Difference between revisions of "GstWebRTC"
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Start navigating this wiki by going to the [[ GstWebRTC - WebRTC Fundamentals|WebRTC Fundamentals ]] page in the table of contents. | Start navigating this wiki by going to the [[ GstWebRTC - WebRTC Fundamentals|WebRTC Fundamentals ]] page in the table of contents. | ||
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:- A walkthrough on how to create a custom signaler for your application | :- A walkthrough on how to create a custom signaler for your application | ||
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[[Category:GStreamer]][[Category:GstWebRTC]] | [[Category:GStreamer]][[Category:GstWebRTC]] | ||
Revision as of 10:35, 15 August 2019
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Overview
GstRrWebRTC is a GStreamer plug-in that turns pipelines into WebRTC compliant endpoints, which allows audio and/or video streaming using the WebRTC protocol.
GstRrWebRTC Promo Video
GstRrWebRTC MCU Support
Features
The plug-in is equipped with three elements:
- GstRrWebRTCSrc: Unidirectional video/audio source.
- GstRrWebRTCSink: Unidirectional video/audio sink.
- GstRrWebRTCBin: Bidirectional video/audio source/sink.
Supported Formats
Audio
- Opus
- G722
- PCMU
- PCMA
Video
- VP8
- H264
Getting Started
Start navigating this wiki by going to the WebRTC Fundamentals page in the table of contents.
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