WebRTC developer tool

WebRTC Codec Test

Check WebRTC codec support in your browser. Inspect advertised audio and video send/receive capabilities for VP8, VP9, H.264, AV1, Opus, G.711 and more.

Video send capabilities

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CodecClock rateChannelsFMTP

Video receive capabilities

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CodecClock rateChannelsFMTP

Audio send capabilities

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CodecClock rateChannelsFMTP

Audio receive capabilities

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CodecClock rateChannelsFMTP
These are WebRTC capabilities advertised by the browser through getCapabilities(). They do not prove that hardware acceleration, encoding, decoding or successful negotiation will work for every codec/profile.

What does the WebRTC codec test show?

This tool reads the browser's advertised WebRTC RTP capabilities for audio and video. It separates sender and receiver capability lists so developers can inspect which MIME types, clock rates, channel counts and codec formatting parameters are exposed for negotiation.

Browser capability is not the same as successful media processing

A codec appearing in the table means the browser advertises that capability through WebRTC APIs. It does not guarantee hardware acceleration, a particular profile, smooth performance, or successful negotiation with every remote peer. Use the result as a browser capability inventory rather than a full encode/decode benchmark.

Related diagnostics

Use the WebRTC SDP parser to see which codecs are actually offered in a session description, the WebRTC stats viewer to inspect the negotiated codec during a live local test, or the Data Channel test for non-media WebRTC diagnostics.

WebRTC codec test FAQ

Does codec support mean hardware acceleration is available?

No. This tool reports codec capabilities exposed by the browser WebRTC APIs. It does not prove hardware acceleration, successful encoding or decoding, or negotiation with a remote peer.

Why can the same codec appear more than once?

Browsers can expose multiple payload configurations or codec profiles. Send and receive capability lists can also differ, so the table keeps those directions separate.

Which WebRTC codecs should I look for?

Opus is commonly used for WebRTC audio, while VP8, VP9, H.264 and AV1 may appear for video depending on browser and platform support. Actual call negotiation still depends on both peers.