WebRTC Test – Check Camera, Microphone & Connection
Test your browser, camera, microphone, ICE and STUN connectivity before a video call. Run these checks locally in your browser free, with no signup required.
WebRTC readiness
Camera, microphone & screen
Webcam Test
Preview your camera and inspect resolution, frame rate and device details.
CameraCamera Resolution Test
Probe common webcam resolutions from 240p through 4K.
CameraWebcam FPS Test
Measure actual delivered camera frame rate over a timed sample.
Audio inputMicrophone Test
Check live input level, sample rate, channels and browser audio processing.
Audio outputSpeaker Test
Verify browser audio playback and left/right channel output.
Display captureScreen Share Test
Preview screen sharing and inspect source type, resolution and frame rate.
Connectivity & privacy
ICE Test
Gather host, server-reflexive and relay ICE candidates.
ConnectivitySTUN Test
Check whether a STUN server can discover a server-reflexive candidate.
ConnectivityTURN Test
Verify TURN credentials and relay candidate allocation.
PrivacyWebRTC Leak Test
Inspect ICE candidates and see which addresses WebRTC exposes.
Developer tools
ICE Candidate Parser
Decode candidate type, protocol, priority, address and related fields.
DeveloperSDP Parser
Parse WebRTC media sections, codecs, ICE credentials and DTLS fingerprints.
DeveloperWebRTC Codec Test
Inspect advertised WebRTC audio and video sender/receiver codec capabilities.
DeveloperData Channel Test
Verify RTCDataChannel text, binary transfer, RTT and local loopback throughput.
DeveloperWebRTC Stats Viewer
Inspect live getStats() bitrate, packets, frames, codec and raw JSON.
What does this WebRTC test check?
This WebRTC test checks whether your browser exposes the core APIs needed for real-time communication, whether camera and microphone capture can start successfully, and whether ICE candidate gathering works through a public STUN server. These checks cover the most common browser-side conditions that affect video meetings, browser calling, customer-support sessions and WebRTC development.
Camera and microphone access
The readiness test requests camera and microphone permission so it can verify that the browser can open both media devices. If either device fails, use the dedicated webcam test or microphone test to inspect the device in more detail. For deeper camera checks, use the camera resolution test and webcam FPS test.
ICE, STUN and TURN connectivity
WebRTC uses ICE candidates to find a network path between peers. Developers can inspect candidates with the WebRTC ICE test, test a custom STUN server, verify relay allocation with the TURN server test, or decode raw data with the ICE candidate parser.
Developer debugging tools
Use the SDP parser to inspect negotiated media descriptions, the WebRTC codec test to inventory browser-advertised RTP capabilities, the RTCDataChannel test for browser-level loopback transfer diagnostics, and the WebRTC stats viewer to inspect live getStats() metrics.
Screen sharing and privacy
Use the screen share test to verify browser display capture before a call. ICE candidates may reveal network address information depending on the browser, operating system and network configuration; use the WebRTC leak test when you specifically want to inspect candidate addresses and privacy behavior.
WebRTC test FAQ
Does this WebRTC test upload my camera or microphone?
The readiness test opens local browser media streams only long enough to confirm that capture works, then stops the tracks. The current media checks do not upload your camera or microphone stream.
Why can WebRTC work even if STUN fails?
Some local or constrained scenarios can still produce host candidates, but internet peer-to-peer connectivity commonly relies on STUN discovery and, when direct connectivity fails, a TURN relay.
Why should I test TURN separately?
A basic readiness test can confirm WebRTC and STUN behavior, but TURN needs valid server credentials and successful relay allocation. The dedicated TURN test checks for relay candidates.