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. Any device running GStreamer can connect to another WebRTC peer and share Audio, Video and Data content. You specify the pipeline, we do the hard work.

 

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 these criteria. As such, the plug-in is ideal for:

• Embedded platforms

• Existing media servers/applications

• Advanced multimedia solutions
  
  Learn more about GstWebRTC