CN110809097A - Main and standby framework VoIP radio station gateway based on ED137 protocol - Google Patents

Main and standby framework VoIP radio station gateway based on ED137 protocol Download PDF

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Publication number
CN110809097A
CN110809097A CN201911080931.6A CN201911080931A CN110809097A CN 110809097 A CN110809097 A CN 110809097A CN 201911080931 A CN201911080931 A CN 201911080931A CN 110809097 A CN110809097 A CN 110809097A
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radio station
voip
gateway
voip radio
session
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CN110809097B (en
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张朋
王虎
孙英晖
杨康
朱昆
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Nanjing Lesi Electronic Equipment Co Ltd
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Nanjing Lesi Electronic Equipment Co Ltd
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Priority to PCT/CN2020/090123 priority patent/WO2021088345A1/en
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L41/00—Arrangements for maintenance, administration or management of data switching networks, e.g. of packet switching networks
    • H04L41/06—Management of faults, events, alarms or notifications
    • H04L41/0654—Management of faults, events, alarms or notifications using network fault recovery
    • H04L41/0663—Performing the actions predefined by failover planning, e.g. switching to standby network elements
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10—Architectures or entities
    • H04L65/102—Gateways
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1073—Registration or de-registration
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • H04M7/0081—Network operation, administration, maintenance, or provisioning

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  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
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  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Telephonic Communication Services (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

The invention provides a VoIP radio station gateway of a main and standby framework based on an ED137 protocol, and VoIP radio station gateway equipment adopts a VoIP design method and designs a VoIP radio station access interface, a VCCS access scheme and a main and standby redundancy function strictly according to the ED137 protocol. And adding various additional functions such as working modes, gains, time delay and the like. Specific functions such as CLIMAX, BSS, dynamic delay compensation and the like are realized, and the advantages of the VoIP radio station are greatly reflected. The VoIP radio station gateway can be accessed to a VoIP radio station receiver, a VoIP radio station transmitter, a VoIP radio station transceiver integrated machine, a VoIP radio station remote control device and the like. The VoIP radio station gateway is provided with a friendly front-end management terminal, and all functions can be managed and configured through web access. The invention not only meets the requirements of the air traffic control voice system on the VoIP radio station, but also meets the requirements of air traffic control safety redundancy.

Description

Main and standby framework VoIP radio station gateway based on ED137 protocol
Technical Field
The invention belongs to the technical field of voice communication engineering, and particularly relates to a master and backup framework VoIP radio station gateway based on an ED137 protocol.
Background
The european civil aviation equipment organization (europae) is an international organization for specifically setting technical specifications of civil aviation electronic equipment, and belongs to the european aviation safety organization. It is a non-profit organization consisting of air stakeholders in europe and other regions. The EUROCAE WG67 working group successively issued ED137 protocol versions, such as February (2009), ED-137A (2010), ED-137B (2012), ED-137C (2017) and the like. The protocol is a standard and interface specification for international aviation service providers to access VoIP voice, particularly VoIP radio stations.
In recent years, in technical demand indexes for international airport voice switching and control system (VCCS) bidding, it is clearly required that the VCCS system supports the ED137 protocol and can access a VoIP station. However, only international providers such as Frequentis, RS, etc. are available internationally. Several companies of domestic voice systems are also in the beginning of this respect. Therefore, the implementation of the ED137 protocol access of the voice system and being able to perfectly fit with the main/standby architecture of the voice system is imminent.
Disclosure of Invention
The purpose of the invention is as follows: the invention belongs to the technical field of air traffic control voice communication engineering, and solves the main problem that the radio station access mode of the traditional air traffic control voice exchange system is an EM (effective man) analog access mode, so that the urgent requirement of the current VoIP radio station communication cannot be met.
In order to solve the technical problem, the invention discloses a host and backup architecture VoIP (Voice over Internet Protocol) Radio station gateway based on an ED137 Protocol (VoIP ATM Radio component Interoperability standard for VoIP ATM Radio Components), which increases various functions such as working mode, gain, delay and the like, adopts a background service design mode, is provided with a web management terminal, has an access VCCS (Voice over Internet Protocol), and can realize the purpose of accessing a VoIP Radio station to a VCCS system of the host and backup architecture. The requirements of an air traffic control voice system on a VoIP radio station can be met, and the requirements of the main VCCS interface and the standby VCCS interface are also met. The VoIP radio station gateway comprises a core switching module, a main server and standby server selection module, a media signaling interaction module and a system operation and maintenance module;
the core switching module is used for realizing voice switching of a VoIP radio station gateway;
the active/standby server selection module is used for automatically switching to the standby server when the active server selection module detects that the media switching is overtime after the active server is hung up;
the media signaling interaction module is used for state machine management of SIP signaling (Session initiation protocol) and media transceiving of a voice channel;
the system operation and maintenance module is used for providing functions of webpage terminal management and network state detection.
The core switching module is used for realizing voice switching of a VoIP radio station gateway and realizing radio station conference, transceiving delay buffering, transceiving gain adjustment and dynamic delay compensation through a VoIP voice switching technology;
when the core exchange module realizes the radio station conference, the RTP (Real time protocol) protocol is used for interacting voice and radio station control commands, and the radio station control commands are in an extended RTP header;
the receiving and transmitting delay buffering range is 0-600 ms;
the range of the receiving and transmitting gain adjustment is-25 db- +25 db;
the dynamic delay compensation is to calculate the maximum delay compensation time of a CLIMAX (radio CLIMAX) group according to the network delay between a VoIP radio station and a VoIP radio station gateway, control the delay time of the VoIP radio station to achieve synchronous transmission of the VoIP radio station in the group, wherein the dynamic delay compensation range is 0-254 ms, and the precision is 2 ms.
The core switching module is used for realizing Voice switching of a VoIP radio station gateway and also comprises media information used for switching a VoIP radio station session and a VCCS (Voice Communication Control System) session;
the VoIP radio station session is a precondition of VCCS session, and after the VoIP radio station session is established, an R2S (a transmission format) keep-alive packet or a voice packet carrying Payload (voice load) is transmitted;
when a session of the VoIP radio station is established, negotiating R2S-keepalive period (R2S keep-alive interval) and R2S-keepalive Multiplier (R2S keep-alive packet number), wherein the product of the two is overtime time, and disconnecting the VoIP radio station and a VoIP radio station gateway if R2S is lost or a voice packet exceeds the overtime time;
R2S keep-alive packets are typically longer in interval and fewer bytes than voice packets. When the VoIP radio station does not work, an R2S keep-alive packet is transmitted between the VoIP radio station and the VoIP radio station gateway;
when the VoIP radio station receiver receives the carrier wave, the VoIP radio station receiver transmits a voice packet carrying Payload to a VoIP radio station gateway;
when a VCCS system needs to transmit a radio station, a voice packet carrying Payload is transmitted to a VoIP radio station transmitter, and a VoIP radio station gateway can establish a session with the VCCS only after the session of the VoIP radio station is established;
the VoIP radio station gateway sends registration information of a connected VoIP radio station to the VCCS, and after receiving the registration information, the VCCS initiates a session establishment request to the VoIP radio station gateway to establish a VCCS session of a corresponding type; after the VCCS conversation is established, transmitting voice by using a standard RTP, transmitting silence without using a VoIP radio station, and representing a PTT (push ToTalk) signal and a carrier signal by using a Mark bit (Mark bit) of the RTP; when a VoIP station fails or drops, the VoIP station gateway may send a request to the VCCS to end the session for the VoIP station.
The media signaling interaction module uses an SIP state machine mechanism to realize state machine management of signaling and media transceiving of a channel, and specifically comprises the following steps:
step a1, the SIP state machine independently manages each VoIP radio station state, and the initial state is idle state;
step a2, after the VoIP radio station gateway is connected with the VoIP radio station, sending a session establishment request, and entering a request state;
step a3, the VoIP radio station responds to the request, if the session invitation is accepted, the SIP state machine enters the session state; if the session invitation is refused, the SIP state machine re-enters an idle state and waits for the next connection;
step a4, after the SIP state machine enters the session state, opening the VoIP radio station channel media receiving and sending.
The system operation and maintenance module is used for providing functions of web page terminal management and network state detection, wherein the web page terminal management comprises radio station configuration, gateway configuration and additional function configuration; through network state detection, the VoIP radio station gateway can report the dual-network state information to the server in real time.
The system operation and maintenance module is provided with a web (webpage) front-end configuration page, and radio station configuration, gateway configuration and additional function configuration are completed through the web front-end configuration page;
wherein the station configuration comprises station name, station IP and port, station type, PTT mode, station SIP account password (for example, configuration is as radio Rx, 192.168.1.100, 5060, Rx, Normal, 500, 123456);
the gateway configuration comprises server IP port information and gateway SIP account password (for example: configuration is 192.168.1.6, 5060, 9000 and 123456);
additional functional configurations include transceiver gain, transceiver delay, transceiver mode, mode suppression amount, and CLIMAX group (e.g., configured as receive 10db, transmit-5 db, receive 40ms, transmit 20ms, transceiver suppression mode, group 1);
the pair of VoIP transceiver stations can configure the transceiving mode as transceiving isolation, transceiving inhibition and transceiving sound reduction;
the receiving and transmitting isolation means that the transceiver is in a full duplex mode; the transmit-receive suppression means that the transceiver is in a half-duplex mode; when the receiving and sending sound reduction means that the transmitter of the VoIP radio station transmits, the receiver of the VoIP radio station carries out suppression attenuation according to the mode suppression quantity;
VoIP stations that are collocated in the same CLIMAX group are able to transmit synchronously.
The signaling state machine of the media signaling interaction module is responsible for carrying out SIP signaling interaction with a VCCS server, a VoIP Radio station, RRCE (Remote Radio Control Equipment) and a web equipped by a system operation and maintenance module; the network media transceiving realizes the media transceiving of the VoIP radio station and the RRCE; the interaction of VCCS media is realized by the switching of the main and standby media and the receiving and sending of network media; after dynamic compensation measurement of the VoIP radio station, feeding back to the CLIMAX group, wherein the CLIMAX group controls 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; and (4) independently detecting and reporting the double-network state to a server.
The process of establishing SIP and RTP for the VoIP radio station gateway session comprises the following steps:
step b1, after configuring the radio station name, IP, port, SIP account number, SIP password, the VoIP radio station gateway will periodically initiate session request to VoIP radio station, if the session is established with VoIP radio station, no request is sent;
step b2, after receiving the request, if the configuration is correct, the VoIP radio station will automatically respond to the request, reply 100Trying (receiving response message) and 200OK (normal response message), the VoIP radio station gateway replies ACK (response acknowledgement message) to confirm the session establishment;
step b3, starting to transmit RTP message in two directions between the VoIP radio station gateway and the VoIP radio station, when the VoIP radio station does not transmit, transmitting keep alive packet without load in two directions; when the VoIP radio station transmits, the VoIP radio station gateway sends a RTP voice packet carrying a load to the VoIP radio station; when the VoIP radio station receives, the VoIP radio station sends an RTP voice packet carrying load to a VoIP radio station gateway;
step b4, after the session is established by the VoIP radio station, registering the SIP account of the VoIP radio station to the server; when the first registration is successful, the server initiates a session request to a VoIP radio station gateway; after the session is established, an RTP voice packet is sent between the server and the VoIP radio station gateway, and a Mark bit is used for representing a PTT signal and a carrier signal;
step b5, when the VoIP radio station is out of order or disconnected, the VoIP radio station gateway and VoIP radio station session is interrupted, and the station session with VCCS is ended.
The VoIP radio gateway provided by the invention is a VoIP radio gateway device which is used for a voice communication system (VCCS) and conforms to ED137 protocol. The VCCS accessed to the main and standby redundant server architecture can be supported, the main and standby server selection module provides automatic detection server media for the VoIP radio station gateway, and the main and standby server media are switched after time out. The 60ms detection and switching interval allows the VoIP radio gateway to provide continuous voice and signaling for the VCCS. The VoIP radio station is divided into a single VoIP radio station receiver Rx, a single VoIP radio station transmitter Tx and a VoIP radio station transceiver RTx according to the radio station types. Different types of VoIP radio stations can be connected to the VoIP radio gateway. The VoIP radio gateway supports 32 VoIP radio channels at maximum, including 16 receiving channels and 16 sending channels. The core switches include station voice switches and server voice switches. The VCCS establishes a session over standard SIP, using Mark bits of the RTP header to represent PTT signal and carrier signal transmissions. The VoIP radio station is accessed to the VoIP radio station gateway through the ED137, and VoIP radio station control signaling and voice are transmitted through an RTP packet with an extension header. The VoIP radio station gateway can be transplanted in a cross-platform mode, and supports windows, Linux, cross compiling arm-Linux and the like. The VoIP radio station gateway is provided with a parameter configuration management service, and accesses a VoIP radio station gateway server through a web page to perform radio station parameter configuration, channel configuration, gateway equipment configuration, server parameter configuration and the like. The VoIP radio gateway supports multi-network card equipment, the equipment can be used for network port aggregation, and the VoIP radio gateway can realize isolation of a VoIP radio network and a VCCS network and provides safety guarantee for communication of the VCCS network. The VoIP radio station gateway provides various channel expansion functions for the VCCS, including channel receiving and transmitting gain adjustment, channel receiving and transmitting delay, three channel receiving and transmitting modes and the like.
The active/standby server selection module is an important function module of the active/standby architecture VoIP radio station gateway based on the ED137 protocol. The VoIP radio station gateway can be accessed to a single server VCCS and can also be accessed to a main backup redundant VCCS of the server. The VoIP radio station gateway receives SIP information sent by any server, and the media is two paths of session media which are simultaneously established. When the main service media is not received in more than 60ms, the VoIP radio station gateway switches the media to ensure the redundancy effect of the main server and the standby server. The characteristic can realize that when any one server breaks down, the system can be used for the VoIP radio station without interruption, and even the difference can not be heard by human ears. The VoIP Radio gateway may also establish a session with Radio Remote Control Equipment (RRCE) to complete the primary and standby functions of the VoIP Radio. The ED137 protocol specifies the main and standby functions of the RRCE. Two modes applicable to RRCE: single frequency mode and dual frequency mode. Two pairs of active and standby receiving and transmitting VoIP radio stations can be controlled. Signal quality information for a VoIP station receiver may also be communicated.
The media signaling interaction module is a basic function module of the gateway of the VoIP radio station of the main and standby framework based on the ED137 protocol. SIP signaling is the underlying protocol for VoIP voice systems. The session is established by the session initiator sending an INVITE request, the invitee replying a 100Tying response and a 200OK response, then the initiator replying an ACK response, the session is established, and the two parties start to send keep-alive packets or media. When one party needs to finish the session, a BYE session finishing request is sent, the other party replies a 200OK response, and the two parties clean up resources. Because the SIP protocol is an application layer protocol, belongs to a UDP protocol at a transport layer, and the UDP protocol is an unreliable transport protocol, a packet may be lost during transmission, the SIP protocol has a timeout retransmission mechanism to ensure communication reliability. The ED137 protocol specifies the CLIMAX delay and dynamic delay compensation functions. The transmitter of the VoIP radio station of the same CLIMAX group receives the delay detection request message (RMM) from the gateway of the VoIP radio station at regular time and replies a measurement response message (MAM). The VoIP radio station gateway calculates the network delay from VCCS to VoIP radio station and the transmission delay of VoIP radio station through RMM and MAM. The same set of VoIP radio transmitters adjust the transmission delay. The CLIMAX function can be realized (the same group of VoIP radio station transmitters synchronously transmit sound to the air).
Has the advantages that: the invention discloses a VoIP radio station gateway of a main and standby framework based on an ED137 protocol, which can meet the access requirement of a VoIP radio station (or RRCE) conforming to the ED137 protocol, can be accessed into a VCCS system of the main and standby framework, and also provides the functions of receiving and transmitting gain adjustment, receiving and transmitting delay buffering, dynamic delay compensation, a radio station receiving and transmitting mode, a CLIMAX group and the like on the basis. The gateway can meet the control requirements of various civil aviation airports on VoIP radio stations at home and abroad. The additional functionality also greatly improves the availability of the VoIP radio gateway.
Drawings
The foregoing and/or other advantages of the invention will become further apparent from the following detailed description of the invention when taken in conjunction with the accompanying drawings.
Fig. 1 is a block diagram of a VoIP radio gateway of an active/standby architecture based on ED137 protocol.
Fig. 2 is a composition diagram of a logical structure of a VoIP radio gateway of an active/standby architecture based on the ED137 protocol.
Fig. 3 is a flow chart of the SIP/RTP established by the gateway session of the VoIP station with the main/standby architecture based on the ED137 protocol.
Fig. 4 is a CLIMAX code diagram of a VoIP radio station gateway based on the ED137 protocol.
Fig. 5 is a dynamic delay detection code diagram of a master/slave frame VoIP radio gateway based on ED137 protocol.
Fig. 6 is a schematic diagram of the delay of the gateway of the VoIP radio station of the active/standby architecture based on the ED137 protocol.
Fig. 7 is a remote control code diagram of a gateway radio station of a VoIP radio station with an active/standby architecture based on the ED137 protocol.
Fig. 8 is a connection block diagram of a voice communication system.
Detailed Description
The invention provides a VoIP radio station gateway of a main and standby framework based on an ED137 protocol, which is designed for a blank pipe voice system of a main and standby redundant framework, provides a VoIP radio station which is accessed to the voice system and accords with the ED137 protocol, and supports the dynamic delay compensation of the ED137 protocol and the access function of remote control equipment of the radio station. In addition, the VoIP radio station gateway provides rich voice additional functions, and can provide VoIP radio station gateway radio station gain, radio station delay, radio station transceiving mode and CLIMAX group to realize dynamic compensation.
1. A block diagram of the VoIP station gateway is shown in fig. 1. The VoIP radio station gateway comprises a core switching module, a main server and standby server selection module, a media signaling interaction module and a system operation and maintenance module.
The core switching module is a voice switching module of a gateway of the VoIP station. The method comprises the steps of radio station meeting, receiving and transmitting delay buffering, receiving and transmitting gain adjustment and dynamic delay compensation. Station conferencing uses the RTP protocol for interactive voice and station control commands. The station control commands are in the extended RTP header, such as PTT information and carrier information. Transmit-receive delay buffering and transmit-receive gain adjustment are additional functions of the VoIP radio gateway. The receiving gain and the transmitting gain of each VoIP radio station can be realized, so that the voice input balance can be realized aiming at the physical characteristics of the unavailable VoIP radio stations. The range of gain adjustment is (-25db to +25 db). The receiving and transmitting delay buffering is based on the dynamic delay, and the receiving and transmitting delay function can be added. The transmission and reception delay buffering range is (0-600 ms). The dynamic delay compensation is to calculate the maximum delay compensation time of the CLIMAX group according to the network delay between the VoIP radio station and the VoIP radio station gateway, and control the delay time of the VoIP radio station to achieve the synchronous transmission of the VoIP radio station in the group. The dynamic compensation range is (0 to 254ms, precision 2ms)
The main and standby server selection module is a functional module of the VoIP radio station gateway connected with the main and standby voice servers. After the main server is hung, the VoIP radio station gateway detects that the media switching is overtime (60ms), and can automatically switch to the standby server, so that the media continuity and the voice uninterrupted can be ensured. The main and standby redundancy functions of the server are realized. The main and standby signaling is in a mode of dual-sending and main reply of the server.
The media signaling interaction module comprises the state machine management of signaling and the media transceiving of the channel. SIP is managed using SIP state machine mechanisms to establish sessions. The media transceiving can realize large-scale media transceiving.
The system operation and maintenance module comprises webpage terminal management and network state detection. The station configuration, the gateway configuration and the additional function configuration can be logged in and configured through a webpage. The VoIP radio gateway can report the double-network state information to the server in real time.
2. The logical structure of the VoIP station gateway is shown in figure 2. The signaling state machine is responsible for SIP signaling interaction with VCCS servers, VoIP stations, RRCEs, and the web. The network media transceiving realizes the media transceiving of the VoIP radio station and the RRCE. The interaction of VCCS media is realized by the switching of the main and standby media and the receiving and sending of network media. And after the dynamic compensation measurement of the VoIP radio station, feeding back to the CLIMAX group, wherein 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. And (4) independently detecting and reporting the double-network state to a server.
3. A VoIP station gateway session establishment SIP/RTP flow diagram is shown in figure 3. After configuring attributes such as a radio station name, an IP (Internet protocol), a port, an SIP (Session initiation protocol) account number, an SIP password and the like, the VoIP radio station gateway can initiate a session request to the VoIP radio station regularly (10 s). If a session has been established with a VoIP station, no further requests are sent. After receiving the request, the VoIP radio station will automatically reply the request, and reply 100Trying and 200OK if the configuration is correct. The VoIP station gateway replies an ACK acknowledging the session establishment. Then, the VoIP radio station gateway and the VoIP radio station start to transmit and receive RTP messages in two directions. When the VoIP radio station does not transmit and receive, the bidirectional transmission is the keep-alive packet without load; when the VoIP radio station transmits, the VoIP radio station gateway sends a RTP voice packet carrying a load to the VoIP radio station; when the VoIP radio station receives, the VoIP radio station sends RTP voice packets carrying payload to a VoIP radio station gateway. After establishing session, the VoIP radio station registers SIP account number of the VoIP radio station to the server. When the first registration is successful, the server initiates a session request to the VoIP radio station gateway. And after the session is established, an RTP voice packet is sent between the server and the VoIP radio station gateway. Mark bits are used to represent the PTT signal and the carrier signal. When the VoIP radio station has faults or network disconnection or other abnormalities, the session between the VoIP radio station gateway and the VoIP radio station is interrupted, and the station session between the VoIP radio station gateway and the VCCS is ended.
4. A VoIP station gateway CLIMAX time delay (CLD) code map is shown in fig. 4. No matter the VoIP radio station gateway and the VoIP radio station are keep-alive packets or RTP voice packets, the keep-alive packets or the RTP voice packets belong to RTP messages, the RTP messages comprise RTP headers and loads, and the loads can be optional. The RTP header includes a normal RTP header and an extended RTP header. As shown in fig. 4, i.e. the format of the extended RTP header, the meaning of each field is as follows:
PTT type: the method is used for a VoIP radio transmitter or a VoIP radio transceiver, and defines the types of PTT, wherein the types of PTT comprise normal PTT, priority PTT, coupled PTT and emergency PTT. May be set up on the VoIP radio gateway web.
SQU: carrier state information is defined with a VoIP radio receiver or VoIP radio transceiver.
PTT-ID: the PTT ID is used for a VoIP radio transmitter or a VoIP radio transceiver and is distributed to a user accessing the VoIP radio. For the user to know which user is currently transmitting on the VoIP station.
PM: PTT Mute used for multi-user accessing VoIP radio station.
PTTS: PTT Summation, for simultaneous transmission at the same priority.
SCT: simultaneous Call Transmissions for multi-user Simultaneous transmission of station fields.
Reserved: a reserved field.
X: indicating whether the extension information field is used or not.
TYPE: an extension field type.
LENGTH: the field length is extended.
VALUE: the field contents are extended.
The CLD is as follows: CLIMAX delay, bit24 is 0 for relative delay and 1 for absolute delay. bit 25-bit 31 represent delay time, unit 2ms, range 0-127 (0 ms-254 ms).
5. The VoIP station gateway dynamic delay detection code pattern is shown in fig. 5. The VoIP radio station gateway sends a keep-alive packet or an RTP voice packet to a VoIP radio station at regular time (4s) and carries a dynamic delay measurement message (RMM), and a keep-alive packet or an RTP voice packet replied by the VoIP radio station carries a dynamic delay measurement response message (MAM), wherein the significance of each field is as follows:
TQV: the gateway measures whether the time is relative time or absolute time on behalf of the VoIP station.
T1: is the time sent by the gateway RMM of the VoIP station in 125 us.
TQG: the gateway measures whether the time is relative time or absolute time on behalf of the VoIP station.
NMR: representing whether a new measurement, such as jitter buffer failure, is required for the VoIP station.
T2: is the time when the VoIP station receives the RMM in 125 us.
Tsd: the interval between sending MAM and receiving RMM by VoIP station is 125 us.
Tj 1: is the jitter buffer delay, in units of 125 us.
And (2) Tid: is the internal delay of the VoIP station, the delay from the jitter buffer to the antenna, in units of 125 us.
6. A schematic diagram of the VoIP station gateway delay is shown in fig. 6. The CLIMAX group synchronous transmission function requires that the VoIP station transmission of the same CLIMAX group is synchronous transmission, namely, the time synchronization of the transmission to the antenna. The VCCS system delays of all VoIP stations are equal, i.e., Tv1 is equal. As long as TdTx is equal for all VoIP stations. The VoIP station gateway packetization delay and jitter buffer delay are equal for all VoIP stations, i.e., Tp 1.
The network delay is as follows:
Figure BDA0002263927470000091
where T3 is the time when the MAM was sent by the VoIP radio and T4 is the time when the MAM was received by the gateway of the VoIP radio.
The internal delay Tid of the VoIP station is Td1+ Ts 1.
The delay from the VoIP station gateway to the antenna is therefore: TdTx-Tp1 Tn1+ Tj1+ Tid.
The delay from the gateway to the antenna of the VoIP station can be measured by the RMM and the MAM. And then the sending time delay of the VoIP radio station is dynamically controlled by CLD to achieve the sending synchronization.
7. The VoIP radio station gateway can be accessed to the RRCE, and the RRCE is connected with the VoIP radio station, so that the main and standby functions of the VoIP radio station can be realized. The remote control code pattern of the VoIP station is shown in fig. 7. The meaning of each field is as follows:
BSS-qidx: and comparing and selecting the signal quality.
BSS-qidx-ml: and (4) a comparison and selection method.
MSTxF 1: f1 frequency point master-slave VoIP radio station transmitter enabling mark of RRCE.
MSRxF 1: f1 frequency point master-slave VoIP radio station receiver enabling mark of RRCE.
MSTxF 2: f2 frequency point master-slave VoIP radio station transmitter enabling mark of RRCE.
MSRxF 2: f2 frequency point master-slave VoIP radio station receiver enabling mark of RRCE.
SelTxF 1: RRCE's F1 frequency point VoIP radio transmitter PTT enable flag.
SelTxF 2: RRCE's F2 frequency point VoIP radio transmitter PTT enable flag.
MuRxF 1: f1 frequency point VoIP station receiver mute enable flag of RRCE.
MuRxF 2: f2 frequency point VoIP station receiver mute enable flag of RRCE.
SQF 1: f1 frequency point carrier enabling mark of RRCE.
SQF 2: f2 frequency point carrier enabling mark of RRCE.
reserved: and (6) reserving.
F1 SQI: and F1 frequency point of RRCE receives signal quality.
F2 SQI: and F2 frequency point of RRCE receives signal quality.
Examples
Depending on the particular application scenario, a particular voice communication system is shown in FIG. 8. The set of voice communication system comprises a Server (Server), a monitoring system (RCMS), a Switch (Switch), a seat (CWP), a Line interface unit (Line interface unit), a VoIP telephone gateway, a VoIP radio gateway, a VoIP telephone, a VoIP radio and the like. The server and the network switch are in a dual redundancy design, and other equipment and the server interface are connected through a dual network. The VoIP radio gateway/VoIP phone gateway isolates the VoIP radio/VoIP phone from the voice system network. The VoIP radio gateway is used for realizing the access of a VoIP radio to the voice communication system. The system and the gateway of the VoIP radio station are realized and put into use.
The present invention provides a primary and standby architecture VoIP radio station gateway based on ED137 protocol, and there are many methods and ways for implementing the technical solution, and the above description is only a preferred embodiment of the present invention, and it should be noted that, for those skilled in the art, without departing from the principle of the present invention, several improvements and modifications may be made, and these improvements and modifications should also be regarded as the protection scope of the present invention. All the components not specified in the present embodiment can be realized by the prior art.

Claims (8)

1. A VoIP radio gateway based on an ED137 protocol and a main and standby framework is characterized by comprising a core switching module, a main and standby server selection module, a media signaling interaction module and a system operation and maintenance module;
the core switching module is used for realizing voice switching of a VoIP radio station gateway;
the active/standby server selection module is used for automatically switching to the standby server when the active server selection module detects that the media switching is overtime after the active server is hung up;
the media signaling interaction module is used for the state machine management of SIP signaling and the media receiving and sending of a voice channel;
the system operation and maintenance module is used for providing functions of webpage terminal management and network state detection.
2. The VoIP gateway of claim 1, wherein the core switch module is configured to implement voice switching of a VoIP gateway, and implement station conferencing, transmit-receive delay buffering, transmit-receive gain adjustment, and dynamic delay compensation through VoIP voice switching technology;
when the core exchange module realizes the radio station conference, the core exchange module uses an RTP protocol to interact voice and a radio station control command, and the radio station control command is in an extended RTP head;
the receiving and transmitting delay buffering range is 0-600 ms;
the range of the receiving and transmitting gain adjustment is-25 db- +25 db;
the dynamic delay compensation is to calculate the maximum delay compensation time of the CLIMAX group according to the network delay between the VoIP radio station and the VoIP radio station gateway, control the delay time of the VoIP radio station, and achieve the synchronous transmission of the VoIP radio station in the group, wherein the dynamic delay compensation range is 0-254 ms, and the precision is 2 ms.
3. The gateway of claim 2, wherein the core switch module is configured to implement voice switching of a VoIP radio gateway, and further includes a media information for exchanging a VoIP radio session with a VCCS session;
the radio station session is a precondition of VCCS session, and after the radio station session is established, an R2S keep-alive packet or a voice packet carrying Payload is transmitted;
when the radio station session is established, negotiating R2S-keepalivePeriod and R2S-keepaliveMultiplier, wherein the product of the two is overtime, and disconnecting the radio station if R2S is lost between the VoIP radio station and the VoIP radio station gateway or a voice packet exceeds the overtime;
when the VoIP radio station does not work, an R2S keep-alive packet is transmitted between the VoIP radio station and the VoIP radio station gateway;
when the VoIP radio station receiver receives the carrier wave, the VoIP radio station receiver transmits a voice packet carrying Payload to a VoIP radio station gateway;
when a VCCS system needs to transmit a radio station, a voice packet carrying Payload is transmitted to a VoIP radio station transmitter, and a VoIP radio station gateway can establish a session with the VCCS only after a radio station session is established;
the VoIP radio station gateway sends registration information of a connected VoIP radio station to the VCCS, and after receiving the registration information, the VCCS initiates a session establishment request to the VoIP radio station gateway to establish a VCCS session of a corresponding type; after the VCCS conversation is established, transmitting voice by using a standard RTP, transmitting silence without using a VoIP radio station, and representing a PTT signal and a carrier signal by using a Mark bit of the RTP; when a VoIP station fails or drops, the VoIP station gateway may send a request to the VCCS to end the station session.
4. The gateway of claim 3, wherein the media signaling interaction module implements state machine management of signaling and media transceiving of channels by using an SIP state machine mechanism, and specifically comprises the following steps:
step a1, the SIP state machine independently manages each VoIP radio station state, and the initial state is idle state;
step a2, after the VoIP radio station gateway is connected with the VoIP radio station, sending a session establishment request, and entering a request state;
step a3, the VoIP radio station responds to the request, if the session invitation is accepted, the SIP state machine enters the session state; if the session invitation is refused, the SIP state machine re-enters an idle state and waits for the next connection;
step a4, after the SIP state machine enters the session state, opening the VoIP radio station channel media receiving and sending.
5. The VoIP gateway as claimed in claim 4, wherein the system operation and maintenance module is configured to provide web page terminal management and network status detection functions, wherein the web page terminal management includes radio station configuration, gateway configuration, and additional function configuration; through network state detection, the VoIP radio station gateway can report the dual-network state information to the server in real time.
6. The VoIP gateway as claimed in claim 5, wherein the system operation and maintenance module is configured with a web front-end configuration page, and performs station configuration, gateway configuration and additional function configuration via the web front-end configuration page;
the radio station configuration comprises a radio station name, a radio station IP and port, a radio station type, a PTT mode and a radio station SIP account password;
the gateway configuration comprises server IP port information and a gateway SIP account password;
the additional function configuration comprises a transceiving gain, a transceiving delay, a transceiving mode, a mode suppression amount and a CLIMAX group;
the pair of VoIP transceiver stations can configure the transceiving mode as transceiving isolation, transceiving inhibition and transceiving sound reduction;
the receiving and transmitting isolation means that the transceiver is in a full duplex mode; the transmit-receive suppression means that the transceiver is in a half-duplex mode; when the receiving and sending sound reduction means that the transmitter of the VoIP radio station transmits, the receiver of the VoIP radio station carries out suppression attenuation according to the mode suppression quantity;
VoIP stations that are collocated in the same CLIMAX group are able to transmit synchronously.
7. The gateway of claim 6, wherein the signaling state machine of the media signaling interaction module is responsible for SIP signaling interaction with the VCCS server, the VoIP station, the RRCE, and the web equipped by the system operation and maintenance module; the network media transceiving realizes the media transceiving of the VoIP radio station and the RRCE; the interaction of VCCS media is realized by the switching of the main and standby media and the receiving and sending of network media; after dynamic compensation measurement of the VoIP radio station, feeding back to the CLIMAX group, wherein the CLIMAX group controls 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; and (4) independently detecting and reporting the double-network state to a server.
8. The VoIP gateway of claim 7, wherein the session establishment SIP and RTP of the VoIP gateway includes the following steps:
step b1, after configuring the radio station name, IP, port, SIP account number, SIP password, the VoIP radio station gateway will periodically initiate session request to VoIP radio station, if the session is established with VoIP radio station, no request is sent;
step b2, after receiving the request, if the configuration is correct, the VoIP radio station will automatically respond to the request, reply 100Trying and 200OK, and the VoIP radio station gateway replies ACK to confirm the session establishment;
step b3, starting to transmit RTP message in two directions between the VoIP radio station gateway and the VoIP radio station, when the VoIP radio station does not transmit, transmitting keep alive packet without load in two directions; when the VoIP radio station transmits, the VoIP radio station gateway sends a RTP voice packet carrying a load to the VoIP radio station; when the VoIP radio station receives, the VoIP radio station sends an RTP voice packet carrying load to a VoIP radio station gateway;
step b4, after the session is established by the VoIP radio station, registering the SIP account of the VoIP radio station to the server; when the first registration is successful, the server initiates a session request to a VoIP radio station gateway; after the session is established, an RTP voice packet is sent between the server and the VoIP radio station gateway, and a Mark bit is used for representing a PTT signal and a carrier signal;
step b5, when the VoIP radio station is out of order or disconnected, the VoIP radio station gateway and VoIP radio station session is interrupted, and the VoIP radio station session with VCCS is ended.
CN201911080931.6A 2019-11-07 2019-11-07 Main and standby framework VoIP radio station gateway based on ED137 protocol Active CN110809097B (en)

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