Reducing delay in RTP streaming

[Tim Müller]

On Tue, 2017-01-03 at 14:55 +0100, Kévin Aupée wrote:

Hi,

> I'm using Gstreamer for RTP streaming with this pipeline :
> 
> > gst-launch-1.0 udpsrc port=5000 caps=application/x-rtp !
> > rtpjitterbuffer latency=50 drop-on-latency=true ! rtpopusdepay !
> > opusdec ! alsasink sync=false
> 
> The delay is about 300ms. It is not so bad but I want to reduce it as
> much as I can.
> I don't really understand all the properties of the elements and I
> would like to know which ones can help reduce the latency (besides
> the ones I already use).

Maybe also have a look at the various alsasink properties that affect
the audio sink internal latency and buffer size. You should probably
drop the sync=false though.

Cheers
 -Tim

-- 
Tim Müller, Centricular Ltd - http://www.centricular.com