Codecs & Media

Opus

Opus (RFC 6716) is a low-latency, highly adaptive audio codec covering bitrates from 6 kbps narrowband speech to 510 kbps fullband stereo music. It is the default codec for WebRTC and the strongly preferred codec for any modern VoIP stack.

Specs

• Sample rates: 8 / 12 / 16 / 24 / 48 kHz
• Bitrate: 6-510 kbps (variable, adapts in real time)
• Frame size: 2.5 / 5 / 10 / 20 / 40 / 60 ms
• MOS at 24 kbps wideband: ~4.5

Why prefer Opus

• Quality: at 24-32 kbps wideband, indistinguishable from uncompressed.
• Adaptive: adjusts bitrate in real time based on packet loss and bandwidth feedback (RTCP).
• FEC: in-band forward error correction recovers from up to 30% packet loss without re-transmits.
• Royalty-free.

PSTN interop

The PSTN cannot carry Opus — calls landing on a TDM trunk are transcoded to G.711. DIDHub does this transcode at the SBC, so you can offer Opus end-to-end inside your network and we handle the gateway.

Related terms