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Interactive protocol lab

See a voice session happen.

Make a deterministic call entirely inside this tab. Hear both media directions, press keys, barge in, and watch generated RTVBP frames travel between a phone and an application while the browser reports real WebRTC health.

No accountNo backendNo microphoneNo telemetry

Runs in this tab

Browser phone

Ready — no network or microphone neededEverything stays in your browser
Media path
Caller → app
App → caller

Glass-box view

Protocol timeline

  1. Ready when you are.Call to watch profile negotiation, typed frames, media, and shutdown.

Browser-reported

WebRTC health​

Live statistics
Codec
Not started
Connection
Idle
ICE
Idle
Selected pair
Not selected
Bitrate
—
RTT
—
Jitter
—
Packets lost
—

A dash means this browser did not report the statistic. Candidate addresses and SDP are never displayed.

Spec-generated proof

Generated conformance scenario​

1 / 4

The voice peer offers audio, the application selects it, then receives session state and DTMF.

Open generated flow reference

What is real here?

The SDK, frames, media, and statistics.​

Simulation mode runs two actual TypeScript SDK sessions, generated role adapters, typed peers, the generated classic.v1 envelope, and a native local WebRTC peer pair. The phone UI is the only simulation layer.

Bring an endpoint when you are ready.

Advanced live mode accepts a caller-supplied secure endpoint and optional runtime credentials. The site ships with neither and never persists what you enter.

Build the browser integration