mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-14 13:21:28 +00:00
123 lines
5.1 KiB
Markdown
123 lines
5.1 KiB
Markdown
# GStreamer WebRTC demos
|
|
|
|
All demos use the same signalling server in the `signalling/` directory
|
|
|
|
## Downloading GStreamer
|
|
|
|
The GStreamer WebRTC implementation has now been merged upstream, and is in the GStreamer 1.14 release. Binaries can be found here:
|
|
|
|
https://gstreamer.freedesktop.org/download/
|
|
|
|
## Building GStreamer from source
|
|
|
|
If you don't want to use the binaries provided by GStreamer or on your Linux distro, you can build GStreamer from source.
|
|
|
|
The easiest way to build the webrtc plugin and all the plugins it needs, is to [use Cerbero](https://gstreamer.freedesktop.org/documentation/installing/building-from-source-using-cerbero.html). These instructions should work out of the box for all platforms, including cross-compiling for iOS and Android.
|
|
|
|
## Building GStreamer manually from source
|
|
|
|
For hacking on the webrtc plugin, you may want to build manually using the git repositories:
|
|
|
|
- http://gitlab.freedesktop.org/gstreamer/gstreamer
|
|
- http://gitlab.freedesktop.org/gstreamer/gst-plugins-base
|
|
- http://gitlab.freedesktop.org/gstreamer/gst-plugins-good
|
|
- http://gitlab.freedesktop.org/gstreamer/gst-plugins-bad
|
|
- http://gitlab.freedesktop.org/libnice/libnice
|
|
|
|
Or with Meson gst-build:
|
|
|
|
https://gitlab.freedesktop.org/gstreamer/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, called gstreamer1.0-nice Debian)
|
|
|
|
### Ubuntu 18.04
|
|
|
|
Here are the commands for Ubuntu 18.04.
|
|
|
|
```
|
|
sudo apt-get install -y gstreamer1.0-tools gstreamer1.0-nice gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly gstreamer1.0-plugins-good libgstreamer1.0-dev git libglib2.0-dev libgstreamer-plugins-bad1.0-dev libsoup2.4-dev libjson-glib-dev
|
|
```
|
|
|
|
## Filing bugs
|
|
|
|
Please only file bugs about the demos here. Bugs about GStreamer's WebRTC implementation should be filed on the [GStreamer gitlab](https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/new).
|
|
|
|
You can also find us on IRC by joining #gstreamer @ FreeNode.
|
|
|
|
## Documentation
|
|
|
|
Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:
|
|
|
|
http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
|
|
|
|
## Examples
|
|
|
|
### Building
|
|
|
|
Most of the examples that require a build process can be built using the meson build system in the top-level gst-examples directory by using the following commands:
|
|
|
|
```console
|
|
cd /path/to/gst-examples
|
|
meson _builddir
|
|
ninja -C _builddir
|
|
```
|
|
|
|
Build outputs will be placed in the directory `_builddir`.
|
|
|
|
### 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
|
|
|
|
* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
|
|
|
|
#### Running the C version
|
|
|
|
* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
|
|
|
|
#### Running the Python version
|
|
|
|
* python3 -m pip install --user websockets
|
|
* run `python3 sendrecv/gst/webrtc_sendrecv.py ID` with the `id` from the browser. You will see state changes and an SDP exchange.
|
|
|
|
> The python version requires at least version 1.14.2 of gstreamer and its plugins.
|
|
|
|
#### Running the Rust version
|
|
|
|
* Install a recent Rust toolchain, e.g. via [rustup](https://rustup.rs/).
|
|
* Run `cargo build` for building the executable.
|
|
* Run `cargo run -- --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
|
|
|
|
With all versions, you will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app.
|
|
|
|
You can pass a --server argument to all versions, for example `--server=wss://127.0.0.1:8443`.
|
|
|
|
#### Running the Java version
|
|
|
|
`cd sendrecv/gst-java`\
|
|
`./gradlew build`\
|
|
`java -jar build/libs/gst-java.jar --peer-id=ID` with the `id` from the browser.
|
|
|
|
You can optionally specify the server URL too (it defaults to wss://webrtc.nirbheek.in:8443):
|
|
|
|
`java -jar build/libs/gst-java.jar --peer-id=1 --server=ws://localhost:8443`
|
|
|
|
### multiparty-sendrecv: Multiparty audio conference with N peers
|
|
|
|
* Run `_builddir/multiparty-sendrecv/gst/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.
|
|
|
|
### TODO: Selective Forwarding Unit (SFU) example
|
|
|
|
* Server routes media between peers
|
|
* Participant sends 1 stream, receives n-1 streams
|
|
|
|
### TODO: Multipoint Control Unit (MCU) example
|
|
|
|
* Server mixes media from all participants
|
|
* Participant sends 1 stream, receives 1 stream
|