[opus] Estimating bitrate during a real-time voip call
manpreets7 at gmail.com
Mon Dec 15 22:54:02 PST 2014
Although this maybe considered out of scope here, but I'll ask anyway.
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?
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