# gst-meet: Integrate Jitsi Meet conferences with GStreamer pipelines Note: gst-meet is in an **alpha** state and is under active development. The command-line options and the lib-gst-meet API are subject to change, and some important features (simulcast, RTX, TCC) are not yet fully functional. gst-meet provides a library and tool for integrating Jitsi Meet conferences with GStreamer pipelines. You can pipe audio and video into a conference as a participant, and pipe out other participants' audio and video streams. Thanks to GStreamer's flexibility and wide range of plugins, this enables many new possibilities. ## Dependencies * `glib` * `gstreamer` 1.19 or later. Most distributions do not package this version yet, so you may need to build from source * `gst-plugins-bad` (same version as `gstreamer`) and any other plugins that you want to use in your pipelines * `libnice` * `pkg-config` * A TLS library (usually installed by default, but you may need to install `openssl-dev` or similar on minimal Linux distributions) * A Rust toolchain ([rustup](https://rustup.rs/) is the easiest way to install one) ## Installation `cargo install --force gst-meet` (`--force` will upgrade `gst-meet` if you have already installed it.) To integrate gst-meet into your own application, add a Cargo dependency on `lib-gst-meet`. ## Docker A `Dockerfile` is provided that uses AVStack-built Alpine APKs for gstreamer 1.19.x. ## Nix For nix users, a `shell.nix` is provided. Within the repository, run `nix-shell --pure` to get a shell with access to all needed dependencies (and nothing else). ## Development Install the dependencies described above, along with their `-dev` packages if your distribution uses them. `cargo build` should then successfully build the libraries and `gst-meet` binary. ## Pipeline Structure You can pass two different pipeline fragments to gst-meet. `--send-pipeline` is for sending audio and video. If it contains an element named `audio`, this audio will be streamed to the conference. The audio codec must be 48kHz Opus. If it contains an element named `video`, this video will be streamed to the conference. The video codec must match the codec passed to `--video-codec`, which is VP8 by default. `--recv-pipeline-participant-template` is for receiving audio and video from other participants. This pipeline will be created once for each other participant in the conference. If it contains an element named `audio`, the participant's audio (48kHz Opus) will be sent to that element. If it contains an element named `video`, the participant's video (encoded with the codec selected by `--video-codec`) will be sent to that element. The strings `{jid}`, `{jid_user}`, `{participant_id}` and `{nick}` are replaced in the template with the participant's full JID, user part, MUC JID resource part (a.k.a. participant/occupant ID) and nickname respectively. ## Examples A few examples of `gst-meet` usage are below. The GStreamer reference provides full details on available pipeline elements. `gst-meet --help` lists full usage information. Stream an Opus audio file to the conference. This is very efficient; the Opus data in the file is streamed directly without transcoding: ``` gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \ --xmpp-domain=your.jitsi.domain \ --room-name=roomname \ --send-pipeline="filesrc location=sample.opus ! queue ! oggdemux name=audio" ``` Stream a FLAC audio file to the conference, transcoding it to Opus: ``` gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \ --xmpp-domain=your.jitsi.domain \ --room-name=roomname \ --send-pipeline="filesrc location=shake-it-off.flac ! queue ! flacdec ! audioconvert ! audioresample ! opusenc name=audio" ``` Stream a .webm file containing VP8 video and Vorbis audio to the conference. This pipeline passes the VP8 stream through efficiently without transcoding, and transcodes the audio from Vorbis to Opus: ``` gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \ --xmpp-domain=your.jitsi.domain \ --room-name=roomname \ --send-pipeline="filesrc location=big-buck-bunny_trailer.webm ! queue ! matroskademux name=demuxer demuxer.video_0 ! queue name=video demuxer.audio_0 ! queue ! vorbisdec ! audioconvert ! audioresample ! opusenc name=audio" ``` Stream the default video & audio inputs to the conference, encoding as VP8 and Opus, composite incoming video streams and play back incoming audio (a very basic, but completely native, Jitsi Meet conference!): ``` gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \ --xmpp-domain=your.jitsi.domain \ --room-name=roomname \ --send-pipeline="autovideosrc ! queue ! videoconvert ! vp8enc buffer-size=1000 deadline=1 name=video autoaudiosrc ! queue ! audioconvert ! audioresample ! opusenc name=audio" \ --recv-pipeline-participant-template="opusdec name=audio ! autoaudiosink vp8dec name=video ! videoconvert ! autovideosink" ``` Record a .webm file for each other participant, containing VP8 video and Opus audio, without needing to do any transcoding: ``` gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \ --xmpp-domain=your.jitsi.domain \ --room-name=roomname \ --recv-pipeline-participant-template="webmmux name=muxer ! filesink location={participant_id}.webm opusparse name=audio ! muxer.audio_0 capsfilter caps=video/x-vp8 name=video ! muxer.video_0" ``` Play a YouTube video in the conference. By requesting Opus audio and VP9 video from YouTube, and setting the Jitsi Meet video codec to VP9, no transcoding is necessary: ``` YOUTUBE_URL="https://www.youtube.com/watch?v=vjV_2Ri2rfE" gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \ --xmpp-domain=your.jitsi.domain \ --room-name=roomname \ --video-codec=vp9 \ --send-pipeline="curlhttpsrc location=\"$(youtube-dl -g $YOUTUBE_URL -f 'bestaudio[acodec=opus]')\" ! queue ! matroskademux name=audiodemux curlhttpsrc location=\"$(youtube-dl -g $YOUTUBE_URL -f 'bestvideo[vcodec=vp9]')\" ! queue ! matroskademux name=videodemux audiodemux.audio_0 ! queue ! clocksync name=audio videodemux.video_0 ! queue ! clocksync name=video" ``` ## Feature flags By default, the system native TLS library is used. This can be turned off by passing `--no-default-features` to Cargo, and one of the following features can be enabled: ``` tls-native (the default) link to the system native TLS library tls-native-vendored automatically build a copy of OpenSSL and statically link to it tls-rustls use rustls for TLS (note that rustls may not accept some server certificates) ``` These options only affect the TLS library used for the WebSocket connections (to the XMPP server and to the JVB). Gstreamer uses its own choice of TLS library for its elements, including DTLS-SRTP (the media streams). ## Debugging It can sometimes be tricky to get GStreamer pipeline syntax and structure correct. To help with this, you can try setting the `GST_DEBUG` environment variable (for example, `3` is modestly verbose, while `6` produces copious per-packet output). You can also set `GST_DEBUG_DUMP_DOT_DIR` to the relative path to a directory (which must already exist). `.dot` files containing the pipeline graph will be saved to this directory, and can be converted to `.png` with the `dot` tool from GraphViz; for example `dot filename.dot -Tpng > filename.png`. ## License `gst-meet`, `lib-gst-meet`, `nice` and `nice-sys` are licensed under either of * Apache License, Version 2.0, ([LICENSE-APACHE](LICENSE-APACHE) or http://www.apache.org/licenses/LICENSE-2.0) * MIT license ([LICENSE-MIT](LICENSE-MIT) or http://opensource.org/licenses/MIT) at your option. The dependency `xmpp-parsers` is licensed under the Mozilla Public License, Version 2.0, https://www.mozilla.org/en-US/MPL/2.0/ The dependency `gstreamer` is licensed under the GNU Lesser General Public License, Version 2.1, https://www.gnu.org/licenses/old-licenses/lgpl-2.1.en.html. ## Contribution Any kinds of contributions are welcome as a pull request. Unless you explicitly state otherwise, any contribution intentionally submitted for inclusion in these crates by you, as defined in the Apache-2.0 license, shall be dual licensed as above, without any additional terms or conditions. ## Acknowledgements `gst-meet` development is sponsored by [AVStack](https://www.avstack.io/). We provide globally-distributed, scalable, managed Jitsi Meet backends.