The Sovereign Voice & Video Platform is a self-hosted real-time communication system — voice calls, video calls, small-group video meetings — running entirely on your own infrastructure, with no per-minute fees, no recording fees, and no third-party visibility into the call content. Tightly integrated with the messaging platform so the same address book and presence state work everywhere. In active production.
The critical piece that most “just use WebRTC” projects fail at is the NAT traversal relay: most users are behind a home router, mobile carrier NAT, or corporate firewall, and a naive peer-to-peer call simply does not connect. A properly-operated relay is what makes voice and video work reliably across the real internet.
| Property | Value |
|---|---|
| Hosting | Self-hosted on your own infrastructure |
| Foundation | Open standards (WebRTC, ICE, STUN, TURN) |
| Call types | 1-to-1 voice, 1-to-1 video, small-group video meetings, screen sharing |
| Encryption | End-to-end media encryption (DTLS-SRTP) |
| Network resilience | Works through any NAT, corporate firewall, or mobile carrier — proper TURN relay |
| Clients | Same as the messaging platform: mobile, desktop, browser |
| Cost model | Fixed infrastructure cost — no per-minute, per-participant, or per-month-per-user fees |
| Deployment | In production |
flowchart LR
A[Client A] -.->|signaling| B[Messaging server]
C[Client B] -.->|signaling| B
A -->|WebRTC media| D{"NAT / firewall?"}
D -->|direct| C
D -->|relay| E[TURN relay]
E --> C
Public-cloud meeting platforms — Zoom, Microsoft Teams, Google Meet, Webex — solved the practical problem of getting voice and video working across the modern internet, but they solved it as a vendor business model: per-seat licensing, participant caps on lower tiers, recording-feature paywalls, transcription-as-a-service, and the entire call audio routed through their infrastructure.
The technical building blocks for self-hosted real-time communication have existed for over a decade. What is rarely done well is operating the relay component: the small piece of infrastructure that bridges callers whose networks will not let them connect peer-to-peer.
The Sovereign Voice & Video Platform is the missing operational discipline around the open standards: a properly-configured relay, integrated authentication and authorization, monitoring, capacity management, and tight integration with the messaging platform so that calling someone is one click from the chat window.
Built on a mature set of open standards. The hard operational work is in running the relay component well — capacity-sized correctly, geographically placed correctly, and monitored properly.
| Layer | Implementation |
|---|---|
| Real-time media | WebRTC — open standard implemented in every modern browser and mobile SDK |
| Media encryption | DTLS-SRTP (standard, end-to-end on the media path) |
| NAT-traversal relay | coturn (industry-standard open-source TURN/STUN server) |
| Signaling protocol | XMPP Jingle (XEP-0166 / XEP-0167 / XEP-0176), shared with the messaging platform |
| Multi-party conferencing | Jitsi Videobridge (open-source SFU) for small-group video meetings |
| Identity integration | Shared with messaging and collaboration platforms — LDAP / SAML / OIDC |
| Mobile clients | Same client apps as the messaging platform (Conversations, Snikket, Monal) |
| Desktop clients | Same client apps as the messaging platform |
| Browser client | Jitsi Meet web interface for browser-based meetings |
| Network capacity planning | Sized for concurrent-call peak, with TURN-relay bandwidth as the limiting factor |
Reach out and we will scope a deployment for your typical concurrent-call volume, network topology, and integration requirements with the rest of your communications stack.