Command Communication System VHF Technical Implementation Solution
The Command Communication System VHF Technical Implementation Solution adopts a converged architecture of "VHF radio + SIP voice gateway + SIP dispatch platform" to achieve cross-regional two-way communication between Nanjing Tower VHF signals and Beijing AOC positions, balancing low latency, high reliability, and scalability. The system networking diagram is as follows:
SIP VHF radio gateways are deployed at locations requiring VHF communication access to the tower; they connect to the operation command system via audio and PTT signaling.
One VHF radio can be connected to multiple SIP VHF radio gateways in parallel through a VHF gateway access box, allowing multiple airlines to share the VHF radio on a time-division basis.
Access to carrier and enterprise telephone networks is provided via SIP E1 access gateways.
The specific implementation is as follows:
(1) Signal Access and Low-Latency SIP Transmission
A domestically produced SIP media gateway is deployed at Nanjing Tower to convert VHF analog voice signals into standard SIP/RTP IP voice streams, completing seamless conversion from analog to digital signals while supporting SIP protocol standard interaction to ensure compatibility with AOC position SIP terminals;
An encrypted tunnel is used between Beijing and Nanjing to transmit SIP/RTP voice streams, with QoS traffic priority policies configured to set VHF-related SIP voice streams at the highest priority, avoiding latency and packet loss caused by network congestion and ensuring end-to-end latency ≤100ms to meet real-time air-ground communication requirements;
Dual active/standby transmission links are deployed. When the primary link latency exceeds 100ms or packet loss rate exceeds 5%, the router automatically switches to the backup link with a switchover time of <1s. During switchover, SIP voice streams are seamlessly handed over, ensuring uninterrupted voice and continuous monitoring.
(2) SIP Voice Insertion and Tiered Authorization Control
Voice insertion permission adopts a three-tier control mechanism of "role authorization + secondary approval + operation traceability," leveraging the SIP dispatch platform for fine-grained permission management. Only AOC duty personnel with designated permissions can initiate voice insertion requests through their own SIP phones/soft terminals, and the request must be secondarily approved by the duty manager on the SIP core management console before voice insertion permission is activated;
During voice insertion, audio signals from AOC position SIP terminals are encapsulated via the SIP protocol and pushed through the encrypted tunnel to the Nanjing Tower SIP media gateway. The gateway restores the SIP/RTP voice stream to analog audio, driving the VHF radio to transmit, achieving two-way communication between AOC positions and Nanjing Tower. The insertion process does not interfere with the existing air-ground communication link, ensuring communication order;
All SIP terminal authorization operations, voice insertion operations, and call content generate detailed audit logs through the SIP dispatch platform, recording operator, operation time, operation content, call duration, SIP terminal information, etc. Logs are stored for ≥3 years, supporting traceability, query, and audit to meet compliance requirements.
(3) Multi-Base SIP Expansion Architecture
The system adopts a "center-spoke" star topology architecture, deploying the SIP core dispatch and management platform at the Beijing AOC center as the core node of the entire VHF-SIP converged system; new bases only need to deploy standardized VHF radios + SIP media gateways, with gateways connecting to the central management platform via SIP standard interfaces and encrypted tunnels, without modifying core equipment at the center.
When expanding to a new base, only local VHF radio and SIP gateway linkage configuration and network encrypted tunnel setup are required, enabling remote monitoring of VHF signals at the new base, voice insertion, and SIP intercommunication between multiple bases, truly achieving smooth expansion and collaborative support.
Functions (Adapted to VHF+SIP Architecture)
Intelligent Voice Detection and Marking: Based on real-time SIP voice stream parsing technology, supports configuration of civil aviation professional keyword/callsign voice detection. When specified keywords are monitored, the system automatically marks the SIP voice recording and triggers alarms, while synchronizing the marking points to the recording playback interface, eliminating manual segment-by-segment review and greatly improving monitoring efficiency;
Multi-Position Simultaneous Monitoring and Speaking Queue: Through the SIP dispatch platform, concurrent management of multiple terminals is achieved, supporting multiple AOC position SIP terminals to monitor the same VHF signal simultaneously; when multiple positions initiate voice insertion requests, the SIP dispatch platform automatically implements speaking queue management, displaying queue status in real time on each terminal interface, effectively avoiding voice conflicts and ensuring orderly communication;
VHF Signal Status Visualization: A VHF-SIP converged status monitoring dashboard is built, displaying in real time VHF signal strength, signal-to-noise ratio, SIP voice stream transmission latency, link connectivity status, and SIP gateway operating status at each base. When signal anomalies, link interruptions, or SIP terminal faults occur, multi-channel alarms such as SMS and platform pop-ups are triggered, facilitating timely troubleshooting and handling by operations and maintenance personnel.