Voice, video, recording, and AI-powered communication built for modern applications. Production-ready WebRTC platform powering telemedicine, CRM meetings, customer support, and AI voice experiences.
A complete communications stack — not another abstraction layer on top of a third-party service.
HD video with adaptive simulcast. Automatically switches quality layers based on network conditions.
Crystal-clear audio with echo cancellation, noise suppression, and active speaker detection.
Full screen and application window sharing on Web and Android, streamed in real time.
Per-participant recording via FFmpeg. Store locally or push to Cloudflare R2 or AWS S3.
Real-time speech-to-text via Whisper. Ready to pipe into summaries, CRM notes, or agents.
1-on-1 calls get full HD. Group calls automatically downgrade layers to save bandwidth.
JWT auth, HMAC TURN credentials, DTLS-SRTP encryption, and per-route rate limiting.
Full control over your data, privacy, and compliance. Deploy on any VPS or cloud provider.
Most platforms hide their architecture. We embrace it. Every layer is replaceable, inspectable, and runs on your infrastructure.
Web · Android · iOS (coming soon)
Real-time transport negotiation and room events
4 workers · simulcast · VP8 / H.264 / Opus
Room state, node heartbeats, pub/sub routing
Per-participant capture via RTP PlainTransport
Cloudflare R2 · AWS S3 · local disk — pluggable
Purpose-built for high-stakes, high-reliability communication use cases.
HIPAA-ready architecture. Recorded consultations, screen sharing for results, and AI transcription for clinical notes.
Embedded video directly inside your CRM. Automatic call recording and transcription synced to contact records.
Voice and video support with live transcription, agent coaching overlays, and post-call summaries.
Multi-participant sessions with screen sharing, recording for playback, and active speaker detection.
WebRTC-powered voice pipeline for real-time AI agents — low latency audio in, processed speech out.
Self-hosted voice and video for teams who need full data sovereignty and no external dependencies.
Built by engineers, for engineers. Every design decision has a reason behind it.
Run on your own VPS. No vendor lock-in. No data leaving your infrastructure.
Direct mediasoup integration. No abstraction tax. Full simulcast and consumer control.
Built on stream-webrtc-android 1.3.8. No abandoned libraries. Kotlin-first API.
FFmpeg records directly from mediasoup PlainTransport. No external recording service required.
Zero egress fees for recorded files. Pluggable storage — swap R2, S3, or local disk with one env var.
Whisper transcription pipeline included. Drop in your model, point at the audio stream, done.
Shipping in production. Building what's next.
Try the live demo or reach out to discuss your use case.