Difference between revisions of "GstWebRTC"
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The plug-in is equipped with three elements: | The plug-in is equipped with three elements: | ||
| − | * GstWebRTCSrc | + | * GstWebRTCSrc: Unidirectional video/audio source. |
| − | * GstWebRTCSink | + | * GstWebRTCSink: Unidirectional video/audio sink. |
| − | * GstWebRTCBin | + | * GstWebRTCBin: Bidirectional video/audio source/sink. |
| − | |||
== Supported Formats == | == Supported Formats == | ||
Revision as of 13:23, 6 July 2017
Overview
GstWebRTC is a GStreamer plug-in that turns pipelines into WebRTC compliant endpoints, which allows audio and/or video streaming using the WebRTC protocol.
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Template:Eval SDK Download, Demo Image download and Contact Us buttons |
GstWebRTC Promo Video
Features
The plug-in is equipped with three elements:
- GstWebRTCSrc: Unidirectional video/audio source.
- GstWebRTCSink: Unidirectional video/audio sink.
- GstWebRTCBin: Bidirectional video/audio source/sink.
Supported Formats
Audio
- Opus
Video
- Vp8
- H264
Getting Started
Start navigating this wiki by going to the WebRTC Fundamentals page in the table of contents.
