Opus
Opus is an open, royalty-free audio codec designed for both speech and music over the internet. Opus works well at very low bitrates and adds very little delay, which makes it ideal for real-time communication. It is the default audio codec for WebRTC and is common in WebM files. Opus was standardized by the IETF in 2012.
Why it matters
Opus keeps voices clear when networks are weak and adds very little delay, which is what makes video calls and live audio feel natural. Choosing it for real-time apps reduces dropouts and talk-over, and it costs nothing in royalties.
How it works
- Opus captures sound and splits it into short frames, often 20 milliseconds long.
- It uses a speech-focused mode, a music-focused mode, or both together, depending on the content.
- The bitrate can change on the fly to match the network.
- The receiving app decodes each frame as it arrives, keeping delay low.
Opus vs AAC: which is better?
Opus is better for real-time voice and low bitrates, while AAC is better for broad playback in on-demand streaming. Opus adds little delay and sounds good even at low bitrates, which is ideal for calls. AAC has wider support in HLS players, Apple devices and smart TVs.
What bitrate should I use for Opus?
For speech, Opus sounds clear at roughly 16 to 32 kbps, and for stereo music around 96 to 128 kbps works well. The codec supports anything from 6 kbps to 510 kbps. Real-time apps often let Opus adjust its bitrate to match the caller's network.
Where it's used
Video call apps, telehealth platforms and live audio rooms built on WebRTC use Opus so voices stay clear and in sync on weak mobile networks.
By the numbers
- Opus is standardized in RFC 6716, published by the IETF in September 2012. (IETF RFC 6716)
- Opus supports all bitrates from 6 kbit/s to 510 kbit/s, with algorithmic delays from 5 ms to 65.2 ms. (IETF RFC 6716)
