webrtc: Add Janus video-room example

This Rust crate provides a program able to connect to a Janus instance using
WebSockets and send a live video stream to the videoroom plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/15>

4 files changed, 2017 insertions, 0 deletions

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new file mode 100644
index 0000000000..3cde9a5a99
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diff --git a/webrtc/janus/rust/Cargo.toml b/webrtc/janus/rust/Cargo.toml
new file mode 100644
index 0000000000..29c6aad29d
--- /dev/null
+++ b/webrtc/janus/rust/Cargo.toml
@@ -0,0 +1,25 @@
+[package]
+name = "janus-video-room"
+version = "0.1.0"
+authors = ["Philippe Normand <philn@igalia.com>"]
+edition = "2018"
+license = "LGPL"
+
+[dependencies]
+futures = "0.3"
+structopt = { version = "0.3", default-features = false }
+anyhow = "1"
+url = "2"
+rand = "0.7"
+async-tungstenite = { version = "0.7", features = ["gio-runtime"] }
+gst = { package = "gstreamer", version = "0.15", features = ["v1_14"] }
+gst-webrtc = { package = "gstreamer-webrtc", version = "0.15" }
+gst-sdp = { package = "gstreamer-sdp", version = "0.15", features = ["v1_14"] }
+serde = "1"
+serde_derive = "1"
+serde_json = "1.0.53"
+http = "0.2"
+glib = "0.9"
+gio = "0.8"
+log = "0.4.8"
+env_logger = "0.7.1"
diff --git a/webrtc/janus/rust/src/janus.rs b/webrtc/janus/rust/src/janus.rs
new file mode 100644
index 0000000000..e37adf80b2
--- /dev/null
+++ b/webrtc/janus/rust/src/janus.rs
@@ -0,0 +1,707 @@
+// GStreamer
+//
+// Copyright (C) 2018 maxmcd <max.t.mcdonnell@gmail.com>
+// Copyright (C) 2019 Sebastian Dröge <sebastian@centricular.com>
+// Copyright (C) 2020 Philippe Normand <philn@igalia.com>
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Library General Public
+// License as published by the Free Software Foundation; either
+// version 2 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+// Library General Public License for more details.
+//
+// You should have received a copy of the GNU Library General Public
+// License along with this library; if not, write to the
+// Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+// Boston, MA 02110-1301, USA.
+
+use {
+    anyhow::{anyhow, bail, Context},
+    async_tungstenite::{gio::connect_async, tungstenite},
+    futures::channel::mpsc,
+    futures::sink::{Sink, SinkExt},
+    futures::stream::{Stream, StreamExt},
+    gst::gst_element_error,
+    gst::prelude::*,
+    http::Request,
+    rand::prelude::*,
+    serde_derive::{Deserialize, Serialize},
+    serde_json::json,
+    std::sync::{Arc, Mutex, Weak},
+    structopt::StructOpt,
+    tungstenite::Message as WsMessage,
+};
+
+// upgrade weak reference or return
+#[macro_export]
+macro_rules! upgrade_weak {
+    ($x:ident, $r:expr) => {{
+        match $x.upgrade() {
+            Some(o) => o,
+            None => return $r,
+        }
+    }};
+    ($x:ident) => {
+        upgrade_weak!($x, ())
+    };
+}
+
+#[derive(Debug)]
+struct VideoParameter {
+    encoder: &'static str,
+    encoding_name: &'static str,
+    payloader: &'static str,
+}
+
+const VP8: VideoParameter = VideoParameter {
+    encoder: "vp8enc target-bitrate=100000 overshoot=25 undershoot=100 deadline=33000 keyframe-max-dist=1",
+    encoding_name: "VP8",
+    payloader: "rtpvp8pay picture-id-mode=2"
+};
+
+const H264: VideoParameter = VideoParameter {
+    encoder: "x264enc tune=zerolatency",
+    encoding_name: "H264",
+    payloader: "rtph264pay",
+};
+
+impl std::str::FromStr for VideoParameter {
+    type Err = anyhow::Error;
+
+    fn from_str(s: &str) -> Result<Self, Self::Err> {
+        match s {
+            "vp8" => Ok(VP8),
+            "h264" => Ok(H264),
+            _ => Err(anyhow!(
+                "Invalid video parameter: {}. Use either vp8 or h264",
+                s
+            )),
+        }
+    }
+}
+
+#[derive(Debug, StructOpt)]
+pub struct Args {
+    #[structopt(short, long, default_value = "wss://janus.conf.meetecho.com/ws:8989")]
+    server: String,
+    #[structopt(short, long, default_value = "1234")]
+    room_id: u32,
+    #[structopt(short, long, default_value = "1234")]
+    feed_id: u32,
+    #[structopt(short, long, default_value = "vp8")]
+    webrtc_video_codec: VideoParameter,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct Base {
+    janus: String,
+    transaction: Option<String>,
+    session_id: Option<i64>,
+    sender: Option<i64>,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct DataHolder {
+    id: i64,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct PluginDataHolder {
+    videoroom: String,
+    room: i64,
+    description: Option<String>,
+    id: Option<i64>,
+    configured: Option<String>,
+    video_codec: Option<String>,
+    unpublished: Option<i64>,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct PluginHolder {
+    plugin: String,
+    data: PluginDataHolder,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct IceHolder {
+    candidate: String,
+    #[serde(rename = "sdpMLineIndex")]
+    sdp_mline_index: u32,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct JsepHolder {
+    #[serde(rename = "type")]
+    type_: String,
+    sdp: Option<String>,
+    ice: Option<IceHolder>,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+struct JsonReply {
+    #[serde(flatten)]
+    base: Base,
+    data: Option<DataHolder>,
+    #[serde(rename = "plugindata")]
+    plugin_data: Option<PluginHolder>,
+    jsep: Option<JsepHolder>,
+}
+
+fn transaction_id() -> String {
+    thread_rng()
+        .sample_iter(&rand::distributions::Alphanumeric)
+        .take(30)
+        .collect()
+}
+
+// Strong reference to the state of one peer
+#[derive(Debug, Clone)]
+struct Peer(Arc<PeerInner>);
+
+// Weak reference to the state of one peer
+#[derive(Debug, Clone)]
+struct PeerWeak(Weak<PeerInner>);
+
+impl PeerWeak {
+    // Try upgrading a weak reference to a strong one
+    fn upgrade(&self) -> Option<Peer> {
+        self.0.upgrade().map(Peer)
+    }
+}
+
+// To be able to access the Peers's fields directly
+impl std::ops::Deref for Peer {
+    type Target = PeerInner;
+
+    fn deref(&self) -> &PeerInner {
+        &self.0
+    }
+}
+
+#[derive(Clone, Copy, Debug)]
+struct ConnectionHandle {
+    id: i64,
+    session_id: i64,
+}
+
+// Actual peer state
+#[derive(Debug)]
+struct PeerInner {
+    handle: ConnectionHandle,
+    bin: gst::Bin,
+    webrtcbin: gst::Element,
+    send_msg_tx: Arc<Mutex<mpsc::UnboundedSender<WsMessage>>>,
+}
+
+impl Peer {
+    // Downgrade the strong reference to a weak reference
+    fn downgrade(&self) -> PeerWeak {
+        PeerWeak(Arc::downgrade(&self.0))
+    }
+
+    // Whenever webrtcbin tells us that (re-)negotiation is needed, simply ask
+    // for a new offer SDP from webrtcbin without any customization and then
+    // asynchronously send it to the peer via the WebSocket connection
+    fn on_negotiation_needed(&self) -> Result<(), anyhow::Error> {
+        info!("starting negotiation with peer");
+
+        let peer_clone = self.downgrade();
+        let promise = gst::Promise::new_with_change_func(move |res| {
+            let s = res.expect("no answer");
+            let peer = upgrade_weak!(peer_clone);
+
+            if let Err(err) = peer.on_offer_created(&s.to_owned()) {
+                gst_element_error!(
+                    peer.bin,
+                    gst::LibraryError::Failed,
+                    ("Failed to send SDP offer: {:?}", err)
+                );
+            }
+        });
+
+        self.webrtcbin
+            .emit("create-offer", &[&None::<gst::Structure>, &promise])?;
+
+        Ok(())
+    }
+
+    // Once webrtcbin has create the offer SDP for us, handle it by sending it to the peer via the
+    // WebSocket connection
+    fn on_offer_created(&self, reply: &gst::Structure) -> Result<(), anyhow::Error> {
+        let offer = reply
+            .get_value("offer")?
+            .get::<gst_webrtc::WebRTCSessionDescription>()
+            .expect("Invalid argument")
+            .expect("Invalid offer");
+        self.webrtcbin
+            .emit("set-local-description", &[&offer, &None::<gst::Promise>])?;
+
+        info!("sending SDP offer to peer: {:?}", offer.get_sdp().as_text());
+
+        let transaction = transaction_id();
+        let sdp_data = offer.get_sdp().as_text()?;
+        let msg = WsMessage::Text(
+            json!({
+                 "janus": "message",
+                 "transaction": transaction,
+                 "session_id": self.handle.session_id,
+                 "handle_id": self.handle.id,
+                 "body": {
+                     "request": "publish",
+                     "audio": true,
+                     "video": true,
+                 },
+                 "jsep": {
+                     "sdp": sdp_data,
+                     "trickle": true,
+                     "type": "offer"
+                 }
+            })
+            .to_string(),
+        );
+        self.send_msg_tx
+            .lock()
+            .expect("Invalid message sender")
+            .unbounded_send(msg)
+            .with_context(|| "Failed to send SDP offer".to_string())?;
+
+        Ok(())
+    }
+
+    // Once webrtcbin has create the answer SDP for us, handle it by sending it to the peer via the
+    // WebSocket connection
+    fn on_answer_created(&self, reply: &gst::Structure) -> Result<(), anyhow::Error> {
+        let answer = reply
+            .get_value("answer")?
+            .get::<gst_webrtc::WebRTCSessionDescription>()
+            .expect("Invalid argument")
+            .expect("Invalid answer");
+        self.webrtcbin
+            .emit("set-local-description", &[&answer, &None::<gst::Promise>])?;
+
+        info!(
+            "sending SDP answer to peer: {:?}",
+            answer.get_sdp().as_text()
+        );
+
+        Ok(())
+    }
+
+    // Handle incoming SDP answers from the peer
+    fn handle_sdp(&self, type_: &str, sdp: &str) -> Result<(), anyhow::Error> {
+        if type_ == "answer" {
+            info!("Received answer:\n{}\n", sdp);
+
+            let ret = gst_sdp::SDPMessage::parse_buffer(sdp.as_bytes())
+                .map_err(|_| anyhow!("Failed to parse SDP answer"))?;
+            let answer =
+                gst_webrtc::WebRTCSessionDescription::new(gst_webrtc::WebRTCSDPType::Answer, ret);
+
+            self.webrtcbin
+                .emit("set-remote-description", &[&answer, &None::<gst::Promise>])?;
+
+            Ok(())
+        } else if type_ == "offer" {
+            info!("Received offer:\n{}\n", sdp);
+
+            let ret = gst_sdp::SDPMessage::parse_buffer(sdp.as_bytes())
+                .map_err(|_| anyhow!("Failed to parse SDP offer"))?;
+
+            // And then asynchronously start our pipeline and do the next steps. The
+            // pipeline needs to be started before we can create an answer
+            let peer_clone = self.downgrade();
+            self.bin.call_async(move |_pipeline| {
+                let peer = upgrade_weak!(peer_clone);
+
+                let offer = gst_webrtc::WebRTCSessionDescription::new(
+                    gst_webrtc::WebRTCSDPType::Offer,
+                    ret,
+                );
+
+                peer.0
+                    .webrtcbin
+                    .emit("set-remote-description", &[&offer, &None::<gst::Promise>])
+                    .expect("Unable to set remote description");
+
+                let peer_clone = peer.downgrade();
+                let promise = gst::Promise::new_with_change_func(move |reply| {
+                    let s = reply.expect("No answer");
+                    let peer = upgrade_weak!(peer_clone);
+
+                    if let Err(err) = peer.on_answer_created(&s.to_owned()) {
+                        gst_element_error!(
+                            peer.bin,
+                            gst::LibraryError::Failed,
+                            ("Failed to send SDP answer: {:?}", err)
+                        );
+                    }
+                });
+
+                peer.0
+                    .webrtcbin
+                    .emit("create-answer", &[&None::<gst::Structure>, &promise])
+                    .expect("Unable to create answer");
+            });
+
+            Ok(())
+        } else {
+            bail!("Sdp type is not \"answer\" but \"{}\"", type_)
+        }
+    }
+
+    // Handle incoming ICE candidates from the peer by passing them to webrtcbin
+    fn handle_ice(&self, sdp_mline_index: u32, candidate: &str) -> Result<(), anyhow::Error> {
+        info!(
+            "Received remote ice-candidate {} {}",
+            sdp_mline_index, candidate
+        );
+        self.webrtcbin
+            .emit("add-ice-candidate", &[&sdp_mline_index, &candidate])?;
+
+        Ok(())
+    }
+
+    // Asynchronously send ICE candidates to the peer via the WebSocket connection as a JSON
+    // message
+    fn on_ice_candidate(&self, mlineindex: u32, candidate: String) -> Result<(), anyhow::Error> {
+        let transaction = transaction_id();
+        info!("Sending ICE {} {}", mlineindex, &candidate);
+        let msg = WsMessage::Text(
+            json!({
+                "janus": "trickle",
+                "transaction": transaction,
+                "session_id": self.handle.session_id,
+                "handle_id": self.handle.id,
+                "candidate": {
+                    "candidate": candidate,
+                    "sdpMLineIndex": mlineindex
+                },
+            })
+            .to_string(),
+        );
+        self.send_msg_tx
+            .lock()
+            .expect("Invalid message sender")
+            .unbounded_send(msg)
+            .with_context(|| "Failed to send ICE candidate".to_string())?;
+
+        Ok(())
+    }
+}
+
+// At least shut down the bin here if it didn't happen so far
+impl Drop for PeerInner {
+    fn drop(&mut self) {
+        let _ = self.bin.set_state(gst::State::Null);
+    }
+}
+
+type WsStream =
+    std::pin::Pin<Box<dyn Stream<Item = Result<WsMessage, tungstenite::error::Error>> + Send>>;
+type WsSink = std::pin::Pin<Box<dyn Sink<WsMessage, Error = tungstenite::error::Error> + Send>>;
+
+pub struct JanusGateway {
+    ws_stream: Option<WsStream>,
+    ws_sink: Option<WsSink>,
+    handle: ConnectionHandle,
+    peer: Mutex<Peer>,
+    send_ws_msg_rx: Option<mpsc::UnboundedReceiver<WsMessage>>,
+}
+
+impl JanusGateway {
+    pub async fn new(pipeline: gst::Bin) -> Result<Self, anyhow::Error> {
+        let args = Args::from_args();
+        let request = Request::builder()
+            .uri(&args.server)
+            .header("Sec-WebSocket-Protocol", "janus-protocol")
+            .body(())?;
+
+        let (mut ws, _) = connect_async(request).await?;
+
+        let transaction = transaction_id();
+        let msg = WsMessage::Text(
+            json!({
+                "janus": "create",
+                "transaction": transaction,
+            })
+            .to_string(),
+        );
+        ws.send(msg).await?;
+
+        let msg = ws
+            .next()
+            .await
+            .ok_or_else(|| anyhow!("didn't receive anything"))??;
+        let payload = msg.to_text()?;
+        let json_msg: JsonReply = serde_json::from_str(payload)?;
+        assert_eq!(json_msg.base.janus, "success");
+        assert_eq!(json_msg.base.transaction, Some(transaction));
+        let session_id = json_msg.data.expect("no session id").id;
+
+        let transaction = transaction_id();
+        let msg = WsMessage::Text(
+            json!({
+                "janus": "attach",
+                "transaction": transaction,
+                "plugin": "janus.plugin.videoroom",
+                "session_id": session_id,
+            })
+            .to_string(),
+        );
+        ws.send(msg).await?;
+
+        let msg = ws
+            .next()
+            .await
+            .ok_or_else(|| anyhow!("didn't receive anything"))??;
+        let payload = msg.to_text()?;
+        let json_msg: JsonReply = serde_json::from_str(payload)?;
+        assert_eq!(json_msg.base.janus, "success");
+        assert_eq!(json_msg.base.transaction, Some(transaction));
+        let handle = json_msg.data.expect("no session id").id;
+
+        let transaction = transaction_id();
+        let msg = WsMessage::Text(
+            json!({
+                "janus": "message",
+                "transaction": transaction,
+                "session_id": session_id,
+                "handle_id": handle,
+                "body": {
+                    "request": "join",
+                    "ptype": "publisher",
+                    "room": args.room_id,
+                    "id": args.feed_id,
+                },
+            })
+            .to_string(),
+        );
+        ws.send(msg).await?;
+
+        let webrtcbin = pipeline
+            .get_by_name("webrtcbin")
+            .expect("can't find webrtcbin");
+
+        let webrtc_codec = &args.webrtc_video_codec;
+        let bin_description = &format!(
+            "{encoder} name=encoder ! {payloader} ! queue ! capsfilter name=webrtc-vsink caps=\"application/x-rtp,media=video,encoding-name={encoding_name},payload=96\"",
+            encoder=webrtc_codec.encoder, payloader=webrtc_codec.payloader,
+            encoding_name=webrtc_codec.encoding_name
+        );
+
+        let encode_bin = gst::parse_bin_from_description(bin_description, false)?;
+        encode_bin.set_name("encode-bin")?;
+
+        pipeline.add(&encode_bin).expect("Failed to add encode bin");
+
+        let video_queue = pipeline.get_by_name("vqueue").expect("No vqueue found");
+        let encoder = encode_bin.get_by_name("encoder").expect("No encoder");
+
+        let srcpad = video_queue
+            .get_static_pad("src")
+            .expect("Failed to get video queue src pad");
+        let sinkpad = encoder
+            .get_static_pad("sink")
+            .expect("Failed to get sink pad from encoder");
+
+        if let Ok(video_ghost_pad) = gst::GhostPad::new(Some("video_sink"), &sinkpad) {
+            encode_bin.add_pad(&video_ghost_pad)?;
+            srcpad.link(&video_ghost_pad)?;
+        }
+
+        let sinkpad2 = webrtcbin
+            .get_request_pad("sink_%u")
+            .expect("Unable to request outgoing webrtcbin pad");
+        let vsink = encode_bin
+            .get_by_name("webrtc-vsink")
+            .expect("No webrtc-vsink found");
+        let srcpad = vsink
+            .get_static_pad("src")
+            .expect("Element without src pad");
+        if let Ok(webrtc_ghost_pad) = gst::GhostPad::new(Some("webrtc_video_src"), &srcpad) {
+            encode_bin.add_pad(&webrtc_ghost_pad)?;
+            webrtc_ghost_pad.link(&sinkpad2)?;
+        }
+
+        if let Ok(transceiver) = webrtcbin.emit("get-transceiver", &[&0.to_value()]) {
+            if let Some(t) = transceiver {
+                if let Ok(obj) = t.get::<glib::Object>() {
+                    obj.expect("Invalid transceiver")
+                        .set_property("do-nack", &true.to_value())?;
+                }
+            }
+        }
+
+        let (send_ws_msg_tx, send_ws_msg_rx) = mpsc::unbounded::<WsMessage>();
+
+        let connection_handle = ConnectionHandle {
+            id: handle,
+            session_id,
+        };
+
+        let peer = Peer(Arc::new(PeerInner {
+            handle: connection_handle,
+            bin: pipeline,
+            webrtcbin,
+            send_msg_tx: Arc::new(Mutex::new(send_ws_msg_tx)),
+        }));
+
+        // Connect to on-negotiation-needed to handle sending an Offer
+        let peer_clone = peer.downgrade();
+        peer.webrtcbin
+            .connect("on-negotiation-needed", false, move |_| {
+                let peer = upgrade_weak!(peer_clone, None);
+                if let Err(err) = peer.on_negotiation_needed() {
+                    gst_element_error!(
+                        peer.bin,
+                        gst::LibraryError::Failed,
+                        ("Failed to negotiate: {:?}", err)
+                    );
+                }
+
+                None
+            })?;
+
+        // Whenever there is a new ICE candidate, send it to the peer
+        let peer_clone = peer.downgrade();
+        peer.webrtcbin
+            .connect("on-ice-candidate", false, move |values| {
+                let mlineindex = values[1]
+                    .get::<u32>()
+                    .expect("Invalid argument")
+                    .expect("Invalid type");
+                let candidate = values[2]
+                    .get::<String>()
+                    .expect("Invalid argument")
+                    .expect("Invalid type");
+
+                let peer = upgrade_weak!(peer_clone, None);
+                if let Err(err) = peer.on_ice_candidate(mlineindex, candidate) {
+                    gst_element_error!(
+                        peer.bin,
+                        gst::LibraryError::Failed,
+                        ("Failed to send ICE candidate: {:?}", err)
+                    );
+                }
+
+                None
+            })?;
+
+        // Split the websocket into the Sink and Stream
+        let (ws_sink, ws_stream) = ws.split();
+
+        Ok(Self {
+            ws_stream: Some(ws_stream.boxed()),
+            ws_sink: Some(Box::pin(ws_sink)),
+            handle: connection_handle,
+            peer: Mutex::new(peer),
+            send_ws_msg_rx: Some(send_ws_msg_rx),
+        })
+    }
+
+    pub async fn run(&mut self) -> Result<(), anyhow::Error> {
+        if let Some(ws_stream) = self.ws_stream.take() {
+            // Fuse the Stream, required for the select macro
+            let mut ws_stream = ws_stream.fuse();
+
+            // Channel for outgoing WebSocket messages from other threads
+            let send_ws_msg_rx = self
+                .send_ws_msg_rx
+                .take()
+                .expect("Invalid message receiver");
+            let mut send_ws_msg_rx = send_ws_msg_rx.fuse();
+
+            let timer = glib::interval_stream(10_000);
+            let mut timer_fuse = timer.fuse();
+
+            let mut sink = self.ws_sink.take().expect("Invalid websocket sink");
+            loop {
+                let ws_msg = futures::select! {
+                    // Handle the WebSocket messages here
+                    ws_msg = ws_stream.select_next_some() => {
+                        match ws_msg? {
+                            WsMessage::Close(_) => {
+                                info!("peer disconnected");
+                                break
+                            },
+                            WsMessage::Ping(data) => Some(WsMessage::Pong(data)),
+                            WsMessage::Pong(_) => None,
+                            WsMessage::Binary(_) => None,
+                            WsMessage::Text(text) => {
+                                if let Err(err) = self.handle_websocket_message(&text) {
+                                    error!("Failed to parse message: {} ... error: {}", &text, err);
+                                }
+                                None
+                            },
+                        }
+                    },
+                    // Handle WebSocket messages we created asynchronously
+                    // to send them out now
+                    ws_msg = send_ws_msg_rx.select_next_some() => Some(ws_msg),
+
+                    // Handle keepalive ticks, fired every 10 seconds
+                    ws_msg = timer_fuse.select_next_some() => {
+                        let transaction = transaction_id();
+                        Some(WsMessage::Text(
+                            json!({
+                                "janus": "keepalive",
+                                "transaction": transaction,
+                                "handle_id": self.handle.id,
+                                "session_id": self.handle.session_id,
+                            }).to_string(),
+                        ))
+                    },
+                    // Once we're done, break the loop and return
+                    complete => break,
+                };
+
+                // If there's a message to send out, do so now
+                if let Some(ws_msg) = ws_msg {
+                    sink.send(ws_msg).await?;
+                }
+            }
+        }
+        Ok(())
+    }
+
+    fn handle_jsep(&self, jsep: &JsepHolder) -> Result<(), anyhow::Error> {
+        if let Some(sdp) = &jsep.sdp {
+            assert_eq!(jsep.type_, "answer");
+            let peer = self.peer.lock().expect("Invalid peer");
+            return peer.handle_sdp(&jsep.type_, &sdp);
+        } else if let Some(ice) = &jsep.ice {
+            let peer = self.peer.lock().expect("Invalid peer");
+            return peer.handle_ice(ice.sdp_mline_index, &ice.candidate);
+        }
+
+        Ok(())
+    }
+
+    // Handle WebSocket messages, both our own as well as WebSocket protocol messages
+    fn handle_websocket_message(&self, msg: &str) -> Result<(), anyhow::Error> {
+        trace!("Incoming raw message: {}", msg);
+        let json_msg: JsonReply = serde_json::from_str(msg)?;
+        let payload_type = &json_msg.base.janus;
+        if payload_type == "ack" {
+            trace!(
+                "Ack transaction {:#?}, sessionId {:#?}",
+                json_msg.base.transaction,
+                json_msg.base.session_id
+            );
+        } else {
+            debug!("Incoming JSON WebSocket message: {:#?}", json_msg);
+        }
+        if payload_type == "event" {
+            if let Some(_plugin_data) = json_msg.plugin_data {
+                if let Some(jsep) = json_msg.jsep {
+                    return self.handle_jsep(&jsep);
+                }
+            }
+        }
+        Ok(())
+    }
+}
diff --git a/webrtc/janus/rust/src/main.rs b/webrtc/janus/rust/src/main.rs
new file mode 100644
index 0000000000..eace4d8fcb
--- /dev/null
+++ b/webrtc/janus/rust/src/main.rs
@@ -0,0 +1,186 @@
+// GStreamer
+//
+// Copyright (C) 2019 Sebastian Dröge <sebastian@centricular.com>
+// Copyright (C) 2020 Philippe Normand <philn@igalia.com>
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Library General Public
+// License as published by the Free Software Foundation; either
+// version 2 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+// Library General Public License for more details.
+//
+// You should have received a copy of the GNU Library General Public
+// License along with this library; if not, write to the
+// Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+// Boston, MA 02110-1301, USA.
+
+#![recursion_limit = "256"]
+
+use anyhow::bail;
+use gst::gst_element_error;
+use gst::prelude::*;
+use std::sync::{Arc, Weak};
+
+#[macro_use]
+extern crate log;
+
+mod janus;
+
+// Strong reference to our application state
+#[derive(Debug, Clone)]
+struct App(Arc<AppInner>);
+
+// Weak reference to our application state
+#[derive(Debug, Clone)]
+struct AppWeak(Weak<AppInner>);
+
+// Actual application state
+#[derive(Debug)]
+struct AppInner {
+    pipeline: gst::Pipeline,
+}
+
+// To be able to access the App's fields directly
+impl std::ops::Deref for App {
+    type Target = AppInner;
+
+    fn deref(&self) -> &AppInner {
+        &self.0
+    }
+}
+
+impl AppWeak {
+    // Try upgrading a weak reference to a strong one
+    fn upgrade(&self) -> Option<App> {
+        self.0.upgrade().map(App)
+    }
+}
+
+impl App {
+    // Downgrade the strong reference to a weak reference
+    fn downgrade(&self) -> AppWeak {
+        AppWeak(Arc::downgrade(&self.0))
+    }
+
+    fn new() -> Result<Self, anyhow::Error> {
+        let pipeline = gst::parse_launch(
+            &"webrtcbin name=webrtcbin stun-server=stun://stun.l.google.com:19302 \
+             videotestsrc pattern=ball ! videoconvert ! queue name=vqueue"
+                .to_string(),
+        )?;
+
+        let pipeline = pipeline
+            .downcast::<gst::Pipeline>()
+            .expect("Couldn't downcast pipeline");
+
+        let bus = pipeline.get_bus().unwrap();
+        let app = App(Arc::new(AppInner { pipeline }));
+
+        let app_weak = app.downgrade();
+        bus.add_watch_local(move |_bus, msg| {
+            let app = upgrade_weak!(app_weak, glib::Continue(false));
+
+            if app.handle_pipeline_message(msg).is_err() {
+                return glib::Continue(false);
+            }
+            glib::Continue(true)
+        })
+        .expect("Unable to add bus watch");
+        Ok(app)
+    }
+
+    fn handle_pipeline_message(&self, message: &gst::Message) -> Result<(), anyhow::Error> {
+        use gst::message::MessageView;
+
+        match message.view() {
+            MessageView::Error(err) => bail!(
+                "Error from element {}: {} ({})",
+                err.get_src()
+                    .map(|s| String::from(s.get_path_string()))
+                    .unwrap_or_else(|| String::from("None")),
+                err.get_error(),
+                err.get_debug().unwrap_or_else(|| String::from("None")),
+            ),
+            MessageView::Warning(warning) => {
+                println!("Warning: \"{}\"", warning.get_debug().unwrap());
+            }
+            _ => (),
+        }
+        Ok(())
+    }
+
+    pub async fn run(&self) -> Result<(), anyhow::Error> {
+        let bin = self.pipeline.clone().upcast::<gst::Bin>();
+        let mut gw = janus::JanusGateway::new(bin).await?;
+
+        // Asynchronously set the pipeline to Playing
+        self.pipeline.call_async(|pipeline| {
+            // If this fails, post an error on the bus so we exit
+            if pipeline.set_state(gst::State::Playing).is_err() {
+                gst_element_error!(
+                    pipeline,
+                    gst::LibraryError::Failed,
+                    ("Failed to set pipeline to Playing")
+                );
+            }
+        });
+
+        gw.run().await?;
+        Ok(())
+    }
+}
+
+// Make sure to shut down the pipeline when it goes out of scope
+// to release any system resources
+impl Drop for AppInner {
+    fn drop(&mut self) {
+        let _ = self.pipeline.set_state(gst::State::Null);
+    }
+}
+
+// Check if all GStreamer plugins we require are available
+fn check_plugins() -> Result<(), anyhow::Error> {
+    let needed = [
+        "videotestsrc",
+        "videoconvert",
+        "autodetect",
+        "vpx",
+        "webrtc",
+        "nice",
+        "dtls",
+        "srtp",
+        "rtpmanager",
+        "rtp",
+    ];
+
+    let registry = gst::Registry::get();
+    let missing = needed
+        .iter()
+        .filter(|n| registry.find_plugin(n).is_none())
+        .cloned()
+        .collect::<Vec<_>>();
+
+    if !missing.is_empty() {
+        bail!("Missing plugins: {:?}", missing);
+    } else {
+        Ok(())
+    }
+}
+
+async fn async_main() -> Result<(), anyhow::Error> {
+    gst::init()?;
+    check_plugins()?;
+    let app = App::new()?;
+    app.run().await?;
+    Ok(())
+}
+
+fn main() -> Result<(), anyhow::Error> {
+    env_logger::init();
+    let main_context = glib::MainContext::default();
+    main_context.block_on(async_main())
+}