Opus Codec
Opus is a modern, royalty-free audio codec that adapts its bitrate and bandwidth to network conditions. It delivers wideband voice quality and handles packet loss well.
Published 8 November 2026 · 1 min read · By SpaGreen Creative
Short answer
Opus is a royalty-free audio codec, defined in RFC 6716, that covers narrowband to fullband audio at bitrates from about 6 to 510 kbit/s and adapts to network conditions during a call.
Opus is an audio codec standardised by the IETF in RFC 6716 and available without licence fees. It combines a speech-oriented coder with a general music coder and can switch between them, which lets it cover everything from low-bitrate voice to high-fidelity audio. In SIP it is carried over RTP as described in RFC 7587, with a clock rate of 48,000 Hz.
For telephony its useful properties are adaptability and robustness. The encoder can change bitrate from one frame to the next, so a softphone can lower it when the network degrades and raise it again when conditions improve. Built-in forward error correction and packet loss concealment keep speech intelligible when packets go missing. At typical voice bitrates of 16 to 32 kbit/s, Opus sounds clearer than G.711 while using less bandwidth.
Support on the server side is uneven. Recent Asterisk releases include an Opus module, and many PBX distributions such as Issabel and FreePBX can offer it, but older systems and some trunk providers do not. When Opus is not available, the call falls back to another codec in the list, usually G.711.
SIPdia includes Opus alongside G.711 ulaw and alaw, through the baresip engine.
Sources and further reading
- RFC 6716: Definition of the Opus Audio Codecwww.rfc-editor.org
- RFC 7587: RTP Payload Format for the Opus Speech and Audio Codecwww.rfc-editor.org
This article is published by SpaGreen Creative, the maker of SIPdia. Product names mentioned belong to their owners. Where it describes other software, treat the steps as general guidance and check the official documentation linked above.
