GstWebRTC - GstWebRTC Basics: Difference between revisions
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[[GstWebRTC - WebRTC Fundamentals|WebRTC Fundamentals]]| | [[GstWebRTC - WebRTC Fundamentals|WebRTC Fundamentals]]| | ||
[[GstWebRTC - Signaler | Signaler]]| | [[GstWebRTC - Signaler | Signaler]]| | ||
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This page describes the basic features of Ridgerun's GstRrWebRTC Gstreamer plugin. | This page describes the basic features of Ridgerun's GstRrWebRTC Gstreamer plugin. | ||
== What is GstRrWebRTC? == | == What is GstRrWebRTC? == |
Revision as of 19:52, 18 January 2019
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This page describes the basic features of Ridgerun's GstRrWebRTC Gstreamer plugin.
What is GstRrWebRTC?
GstRrWebRTC is a GStreamer plug-in that turns pipelines into WebRTC compliant endpoints, in order to allow audio and/or video streaming using the WebRTC protocol.
Why GstRrWebRTC?
Other WebRTC solutions will automatically detect the video and audio sources, as well as the decoders/encoders and other elements to be used to build the pipeline. This may be convenient for many applications, but result limiting for several other use cases. To mention some of them:
- Extend existing pipeline to support WebRTC streaming
- Use non-standard pipeline configurations
- High performance pipeline tuning for resource critical systems
- Dynamic stream handling in a running pipeline.
- Fine grained pipeline control
- Quick gst-launch prototyping
GstRrWebRTC was developed based on this criteria. As such, the plug-in is ideal for:
- Embedded platforms
- Existing media servers/applications
- Advanced multimedia solutions
Plugin Overview
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