Modern audio codec for clear voice and music with low delay and bandwidth.
An agent uses a browser phone to call a customer. The call uses Opus at about 24 kilobits per second. When the home network gets busy, Opus lowers its bitrate and the call still sounds clear. The contact center sees fewer dropouts than with older audio formats.
Opus is an open, royalty free audio format that works well for speech and music. It delivers high quality at low bitrates, typically 16 to 32 kilobits per second for voice, and keeps delay very low so conversations feel natural. Opus adapts on the fly as network conditions change. It can reduce or raise bitrate, and it includes tools that hide brief losses and fill small gaps so callers hear fewer glitches. It handles a wide range of audio frequencies, from narrowband speech to fullband, which improves clarity and reduces listening fatigue.
For contact centers, Opus saves bandwidth, improves quality for remote agents on home Wi Fi, and is supported in all modern browsers. Many cloud providers prefer it for browser based calling. When calls must connect to older phone systems that only speak legacy formats, the edge gateway can convert the audio, which may add some processing load. Proper monitoring of loss, delay, and jitter still matters, but Opus gives a strong baseline for clear, reliable conversations.