diff --git a/docs/plugins/gst_plugins_cache.json b/docs/plugins/gst_plugins_cache.json
index f3837d0d..ef0f4a11 100644
--- a/docs/plugins/gst_plugins_cache.json
+++ b/docs/plugins/gst_plugins_cache.json
@@ -7319,6 +7319,38 @@
                 },
                 "rank": "none"
             },
+            "livekitwebrtcsrc": {
+                "author": "Jordan Yelloz <jordan.yelloz@collabora.com>",
+                "description": "WebRTC source with LiveKit signaller",
+                "hierarchy": [
+                    "GstLiveKitWebRTCSrc",
+                    "GstBaseWebRTCSrc",
+                    "GstBin",
+                    "GstElement",
+                    "GstObject",
+                    "GInitiallyUnowned",
+                    "GObject"
+                ],
+                "interfaces": [
+                    "GstChildProxy"
+                ],
+                "klass": "Source/Network/WebRTC",
+                "pad-templates": {
+                    "audio_%%u": {
+                        "caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
+                        "direction": "src",
+                        "presence": "sometimes",
+                        "type": "GstWebRTCSrcPad"
+                    },
+                    "video_%%u": {
+                        "caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
+                        "direction": "src",
+                        "presence": "sometimes",
+                        "type": "GstWebRTCSrcPad"
+                    }
+                },
+                "rank": "none"
+            },
             "webrtcsink": {
                 "author": "Mathieu Duponchelle <mathieu@centricular.com>",
                 "description": "WebRTC sink with custom protocol signaller",
diff --git a/net/webrtc/src/webrtcsrc/imp.rs b/net/webrtc/src/webrtcsrc/imp.rs
index b978a0b4..8a152aa7 100644
--- a/net/webrtc/src/webrtcsrc/imp.rs
+++ b/net/webrtc/src/webrtcsrc/imp.rs
@@ -2,6 +2,7 @@
 
 use gst::prelude::*;
 
+use crate::livekit_signaller::LiveKitSignaller;
 use crate::signaller::{prelude::*, Signallable, Signaller};
 use crate::utils::{Codec, Codecs, NavigationEvent, AUDIO_CAPS, RTP_CAPS, VIDEO_CAPS};
 use crate::webrtcsrc::WebRTCSrcPad;
@@ -1290,3 +1291,44 @@ impl ObjectSubclass for WhipServerSrc {
     type Type = super::WhipServerSrc;
     type ParentType = super::BaseWebRTCSrc;
 }
+
+#[derive(Default)]
+pub struct LiveKitWebRTCSrc;
+
+impl ObjectImpl for LiveKitWebRTCSrc {
+    fn constructed(&self) {
+        self.parent_constructed();
+        let element = self.obj();
+        let ws = element.upcast_ref::<super::BaseWebRTCSrc>().imp();
+
+        let _ = ws.set_signaller(LiveKitSignaller::new_consumer().upcast());
+    }
+}
+
+impl GstObjectImpl for LiveKitWebRTCSrc {}
+
+impl BinImpl for LiveKitWebRTCSrc {}
+
+impl ElementImpl for LiveKitWebRTCSrc {
+    fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
+        static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
+            gst::subclass::ElementMetadata::new(
+                "LiveKitWebRTCSrc",
+                "Source/Network/WebRTC",
+                "WebRTC source with LiveKit signaller",
+                "Jordan Yelloz <jordan.yelloz@collabora.com>",
+            )
+        });
+
+        Some(&*ELEMENT_METADATA)
+    }
+}
+
+impl BaseWebRTCSrcImpl for LiveKitWebRTCSrc {}
+
+#[glib::object_subclass]
+impl ObjectSubclass for LiveKitWebRTCSrc {
+    const NAME: &'static str = "GstLiveKitWebRTCSrc";
+    type Type = super::LiveKitWebRTCSrc;
+    type ParentType = super::BaseWebRTCSrc;
+}
diff --git a/net/webrtc/src/webrtcsrc/mod.rs b/net/webrtc/src/webrtcsrc/mod.rs
index 45bef4ba..6b57f607 100644
--- a/net/webrtc/src/webrtcsrc/mod.rs
+++ b/net/webrtc/src/webrtcsrc/mod.rs
@@ -53,6 +53,10 @@ glib::wrapper! {
     pub struct WhipServerSrc(ObjectSubclass<imp::WhipServerSrc>) @extends BaseWebRTCSrc, gst::Bin, gst::Element, gst::Object, @implements gst::URIHandler, gst::ChildProxy;
 }
 
+glib::wrapper! {
+    pub struct LiveKitWebRTCSrc(ObjectSubclass<imp::LiveKitWebRTCSrc>) @extends BaseWebRTCSrc, gst::Bin, gst::Element, gst::Object, @implements gst::URIHandler, gst::ChildProxy;
+}
+
 glib::wrapper! {
     pub struct WebRTCSrcPad(ObjectSubclass<pad::WebRTCSrcPad>) @extends gst::GhostPad, gst::ProxyPad, gst::Pad, gst::Object;
 }
@@ -76,5 +80,60 @@ pub fn register(plugin: Option<&gst::Plugin>) -> Result<(), glib::BoolError> {
         WhipServerSrc::static_type(),
     )?;
 
+    /**
+     * element-livekitwebrtcsrc:
+     *
+     * The `livekitwebrtcsrc` plays streams from a LiveKit room.
+     *
+     * The element can either subscribe to the streams published by a single
+     * peer in the room using the same `signaller::producer-peer-id` child
+     * property that other webrtcsrc elements use or auto-subscribe to all peers
+     * in a room by not specifying anything for that property. When in
+     * auto-subscribe mode, you can use
+     * `signaller::excluded-producer-peer-ids=<a,b,c>` to ignore peers `a`, `b`,
+     * and `c` while subscribing to all other members of the room.
+     *
+     * ## Sample Pipeline
+     *
+     * First, start the livekit server with the `--dev` flag to enable the test credentials.
+     *
+     * Next, publish a stream:
+     *
+     * ```shell
+     * gst-launch-1.0 \
+     * videotestsrc is-live=1 \
+     * ! video/x-raw,width=640,height=360,framerate=15/1 \
+     * ! timeoverlay ! videoconvert ! queue \
+     * ! livekitwebrtcsink name=sink \
+     *     signaller::ws-url=ws://127.0.0.1:7880 \
+     *     signaller::api-key=devkey \
+     *     signaller::secret-key=secret \
+     *     signaller::room-name=testroom \
+     *     signaller::identity=gst-producer \
+     *     signaller::participant-name=gst-producer \
+     *     video-caps='video/x-vp8'
+     * ```
+     *
+     * Finally, watch the stream:
+     *
+     * ```shell
+     * gst-launch-1.0 \
+     * livekitwebrtcsrc \
+     *   signaller::ws-url=ws://127.0.0.1:7880 \
+     *   signaller::api-key=devkey \
+     *   signaller::secret-key=secret \
+     *   signaller::room-name=testroom \
+     *   signaller::identity=gst-consumer \
+     *   signaller::participant-name=gst-consumer \
+     * ! queue ! videoconvert ! autovideosink
+     * ```
+     */
+    gst::Element::register(
+        plugin,
+        "livekitwebrtcsrc",
+        gst::Rank::NONE,
+        LiveKitWebRTCSrc::static_type(),
+    )?;
+
     Ok(())
 }