WebRTC OPUS codec : Minimum Bandwidth for good audio

Question

In my WebRTC application, OPUS codec has been used to compress the audio stream and I was wondering what is the minimum viable bandwidth that should be allocated for audio stream without jitter?

Answer 1

For Opus voice encoding, mono 16KHz sample rate:

• 6Kbps is a minimum, when voice is still recognizible
• 16Kbps is a medium - good enough
• 32Kbps is a maximum - you wont see big difference if encode at higher bitrate (higher than 32)

Answer 2

From what I tested a few hundred Kbps (bits, not bytes), approximately 300-400Kbps should be enough for good audio quality, not only voice, but music too. But more important is the network latency, which should be under 20-25ms.

For decent voice audio a tenth (30-40Kbps) should be enough. But this is for one peer only. The latency can be much higher but you'll hear small skips now and then, which should acceptable for conversations.