[opus] Estimating bitrate during a real-time voip call
manpreets7 at gmail.com
Tue Dec 16 11:41:33 PST 2014
The issue is that not all packet loss maybe congestion related. Often,
reducing bitrate seems to have no impact on improving packet loss.
On Tue, Dec 16, 2014 at 2:09 AM, Dragos Oancea <droancea at yahoo.com> wrote:
> 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 .
> 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
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