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
G.711 (mu-law and A-law)
G.729
WebRTC (Web Real-Time Communications)
SDP (Session Description Protocol)
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