GstWebRTC - GstWebRTC Basics
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__NO_TOC__ This page describes the basic features of Ridgerun's GstWebRTC Gstreamer plugin.
Contents
What is GstWebRTC?
GstWebRTC 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 GstWebRTC?
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
GstWebRTC 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
The plug-in is equipped with three elements:
GstWebRTC Requirements
Signaler
Predefined or custom signalers are required for GstWebRTC to work.
Predefined
Currently OpenWebRTC signaler is supported, more predefined signalers are going to be supported as the project grows.
To set a predefined signaler, the plugin's elements property signaler is used.
Custom
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