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

Dragos Oancea 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 .
Regards,Dragos Oancea
      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?

opus mailing list
opus at xiph.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.xiph.org/pipermail/opus/attachments/20141216/206e3a43/attachment.htm 

More information about the opus mailing list