[opus] Estimating bitrate during a real-time voip call
droancea at yahoo.com
Tue Dec 16 02:09:51 PST 2014
You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if there is packet loss. You know if there's packet loss if you receive RTCP . Linphone does this .
From: Manpreet Singh <manpreets7 at gmail.com>
To: opus at xiph.org
Sent: Tuesday, December 16, 2014 7:54 AM
Subject: [opus] Estimating bitrate during a real-time voip call
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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