mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-04-29 04:54:51 +00:00

History

Nirbheek Chauhan 492d13a7c9 README: link to blog post, document multiparty example Also add TODO stubs for MCU and SFU		2018-02-17 08:13:36 +05:30
..
multiparty-sendrecv/gst	Update to new promise API	2017-11-22 22:28:55 +10:00
sendrecv	webrtc-sendrecv: define GST_USE_UNSTABLE_API to avoid compiler warnings	2018-02-02 08:39:04 +00:00
signalling	Update Protocol.md	2017-10-29 04:08:45 +05:30
.gitignore	Add sendrecv implementation in js and gst webrtc	2017-10-21 20:02:19 +05:30
LICENSE	Initial commit	2017-10-21 19:43:01 +05:30
README.md	README: link to blog post, document multiparty example	2018-02-17 08:13:36 +05:30

GStreamer WebRTC demos

All demos use the same signalling server in the signalling/ directory

Build

The GStreamer WebRTC implementation has now been merged upstream, so all you need is the latest GStreamer git master, as of 2 February 2018 or later:

You can build these with either Autotools gst-uninstalled:

Or with Meson gst-build:

You may need to install the following packages using your package manager:

json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice)

Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:

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 the sources in the gst/ directory on your machine

$ 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

TODO: Port to Python and Rust.

$ gcc mp-webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o mp-webrtc-sendrecv

Run mp-webrtc-sendrecv --room-id=ID with ID as a room name. The peer will connect to the signalling server and setup a conference room.
Run this as many times as you like, each will spawn a peer that sends red noise and outputs the red noise it receives from other peers.
- To change what a peer sends, find the audiotestsrc element in the source and change the wave property.
- You can, of course, also replace audiotestsrc itself with autoaudiosrc (any platform) or pulsesink (on linux).
TODO: implement JS to do the same, derived from the JS for the sendrecv example.