GstWebRTC - GstWebRTC Basics: Difference between revisions
mNo edit summary |
mNo edit summary |
||
(16 intermediate revisions by 3 users not shown) | |||
Line 1: | Line 1: | ||
GstWebRTC | {{GstWebRTC/Head|previous=WebRTC Fundamentals|next=Signaling|metakeywords=Gstreamer WebRTC Basics, Gstreamer WebRTC Plugin Overview,WebRTC Basics,signaling}} | ||
== | __TOC__ | ||
This page describes the basics, features and examples of Ridgerun's GstRrWebRTC [https://www.ridgerun.com/gstreamer GStreamer] plugin. | |||
== What is GstRrWebRTC? == | |||
GstRrWebRTC is a GStreamer plug-in that turns pipelines into [[GstWebRTC_-_WebRTC_Fundamentals|WebRTC]] compliant endpoints, in order to allow audio and/or video streaming using the WebRTC protocol. | |||
== GstRrWebRTC Use Cases == | |||
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: | 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: | ||
Line 18: | Line 18: | ||
* Quick gst-launch prototyping | * Quick gst-launch prototyping | ||
GstRrWebRTC was developed based on this criteria. As such, the plug-in is ideal for: | |||
* Embedded platforms | * Embedded platforms | ||
* Existing media servers/applications | * Existing media servers/applications | ||
* Advanced multimedia solutions | * Advanced multimedia solutions | ||
== GStreamer WebRTC Tutorial and Plugin Overview == | |||
== Plugin Overview == | <br> | ||
<center> | <center> | ||
<embedvideo service="vimeo">https://vimeo.com/190030003</embedvideo> | <embedvideo service="vimeo">https://vimeo.com/190030003</embedvideo> | ||
</center> | </center> | ||
| | {{GstWebRTC/Foot|previous=WebRTC Fundamentals|next=Signaling}} |
Latest revision as of 15:12, 9 March 2023
GstWebRTC | ||||||||
---|---|---|---|---|---|---|---|---|
WebRTC Fundamentals | ||||||||
GstWebRTC Basics | ||||||||
|
||||||||
Evaluating GstWebRTC | ||||||||
Getting the code | ||||||||
Building GstWebRTC | ||||||||
Examples | ||||||||
|
||||||||
MCU Demo Application | ||||||||
Contact Us |
This page describes the basics, features and examples 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.
GstRrWebRTC Use Cases
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
GStreamer WebRTC Tutorial and Plugin Overview