin workstation uses UDT protocol to multicast and share conference room information to other cabins through a cross-cabin encryption link or an autonomous network link;
step 3-4: all client seat client software respectively carries out compression coding on local microphone voice data according to a contracted compression format;
step 3-5: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 3-6: each cabin voice processing workstation stores the voice data according to the RTP voice data;
step 3-7: each cabin voice processing workstation determines a conference shelter according to the RTP extension header data, then carries out subpackaging processing, and starts cross-cabin encryption forwarding;
step 3-8: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
step 3-9: after receiving the data, the client decodes the RTP audio data;
step 3-10: and the client sends the decoded data to the earphone for playing.
Optionally, the single-cabin client and multi-station communication method comprises the following steps:
Step 4-1: logging in a radio station through a system management interface;
step 4-2: the voice processing workstation acquires the information of the online radio station through the heartbeat information;
step 4-3: the seat voice system client software is connected with a digital radio station or a voice processing terminal through an SIP protocol with a radio station type number, and initiates an Invite request to start calling;
step 4-4: the digital radio station or the voice processing terminal returns a response indicating successful connection to the seat 1 client;
step 4-5: the digital radio station or the voice processing terminal returns a response indicating successful connection to the response information of the seat 1 client;
step 4-6: after receiving the information, the seat client sends ACK information to the digital radio station or the voice processing terminal for confirmation;
step 4-7: the communication connection is established between the client and the digital radio or the voice processing terminal, and the communication is started;
step 4-8: the digital radio station or the voice processing terminal compresses the received voice according to a contracted compression format and sends the compressed voice to a seat 1 client;
step 4-9: the seat client side continuously receives the voice of the digital radio station or the voice processing terminal through configuration;
step 4-10: the seat client sends RTP heartbeat information to keep connection when the PTT is lifted;
Step 4-11: the seat client transmits an extended RTP data packet with ED137 to a digital radio station or a voice processing terminal when the PTT is pressed down;
step 4-12: the digital radio station or the voice processing terminal receives the voice and converts the voice into analog audio to be sent by the radio station;
step 4-13: the voice processing workstation receives and stores the voice realization sent by the client and the digital radio or the voice processing terminal.
According to another aspect of the embodiment of the present invention, there is provided a voice communication system based on a ground station networking, for implementing any of the methods, including: the system comprises a voice processing workstation, a voice system client, a voice processing terminal based on an analog voice link, a cross-cabin communication link, a telephone gateway, a digital radio station and a network switch; the voice processing workstation is connected with other equipment through switch network equipment, and the internal software of the voice processing workstation comprises an internal call communication module, a conference communication module, a digital radio communication module, an analog radio communication module, a program-controlled telephone communication module, a voice recognition module, a voice synthesis module, a cross-cabin communication module and an autonomous network link module;
the internal call communication module refers to point-to-point communication of voice system client software, the client software and the client software operate through an interface and execute a Session Initiation Protocol (SIP) protocol to dial, when two clients are connected, voice communication is started through an RTP protocol, an RTP voice data stream passes through a voice processing workstation, and the voice processing workstation completes voice recording, management of a communication path protocol, link load management and volume adjustment;
The conference communication module is used for carrying out multiparty voice communication in a conference room by a plurality of clients, and the voice processing workstation is used for carrying out voice mixing processing on voice data of a conference according to set requirements besides completing management of a communication path, link load control and voice recording;
the digital radio station communication module is used as a recorder for voice recording of the voice system client software and the digital radio station;
the analog radio station communication module, the voice processing workstation is used as a recorder for voice recording of the voice system client software and the digital radio station;
the program-controlled telephone communication module is used for communicating the workstation with the program-controlled telephone and the telecommunication exchanger, sending out a connection instruction and voice data through the control of the SIP protocol by the client software, completing the dialing of the program-controlled telephone and the external telephone by the voice processing workstation, transmitting voice to the telephone gateway through the RTP protocol, and being used for the management, the link load control and the voice recording of the dialing of the telephone and the communication path by the client software;
the cross-cabin communication module is used for performing cross-cabin protocol coding on the VoIP communication data by the voice processing workstation, converting the communication data into a communication protocol of a cross-cabin encryption transmission terminal, and analyzing the communication protocol by another voice processing workstation after the communication protocol is transmitted by a link, wherein the communication protocol is used for VoIP communication based on a cross-cabin link;
The voice processing workstation carries out autonomous network Ad Hoc link protocol coding on VoIP communication data, converts the VoIP communication data into an Ad Hoc link communication protocol, connects a remote station group with a current station group through an autonomous network and is used for voice, conference, file and text communication among autonomous network nodes;
the voice recognition module is provided with a plurality of voice recognition services in the voice processing workstation and is used for commanding the recognition to simplify operation and free grammar and converting stored audio into text;
and the voice synthesis module is used for synthesizing voice in the voice processing workstation, the external equipment sends the alarm information to the voice processing workstation through an interface protocol, and the voice processing workstation converts the information into audio and sends the audio to the client in the workstation to realize voice alarm.
Optionally, the cross-cabin communication link includes a fiber optic encryption link, an inter-drone, an air-to-ground autonomous network encryption link, and a UV encryption link.
Optionally, the client software of the voice system in the voice system further comprises an interface display module, an address book module, a point-to-point voice call module, a conference call module, a digital radio call module, an analog radio call module, and a file transmission module, wherein
The interface display module provides a UI interaction interface;
the address book module is used for immediately synchronizing the client information, entering an organization address book, checking the online/offline information of the client, rapidly positioning the client to be searched, and checking the detailed information of the client;
the point-to-point voice call module is characterized in that client software is connected with other clients in the system through an SIP protocol, and after the other clients are connected, the client software realizes the mutual voice call of the clients through links formed by a plurality of workstations and cross-cabin connection guiding equipment through an RTP protocol;
the conference call module is used for supporting the mutual call among a plurality of client sections after entering a preset conference room through client software, and sending characters and shared files;
the voice system client software expands the VoIP protocol into ED137 protocol, and the voice system client software transmits SIP, RTP, ED137 expansion frames to be directly connected with the digital radio for realizing PTT function and voice transmission, and simultaneously transmits voice data to the voice processing workstation to complete voice recording, wherein the client software is used for voice receiving and PTT transmitting of a single radio and also receives voice mixing signals of multiple radio;
The simulation radio station communication module is characterized in that voice system client software expands a VoIP protocol into an ED137 protocol, the voice system client software sends SIP, RTP, ED expansion frames to a voice processing hardware terminal, the voice processing hardware terminal can automatically complete connection with the client software, and a voice processing workstation starts voice recording after connection;
the file transmission module is used for transmitting common files, video files and audio files;
optionally, the voice processing based on the analog voice link is provided with a standard ED137 protocol, and the voice processing based on the analog voice link is internally composed of a data processing unit and an analog audio impedance matching unit. The multi-channel analog voice equipment can be used for converting analog voice into ED137RTP audio through a data processing unit after impedance matching or converting ED137RTP audio input by the system into analog voice to be sent to external equipment, so that the traditional radio station and voice equipment can be connected into the system related to the large-scale ground station networking voice communication method, wherein
The data processing unit is used for digitizing analog audio, converting the input analog audio into a protocol supported by VoIP and completing digital audio coding;
The impedance matching is adapted to analog audio input and output of various relay stations, ground stations and voice terminals;
the cross-cabin communication link is based on an optical fiber, an encrypted cross-cabin circuit of an Ethernet and an autonomous network link based on Ad Hoc, and a remote station group and a current station group are connected to form a voice communication network;
the telephone network is connected with the traditional telecommunication line and the voice system and is used for communicating the seat client with the program-controlled telephone and the external telephone, and the heterogeneous network based on the analog signaling interaction model and the communication transmission medium is comprehensively accessed into the unified system.
The system mainly comprises a voice processing terminal, a voice processing workstation, a voice system client, a cross-cabin link, a telephone gateway, a digital radio station, an analog radio station and a gigabit network switch.
The voice communication method based on the large ground station networking comprises a single-cabin point-to-point communication method, a cross-cabin point-to-point communication method and a conference communication method among multiple cabin clients. A communication method of a single cabin client and multiple radio stations.
The invention relates to voice processing workstations which are distributed in a plurality of shelter and are voice access equipment with powerful functions, so that radio stations, voice system clients, voice processing terminals, program control gateways, autonomous network links such as unmanned aerial vehicles, air spaces and the like, cross-cabin encryption leading links and the like are conveniently integrated and are accessed into a system together, and the functions of inter-cabin voice communication, radio station communication, various cross-cabin link communication, text communication, file sharing, voice recording, historical voice tracing and the like are realized. The voice processing workstation is also integrated with a voice recognition function, a free text-to-electricity function, a voice synthesis function, a voice alarm function, and the functions of voice control, text retrieval of historical voice records, voice playing of alarm signals and the like of the ground station seat user of the unmanned plane can be realized through cross-linking with other ground station equipment.
The invention relates to a voice processing hardware terminal which provides a connection interface for an unmanned aerial vehicle analog voice link and converts multiple paths of traditional analog voices into VoIP for transmission according to a standard ED137 protocol. The voice processing terminal is provided with 6 gigabit Ethernet, and 6 analog voice input and output with configurable electrical interfaces. 6 RS422 interactive interfaces, 6 physical PTT output interfaces. The voice processing terminal can be connected with 6 unmanned aerial vehicle ground radio stations, a relay radio station and an analog terminal at most. The voice system client may listen to the audio of one or more stations through the interface settings. The terminal for converting the traditional analog voice terminal into the ED137 digital radio station has the characteristic of modularization, and the economy of the terminal can also meet the reconstruction application requirements of the ground station of the traditional unmanned aerial vehicle.
The invention relates to voice system client software which is installed on a seat computer of an unmanned aerial vehicle ground station, and each seat has the functions of password login, point-to-point communication, conference communication, radio station communication and the like.
1. Compared with the traditional VoIP, the technical scheme of the invention is more flexible;
2. the bandwidth actually occupied by the technical scheme of the invention in the communication process is lower;
3. the protocol is easy to define, and is convenient to be compatible with the business data frame;
4. The protocol strategy takes precedence of digital voice transmission, and ensures accurate preferential restoration of sound;
5. the protocol is customized, and the security of service data is high;
6. a voice recognition and voice control database is reserved, so that later voice authentication is facilitated;
drawings
Fig. 1 is a topology diagram of a voice communication method based on a ground station networking according to an embodiment of the present invention.
Fig. 2 is a single-station connection diagram of a voice communication method based on a ground station networking according to an embodiment of the present invention.
Fig. 3 is a communication module data flow diagram of a voice communication method based on a ground station networking according to an embodiment of the present invention.
Fig. 4 is a diagram of single cabin point-to-point voice communication steps of a call in a voice communication method based on a ground station networking according to an embodiment of the present invention.
Fig. 5 is a cross-cabin point-to-point voice communication step diagram of a voice communication method based on ground station networking according to an embodiment of the present invention.
Fig. 6 is a diagram of conference communication steps between multiple cabin clients of a voice communication method based on ground station networking according to an embodiment of the present invention.
Fig. 7 is a diagram of single-cabin client and multi-station communication steps of a voice communication method based on ground station networking according to an embodiment of the present invention.
Detailed Description
The following description of the embodiments of the present invention will be made clearly and completely with reference to the accompanying drawings, in which it is apparent that the embodiments described are only some embodiments of the present invention, but not all embodiments. All other embodiments, which can be made by those skilled in the art based on the embodiments of the invention without making any inventive effort, are intended to be within the scope of the invention.
The invention provides a voice communication method based on ground station networking, which comprises the following steps:
step 2-1: the seat of the cabin A logs in a voice processing workstation of the cabin A through voice system client software;
step 2-2: the voice processing workstation of the cabin A acquires information of all seat 1 clients of the cabin, and multicast sharing is carried out on the information to other cabins by using a UDT protocol through a cross-cabin encryption link or an autonomous network link;
step 2-3: the voice processing workstation of the cabin B acquires and stores information of the seat 1 client side of the cabin A;
step 2-4: the voice system client software of the cabin A seat sends an SIP command with the cabin B client number to the cabin A voice processing workstation through an SIP protocol according to the user information of the cabin B client to realize sending of a cross-cabin Invite request to start calling;
Step 2-5: after the cabin A voice processing workstation confirms that the user authentication is passed, the voice processing workstation inserts an own address in a Via header field of the request message, packages the Invite message according to a cross-cabin or autonomous network protocol, and distributes the Invite message to a cross-cabin encryption link receiving device according to a load requirement of a cross-cabin encryption or autonomous network link;
step 2-6: the cross-cabin encryption link guiding equipment encrypts the sub-packet data and sends the encrypted sub-packet data to the cabin B;
step 2-7: the cabin B cross-cabin encryption link receiving equipment decrypts and packetizes and sends the decryption and the packetization to a cabin B voice processing workstation according to an encryption protocol;
step 2-8: the cabin B voice processing workstation receives the sub-packet data packet sent by the cross-cabin encryption link and then sends an Invite request To a cabin B seat client according To the To domain of the message;
step 2-9: the cabin B seat client sends response information with SIP cross-cabin expansion frames in call processing to the cabin B voice processing workstation;
step 2-10: the cabin B voice processing workstation receives SIP cabin-crossing response information sent by the equipment to the cabin B cabin-crossing encryption link; and sub-packaging;
step 2-11: the cabin B sends encrypted data to the cabin A after being encrypted by the cabin B cross-cabin encryption link guide receiving equipment;
step 2-12: the cabin A cross-cabin encryption link guide equipment performs decryption and then sends the decrypted decryption to a voice processing workstation of the cabin A;
Step 2-13: the voice processing workstation of the cabin A sends SIP cross-cabin response information in the software call processing of the seat client of the cabin A;
step 2-14: the cabin B seat client side indicates the called user to ring, and after the user rings, ringing information is sent to the cabin B voice processing workstation;
step 2-15: the cabin B voice processing workstation executes the encryption forwarding steps from the step 2-9 to the step 2-12 to forward the ringing information of the called user to the cabin A seat client;
step 2-16: the cabin B seat client returns a response indicating that the connection is successful to the cabin B voice processing workstation;
step 2-17: after the cabin B voice processing workstation divides the packets, forwarding the success indication to the cabin A client of the cabin A voice processing workstation through a cross-cabin encryption link;
step 2-18: the cabin A voice processing workstation forwards the success indication to the cabin A client;
step 2-19: after receiving the information, the cabin A client sends ACK information to the cabin A voice processing workstation for confirmation;
step 2-20: the cabin A voice processing workstation is used for transmitting the packetized data to the cabin B voice processing workstation through a cross-cabin encryption link;
step 2-21: the cabin B voice processing workstation forwards the ACK confirmation message to the cabin B client;
step 2-22: and the calling user and the called user establish communication connection and start talking. The client software of the seat compresses the voice data according to a contracted compression format;
Step 2-23: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 2-24: each cabin voice processing workstation stores voice data according to RTP voice data;
step 2-25: and the voice processing workstations of each cabin carry out sub-packaging processing according to the cabin bit of the RTP extension header data receiving party and the client, and start cross-cabin encryption forwarding.
Step 2-26: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
step 2-27: after receiving the data, the client decodes the RTP audio data;
step 2-28: and the client sends the decoded data to the sound card for playing.
Further, the text and file transmission communication method comprises the following steps in addition to the steps 2-1 to 2-20:
step 2-29: after communication connection is established between the clients, cabin A client software sends relevant text information and file data to be subjected to lossless compression and then sends the compressed text information and file data to a cabin A voice processing workstation according to a UDT format so as to reduce bandwidth occupation;
step 2-30: the cabin A voice processing workstation stores text information and file data according to the UDT data;
Step 2-31: and the cabin A voice processing workstation determines a client receiving the shelter according to the UDT extension header data to forward. The sub-packaging is executed under the condition of crossing the cabin, and the sub-packaging is sent to the cabin A guiding equipment for encryption transmission;
step 2-32: the cabin B voice processing workstation receives decryption data of the cabin B guiding equipment and then packs the decryption data;
step 2-33: the cabin B voice processing workstation stores the packed data and forwards the data to a client of the cabin B;
step 2-34: after receiving the data, the client of the cabin B decompresses the UDT text information and the file data;
step 2-35: and displaying text information or storing files on the client side of the cabin B.
Further, a single-cabin point-to-point voice communication method, the method further comprising:
step 1-1: logging in a voice processing workstation through voice system client software;
step 1-2: the voice processing workstation acquires information of all online clients of the cabin;
step 1-3: the seat 1 voice system client software initiates an Invite request to the voice processing workstation through a standard SIP protocol to initiate a call;
step 1-4: after the voice processing workstation confirms that the user authentication is passed, checking whether the Via header field in the request message contains the address of the user authentication; if yes, the loop back is indicated to occur, and a response indicating the error is returned; if not, the voice processing workstation inserts the self address in the Via header field of the request message and transmits an Invite request To the called seat 2 client indicated by the To field of the Invite message;
Step 1-5: the seat 2 client sends response information in call processing to the voice processing workstation;
step 1-6: the voice processing workstation sends response information in call processing to the seat 1 client;
step 1-7: the seat 2 client side indicates the called user to ring, and after the user rings, the ringing information is sent to the voice processing workstation;
step 1-8: the voice processing workstation forwards the ringing information of the called user to the seat 1 client;
step 1-9: the seat 2 client returns a response indicating that the connection is successful to the voice processing workstation;
step 1-10: the voice processing workstation forwards the success indication to the seat 1 client;
step 1-11: after receiving the information, the seat 1 client sends ACK information to the voice processing workstation for confirmation;
step 1-12: the voice processing workstation forwards the ACK confirmation message to the seat 2 client;
step 1-13: communication connection is established between the calling user and the called user, communication is started, and the client software of the seat compresses voice data according to a contracted compression format;
step 1-14: the seat client software sends the compressed and encoded data to a voice processing workstation according to an RTP format;
step 1-15: the voice processing workstation stores voice data according to the RTP voice data;
Step 1-16: and the voice processing workstation judges the receiver client according to the RTP extension header data and forwards the receiver client.
Step 1-17: after receiving the data, the client decodes the RTP audio data;
step 1-18: the client sends the decoded data to the sound card for playing;
further, the text and file transmission communication method includes the steps of:
step 1-19: the clients are connected in a communication way, and seat 1 client software sends relevant text information and file data to a voice processing workstation according to a UDT format after lossless compression;
step 1-20: the voice processing workstation stores text information and file data according to the UDT data;
step 1-16: the voice processing workstation determines a receiver client according to the UDT extension header data and forwards the receiver client;
step 1-17: after receiving the data, the client decompresses the UDT text information and the file data;
step 1-18: the client displays the text information or saves the file.
Further, the conference communication method between the clients among the multiple cabins comprises the following steps:
step 3-1: the cabin A seat 1 establishes a conference room through voice system client software;
Step 3-2: after receiving the conference room establishment message, the voice processing workstation of the cabin A uses a UDT protocol to multicast and share the conference room information to other cabins through a cross-cabin encryption link or an autonomous network link;
step 3-3: other client software enters a conference room by using a Session Initiation Protocol (SIP) handshake command with a conference number; each cabin workstation uses UDT protocol to multicast and share conference room information to other cabins through a cross-cabin encryption link or an autonomous network link;
step 3-4: all client seat client software respectively carries out compression coding on local microphone voice data according to a contracted compression format;
step 3-5: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 3-6: each cabin voice processing workstation stores the voice data according to the RTP voice data;
step 3-7: each cabin voice processing workstation determines a conference shelter according to the RTP extension header data, then carries out subpackaging processing, and starts cross-cabin encryption forwarding;
step 3-8: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
Step 3-9: after receiving the data, the client decodes the RTP audio data;
step 3-10: and the client sends the decoded data to the earphone for playing.
Further, the single-cabin client and multi-station communication method comprises the following steps:
step 4-1: logging in a radio station through a system management interface;
step 4-2: the voice processing workstation acquires the information of the online radio station through the heartbeat information;
step 4-3: the seat voice system client software is connected with a digital radio station or a voice processing terminal through an SIP protocol with a radio station type number, and initiates an Invite request to start calling;
step 4-4: the digital radio station or the voice processing terminal returns a response indicating successful connection to the seat 1 client;
step 4-5: the digital radio station or the voice processing terminal returns a response indicating successful connection to the response information of the seat 1 client;
step 4-6: after receiving the information, the seat client sends ACK information to the digital radio station or the voice processing terminal for confirmation;
step 4-7: the communication connection is established between the client and the digital radio or the voice processing terminal, and the communication is started;
step 4-8: the digital radio station or the voice processing terminal compresses the received voice according to a contracted compression format and sends the compressed voice to a seat 1 client;
Step 4-9: the seat client side continuously receives the voice of the digital radio station or the voice processing terminal through configuration;
step 4-10: the seat client sends RTP heartbeat information to keep connection when the PTT is lifted;
step 4-11: the seat client transmits an extended RTP data packet with ED137 to a digital radio station or a voice processing terminal when the PTT is pressed down;
step 4-12: the digital radio station or the voice processing terminal receives the voice and converts the voice into analog audio to be sent by the radio station;
step 4-13: the voice processing workstation receives and stores the voice realization sent by the client and the digital radio or the voice processing terminal.
The invention also provides a voice communication system based on the ground station networking, which is used for realizing any method, and comprises the following steps: the system comprises a voice processing workstation, a voice system client, a voice processing terminal based on an analog voice link, a cross-cabin communication link, a telephone gateway, a digital radio station and a network switch; the voice processing workstation is connected with other equipment through switch network equipment, and the internal software of the voice processing workstation comprises an internal call communication module, a conference communication module, a digital radio communication module, an analog radio communication module, a program-controlled telephone communication module, a voice recognition module, a voice synthesis module, a cross-cabin communication module and an autonomous network link module;
The internal call communication module refers to point-to-point communication of voice system client software, the client software and the client software operate through an interface and execute a Session Initiation Protocol (SIP) protocol to dial, when two clients are connected, voice communication is started through an RTP protocol, an RTP voice data stream passes through a voice processing workstation, and the voice processing workstation completes voice recording, management of a communication path protocol, link load management and volume adjustment;
the conference communication module is used for carrying out multiparty voice communication in a conference room by a plurality of clients, and the voice processing workstation is used for carrying out voice mixing processing on voice data of a conference according to set requirements besides completing management of a communication path, link load control and voice recording;
the digital radio station communication module is used as a recorder for voice recording of the voice system client software and the digital radio station;
the analog radio station communication module, the voice processing workstation is used as a recorder for voice recording of the voice system client software and the digital radio station;
the program-controlled telephone communication module is used for communicating the workstation with the program-controlled telephone and the telecommunication exchanger, sending out a connection instruction and voice data through the control of the SIP protocol by the client software, completing the dialing of the program-controlled telephone and the external telephone by the voice processing workstation, transmitting voice to the telephone gateway through the RTP protocol, and being used for the management, the link load control and the voice recording of the dialing of the telephone and the communication path by the client software;
The cross-cabin communication module is used for performing cross-cabin protocol coding on the VoIP communication data by the voice processing workstation, converting the communication data into a communication protocol of a cross-cabin encryption transmission terminal, and analyzing the communication protocol by another voice processing workstation after the communication protocol is transmitted by a link, wherein the communication protocol is used for VoIP communication based on a cross-cabin link;
the voice processing workstation carries out autonomous network Ad Hoc link protocol coding on VoIP communication data, converts the VoIP communication data into an Ad Hoc link communication protocol, connects a remote station group with a current station group through an autonomous network and is used for voice, conference, file and text communication among autonomous network nodes;
the voice recognition module is provided with a plurality of voice recognition services in the voice processing workstation and is used for commanding the recognition to simplify operation and free grammar and converting stored audio into text;
and the voice synthesis module is used for synthesizing voice in the voice processing workstation, the external equipment sends the alarm information to the voice processing workstation through an interface protocol, and the voice processing workstation converts the information into audio and sends the audio to the client in the workstation to realize voice alarm.
Further, the cross-cabin communication links include fiber optic encryption links, inter-unmanned aerial vehicle, air-to-ground autonomous network encryption links, and UV encryption links.
Further, the voice system client software in the voice system further comprises an interface display module, an address book module, a point-to-point voice call module, a conference call module, a digital radio call module, an analog radio call module and a file transmission module, wherein
The interface display module provides a UI interaction interface;
the address book module is used for immediately synchronizing the client information, entering an organization address book, checking the online/offline information of the client, rapidly positioning the client to be searched, and checking the detailed information of the client;
the point-to-point voice call module is characterized in that client software is connected with other clients in the system through an SIP protocol, and after the other clients are connected, the client software realizes the mutual voice call of the clients through links formed by a plurality of workstations and cross-cabin connection guiding equipment through an RTP protocol;
the conference call module is used for supporting the mutual call among a plurality of client sections after entering a preset conference room through client software, and sending characters and shared files;
the voice system client software expands the VoIP protocol into ED137 protocol, and the voice system client software transmits SIP, RTP, ED137 expansion frames to be directly connected with the digital radio for realizing PTT function and voice transmission, and simultaneously transmits voice data to the voice processing workstation to complete voice recording, wherein the client software is used for voice receiving and PTT transmitting of a single radio and also receives voice mixing signals of multiple radio;
The simulation radio station communication module is characterized in that voice system client software expands a VoIP protocol into an ED137 protocol, the voice system client software sends SIP, RTP, ED expansion frames to a voice processing hardware terminal, the voice processing hardware terminal can automatically complete connection with the client software, and a voice processing workstation starts voice recording after connection;
the file transmission module is used for transmitting common files, video files and audio files;
further, the voice processing based on the analog voice link has a standard ED137 protocol, and the voice processing based on the analog voice link is internally composed of a data processing unit and an analog audio impedance matching unit. The multi-channel analog voice equipment can be used for converting analog voice into ED137 RTP audio through a data processing unit after impedance matching or converting ED137 RTP audio input by the system into analog voice to be sent to external equipment, so that the traditional radio station and voice equipment can be connected into the system related to the large-scale ground station networking voice communication method, wherein
The data processing unit is used for digitizing analog audio, converting the input analog audio into a protocol supported by VoIP and completing digital audio coding;
The impedance matching is adapted to analog audio input and output of various relay stations, ground stations and voice terminals;
the cross-cabin communication link is based on an optical fiber, an encrypted cross-cabin circuit of an Ethernet and an autonomous network link based on Ad Hoc, and a remote station group and a current station group are connected to form a voice communication network;
the telephone network is connected with the traditional telecommunication line and the voice system and is used for communicating the seat client with the program-controlled telephone and the external telephone, and the heterogeneous network based on the analog signaling interaction model and the communication transmission medium is comprehensively accessed into the unified system.
The invention is applicable to a network topology architecture of a system related to a voice communication method based on large-scale ground station networking. The mature VoIP communication protocol is applied to improve the integration capability of the unmanned aerial vehicle ground station voice system and other voice equipment (radio stations and telephone gateways), and the practicability and the universality of the unmanned aerial vehicle ground station voice system are improved.
The unmanned aerial vehicle ground station seat computer and various seat headsets are integrated, the number of system equipment is effectively reduced, the complexity of the system is reduced, and the operability and the light-weight level of the unmanned aerial vehicle ground station voice system are improved.
A set of large-scale communication system is formed for seat operators, radio stations, telephone systems, cross-cabin encryption links and unmanned aerial vehicle autonomous network Ad Hoc links, a set of low-flow high-confidentiality communication protocol is established, and interconnection and barrier-free cross-cabin communication are realized.
The method has the function of multimedia communication, and realizes the functions of voice communication, text communication, file sharing and the like in a system related to a voice communication method based on large-scale ground station networking. The method provides more effective processing of the sophisticated tasks under complex battlefield and precise command control battle for operators.
Examples
A voice communication method based on large ground station networking considers communication distance and network architecture, and a single voice processing workstation cannot realize voice communication and control requirements of large ground station networking. Therefore, the system needs to support a large-scale network topology architecture so as to meet the networking voice communication requirement of the ground station of the large unmanned aerial vehicle.
By combining the characteristics of the use environment of the ground station of the unmanned aerial vehicle, the system can support autonomous network links such as the space between the unmanned aerial vehicle and the air space to realize encryption cascade of multiple cabins, and a cascade network structure is formed by a plurality of ground station workstation nodes. At the moment, the voice processing work stations in the ground stations of each unmanned plane can independently meet the voice and text service functions of clients in the stations, and meanwhile, each work station can forward the client information of the station to other stations to finish registration, so that the establishment of multi-station encrypted communication channels is realized. The topology is shown in fig. 1.
A related system of a voice communication method based on large ground station networking mainly comprises a voice processing workstation, a voice system client, a voice processing terminal based on an analog voice link, a cross-cabin communication link, a telephone gateway, a digital radio station and a gigabit network switch.
The voice processing workstation is connected with other devices through network devices such as a gigabit Ethernet switch, and the internal software of the voice processing workstation consists of modules such as internal telephone communication, conference communication, digital radio communication, analog radio communication, program-controlled telephone communication, voice recognition, voice synthesis, cross-cabin communication, inter-machine and air-ground autonomous network links and the like. The single station connection diagram is shown in fig. 2.
The internal call communication module refers to point-to-point communication of the client software of the voice system, the client software and the client software operate through an interface, the SIP protocol is executed to dial, when the two clients are connected, voice communication is started through the RTP protocol, the RTP voice data flow passes through a voice processing workstation, and the voice processing workstation completes the functions of voice recording, management of a communication path protocol, link load management, volume adjustment and the like, and a data flow diagram is shown in figure 3.
And the conference communication module is used for carrying out multiparty voice communication in the set conference room by a plurality of clients. Besides completing management of communication channels, link load control and voice recording, the voice processing workstation also carries out voice mixing processing on voice data of the conference according to set requirements so as to meet the voice requirements of all clients.
The digital radio communication module, the voice processing workstation is used as a recorder to realize the voice recording of the voice system client software and the digital radio.
And the analog radio station communication module, the voice processing workstation is used as a recorder to realize voice recording of the voice system client software and the digital radio station.
The program-controlled telephone communication module is mainly used for communicating a workstation of the unmanned aerial vehicle ground station with a program-controlled telephone and a telecommunication switch, sending a connection instruction and voice data through SIP protocol control by client software, completing dialing of the program-controlled telephone and the external telephone by a voice processing workstation, transmitting voice to a telephone gateway through RTP protocol, realizing the function of dialing the telephone by the client software and realizing the functions of management of a communication path, link load control and voice recording.
The cross-cabin communication module is characterized in that the voice processing workstation carries out cross-cabin protocol coding on VoIP communication data, protocols such as SIP, RTP, UTP are converted into a communication protocol of a cross-cabin encryption transmission terminal, the communication protocol is transmitted through a link and then analyzed by another voice processing workstation, voIP communication based on the cross-cabin link is realized, and a plurality of large unmanned aerial vehicle ground stations have communication functions such as mutual voice, conference, file, text and the like. Wherein the cross-cabin communication link comprises an optical fiber encryption link, an unmanned aerial vehicle inter-aircraft autonomous network encryption link, an air-ground autonomous network encryption link, a UV encryption link and the like.
The voice processing workstation carries out autonomous network Ad Hoc link protocol coding on VoIP communication data, converts SIP, RTP, UTP and other protocols into Ad Hoc link communication protocols, realizes networking of a far-end ground station group and a current ground station group through an air autonomous network of the unmanned aerial vehicle, and realizes communication functions of voice, conference, file, text and the like among autonomous network nodes.
The voice recognition module is provided with 6 paths of voice recognition services in the voice processing workstation, can realize instruction recognition to simplify unmanned aerial vehicle operation of seat operators, can also realize a free text function, converts stored audio into text, and is convenient to search and inquire.
The voice synthesis module is used for synthesizing voice in the voice processing workstation, the external equipment can send the alarm information to the voice processing workstation through an interface protocol, and the voice processing workstation converts the information into audio and sends the audio to the client in the workstation to realize voice alarm.
The voice system client software in the voice system of the unmanned aerial vehicle ground station can be deployed on a seat computer with an operating system such as Windows, android, harmonyOS. The client software has an interface display function, an address book function, a point-to-point voice call function, a conference call function, a digital radio call function, an analog radio call function and a file transmission function module.
The interface display function module is mainly used for providing a simple and convenient UI (user interface) for seat users, wherein the main interfaces comprise a login interface, an address book interface, a text chat interface, a voice call interface, a radio station interface, a conference interface, a setting interface module and the like.
The address book functional module can instantly synchronize the organization structure of the command station system, the ground station and various client information. And entering an organization address book, looking at the ground station list and the on-line/off-line information of the clients under the station, rapidly positioning the clients to be searched, and looking up the detailed information of the clients to conveniently initiate chat session.
The point-to-point voice call function module is that client software is connected with other clients in the system through an SIP protocol, and after the other clients are connected, the client software realizes the mutual voice call of the clients through links formed by a plurality of workstations and cross-cabin connection guiding equipment through an RTP protocol.
The conference call function refers to that after a seat operator enters a preset conference room through client software, a plurality of client sections can communicate with each other and send characters and shared files.
And the voice system client software expands the VoIP protocol into the ED137 protocol, and the voice system client software transmits SIP, RTP, ED expansion frames to be directly connected with the digital radio, so that the PTT function and the voice transmission function are realized, and simultaneously, voice data is transmitted to the voice processing workstation to complete voice recording. The client software can realize the voice receiving and PTT sending of a single radio station and also receive the voice mixing signals of multiple radio stations.
And the voice system client software expands the VoIP protocol into the ED137 protocol, the voice system client software transmits SIP, RTP, ED expansion frames to the voice processing hardware terminal, the voice processing hardware terminal can automatically complete connection with the client software, and a voice processing workstation starts voice recording after connection.
The file transmission function module is used for realizing multimedia data communication between the client and the client through the voice processing workstation by utilizing the UTP protocol, and can send common files, video files and audio files.
The speech processing based on the analog voice link has a standard ED137 protocol, and the inside of the speech processing based on the analog voice link is composed of a data processing unit and an analog audio impedance matching unit. The multi-channel analog voice equipment can be used for converting analog voice into ED137 RTP audio through a data processing unit after impedance matching or converting ED137 RTP audio input by a system into analog voice to be sent to external equipment, so that the equipment such as a traditional radio station, voice and the like can be accessed into the system related to the large-scale ground station networking voice communication method.
The data processing unit has an analog audio digitizing function, converts input analog audio into a protocol supported by VoIP and completes digital audio encoding.
The impedance matching can be adapted to analog audio input and output of various relay stations, ground stations and voice terminals.
The cross-cabin communication link is mainly an encryption cross-cabin circuit based on optical fibers and Ethernet and an unmanned aerial vehicle autonomous network link based on Ad Hoc, and a remote ground station group and a current ground station group are connected to form a large-scale voice communication network.
The telephone network is a bridge for connecting the traditional telecommunication line and the voice system of the unmanned aerial vehicle ground station, is mainly used for the communication between the seat client and the program-controlled telephone and between the seat client and the external telephone, and is used for comprehensively accessing the heterogeneous network based on the analog signaling interaction model and the communication transmission medium into a unified system.
The voice communication method based on the large ground station networking comprises a single cabin point-to-point communication method, a cross cabin point-to-point communication method and a cross cabin client conference communication method. Single-cabin clients and multi-station communication methods.
As shown in fig. 4. The single cabin point-to-point voice communication method specifically comprises the following steps:
step 1-1: a seat operator logs in a voice processing workstation through voice system client software;
step 1-2: the voice processing workstation acquires information of all online clients of the cabin;
Step 1-3: the seat 1 voice system client software initiates an Invite request to the voice processing workstation through a standard SIP protocol to initiate a call;
step 1-4: after the voice processing workstation confirms that the user authentication has passed, it checks whether the Via header field in the request message already contains its address. If so, the instruction loops back, and returns a response indicating the error; if there is no problem, the voice processing workstation inserts its own address in the Via header field of the request message and transmits an Invite request To the called seat 2 client indicated by the To field of the Invite message.
Step 1-5: the seat 2 client sends response information in call processing to the voice processing workstation: 100Trying;
step 1-6: the voice processing workstation sends response information in call processing to the seat 1 client side: 100Trying;
step 1-7: the seat 2 client side indicates the called user to ring, and after the user rings, 180Ringing information is sent to the voice processing workstation;
step 1-8: the voice processing workstation forwards the ringing information of the called user to the seat 1 client;
step 1-9: the seat 2 client returns a response (200 OK) indicating that the connection was successful to the speech processing workstation;
step 1-10: the voice processing workstation forwards the success indication (200 OK) to the seat 1 client;
Step 1-11: after receiving the information, the seat 1 client sends ACK information to the voice processing workstation for confirmation;
step 1-12: the voice processing workstation forwards the ACK confirmation message to the seat 2 client;
step 1-13: and the calling user and the called user establish communication connection and start talking. The seat client software compresses the voice data according to the appointed compression format;
step 1-14: the seat client software sends the compressed and encoded data to a voice processing workstation according to an RTP format;
step 1-15: the voice processing workstation stores voice data according to the RTP voice data;
step 1-16: the voice processing workstation judges which client the receiver is for forwarding according to the RTP extension header data.
Step 1-17: after receiving the data, the client decodes the RTP audio data;
step 1-18: and the client sends the decoded data to the sound card for playing.
The text and file transmission communication method includes the following steps in addition to steps 1-1 to 1-12
Step 1-19: the clients are connected in a communication way, seat 1 client software sends relevant text information and file data to a voice processing workstation according to a UDT format after lossless compression, so that bandwidth occupation is reduced;
Step 1-20: the voice processing workstation stores text information and file data according to the UDT data;
step 1-16: the voice processing workstation judges which client the receiver is for forwarding according to the UDT extension header data.
Step 1-17: after receiving the data, the client decompresses the UDT text information and the file data;
step 1-18: the client displays the text information or saves the file.
As shown in fig. 5. The cross-cabin point-to-point voice communication method specifically comprises the following steps:
step 2-1: the seat operator of the cabin A logs in a voice processing workstation of the cabin A through voice system client software;
step 2-2: the voice processing workstation of the cabin A acquires information of all seat 1 clients of the cabin, and multicast sharing is carried out on other cabins by using UDT protocol through cross-cabin encryption links or autonomous network links such as unmanned aerial vehicle inter-plane links, air-ground links and the like;
step 2-3: the voice processing workstation of the cabin B acquires and stores information of the seat 1 client side of the cabin A;
step 2-4: the voice system client software of the cabin A seat sends an SIP command with the cabin B client number to the cabin A voice processing workstation through an SIP protocol according to the user information of the cabin B client to realize sending of a cross-cabin Invite request to start calling;
Step 2-5: after the cabin A voice processing workstation confirms that the user authentication is passed, the voice processing workstation inserts an own address in a Via header field of the request message, packages the Invite message according to a cross-cabin or autonomous network protocol, and distributes the Invite message to a cross-cabin encryption link receiving device according to a load requirement of a cross-cabin encryption or autonomous network link;
step 2-6: the cross-cabin encryption link guiding equipment encrypts the sub-packet data and sends the encrypted sub-packet data to the cabin B;
step 2-7: the cabin B cross-cabin encryption link receiving equipment decrypts and packetizes and sends the decryption and the packetization to a cabin B voice processing workstation according to an encryption protocol;
step 2-8: the cabin B voice processing workstation receives the sub-packet data packet sent by the cross-cabin encryption link and then sends an Invite request To a cabin B seat client according To the To domain of the message;
step 2-9: the cabin B seat client sends response information with SIP cross-cabin expansion frames in call processing to the cabin B voice processing workstation: 100Trying;
step 2-10: the cabin B voice processing workstation sends SIP cabin-crossing response information to cabin B cabin-crossing encryption link guiding equipment: 100Trying; and sub-packaging;
step 2-11: the cabin B sends encrypted data to the cabin A after being encrypted by the cabin B cross-cabin encryption link guide receiving equipment;
Step 2-12: the cabin A cross-cabin encryption link guide equipment performs decryption and then sends the decrypted decryption to a voice processing workstation of the cabin A;
step 2-13: the cabin A voice processing workstation calls SIP cross-cabin response information in processing to cabin A seat client software: 100Trying;
step 2-14: the cabin B seat client side indicates the called user to ring, and after the user rings, 180Ringing information is sent to the cabin B voice processing workstation;
step 2-15: the cabin B voice processing workstation executes the encryption forwarding steps similar to the steps 2-9 to 2-12 to forward the ringing information of the called user to the cabin A seat client;
step 2-16: the cabin B seat client returns a response (200 OK) indicating that the connection was successful to the cabin B voice processing workstation;
step 2-17: after the cabin B voice processing workstation divides the packets, forwarding the success indication (200 OK) to the cabin A client of the cabin A voice processing workstation through a cross-cabin encryption link;
step 2-18: cabin a speech processing workstation forwards the success indication (200 OK) to the cabin a client
Step 2-19: after receiving the information, the cabin A client sends ACK information to the cabin A voice processing workstation for confirmation;
step 2-20: the cabin A voice processing workstation is used for transmitting the packetized data to the cabin B voice processing workstation through a cross-cabin encryption link;
Step 2-21: the cabin B voice processing workstation forwards the ACK confirmation message to the cabin B client;
step 2-22: and the calling user and the called user establish communication connection and start talking. The seat client software compresses the voice data according to the appointed compression format;
step 2-23: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 2-24: each cabin voice processing workstation stores voice data according to RTP voice data;
step 2-25: and the voice processing workstations of each cabin judge which client of which cabin the receiver is for packetizing according to the RTP extension header data, and start cross-cabin encryption forwarding.
Step 2-26: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
step 2-27: after receiving the data, the client decodes the RTP audio data;
step 2-28: and the client sends the decoded data to the sound card for playing.
The text and file transmission communication method comprises the following steps in addition to the steps 2-1 to 2-20:
Step 2-29: after communication connection is established between the clients, cabin A client software sends relevant text information and file data to be subjected to lossless compression and then sends the compressed text information and file data to a cabin A voice processing workstation according to a UDT format so as to reduce bandwidth occupation;
step 2-30: the cabin A voice processing workstation stores text information and file data according to the UDT data;
step 2-31: the cabin A voice processing workstation judges which cabin the receiver is and which client is for forwarding according to the UDT extension header data. The sub-packaging is executed under the condition of crossing the cabin, and the sub-packaging is sent to the cabin A guiding equipment for encryption transmission;
step 2-32: the cabin B voice processing workstation receives decryption data of the cabin B guiding equipment and then packs the decryption data;
step 2-33: the cabin B voice processing workstation stores the packed data and forwards the data to a client of the cabin B;
step 2-34: after receiving the data, the client of the cabin B decompresses the UDT text information and the file data;
step 2-35: and displaying text information or storing files on the client side of the cabin B.
As shown in fig. 6. The conference communication method between the multi-cabin clients specifically comprises the following steps:
step 3-1: cabin A seat 1 operator establishes a conference room through voice system client software;
Step 3-2: after receiving the conference room establishment message, the voice processing workstation of the cabin A uses a UDT protocol to multicast and share the conference room information to other cabins through a cross-cabin encryption link or an autonomous network link such as an unmanned aerial vehicle, an air space and the like;
step 3-3: other client software enters a conference room by using a Session Initiation Protocol (SIP) handshake command with a conference number; each cabin workstation uses UDT protocol to multicast and share meeting room information to other cabins through a cross-cabin encryption link or an inter-unmanned aerial vehicle, air-ground and other autonomous network links;
step 3-4: all client seat client software respectively compression codes the microphone voice data according to the agreed compression format.
Step 3-5: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 3-6: each cabin voice processing workstation stores the voice data according to the RTP voice data;
step 3-7: each cabin voice processing workstation judges which cabins of the conference party are subjected to sub-packaging processing according to the RTP extension header data, and starts cross-cabin encryption forwarding;
step 3-8: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
Step 3-9: after receiving the data, the client decodes the RTP audio data;
step 3-10: and the client sends the decoded data to the earphone for playing.
As shown in fig. 7. The communication method of the single cabin client and the multi-station specifically comprises the following steps:
step 4-1: a seat operator logs in a radio station through a system management interface;
step 4-2: the voice processing workstation acquires the information of the online radio station through the heartbeat packet;
step 4-3: the seat voice system client software is connected with a digital radio station or a voice processing terminal through an SIP protocol with a radio station ED137 type number, and initiates an Invite request to start calling;
step 4-4: the digital radio or the voice processing terminal returns a response (200 OK) indicating that the connection was successful to the seat 1 client
Step 4-5: the digital radio station or the voice processing terminal returns a response (200 OK) indicating the success of the connection to the response information of the seat 1 client terminal
Step 4-6: seat client receives information
Then, sending ACK information to the digital radio station or the voice processing terminal for confirmation;
step 4-7: and establishing communication connection between the client and the digital radio or the voice processing terminal, and starting communication.
Step 4-8: the digital radio station or the voice processing terminal compresses the received voice according to a contracted compression format and sends the compressed voice to a seat 1 client;
Step 4-9: the seat client can continuously receive the voice of the digital radio station or the voice processing terminal through configuration;
step 4-10: the seat client sends RTP heartbeat packets to keep connection when the PTT is lifted;
step 4-11: the seat client transmits a voice transmission voice with an ED137 extension RTP data packet to the digital radio station or the voice processing terminal when the PTT is pressed;
step 4-12: the digital radio station or the voice processing terminal receives the voice and converts the voice into analog audio to be sent by the radio station.
Step 4-13: the voice processing workstation receives and saves the voice sent by the client and the digital radio or the voice processing terminal.
The foregoing is merely a detailed description of the invention, which is not a matter of routine skill in the art. However, the scope of the present invention is not limited thereto, and any changes or substitutions that can be easily contemplated by those skilled in the art within the scope of the present invention should be included in the scope of the present invention. The protection scope of the present invention shall be subject to the protection scope of the claims.

Claims (10)

1. A voice communication method based on ground station networking is characterized by comprising the following steps of:
Step 2-1: the seat of the cabin A logs in a voice processing workstation of the cabin A through voice system client software;
step 2-2: the voice processing workstation of the cabin A acquires information of all seat 1 clients of the cabin, and multicast sharing is carried out on the information to other cabins by using a UDT protocol through a cross-cabin encryption link or an autonomous network link;
step 2-3: the voice processing workstation of the cabin B acquires and stores information of the seat 1 client side of the cabin A;
step 2-4: the voice system client software of the cabin A seat sends the SIP command entity with the cabin B client number to the cabin A voice processing workstation through the SIP protocol according to the user information of the cabin B client
Sending a cross-cabin Invite request to start calling;
step 2-5: after the cabin A voice processing workstation confirms that the user authentication is passed, the voice processing workstation inserts an own address in a Via header field of the request message, packages the Invite message according to a cross-cabin or autonomous network protocol, and distributes the Invite message to a cross-cabin encryption link receiving device according to a load requirement of a cross-cabin encryption or autonomous network link;
step 2-6: the cross-cabin encryption link guiding equipment encrypts the sub-packet data and sends the encrypted sub-packet data to the cabin B;
step 2-7: the cabin B cross-cabin encryption link receiving equipment decrypts and packetizes and sends the decryption and the packetization to a cabin B voice processing workstation according to an encryption protocol;
Step 2-8: the cabin B voice processing workstation receives the sub-packet data packet sent by the cross-cabin encryption link and then sends an Invite request To a cabin B seat client according To the To domain of the message;
step 2-9: the cabin B seat client sends response information with SIP cross-cabin expansion frames in call processing to the cabin B voice processing workstation;
step 2-10: the cabin B voice processing workstation receives SIP cabin-crossing response information sent by the equipment to the cabin B cabin-crossing encryption link; and sub-packaging;
step 2-11: the cabin B sends encrypted data to the cabin A after being encrypted by the cabin B cross-cabin encryption link guide receiving equipment;
step 2-12: the cabin A cross-cabin encryption link guide equipment performs decryption and then sends the decrypted decryption to a voice processing workstation of the cabin A;
step 2-13: the voice processing workstation of the cabin A sends SIP cross-cabin response information in the software call processing of the seat client of the cabin A;
step 2-14: the cabin B seat client side indicates the called user to ring, and after the user rings, ringing information is sent to the cabin B voice processing workstation;
step 2-15: the cabin B voice processing workstation executes the encryption forwarding steps from the step 2-9 to the step 2-12 to forward the ringing information of the called user to the cabin A seat client;
step 2-16: the cabin B seat client returns a response indicating that the connection is successful to the cabin B voice processing workstation;
Step 2-17: after the cabin B voice processing workstation divides the packets, forwarding the success indication to the cabin A client of the cabin A voice processing workstation through a cross-cabin encryption link;
step 2-18: the cabin A voice processing workstation forwards the success indication to the cabin A client;
step 2-19: after receiving the information, the cabin A client sends ACK information to the cabin A voice processing workstation for confirmation;
step 2-20: the cabin A voice processing workstation is used for transmitting the packetized data to the cabin B voice processing workstation through a cross-cabin encryption link;
step 2-21: the cabin B voice processing workstation forwards the ACK confirmation message to the cabin B client;
step 2-22: and the calling user and the called user establish communication connection and start talking. The client software of the seat compresses the voice data according to a contracted compression format;
step 2-23: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 2-24: each cabin voice processing workstation stores voice data according to RTP voice data;
step 2-25: and the voice processing workstations of each cabin carry out sub-packaging processing according to the cabin bit of the RTP extension header data receiving party and the client, and start cross-cabin encryption forwarding.
Step 2-26: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
step 2-27: after receiving the data, the client decodes the RTP audio data;
step 2-28: and the client sends the decoded data to the sound card for playing.
2. The method according to claim 1, wherein for text and file transfer communication methods, in addition to steps 2-1 to 2-20, the steps of:
step 2-29: after communication connection is established between the clients, cabin A client software sends relevant text information and file data to be subjected to lossless compression and then sends the compressed text information and file data to a cabin A voice processing workstation according to a UDT format so as to reduce bandwidth occupation;
step 2-30: the cabin A voice processing workstation stores text information and file data according to the UDT data;
step 2-31: and the cabin A voice processing workstation determines a client receiving the shelter according to the UDT extension header data to forward. The sub-packaging is executed under the condition of crossing the cabin, and the sub-packaging is sent to the cabin A guiding equipment for encryption transmission;
step 2-32: the cabin B voice processing workstation receives decryption data of the cabin B guiding equipment and then packs the decryption data;
Step 2-33: the cabin B voice processing workstation stores the packed data and forwards the data to a client of the cabin B;
step 2-34: after receiving the data, the client of the cabin B decompresses the UDT text information and the file data;
step 2-35: and displaying text information or storing files on the client side of the cabin B.
3. The method of claim 1, wherein the single pod point-to-point voice communication method further comprises:
step 1-1: logging in a voice processing workstation through voice system client software;
step 1-2: the voice processing workstation acquires information of all online clients of the cabin;
step 1-3: the seat 1 voice system client software initiates an Invite request to the voice processing workstation through a standard SIP protocol to initiate a call;
step 1-4: after the voice processing workstation confirms that the user authentication is passed, checking whether the Via header field in the request message contains the address of the user authentication; if yes, the loop back is indicated to occur, and a response indicating the error is returned; if not, the voice processing workstation inserts the self address in the Via header field of the request message and transmits an Invite request To the called seat 2 client indicated by the To field of the Invite message;
Step 1-5: the seat 2 client sends response information in call processing to the voice processing workstation;
step 1-6: the voice processing workstation sends response information in call processing to the seat 1 client;
step 1-7: the seat 2 client side indicates the called user to ring, and after the user rings, the ringing information is sent to the voice processing workstation;
step 1-8: the voice processing workstation forwards the ringing information of the called user to the seat 1 client;
step 1-9: the seat 2 client returns a response indicating that the connection is successful to the voice processing workstation;
step 1-10: the voice processing workstation forwards the success indication to the seat 1 client;
step 1-11: after receiving the information, the seat 1 client sends ACK information to the voice processing workstation for confirmation;
step 1-12: the voice processing workstation forwards the ACK confirmation message to the seat 2 client;
step 1-13: communication connection is established between the calling user and the called user, communication is started, and the client software of the seat compresses voice data according to a contracted compression format;
step 1-14: the seat client software sends the compressed and encoded data to a voice processing workstation according to an RTP format;
step 1-15: the voice processing workstation stores voice data according to the RTP voice data;
Step 1-16: and the voice processing workstation judges the receiver client according to the RTP extension header data and forwards the receiver client.
Step 1-17: after receiving the data, the client decodes the RTP audio data;
step 1-18: and the client sends the decoded data to the sound card for playing.
4. A method according to claim 3, characterized in that for text and file transfer communication methods, in addition to steps 1-1 to 1-12, the following steps are included:
step 1-19: the clients are connected in a communication way, and seat 1 client software sends relevant text information and file data to a voice processing workstation according to a UDT format after lossless compression;
step 1-20: the voice processing workstation stores text information and file data according to the UDT data;
step 1-16: the voice processing workstation determines a receiver client according to the UDT extension header data and forwards the receiver client;
step 1-17: after receiving the data, the client decompresses the UDT text information and the file data;
step 1-18: the client displays the text information or saves the file.
5. The method of claim 1, wherein the method of conference communication between clients across multiple bays comprises:
Step 3-1: the cabin A seat 1 establishes a conference room through voice system client software;
step 3-2: after receiving the conference room establishment message, the voice processing workstation of the cabin A uses a UDT protocol to multicast and share the conference room information to other cabins through a cross-cabin encryption link or an autonomous network link;
step 3-3: other client software enters a conference room by using a Session Initiation Protocol (SIP) handshake command with a conference number;
each cabin workstation uses UDT protocol to multicast and share conference room information to other cabins through a cross-cabin encryption link or an autonomous network link;
step 3-4: all client seat client software respectively carries out compression coding on local microphone voice data according to a contracted compression format;
step 3-5: the seat client software sends the compressed and encoded data to each cabin voice processing workstation according to the RTP format;
step 3-6: each cabin voice processing workstation stores the voice data according to the RTP voice data;
step 3-7: each cabin voice processing workstation determines a conference shelter according to the RTP extension header data, then carries out subpackaging processing, and starts cross-cabin encryption forwarding;
step 3-8: each cabin voice processing workstation receives RTP data after cross-cabin decryption, packages the RTP data, stores the voice data and sends the voice data to a receiving client of the cabin;
Step 3-9: after receiving the data, the client decodes the RTP audio data;
step 3-10: and the client sends the decoded data to the earphone for playing.
6. A method according to claim 3, wherein the single-cabin client and multi-station communication method comprises:
step 4-1: logging in a radio station through a system management interface;
step 4-2: the voice processing workstation acquires the information of the online radio station through the heartbeat information;
step 4-3: the seat voice system client software is connected with a digital radio station or a voice processing terminal through an SIP protocol with a radio station type number, and initiates an Invite request to start calling;
step 4-4: the digital radio station or the voice processing terminal returns a response indicating successful connection to the seat 1 client;
step 4-5: the digital radio station or the voice processing terminal returns a response indicating successful connection to the response information of the seat 1 client;
step 4-6: after receiving the information, the seat client sends ACK information to the digital radio station or the voice processing terminal for confirmation;
step 4-7: the communication connection is established between the client and the digital radio or the voice processing terminal, and the communication is started; step 4-8: the digital radio station or the voice processing terminal compresses the received voice according to a contracted compression format and sends the compressed voice to a seat 1 client;
Step 4-9: the seat client side continuously receives the voice of the digital radio station or the voice processing terminal through configuration;
step 4-10: the seat client sends RTP heartbeat information to keep connection when the PTT is lifted;
step 4-11: the seat client transmits an extended RTP data packet with ED137 to a digital radio station or a voice processing terminal when the PTT is pressed down;
step 4-12: the digital radio station or the voice processing terminal receives the voice and converts the voice into analog audio to be sent by the radio station;
step 4-13: the voice processing workstation receives and stores the voice realization sent by the client and the digital radio or the voice processing terminal.
7. A voice communication system based on a ground station networking, for implementing any of the methods of claims 1-6, comprising: the system comprises a voice processing workstation, a voice system client, a voice processing terminal based on an analog voice link, a cross-cabin communication link, a telephone gateway, a digital radio station and a network switch; the voice processing workstation is connected with other equipment through switch network equipment, and the internal software of the voice processing workstation comprises an internal call communication module, a conference communication module, a digital radio communication module, an analog radio communication module, a program-controlled telephone communication module, a voice recognition module, a voice synthesis module, a cross-cabin communication module and an autonomous network link module;
The internal call communication module refers to point-to-point communication of voice system client software, the client software and the client software operate through an interface and execute a Session Initiation Protocol (SIP) protocol to dial, when two clients are connected, voice communication is started through an RTP protocol, an RTP voice data stream passes through a voice processing workstation, and the voice processing workstation completes voice recording, management of a communication path protocol, link load management and volume adjustment;
the conference communication module is used for carrying out multiparty voice communication in a conference room by a plurality of clients, and the voice processing workstation is used for carrying out voice mixing processing on voice data of a conference according to set requirements besides completing management of a communication path, link load control and voice recording;
the digital radio station communication module is used as a recorder for voice recording of the voice system client software and the digital radio station;
the analog radio station communication module, the voice processing workstation is used as a recorder for voice recording of the voice system client software and the digital radio station;
the program-controlled telephone communication module is used for communicating the workstation with the program-controlled telephone and the telecommunication exchanger, sending out a connection instruction and voice data through the control of the SIP protocol by the client software, completing the dialing of the program-controlled telephone and the external telephone by the voice processing workstation, transmitting voice to the telephone gateway through the RTP protocol, and being used for the management, the link load control and the voice recording of the dialing of the telephone and the communication path by the client software;
The cross-cabin communication module is used for performing cross-cabin protocol coding on the VoIP communication data by the voice processing workstation, converting the communication data into a communication protocol of a cross-cabin encryption transmission terminal, and analyzing the communication protocol by another voice processing workstation after the communication protocol is transmitted by a link, wherein the communication protocol is used for VoIP communication based on a cross-cabin link;
the voice processing workstation carries out autonomous network Ad Hoc link protocol coding on VoIP communication data, converts the VoIP communication data into an Ad Hoc link communication protocol, connects a remote station group with a current station group through an autonomous network and is used for voice, conference, file and text communication among autonomous network nodes;
the voice recognition module is provided with a plurality of voice recognition services in the voice processing workstation and is used for commanding the recognition to simplify operation and free grammar and converting stored audio into text;
and the voice synthesis module is used for synthesizing voice in the voice processing workstation, the external equipment sends the alarm information to the voice processing workstation through an interface protocol, and the voice processing workstation converts the information into audio and sends the audio to the client in the workstation to realize voice alarm.
8. The system of claim 7, wherein the cross-cabin communication links include fiber optic encryption links, inter-drone, space-to-ground autonomous network encryption links, and UV encryption links.
9. The system of claim 7, wherein the voice system client software in the voice system further comprises an interface display module, an address book module, a point-to-point voice call module, a conference call module, a digital station call module, an analog station call module, a file transfer module, wherein
The interface display module provides a UI interaction interface;
the address book module is used for immediately synchronizing the client information, entering an organization address book, checking the online/offline information of the client, rapidly positioning the client to be searched, and checking the detailed information of the client;
the point-to-point voice call module is characterized in that client software is connected with other clients in the system through an SIP protocol, and after the other clients are connected, the client software realizes the mutual voice call of the clients through links formed by a plurality of workstations and cross-cabin connection guiding equipment through an RTP protocol;
the conference call module is used for supporting the mutual call among a plurality of client sections after entering a preset conference room through client software, and sending characters and shared files;
the voice system client software expands the VoIP protocol into ED137 protocol, and the voice system client software transmits SIP, RTP, ED137 expansion frames to be directly connected with the digital radio for realizing PTT function and voice transmission, and simultaneously transmits voice data to the voice processing workstation to complete voice recording, wherein the client software is used for voice receiving and PTT transmitting of a single radio and also receives voice mixing signals of multiple radio;
The simulation radio station communication module is characterized in that voice system client software expands a VoIP protocol into an ED137 protocol, the voice system client software sends SIP, RTP, ED expansion frames to a voice processing hardware terminal, the voice processing hardware terminal can automatically complete connection with the client software, and a voice processing workstation starts voice recording after connection;
the file transmission module is used for transmitting common files, video files and audio files.
10. The system of claim 9, wherein the analog voice link based voice processing is provided with a standard ED137 protocol, and is internally composed of a data processing unit and an analog audio impedance matching unit.
The multi-channel analog voice equipment can be used for converting analog voice into ED137 RTP audio through a data processing unit after impedance matching or converting ED137 RTP audio input by the system into analog voice to be sent to external equipment, so that the traditional radio station and voice equipment can be connected into the system related to the large-scale ground station networking voice communication method, wherein
The data processing unit is used for digitizing analog audio, converting the input analog audio into a protocol supported by VoIP and completing digital audio coding;
The impedance matching is adapted to analog audio input and output of various relay stations, ground stations and voice terminals;
the cross-cabin communication link is based on an optical fiber, an encrypted cross-cabin circuit of an Ethernet and an autonomous network link based on Ad Hoc, and a remote station group and a current station group are connected to form a voice communication network;
the telephone network is connected with the traditional telecommunication line and the voice system and is used for communicating the seat client with the program-controlled telephone and the external telephone, and the heterogeneous network based on the analog signaling interaction model and the communication transmission medium is comprehensively accessed into the unified system.
CN202311612593.2A 2023-11-29 2023-11-29 Voice communication method and system based on ground station networking Pending CN117692560A (en)

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