WebRTC on Ships: Real-Time Video Communication Without Cloud Infrastructure

The Video Call Problem at Sea

When an engineer on a cargo vessel encounters an unfamiliar fault condition in the engine room, the ideal scenario is a real-time video call with the manufacturer's technical support team ashore. They can see the panel, the readings, the physical state of the equipment — and guide the repair.

This is technically possible and increasingly expected by fleet operators. But "video call" over satellite introduces latency, bandwidth cost, and reliability requirements that most implementations handle poorly.

WebRTC is the technology that changes the calculation.

What WebRTC Is

WebRTC (Web Real-Time Communication) is a W3C standard that enables peer-to-peer real-time audio, video, and data transfer directly between browsers — without a media server in the path.

The signalling (negotiating the connection) goes through a server. The media (actual audio/video packets) goes directly between devices.

For a video call between two users on the same vessel network:

• Signalling: vessel Matrix server
• Media: direct peer-to-peer between the two devices on the local network

No internet connection required. No satellite bandwidth consumed. The call runs over WiFi.

TURN Servers: When Direct P2P Isn't Possible

In some network configurations — NAT rules, firewall restrictions — direct peer-to-peer is blocked. WebRTC uses TURN servers as fallback relays.

For on-vessel WebRTC:

• The TURN server runs on the vessel server (not in the cloud)
• Media packets relay through the vessel TURN server when direct P2P fails
• Still no internet dependency

For vessel-to-shore video calls:

• Internet connectivity is required (obviously)
• TURN servers on both sides handle relay if P2P across NAT fails
• Bandwidth requirement: 500 kbps per call for standard definition, 1.5 Mbps for HD

Satellite connections can support multiple SD video calls simultaneously on a modern VSAT connection.

Maritime Use Cases for WebRTC

Engine room expert consultation — Shore-based technical experts participate in live inspections via video. The crew member holds up a phone; the expert sees the equipment in real time.

Bridge-to-shore operational briefings — Master connects to fleet manager for departure briefing. No telephone call quality; video-quality visual communication.

Medical telemedicine — Crew medical officer consults with shoreside doctor for complex medical situations. The doctor can see the patient via the officer's device.

Safety officer incident review — Remote safety officer observes incident scene without being physically present.

Training delivery — Shore-based trainers conduct live instructional sessions with crew on multiple vessels simultaneously.

Integration with SIP

WebRTC calls can be bridged to SIP phone extensions. A crew member on their smartphone (WebRTC) can call a bridge desk handset (SIP). The media bridge runs on the vessel softswitch.

This provides a unified communication experience: one application handles text messaging (Matrix), push notifications (Web Push), voice calls (SIP or WebRTC), and video calls (WebRTC). The crew member doesn't need to switch applications based on communication type.

Bandwidth Requirements and Optimisation

For vessels with limited satellite bandwidth:

Automatic quality adjustment — WebRTC implementations adapt video quality based on available bandwidth. A 256 kbps connection will receive a lower-resolution feed; a 2 Mbps connection will receive HD.

Audio-only fallback — When bandwidth is insufficient for video, falling back to audio-only WebRTC retains the real-time connection without the video overhead.

Local call bandwidth — On-vessel calls between crew members use zero satellite bandwidth. Only vessel-to-shore calls consume allocated bandwidth.

Deployment Requirements

To deploy WebRTC in a maritime environment:

TURN server on the vessel (Coturn is the standard open-source implementation)

Signalling server — the Matrix homeserver handles WebRTC session negotiation through its spec

TLS certificates — WebRTC requires HTTPS. Manage with Let's Encrypt certificates or your own PKI

Network port access — STUN (3478 UDP), TURN (443 TCP/UDP for relay)

Client support — Any modern browser (Chrome, Firefox, Safari) supports WebRTC natively

The deployment is lightweight. For most vessels the TURN server runs as a container alongside the Matrix homeserver on the vessel server.

Summary

WebRTC provides the missing real-time video capability for maritime communication platforms — without cloud dependency, without satellite bandwidth cost for on-vessel calls, and without requiring separate applications.

Combined with Matrix messaging and SIP voice, it completes a maritime unified communication stack built entirely on open standards.