Opus is a flexible interactive speech and audio codec with a wide operating range defined by RFC 6716. It is an excellent fit for compatible WebRTC and softphone paths, but a UK PSTN leg commonly needs G.711 or transcoding, so use Opus only where the full route and UKDDI profile support it.
Settings used in this guide
These are public service values, not your password or account-specific authentication ID.
| Approved codec order | Confirm in your UKDDI activation email |
|---|---|
| Media security | SRTP |
Why Opus is different
RFC 6716 defines Opus for interactive speech and audio across 6 to 510 kbit/s, with support for narrowband through fullband audio, variable frame sizes and resilience tools. An implementation chooses parameters; the name Opus alone does not tell you the actual bandwidth or quality.
- Well suited to browsers, WebRTC, softphones and changing network conditions.
- Can adapt bitrate and audio bandwidth when the application implements policy correctly.
- Can use forward error correction and packet-loss concealment features, which still do not replace a stable network.
- May require transcoding when entering conventional PSTN or equipment that supports only G.711/G.722/G.729.
Where Opus is a strong or weak fit
The live carrier setting remains Confirm in your UKDDI activation email. A WebRTC gateway can use Opus toward the browser and a carrier-approved codec toward UKDDI, but that is transcoding and needs capacity monitoring.
Browser-to-browser, compatible softphone, collaboration and IP-to-IP calls where every media leg preserves Opus.
PSTN calls, legacy phones, recorders, SBCs and contact-centre applications that may not pass Opus end to end.
Set an intentional Opus policy
- 1Choose the application target
Decide whether the priority is speech quality, music/fullband audio, constrained bandwidth or loss resilience.
- 2Control bitrate and packetisation
Use supported endpoint/SBC settings and calculate complete packet overhead; do not assume Opus always uses less than G.711.
- 3Map transcoding
Identify where an Opus browser call becomes the codec supported by UKDDI/PSTN and confirm CPU/DSP capacity.
- 4Keep security aligned
WebRTC normally uses DTLS-SRTP while a SIP trunk may use a different issued SRTP method. Terminate and re-establish media security only at an authorised trusted component.
Test Opus under changing conditions
- Confirm actual codec, bitrate, audio bandwidth, packet time and encryption on both media legs.
- Test stable wired, normal Wi-Fi and controlled impaired-network conditions.
- Check calls to UK landlines/mobiles, voicemail, recording, hold, transfer and conferences.
- Watch gateway transcoder load at maximum simultaneous calls.
- Keep a G.711 A-law comparison on the same PSTN route where supported.