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Opus is a lossy audio codec standardized by the Internet Engineering Task Force (IETF) for interactive speech and audio. It can encode voice and music across a wide range of bitrates, but choosing Opus in an encoder does not guarantee a live-streaming platform will accept it. The codec, the transport carrying it, and the platform’s ingest requirements are separate parts of the chain.
What is the Opus audio codec?
Opus is an encoding and decoding format for digital audio. The IETF’s RFC 6716 defines it for interactive uses such as voice over IP, videoconferencing, in-game chat, and live distributed music. Its design combines linear prediction (LP), often useful for speech, with modified discrete cosine transform (MDCT) coding, which also supports music. RFC 6716 was published as an IETF Standards Track document in September 2012.
For a streamer, Opus is one choice within a software and delivery workflow. It is not a microphone type, and the codec specification alone does not describe a complete streaming setup. An encoder creates the audio stream; a transport carries it; and a receiving service decides what formats it accepts.
What can Opus do?
Bitrate and delay
RFC 6716 specifies Opus bitrates from 6 to 510 kbit/s and algorithmic delay from 5 to 65.2 milliseconds. These are codec capabilities, not settings every application exposes or a guarantee that higher bitrate will always sound better. The result also depends on the audio, mono or stereo channel choice, bandwidth mode, frame size, encoder settings, and transport constraints.
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For 20 ms frames, RFC 6716 gives these example bitrate “sweet spots.” Treat them as codec guidance for those contexts, not universal live-stream presets:
| Use case | Example bitrate |
|---|---|
| Narrowband speech | 8–12 kbit/s |
| Wideband speech | 16–20 kbit/s |
| Fullband speech | 28–40 kbit/s |
| Fullband mono music | 48–64 kbit/s |
| Fullband stereo music | 64–128 kbit/s |
These example ranges are from the 2012 RFC and are specifically associated with 20 ms frames. A particular encoder, application, network, or receiving platform may impose different limits or offer different controls.
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Bandwidth and sample rate
Opus defines five audio bandwidths. The terms can be confusing because the fullband effective sample rate is 48 kHz, but the codec does not code audio above 20 kHz.
| Opus bandwidth | Audio bandwidth | Effective sample rate |
|---|---|---|
| Narrowband | 4 kHz | 8 kHz |
| Medium-band | 6 kHz | 12 kHz |
| Wideband | 8 kHz | 16 kHz |
| Super-wideband | 12 kHz | 24 kHz |
| Fullband | 20 kHz | 48 kHz |
So “48 kHz” should not be read as 48 kHz of encoded audio bandwidth. It is the effective sample rate associated with Opus fullband mode.
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Channels and changing parameters
Opus can carry mono or stereo frames. Its parameters can change dynamically without stopping the stream, and in-band signaling lets a decoder interpret those changes. This flexibility belongs to the codec; an application’s controls, transport, or receiving service may still limit which combinations work.
Is Opus good for streaming?
Opus is relevant to real-time audio, including WebRTC communication. The IETF WebRTC audio requirements refer to Opus, and RFC 7875 describes Opus implementation as mandatory for WebRTC endpoints. That makes it important for browser-based calling and other real-time communication workflows.
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WebRTC support does not establish that a video-streaming service accepts Opus for live broadcast ingest. The codec may be capable of encoding the audio while the selected transport or platform rejects it. Check the current official encoder or ingest documentation for the specific service and software version before selecting an audio codec. RFC 6716 and the WebRTC specifications are not platform-specific ingest instructions.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.What bitrate should you use for Opus?
Start with the kind of audio and the channel count, then check what your application and receiving service support. The RFC’s 20 ms examples above distinguish speech from mono and stereo music because those uses have different requirements; they are not a substitute for a platform’s current ingest guidance.
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- For speech, choose the relevant bandwidth and speech example as a starting reference, then use the settings your transport and service support.
- For music, account for mono versus stereo and use the RFC’s music examples only when their fullband and 20 ms conditions apply.
- Do not raise bitrate in isolation: a higher value may not improve the result if the source, channel configuration, encoder, or delivery path is the limiting factor.
- For live broadcast, verify ingest compatibility and limits from the receiving platform’s current official documentation. No universal Opus bitrate or platform setting follows from the codec specification alone.
Opus compared with other codecs
There is no context-free “best codec” conclusion here. Compare candidates against the actual workflow:
- Compatibility: Does the transport and receiving platform accept the codec?
- Audio and bitrate: How do the options perform for the same content and channel count?
- Latency: What is the end-to-end delay in the chosen implementation, not just the codec’s algorithmic delay?
- Recording and delivery: Does the workflow require a particular export, playback, or archival format?
Without current, authoritative comparisons for a particular platform and configuration, a universal quality or bitrate ranking would be misleading.
Licensing and commercial use
The Opus project says the specification is freely available and that its reference encoder and decoder may be integrated into applications, including commercial products, at no cost under the project’s stated terms. The project also describes royalty-free patent licenses for compatible use and notes a patent-litigation condition. Read the official Opus licensing page for the applicable conditions; this summary is not legal advice.
Keeping a prerecorded YouTube stream running
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