mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-07 15:02:40 +00:00
32 lines
1.3 KiB
Markdown
32 lines
1.3 KiB
Markdown
|
## GStreamer WebRTC demos
|
||
|
|
|
||
|
|
All demos use the same signalling server in the `signalling/` directory
|
||
|
|
|
||
|
|
You will need the following repositories till the GStreamer WebRTC implementation is merged upstream:
|
||
|
|
|
||
|
|
https://github.com/ystreet/gstreamer/tree/promise
|
||
|
|
|
||
|
|
https://github.com/ystreet/gst-plugins-bad/tree/webrtc
|
||
|
|
|
||
|
|
You can build these with either Autotools gst-uninstalled:
|
||
|
|
|
||
|
|
https://arunraghavan.net/2014/07/quick-start-guide-to-gst-uninstalled-1-x/
|
||
|
|
|
||
|
|
Or with Meson gst-build:
|
||
|
|
|
||
|
|
https://cgit.freedesktop.org/gstreamer/gst-build/
|
||
|
|
|
||
|
|
### sendrecv: Send and receive audio and video
|
||
|
|
|
||
|
|
* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
|
||
|
|
- The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
|
||
|
|
* Build and run the sources in the `gst/` directory on your machine
|
||
|
|
|
||
|
|
```console
|
||
|
|
$ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
|
||
|
|
```
|
||
|
|
|
||
|
|
* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
|
||
|
|
* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
|
||
|
|
* You will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app
|