[opus] Estimating bitrate during a real-time voip call

Timothy B. Terriberry tterribe at xiph.org
Tue Dec 16 04:56:32 PST 2014

Manpreet Singh wrote:
> Opus has remarkable flexibility for changing encoder bitrate during a
> call. Are there any suggestions about how bandwidth/capacity between the
> two endpoints can be measured/estimated during a call so that the
> outgoing bitrate can be adjusted accordingly?

The topic is quite complicated. You can find code implementing the 
algorithm that Firefox and Chrome use in WebRTC at 
<http://www.webrtc.org/native-code/development>, though it is deeply 
integrated with the rest of the media stack there. It is documented at
<https://tools.ietf.org/html/draft-alvestrand-rmcat-congestion>. The 
IETF has chartered a working group to standardize some techniques. You 
can find the other active proposals here: 

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