WO2021088345A1

WO2021088345A1 - Ed137 protocol-based voip station gateway having master-backup architecture

Info

Publication number: WO2021088345A1
Application number: PCT/CN2020/090123
Authority: WO
Inventors: 张朋; 王虎; 孙英晖; 杨康; 朱昆
Original assignee: 南京莱斯电子设备有限公司
Priority date: 2019-11-07
Filing date: 2020-05-14
Publication date: 2021-05-14
Also published as: CN110809097B; CN110809097A

Abstract

The present invention provides an ED137 protocol-based VoIP station gateway having a master-backup architecture. The VoIP station gateway device uses a VoIP design method to design a VoIP station access interface, a VCC access scheme, and a master-backup redundancy function strictly according to the ED137 protocol. Various additional functions, such as working modes, gains, and delay are added. Specific functions such as CLIMAX, BSS, and dynamic delay compensation are realized, greatly embodying the advantages of a VoIP station. The VoIP station gateway can access a VoIP station receiver, a VoIP station transmitter, a VoIP station transceiver, a VoIP station remote control device, etc. The VoIP station gateway is provided with a friendly front-end management terminal, and all functions thereof can be managed and configured by means of web access. The present invention satisfies not only the requirements of an airline management voice system for a VoIP station, but also the requirements of airline management safety redundancy.

Description

A VoIP radio gateway with active and standby architecture based on ED137 protocol

Technical field

The invention belongs to the technical field of voice communication engineering, and particularly relates to a VoIP radio gateway with a master-standby architecture based on the ED137 protocol.

Background technique

The European Civil Aviation Equipment Organization (EUROCAE) is an international organization that specializes in the formulation of technical specifications for civil avionics equipment, and is subordinate to the European Aviation Safety Organization. It is a non-profit organization composed of aviation stakeholders in Europe and other regions. The EUROCAEWG67 working group has successively released versions of the ED137 protocol, such as February (2009), ED-137A (2010), ED-137B (2012) and ED-137C (2017). This agreement is the standard and interface specification for international aviation service providers to access VoIP voice (especially VoIP radio).

In recent years, in the international airport voice switching and control system (VCCS) bidding technical requirements indicators, it is clearly required that the VCCS system supports the ED137 protocol and can access VoIP radio stations. However, only well-known international suppliers such as Frequentis and RS can realize access in the world. Several domestic voice system companies are also in their infancy in this regard. Therefore, the realization of the ED137 protocol access of the voice system and the perfect fit of the main and standby architecture of the voice system is imminent.

Summary of the invention

Objective of the invention: The present invention belongs to the technical field of air traffic control voice communication engineering. The main problem to be solved is that the radio access mode of the traditional air traffic control voice exchange system is the EM analog access mode, which cannot meet the urgent needs of today's VoIP radio communication.

In order to solve the above technical problems, the present invention discloses a VoIP (Voice over Internet Protocol) radio station based on the ED137 protocol (Interoperability Standards for VoIP ATM Radio Components). The gateway adds multiple functions such as working mode, gain, and delay. It adopts a background service design method and is equipped with a web management terminal. The access to VCCS conforms to the standard SIP protocol, which can realize the connection of VoIP radio stations to the VCCS system of the main and standby architecture. It can not only meet the needs of the air traffic management voice system for VoIP radio stations, but also meet the requirements of the main and backup VCCS interfaces. The VoIP radio gateway of the present invention includes a core switching module, an active and standby server selection module, a media signaling interaction module, and a system operation and maintenance module;

The core switching module is used to implement voice switching of the VoIP radio gateway;

The active/standby server selection module is used to automatically switch to the standby server when the active/standby server selection module detects media switching timeout after the main server hangs up;

The media signaling interaction module is used for state machine management of SIP signaling (Session Initiation Protocol, Session Initiation Protocol) and media transceiving of voice channels;

The system operation and maintenance module is used to provide web terminal management and network status detection functions.

The core switching module is used to realize the voice exchange of the VoIP radio gateway, and realize the radio conference, receiving and sending delay buffer, receiving and sending gain adjustment and dynamic delay compensation through the VoIP voice switching technology;

Wherein, the core switching module uses RTP (Real Time Protocol) protocol to exchange voice and radio control commands when implementing radio conferences, and the radio control commands are in the extended RTP header;

The range of the receiving and sending delay buffer is 0～600ms;

The range of the transceiver gain adjustment is -25db～+25db;

The dynamic delay compensation is based on measuring the network delay between the VoIP station and the VoIP station gateway, calculating the maximum delay compensation time of the CLIMAX (radio climax) group, and controlling the VoIP station delay time to achieve synchronization of the VoIP stations in the group Transmission, dynamic delay compensation range is 0～254ms, accuracy is 2ms.

The core switching module is used to implement the voice exchange of the VoIP radio gateway, and also includes the media information used to exchange the VoIP radio session and the VCCS (Voice Communication Control System) session;

The VoIP radio session is the prerequisite of the VCCS session. After the VoIP radio session is established, the R2S (a transmission format) keep-alive packet or the voice packet carrying the payload (voice payload) is transmitted;

When the VoIP radio session is established, negotiate R2S-KeepAlivePeriod (R2S keep-alive interval) and R2S-KeepAliveMultiplier (R2S keep-alive packet number). The product of the two is the timeout period. R2S or voice packets are lost between the VoIP radio and the VoIP radio gateway If the timeout period is exceeded, the device will be disassembled;

R2S keep-alive packets generally have longer intervals and fewer bytes than voice packets. When the VoIP radio is not working, the R2S keep-alive packet is transmitted between the VoIP radio and the VoIP radio gateway;

When the VoIP radio receiver receives the carrier, it will deliver the voice packet carrying the Payload to the VoIP radio gateway;

When the VCCS system needs to transmit the radio, it will deliver the voice packet carrying the payload to the VoIP radio transmitter. Only after the VoIP radio session is established, the VoIP radio gateway can establish a session with the VCCS;

The VoIP radio gateway sends the registration information of the connected VoIP radio station to the VCCS. After receiving the registration information, the VCCS initiates a session establishment request to the VoIP radio gateway to establish a corresponding type of VCCS session; after the VCCS session is established, standard RTP is used to transmit voice. When a VoIP radio station is used, it transmits mute, and the Mark bit (marker bit) of RTP represents PTT (Push To Talk) signal and carrier signal; when the VoIP radio station fails or is disconnected, the VoIP radio gateway will send a message to VCCS to end the VoIP radio session. request.

The media signaling interaction module uses the SIP state machine mechanism to implement signaling state machine management and channel media transceiving, which specifically includes the following steps:

Step a1, the SIP state machine independently manages the state of each VoIP station, and the initial state is the idle state;

Step a2: After the VoIP radio gateway is connected to the VoIP radio, it sends a session establishment request and enters the request state;

Step a3, the VoIP station responds to the request. If it accepts the session invitation, the SIP state machine enters the session state; if it rejects the session invitation, the SIP state machine enters the idle state again and waits for the next connection;

Step a4: After the SIP state machine enters the conversation state, it opens the VoIP radio channel for media transmission and reception.

The system operation and maintenance module is used to provide web terminal management and network status detection functions. The web terminal management includes radio configuration, gateway configuration and additional function configuration; through network status detection, the VoIP radio gateway can report the dual network status to the server in real time information.

The system operation and maintenance module is equipped with a web (web page) front-end configuration page, and the radio configuration, gateway configuration and additional function configuration are completed through the web front-end configuration page;

The radio configuration includes radio name, radio IP and port, radio type, PTT mode, radio SIP account password (for example: configured as RadioRx, 192.168.1.100, 5060, Rx, Normal, 500, 123456);

Gateway configuration includes server IP port information and gateway SIP account password (for example: configured as 192.168.1.6, 5060, 9000, 123456);

Additional function configuration includes transceiver gain, transceiver delay, transceiver mode, mode suppression and CLIMAX group (for example: configured to receive 10db, send -5db, receive 40ms, send 20ms, send and receive suppression mode, group 1);

A pair of transceiver VoIP stations can configure the transceiver mode as transceiver isolation, transceiver suppression, and transceiver tone reduction;

Transceiver isolation means that the transceiver is in full-duplex mode; transceiver suppression means that the transceiver is in half-duplex mode; transceiver attenuation means that when the VoIP radio transmitter is transmitting, the VoIP radio receiver will suppress attenuation according to the mode suppression amount;

The VoIP stations that form the same CLIMAX group can transmit simultaneously.

The signaling state machine of the media signaling interaction module is responsible for the SIP signaling interaction with the VCCS server, VoIP radio, RRCE (Remote Radio Control Equipment) and the web equipped with the system operation and maintenance module; network media transceiver implementation Media transceiving of VoIP radio and RRCE; switching between active and standby media and network media transceiving to achieve VCCS media interaction; dynamic compensation measurement of VoIP radio is fed back to the CLIMAX group, and the CLIMAX group controls the network delay of the same group of VoIP radios; media of VoIP radio After gain adjustment and buffer control, it exchanges with VCCS voice; the dual network status is detected and reported to the server separately.

The process of establishing SIP and RTP for the VoIP radio gateway session includes the following steps:

Step b1: After the station name, IP, port, SIP account, and SIP password are configured, the VoIP station gateway will periodically initiate a session request to the VoIP station. If a session has been established with the VoIP station, no more requests will be sent;

Step b2: After the VoIP station receives the request, if the configuration is correct, it will automatically answer the request, reply 100Trying (receive response message) and 200OK (normal response message), and the VoIP station gateway will reply ACK (response confirmation message) to confirm the establishment of the session;

Step b3, the VoIP station gateway and the VoIP station start to send and receive RTP packets in both directions. When the VoIP station is not sending and receiving, the two-way transmission is a keep-alive packet without load; when the VoIP station is transmitting, the VoIP station gateway sends it to The VoIP radio station is the RTP voice packet carrying the payload; when the VoIP radio station receives it, the VoIP radio station sends the RTP voice packet carrying the payload to the VoIP radio gateway;

Step b4: After the VoIP radio station establishes the session, it registers the VoIP radio station SIP account with the server; when the first registration is successful, the server initiates a session request to the VoIP radio gateway; after the session is established, the server and the VoIP radio gateway send RTP voice packets , Use Mark bit to represent PTT signal and carrier signal;

Step b5: When the VoIP radio fails or the network is disconnected, the VoIP radio gateway and the VoIP radio session are interrupted, and the radio session with the VCCS is ended.

The VoIP radio gateway provided by the present invention is a VoIP radio gateway equipment used in an air traffic control voice communication system (VCCS) and conforms to the ED137 protocol. It can support access to the VCCS of the main and standby redundant server architecture. The main and standby server selection module provides automatic detection of server media for the VoIP radio gateway, and switches between the main and standby server media after timeout. The 60ms detection and switching interval enables the VoIP radio gateway to provide continuous voice and signal to the VCCS. VoIP radio stations are divided into single VoIP radio receiver Rx, single VoIP radio transmitter Tx, and VoIP radio transceiver RTx according to the type of radio. Different types of VoIP radio stations can be connected to the VoIP radio gateway. The VoIP radio gateway supports a maximum of 32 VoIP radio channels, including 16 receiving and 16 sending channels. Its core exchange includes radio voice exchange and server voice exchange. VCCS establishes a session through standard SIP, and uses the Mark bit of the RTP header to represent PTT signal and carrier signal transmission. The VoIP radio station is connected to the VoIP radio gateway through the ED137, and transmits VoIP radio control signaling and voice through the RTP packet with an extension header. The VoIP radio gateway can be transplanted across platforms and supports windows, Linux, cross-compiled arm-linux, etc. The VoIP radio gateway is equipped with a parameter configuration management service, and the VoIP radio gateway server is accessed through the web page for radio parameter configuration, channel configuration, gateway device configuration, server parameter configuration, etc. The VoIP radio gateway supports multiple network card devices, and the equipment can be used for network port aggregation. The VoIP radio gateway can isolate the VoIP radio network from the VCCS network and provide security guarantee for VCCS network communication. The VoIP radio gateway provides a variety of channel expansion functions for VCCS, including channel transceiver gain adjustment, channel transceiver delay, three channel transceiver modes, etc.

The active and standby server selection module is an important functional module of a VoIP radio gateway with an active and standby architecture based on the ED137 protocol. The VoIP radio gateway can be connected to a single-server VCCS or a redundant VCCS with active and standby servers. The VoIP radio gateway receives SIP messages from any server, and the media is to establish two sessions at the same time. When the main service media is not received for more than 60ms, the VoIP radio gateway performs media switching to ensure the redundancy effect of the main and standby servers. This feature can realize that when any server fails, the system can use the VoIP radio station uninterruptedly, and even the human ears can't hear the difference. The VoIP station gateway can also establish a session with Radio Remote Control Equipment (RRCE) to complete the VoIP station master and backup functions. The ED137 protocol stipulates the radio master and backup functions of RRCE. It can be applied to two modes of RRCE: single frequency mode and dual frequency mode. It can control two pairs of main and standby VoIP radio stations. It can also transmit the signal quality information of the VoIP radio receiver.

The media signaling interaction module is a basic functional module of a VoIP radio gateway with a master-standby architecture based on the ED137 protocol. SIP signaling is the basic protocol of the VoIP voice system. The session is established by the initiator of the session sending an INVITE request, the invited party replies with a 100Tying response and a 200OK response, and then the initiator replies with an ACK response, the session is established, and both parties begin to send keep-alive packets or media. When one party needs to end the session, it sends a BYE end session request, the other party replies with a 200 OK response, and both parties clean up resources. Because the SIP protocol is an application layer protocol, it belongs to the UDP protocol at the transport layer. The UDP protocol is an unreliable transmission protocol. Messages may be lost during transmission. Therefore, the SIP protocol has a timeout retransmission mechanism to ensure communication reliability. The ED137 protocol specifies CLIMAX delay and dynamic delay compensation functions. The VoIP radio transmitters of the same CLIMAX group will periodically receive the delayed detection request message (RMM) from the VoIP radio gateway, and reply to the measurement response message (MAM). The VoIP radio gateway calculates the network delay from VCCS to the VoIP radio station and the transmission delay of the VoIP radio station through RMM and MAM. The VoIP radio transmitter of the same group adjusts the transmission delay. The CLIMAX function can be realized (the same group of VoIP radio transmitters simultaneously transmit sound into the air).

Beneficial effects: The present invention discloses a VoIP radio gateway with a master-standby architecture based on the ED137 protocol. This gateway can not only meet the access requirements of VoIP radio stations (or RRCE) that comply with the ED137 protocol, but also can access the main-standby architecture air traffic control VCCS In the system, and on this basis, it also provides functions such as transceiver gain adjustment, transceiver delay buffer, dynamic delay compensation, radio transceiver mode, and CLIMAX group. This gateway can meet the control requirements of various civil aviation airports at home and abroad for VoIP radio. The additional function also greatly improves the usability of the VoIP radio gateway.

Description of the drawings

In the following, the present invention will be further described in detail with reference to the accompanying drawings and specific embodiments, and the above-mentioned and/or other advantages of the present invention will become clearer.

Fig. 1 is a block diagram of the composition of a VoIP radio gateway based on the active and standby architecture of the ED137 protocol of the present invention.

Figure 2 is a diagram of the logical structure of a VoIP radio gateway with a master-standby architecture based on the ED137 protocol of the present invention.

Fig. 3 is a flow chart of SIP/RTP establishment of a VoIP radio gateway session establishment based on the ED137 protocol in the active and standby architecture of the present invention.

Figure 4 is a CLIMAX coding diagram of a VoIP radio gateway with a master-standby architecture based on the ED137 protocol of the present invention.

Figure 5 is a dynamic delay detection coding diagram of a VoIP radio gateway with a master-standby architecture based on the ED137 protocol of the present invention.

Fig. 6 is a schematic diagram of the delay of a VoIP radio gateway with a master-standby architecture based on the ED137 protocol of the present invention.

Fig. 7 is a remote control coding diagram of a VoIP radio gateway radio remote control based on the ED137 protocol of the present invention.

Figure 8 is a connection block diagram of the voice communication system.

Detailed ways

The VoIP radio gateway with active/standby architecture based on ED137 protocol provided by the present invention is designed for air traffic control voice system with active/standby redundant architecture, provides access to VoIP radio stations complying with the ED137 protocol for the voice system, and supports the dynamic delay of the ED137 protocol Compensation and access function of radio remote control equipment. In addition, the VoIP radio gateway provides a wealth of additional voice functions, which can provide VoIP radio gateway radio gain, radio delay, radio transceiver mode, and CLIMAX group to achieve dynamic compensation.

1. The block diagram of the VoIP radio gateway is shown in Figure 1. The VoIP radio gateway consists of a core switching module, an active and standby server selection module, a media signaling interaction module, and a system operation and maintenance module.

The core switching module is the voice switching module of the VoIP radio gateway. Including radio conference, receiving and dispatching delay buffer, receiving and dispatching gain adjustment, dynamic delay compensation. Radio conferences use RTP protocol for interactive voice and radio control commands. The radio control commands are in the extended RTP header, such as PTT information and carrier information. Transceiver delay buffer and transceiver gain adjustment are additional functions of the VoIP radio gateway. It can realize the receiving gain and transmitting gain of each VoIP station, so that it can realize the equalization of voice input according to the physical characteristics of the unreachable VoIP station. The range of gain adjustment is (-25db～+25db). The transceiver delay buffer is based on the dynamic delay, and the transceiver delay function can also be added. The range of the receiving and sending delay buffer is (0～600ms). Dynamic delay compensation is based on measuring the network delay between the VoIP station and the VoIP station gateway, calculating the maximum delay compensation time of the CLIMAX group, and controlling the VoIP station delay time to achieve simultaneous transmission of the VoIP stations in the group. The dynamic compensation range is (0～254ms, accuracy 2ms)

The active/standby server selection module is a functional module that connects the VoIP radio gateway to the air traffic control voice active/standby server. After the main server hangs up, the VoIP radio gateway detects the media switching timeout (60ms) and can automatically switch to the standby server, which can ensure continuous media and uninterrupted voice. Realize the server main and standby redundancy function. The primary and secondary signaling is a dual-transmission mode, and the server master replies.

The media signaling interaction module includes signaling state machine management and channel media transceiving. SIP uses the SIP state machine mechanism to manage the session establishment. Media sending and receiving can realize large-scale media sending and receiving.

The system operation and maintenance module includes web terminal management and network status detection. Radio configuration, gateway configuration, and additional function configuration can all be logged in and configured through the web page. The VoIP radio gateway can report the dual network status information to the server in real time.

2. The logical structure of the VoIP radio gateway is shown in Figure 2. The signaling state machine is responsible for SIP signaling interaction with VCCS server, VoIP radio, RRCE and web. Network media transceiving realizes media transceiving of VoIP radio and RRCE. Active and standby media switching and network media receiving and sending realize VCCS media interaction. After the dynamic compensation measurement of the VoIP radio station, it is fed back to the CLIMAX group, and the CLIMAX group controls the network delay of the VoIP radio station in the same group. The media of the VoIP radio station is exchanged with VCCS voice after gain adjustment and buffer control. The dual network status is detected and reported to the server separately.

3. The SIP/RTP flow chart of VoIP radio gateway session establishment is shown in Figure 3. After the radio station name, IP, port, SIP account, SIP password and other attributes are configured, the VoIP radio gateway will initiate a session request to the VoIP radio station periodically (10s). If a session has been established with the VoIP station, no more requests will be sent. After the VoIP radio station receives the request, if the configuration is correct, it will automatically answer the request and reply 100Trying and 200OK. The VoIP radio gateway replies with an ACK to confirm the establishment of the session. After that, the VoIP radio gateway and the VoIP radio start to send and receive RTP packets in both directions. When the VoIP station is not transmitting and receiving, the two-way transmission is the keep-alive packet without payload; when the VoIP station is transmitting, the VoIP station gateway sends the VoIP station the RTP voice packet carrying the payload; when the VoIP station receives, VoIP What the radio station sends to the VoIP radio gateway is the RTP voice packet carrying the payload. After the VoIP radio station has established a session, it registers the VoIP radio station SIP account with the server. When the first registration is successful, the server initiates a session request to the VoIP radio gateway. After the conversation is established, the RTP voice packet is sent between the server and the VoIP radio gateway. Use Mark bit to represent PTT signal and carrier signal. When the VoIP radio station fails or the network is disconnected, the conversation between the VoIP radio gateway and the VoIP radio station is interrupted, and the radio conversation with the VCCS is ended.

4. The CLIMAX time delay (CLD) coding diagram of the VoIP radio gateway is shown in Figure 4. Whether it is a keep-alive packet or an RTP voice packet between the VoIP station gateway and the VoIP station, it is an RTP message, which includes the RTP header and payload, and the payload is optional. The RTP header includes a normal RTP header and an extended RTP header. As shown in Figure 4, the extended RTP header format, the meaning of each field is as follows:

PTTtype: Used for VoIP radio transmitters or VoIP radio transceivers. It defines the type of PTT. PTT types include ordinary PTT, priority PTT, coupling PTT, and emergency PTT. Can be set on the VoIP radio gateway web.

SQU: Used with VoIP radio receiver or VoIP radio transceiver to define the carrier status information.

PTT-ID: PTT ID used for VoIP radio transmitter or VoIP radio transceiver, allocated to users accessing VoIP radio. Used to let the user know which user the VoIP station is currently transmitting.

PM: PTT Mute, used for PTT mute for multi-user access to VoIP radio.

PTTS: PTT Summation, used for simultaneous transmission of the same priority.

SCT: Simultaneous Call Transmissions, used for multiple users to transmit the station field at the same time.

Reserved: reserved field.

X: Indicates whether the extended information field is used or not.

TYPE: Extended field type.

LENGTH: Extended field length.

VALUE: Extended field content.

CLD: CLIMAX delay, bit24 is 0 is relative delay, 1 is absolute delay. Bit25～bit31 represent the delay time, the unit is 2ms, and the range is 0～127 (0ms～254ms).

5. The dynamic delay detection coding diagram of the VoIP radio gateway is shown in Figure 5. The keep-alive packet or RTP voice packet sent by the VoIP station gateway to the VoIP station at regular intervals (4s) carries a dynamic delay measurement message (RMM). The keep-alive packet or RTP voice packet replies from the VoIP station will carry the dynamic delay. Time measurement response message (MAM), the meaning of each field is as follows:

TQV: Represents whether the time measured by the VoIP radio gateway is a relative time or an absolute time.

T1: It is the time sent by the VoIP radio gateway RMM, the unit is 125us.

TQG: Represents whether the time measured by the VoIP radio gateway is a relative time or an absolute time.

NMR: Represents whether the VoIP station needs a new measurement, such as jitter buffer failure.

T2: It is the time when the VoIP station receives RMM, in 125us.

Tsd: It is the time interval between the VoIP station sending out MAM and receiving RMM, the unit is 125us.

Tj1: is the delay of the jitter buffer, the unit is 125us.

Tid: It is the internal delay of the VoIP radio, the delay from the jitter buffer to the antenna, the unit is 125us.

6. The schematic diagram of VoIP radio gateway delay is shown in Figure 6. The CLIMAX group synchronization transmission function requires that the VoIP radio transmission of the same CLIMAX group is synchronized transmission, that is, the time of transmission to the antenna is synchronized. The VCCS system delays of all VoIP stations are equal, that is, Tv1 is equal. As long as the TdTx of all VoIP stations are equal. The VoIP radio gateway packing delay and jitter buffer delay of all VoIP radio stations are equal, namely Tp1.

The network delay is:

T3 is the time when the VoIP station sends out the MAM, and T4 is the time when the VoIP station gateway receives the MAM.

The internal delay of the VoIP station Tid=Td1+Ts1.

Therefore, the delay from the VoIP radio gateway to the antenna is: TdTx-Tp1=Tn1+Tj1+Tid.

Therefore, the delay from the VoIP radio gateway to the antenna can be measured through RMM and MAM. Then CLD dynamically controls the transmission delay of the VoIP radio station to achieve transmission synchronization.

7. VoIP radio gateway can access RRCE, and RRCE can connect to VoIP radio to realize the main and standby functions of VoIP radio. The VoIP radio remote control coding diagram is shown in Figure 7. The meaning of each field is as follows:

BSS-qidx: compare and select signal quality.

BSS-qidx-ml: Comparison and selection method.

MSTxF1: RRCE F1 frequency point active and standby VoIP radio transmitter enable flag.

MSRxF1: RRCE F1 frequency point active and standby VoIP radio receiver enable flag.

MSTxF2: RRCE F2 frequency point master and backup VoIP radio transmitter enable flag.

MSRxF2: RRCE F2 frequency point master and backup VoIP radio receiver enable flag.

SelTxF1: RRCE F1 frequency point VoIP radio transmitter PTT enable flag.

SelTxF2: RRCE F2 frequency point VoIP radio transmitter PTT enable flag.

MuRxF1: RRCE F1 frequency point VoIP radio receiver mute enable flag.

MuRxF2: RRCE F2 frequency point VoIP radio receiver mute enable flag.

SQF1: F1 frequency point carrier enable flag of RRCE.

SQF2: F2 frequency point carrier enable flag of RRCE.

reserved: reserved.

F1SQI: The received signal quality at F1 frequency of RRCE.

F2SQI: The received signal quality at F2 frequency of RRCE.

Example

According to specific application scenarios, the specific voice communication system is shown in Figure 8. This set of voice communication system includes server (Server), monitoring (RCMS), switch (Switch), seat (CWP), line interface unit (Line Interface Unit), VoIP telephone gateway, VoIP radio gateway, VoIP phone and VoIP radio station. The server and network switch are dual-redundant designs, and other equipment and server interfaces are connected through dual networks. The VoIP radio gateway/VoIP phone gateway separates the VoIP radio/VoIP phone from the voice system network. The VoIP radio gateway is used to realize the VoIP radio access to this voice communication system. This system and VoIP radio gateway have all been implemented and put into use.

The present invention provides a VoIP radio gateway with a master-standby architecture based on the ED137 protocol. There are many specific methods and ways to implement this technical solution. The above are only the preferred embodiments of the present invention. It should be pointed out that for common technologies in this technical field As far as personnel are concerned, without departing from the principle of the present invention, several improvements and modifications can be made, and these improvements and modifications should also be regarded as the protection scope of the present invention. All the components that are not clear in this embodiment can be implemented using existing technology.

Claims

A VoIP radio gateway with a master-standby architecture based on the ED137 protocol, which is characterized by including a core switching module, a master-standby server selection module, a media signaling interaction module, and a system operation and maintenance module;

The core switching module is used to implement voice switching of the VoIP radio gateway;

The active/standby server selection module is used to automatically switch to the standby server when the active/standby server selection module detects media switching timeout after the main server hangs up;

The media signaling interaction module is used for state machine management of SIP signaling and media transceiving of voice channels;

The system operation and maintenance module is used to provide web terminal management and network status detection functions.
A VoIP radio gateway with a master-standby architecture based on the ED137 protocol according to claim 1, wherein the core switching module is used to implement the voice exchange of the VoIP radio gateway, and realize the radio conference, receiving and sending through the VoIP voice exchange technology. Delay buffer, transceiver gain adjustment and dynamic delay compensation;

Wherein, the core switching module uses the RTP protocol for interactive voice and radio control commands when realizing the radio conference, and the radio control commands are in the extended RTP header;

The range of the receiving and sending delay buffer is 0～600ms;

The range of the transceiver gain adjustment is -25db～+25db;

The dynamic delay compensation is based on measuring the network delay between the VoIP station and the VoIP station gateway, calculating the maximum delay compensation time of the CLIMAX group, and controlling the delay time of the VoIP station to achieve the synchronous transmission of the VoIP stations in the group, and the dynamic delay The range of time compensation is 0～254ms, and the accuracy is 2ms.
A VoIP radio gateway with a master-standby architecture based on the ED137 protocol according to claim 2, wherein the core switching module is used to implement the voice exchange of the VoIP radio gateway, and further comprises: for exchanging VoIP radio sessions and VCCS Media information of the session;

The radio session is the prerequisite of the VCCS session. After the radio session is established, the R2S keep-alive packet or the voice packet carrying the payload is transmitted;

When the radio session is established, negotiate R2S-KeepAlivePeriod and R2S-KeepAliveMultiplier. The product of the two is the timeout period. If R2S is lost between the VoIP radio station and the VoIP radio gateway or the voice packet exceeds the timeout period, the phone will be disassembled;

When the VoIP radio is not working, the R2S keep-alive packet is transmitted between the VoIP radio and the VoIP radio gateway;

When the VoIP radio receiver receives the carrier, it will deliver the voice packet carrying the Payload to the VoIP radio gateway;

When the VCCS system needs to transmit the radio, it will deliver the voice packet carrying the payload to the VoIP radio transmitter. Only after the radio session is established, the VoIP radio gateway can establish a session with VCCS;

The VoIP radio gateway sends the registration information of the connected VoIP radio station to the VCCS. After receiving the registration information, the VCCS initiates a session establishment request to the VoIP radio gateway to establish a corresponding type of VCCS session; after the VCCS session is established, standard RTP is used to transmit voice. Use VoIP radio to transmit mute, and use the Mark bit of RTP to represent PTT signal and carrier signal; when the VoIP radio fails or is disconnected, the VoIP radio gateway will send a request to VCCS to end the radio session.
The VoIP radio gateway with a master-standby architecture based on the ED137 protocol according to claim 3, wherein the media signaling interaction module uses the SIP state machine mechanism to implement signaling state machine management and channel media transceiving, specifically Including the following steps:

Step a1, the SIP state machine independently manages the state of each VoIP station, and the initial state is the idle state;

Step a2: After the VoIP radio gateway is connected to the VoIP radio, it sends a session establishment request and enters the request state;

Step a3, the VoIP station responds to the request. If it accepts the session invitation, the SIP state machine enters the session state; if it rejects the session invitation, the SIP state machine enters the idle state again and waits for the next connection;

Step a4: After the SIP state machine enters the conversation state, it opens the VoIP radio channel for media transmission and reception.
A VoIP radio gateway with a master-standby architecture based on the ED137 protocol according to claim 4, wherein the system operation and maintenance module is used to provide web terminal management and network status detection functions, wherein the web terminal management includes radio configuration , Gateway configuration and additional function configuration; through network status detection, the VoIP radio gateway can report dual network status information to the server in real time.
A VoIP radio gateway with a master-standby architecture based on the ED137 protocol according to claim 5, wherein the system operation and maintenance module is equipped with a web front-end configuration page, and radio configuration, gateway configuration and additional functions are completed through the web front-end configuration page Configuration

The radio configuration includes radio name, radio IP and port, radio type, PTT mode, radio SIP account password;

Gateway configuration includes server IP port information and gateway SIP account password;

Additional function configuration includes transceiver gain, transceiver delay, transceiver mode, mode suppression and CLIMAX group;

A pair of transceiver VoIP stations can configure the transceiver mode as transceiver isolation, transceiver suppression, and transceiver tone reduction;

Transceiver isolation means that the transceiver is in full-duplex mode; transceiver suppression means that the transceiver is in half-duplex mode; transceiver attenuation means that when the VoIP radio transmitter is transmitting, the VoIP radio receiver will suppress attenuation according to the mode suppression amount;

The VoIP stations that form the same CLIMAX group can transmit simultaneously.
A VoIP radio gateway with a master-standby architecture based on the ED137 protocol according to claim 6, wherein the signaling state machine of the media signaling interaction module is responsible for communicating with the VCCS server, VoIP radio, RRCE, and system operation and maintenance module. Equipped web for SIP signaling interaction; network media transceiving implements VoIP radio and RRCE media transceiving; master and standby media switching and network media transceiving implement VCCS media interaction; VoIP radio's dynamic compensation measurement is fed back to the CLIMAX group, which is controlled by the CLIMAX group The network delay of the VoIP radio in the same group; the media of the VoIP radio is exchanged with VCCS voice after gain adjustment and buffer control; the dual network status is detected and reported to the server separately.
The ED137 protocol-based active-standby architecture VoIP radio gateway according to claim 7, wherein the process of establishing SIP and RTP for the VoIP radio gateway session includes the following steps:

Step b1: After the station name, IP, port, SIP account, and SIP password are configured, the VoIP station gateway will periodically initiate a session request to the VoIP station. If a session has been established with the VoIP station, no more requests will be sent;

Step b2: After the VoIP radio station receives the request, if the configuration is correct, it will automatically answer the request, reply 100Trying and 200OK, and the VoIP radio gateway will reply ACK to confirm the establishment of the session;

Step b3, the VoIP station gateway and the VoIP station start to send and receive RTP packets in both directions. When the VoIP station is not sending and receiving, the two-way transmission is a keep-alive packet without load; when the VoIP station is transmitting, the VoIP station gateway sends it to The VoIP radio station is the RTP voice packet carrying the payload; when the VoIP radio station receives it, the VoIP radio station sends the RTP voice packet carrying the payload to the VoIP radio gateway;

Step b4: After the VoIP radio station establishes the session, it registers the VoIP radio station SIP account with the server; when the first registration is successful, the server initiates a session request to the VoIP radio gateway; after the session is established, the server and the VoIP radio gateway send RTP voice packets , Use Mark bit to represent PTT signal and carrier signal;

Step b5: When the VoIP station fails or the network is disconnected, the VoIP station gateway and the VoIP station session are interrupted, and the VoIP station session with the VCCS is ended.