GstRTPNetCC: Difference between revisions

GstRTPNetCC (view source)

700 bytes removed , 16 April 2019

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 __NOTOC__
 == Overview ==
 GstRTPNetCC is a GStreamer plug-in that allows estimating the available bandwidth during a single video stream based on the RTP information.
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 Start navigating this wiki by going to the [[ GstRTPNetCC Design|GstRTPNetCC Fundamentals]] page in the table of contents.
-<!--
-= Contents =
-* '''[[WebRTC Basics]]'''
-:- What is WebRTC? Why is it the next big thing? Why should you switch to WebRTC?
-* '''[[Introduction to RidgeRun's GstWebRTC]]'''
-:- How does RidgeRun's WebRTC solution makes things easy
-* '''[[GstWebRTC Release Notes]]'''
-:- What's new in each WebRTC release?
-* '''[[Getting Started with GstWebRTC]]'''
-:- A quick configure, build and installation guide
-* '''[[GstWebRTC Pipelines]]'''
-:- Some example pipelines to get GstRrWebRTC up and running out of the box on the most popular embedded platforms and desktop computers.
-* '''[[GstWebRTC Signaler Developer's Guide]]'''
-:- A walkthrough on how to create a custom signaler for your application
--->
 [[Category:GStreamer]][[Category:GstRTPNetCC]]