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- Integrate with the build system. - Some README updates. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
123 lines
5.1 KiB
Markdown
123 lines
5.1 KiB
Markdown
# GStreamer WebRTC demos
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All demos use the same signalling server in the `signalling/` directory
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## Downloading GStreamer
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The GStreamer WebRTC implementation has now been merged upstream, and is in the GStreamer 1.14 release. Binaries can be found here:
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https://gstreamer.freedesktop.org/download/
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## Building GStreamer from source
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If you don't want to use the binaries provided by GStreamer or on your Linux distro, you can build GStreamer from source.
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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.
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## Building GStreamer manually from source
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For hacking on the webrtc plugin, you may want to build manually using the git repositories:
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- http://gitlab.freedesktop.org/gstreamer/gstreamer
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- http://gitlab.freedesktop.org/gstreamer/gst-plugins-base
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- http://gitlab.freedesktop.org/gstreamer/gst-plugins-good
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- http://gitlab.freedesktop.org/gstreamer/gst-plugins-bad
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- http://gitlab.freedesktop.org/libnice/libnice
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Or with Meson gst-build:
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https://gitlab.freedesktop.org/gstreamer/gst-build/
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You may need to install the following packages using your package manager:
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json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice, called gstreamer1.0-nice Debian)
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### Ubuntu 18.04
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Here are the commands for Ubuntu 18.04.
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```
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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
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```
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## Filing bugs
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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).
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You can also find us on IRC by joining #gstreamer @ FreeNode.
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## Documentation
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Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:
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http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
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## Examples
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### Building
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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:
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```console
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cd /path/to/gst-examples
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meson _builddir
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ninja -C _builddir
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```
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Build outputs will be placed in the directory `_builddir`.
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### sendrecv: Send and receive audio and video
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* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
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- The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
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* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
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#### Running the C version
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* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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#### Running the Python version
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* python3 -m pip install --user websockets
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* run `python3 sendrecv/gst/webrtc_sendrecv.py ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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> The python version requires at least version 1.14.2 of gstreamer and its plugins.
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#### Running the Rust version
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* Install a recent Rust toolchain, e.g. via [rustup](https://rustup.rs/).
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* Run `cargo build` for building the executable.
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* Run `cargo run -- --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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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.
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You can pass a --server argument to all versions, for example `--server=wss://127.0.0.1:8443`.
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#### Running the Java version
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`cd sendrecv/gst-java`\
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`./gradlew build`\
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`java -jar build/libs/gst-java.jar --peer-id=ID` with the `id` from the browser.
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You can optionally specify the server URL too (it defaults to wss://webrtc.nirbheek.in:8443):
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`java -jar build/libs/gst-java.jar --peer-id=1 --server=ws://localhost:8443`
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### multiparty-sendrecv: Multiparty audio conference with N peers
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* 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.
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* 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.
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- To change what a peer sends, find the `audiotestsrc` element in the source and change the `wave` property.
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- You can, of course, also replace `audiotestsrc` itself with `autoaudiosrc` (any platform) or `pulsesink` (on linux).
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* TODO: implement JS to do the same, derived from the JS for the `sendrecv` example.
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### TODO: Selective Forwarding Unit (SFU) example
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* Server routes media between peers
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* Participant sends 1 stream, receives n-1 streams
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### TODO: Multipoint Control Unit (MCU) example
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* Server mixes media from all participants
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* Participant sends 1 stream, receives 1 stream
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