diff --git a/package.json b/package.json
index acf8ca4d..5b1a91a4 100644
--- a/package.json
+++ b/package.json
@@ -67,6 +67,7 @@
 		"husky": "^9.1.7",
 		"prettier": "^3.5.3",
 		"pretty-quick": "^4.1.1",
+		"spacebar-webrtc-types": "github:dank074/spacebar-webrtc-types",
 		"typescript": "^5.8.3"
 	},
 	"dependencies": {
@@ -124,8 +125,6 @@
 	"optionalDependencies": {
 		"@yukikaze-bot/erlpack": "^1.0.1",
 		"jimp": "^1.6.0",
-		"@dank074/medooze-media-server": "1.156.4",
-		"semantic-sdp": "^3.31.1",
 		"mysql": "^2.18.1",
 		"nodemailer-mailgun-transport": "^2.1.5",
 		"nodemailer-mailjet-transport": "github:n0script22/nodemailer-mailjet-transport",
diff --git a/src/webrtc/Server.ts b/src/webrtc/Server.ts
index 6127aef4..40c39658 100644
--- a/src/webrtc/Server.ts
+++ b/src/webrtc/Server.ts
@@ -21,7 +21,12 @@ import dotenv from "dotenv";
 import http from "http";
 import ws from "ws";
 import { Connection } from "./events/Connection";
-import { mediaServer } from "./util/MediaServer";
+import {
+	mediaServer,
+	WRTC_PORT_MAX,
+	WRTC_PORT_MIN,
+	WRTC_PUBLIC_IP,
+} from "./util/MediaServer";
 import { green, yellow } from "picocolors";
 dotenv.config();
 
@@ -77,7 +82,7 @@ export class Server {
 			console.log(`[WebRTC] ${yellow("WEBRTC disabled")}`);
 			return Promise.resolve();
 		}
-		await mediaServer.start();
+		await mediaServer.start(WRTC_PUBLIC_IP, WRTC_PORT_MIN, WRTC_PORT_MAX);
 		if (!this.server.listening) {
 			this.server.listen(this.port);
 			console.log(`[WebRTC] ${green(`online on 0.0.0.0:${this.port}`)}`);
diff --git a/src/webrtc/medooze/MedoozeSignalingDelegate.ts b/src/webrtc/medooze/MedoozeSignalingDelegate.ts
deleted file mode 100644
index 9e05fb45..00000000
--- a/src/webrtc/medooze/MedoozeSignalingDelegate.ts
+++ /dev/null
@@ -1,274 +0,0 @@
-import { CodecInfo, MediaInfo, SDPInfo } from "semantic-sdp";
-import { SignalingDelegate } from "../util/SignalingDelegate";
-import { Codec, WebRtcClient } from "../util/WebRtcClient";
-import { MediaServer, Endpoint } from "@dank074/medooze-media-server";
-import { VoiceRoom } from "./VoiceRoom";
-import { MedoozeWebRtcClient } from "./MedoozeWebRtcClient";
-
-export class MedoozeSignalingDelegate implements SignalingDelegate {
-	private _rooms: Map<string, VoiceRoom> = new Map();
-	private _ip: string;
-	private _port: number;
-	private _endpoint: Endpoint;
-
-	public start(): Promise<void> {
-		MediaServer.enableLog(true);
-
-		this._ip = process.env.PUBLIC_IP || "127.0.0.1";
-
-		try {
-			const range = process.env.WEBRTC_PORT_RANGE || "3690-3960";
-			var ports = range.split("-");
-			const min = Number(ports[0]);
-			const max = Number(ports[1]);
-
-			MediaServer.setPortRange(min, max);
-		} catch (error) {
-			console.error(
-				"Invalid env var: WEBRTC_PORT_RANGE",
-				process.env.WEBRTC_PORT_RANGE,
-				error,
-			);
-			process.exit(1);
-		}
-
-		//MediaServer.setAffinity(2)
-		this._endpoint = MediaServer.createEndpoint(this._ip);
-		this._port = this._endpoint.getLocalPort();
-		return Promise.resolve();
-	}
-
-	public join(
-		rtcServerId: string,
-		userId: string,
-		ws: any,
-		type: "guild-voice" | "dm-voice" | "stream",
-	): WebRtcClient<any> {
-		// make sure user isn't already in a room of the same type
-		// user can be in two simultanous rooms of different type though (can be in a voice channel and watching a stream for example)
-		const rooms = this.rooms
-			.values()
-			.filter((room) =>
-				type === "stream"
-					? room.type === "stream"
-					: room.type === "dm-voice" || room.type === "guild-voice",
-			);
-		let existingClient;
-
-		for (const room of rooms) {
-			let result = room.getClientById(userId);
-			if (result) {
-				existingClient = result;
-				break;
-			}
-		}
-
-		if (existingClient) {
-			console.log("client already connected, disconnect..");
-			this.onClientClose(existingClient);
-		}
-
-		if (!this._rooms.has(rtcServerId)) {
-			console.debug("no channel created, creating one...");
-			this.createRoom(rtcServerId, type);
-		}
-
-		const room = this._rooms.get(rtcServerId)!;
-
-		const client = new MedoozeWebRtcClient(userId, rtcServerId, ws, room);
-
-		room?.onClientJoin(client);
-
-		return client;
-	}
-
-	public async onOffer(
-		client: WebRtcClient<any>,
-		sdpOffer: string,
-		codecs: Codec[],
-	): Promise<string> {
-		const room = this._rooms.get(client.rtc_server_id);
-
-		if (!room) {
-			console.error(
-				"error, client sent an offer but has not authenticated",
-			);
-			Promise.reject();
-		}
-
-		const offer = SDPInfo.parse("m=audio\n" + sdpOffer);
-
-		const rtpHeaders = new Map(offer.medias[0].extensions);
-
-		const getIdForHeader = (
-			rtpHeaders: Map<number, string>,
-			headerUri: string,
-		) => {
-			for (const [key, value] of rtpHeaders) {
-				if (value == headerUri) return key;
-			}
-			return -1;
-		};
-
-		const audioMedia = new MediaInfo("0", "audio");
-		const audioCodec = new CodecInfo(
-			"opus",
-			codecs.find((val) => val.name == "opus")?.payload_type ?? 111,
-		);
-		audioCodec.addParam("minptime", "10");
-		audioCodec.addParam("usedtx", "1");
-		audioCodec.addParam("useinbandfec", "1");
-		audioCodec.setChannels(2);
-		audioMedia.addCodec(audioCodec);
-
-		audioMedia.addExtension(
-			getIdForHeader(
-				rtpHeaders,
-				"urn:ietf:params:rtp-hdrext:ssrc-audio-level",
-			),
-			"urn:ietf:params:rtp-hdrext:ssrc-audio-level",
-		);
-		if (audioCodec.type === 111)
-			// if this is chromium, apply this header
-			audioMedia.addExtension(
-				getIdForHeader(
-					rtpHeaders,
-					"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
-				),
-				"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
-			);
-
-		const videoMedia = new MediaInfo("1", "video");
-		const videoCodec = new CodecInfo(
-			"H264",
-			codecs.find((val) => val.name == "H264")?.payload_type ?? 102,
-		);
-		videoCodec.setRTX(
-			codecs.find((val) => val.name == "H264")?.rtx_payload_type ?? 103,
-		);
-		videoCodec.addParam("level-asymmetry-allowed", "1");
-		videoCodec.addParam("packetization-mode", "1");
-		videoCodec.addParam("profile-level-id", "42e01f");
-		videoCodec.addParam("x-google-max-bitrate", "2500");
-		videoMedia.addCodec(videoCodec);
-
-		videoMedia.addExtension(
-			getIdForHeader(
-				rtpHeaders,
-				"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
-			),
-			"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
-		);
-		videoMedia.addExtension(
-			getIdForHeader(rtpHeaders, "urn:ietf:params:rtp-hdrext:toffset"),
-			"urn:ietf:params:rtp-hdrext:toffset",
-		);
-		videoMedia.addExtension(
-			getIdForHeader(
-				rtpHeaders,
-				"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
-			),
-			"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
-		);
-		videoMedia.addExtension(
-			getIdForHeader(
-				rtpHeaders,
-				"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
-			),
-			"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
-		);
-
-		if (audioCodec.type === 111)
-			// if this is chromium, apply this header
-			videoMedia.addExtension(
-				getIdForHeader(rtpHeaders, "urn:3gpp:video-orientation"),
-				"urn:3gpp:video-orientation",
-			);
-
-		offer.medias = [audioMedia, videoMedia];
-
-		const transport = this._endpoint.createTransport(offer);
-
-		transport.setRemoteProperties(offer);
-
-		room?.onClientOffer(client, transport);
-
-		const dtls = transport.getLocalDTLSInfo();
-		const ice = transport.getLocalICEInfo();
-		const fingerprint = dtls.getHash() + " " + dtls.getFingerprint();
-		const candidates = transport.getLocalCandidates();
-		const candidate = candidates[0];
-
-		const answer =
-			`m=audio ${this.port} ICE/SDP\n` +
-			`a=fingerprint:${fingerprint}\n` +
-			`c=IN IP4 ${this.ip}\n` +
-			`a=rtcp:${this.port}\n` +
-			`a=ice-ufrag:${ice.getUfrag()}\n` +
-			`a=ice-pwd:${ice.getPwd()}\n` +
-			`a=fingerprint:${fingerprint}\n` +
-			`a=candidate:1 1 ${candidate.getTransport()} ${candidate.getFoundation()} ${candidate.getAddress()} ${candidate.getPort()} typ host\n`;
-
-		return Promise.resolve(answer);
-	}
-
-	public onClientClose = (client: WebRtcClient<any>) => {
-		this._rooms.get(client.rtc_server_id)?.onClientLeave(client);
-	};
-
-	public updateSDP(offer: string): void {
-		throw new Error("Method not implemented.");
-	}
-
-	public createRoom(
-		rtcServerId: string,
-		type: "guild-voice" | "dm-voice" | "stream",
-	): void {
-		this._rooms.set(rtcServerId, new VoiceRoom(rtcServerId, type, this));
-	}
-
-	public disposeRoom(rtcServerId: string): void {
-		const room = this._rooms.get(rtcServerId);
-		room?.dispose();
-		this._rooms.delete(rtcServerId);
-	}
-
-	get rooms(): Map<string, VoiceRoom> {
-		return this._rooms;
-	}
-
-	public getClientsForRtcServer(rtcServerId: string): Set<WebRtcClient<any>> {
-		if (!this._rooms.has(rtcServerId)) {
-			return new Set();
-		}
-
-		return new Set(this._rooms.get(rtcServerId)?.clients.values())!;
-	}
-
-	private getClientForUserId = (
-		userId: string,
-	): MedoozeWebRtcClient | undefined => {
-		for (const channel of this.rooms.values()) {
-			let result = channel.getClientById(userId);
-			if (result) {
-				return result;
-			}
-		}
-		return undefined;
-	};
-
-	get ip(): string {
-		return this._ip;
-	}
-	get port(): number {
-		return this._port;
-	}
-
-	get endpoint(): Endpoint {
-		return this._endpoint;
-	}
-
-	public stop(): Promise<void> {
-		return Promise.resolve();
-	}
-}
diff --git a/src/webrtc/medooze/MedoozeWebRtcClient.ts b/src/webrtc/medooze/MedoozeWebRtcClient.ts
deleted file mode 100644
index c6defaa3..00000000
--- a/src/webrtc/medooze/MedoozeWebRtcClient.ts
+++ /dev/null
@@ -1,152 +0,0 @@
-import {
-	IncomingStream,
-	OutgoingStream,
-	Transport,
-} from "@dank074/medooze-media-server";
-import { SSRCs, WebRtcClient } from "webrtc/util";
-import { VoiceRoom } from "./VoiceRoom";
-
-export class MedoozeWebRtcClient implements WebRtcClient<any> {
-	websocket: any;
-	user_id: string;
-	rtc_server_id: string;
-	webrtcConnected: boolean;
-	public transport?: Transport;
-	public incomingStream?: IncomingStream;
-	public outgoingStream?: OutgoingStream;
-	public room?: VoiceRoom;
-	public isStopped?: boolean;
-
-	constructor(
-		userId: string,
-		rtcServerId: string,
-		websocket: any,
-		room: VoiceRoom,
-	) {
-		this.user_id = userId;
-		this.rtc_server_id = rtcServerId;
-		this.websocket = websocket;
-		this.room = room;
-		this.webrtcConnected = false;
-		this.isStopped = false;
-	}
-
-	public isProducingAudio(): boolean {
-		if (!this.webrtcConnected) return false;
-		const audioTrack = this.incomingStream?.getTrack(
-			`audio-${this.user_id}`,
-		);
-
-		if (audioTrack) return true;
-
-		return false;
-	}
-
-	public isProducingVideo(): boolean {
-		if (!this.webrtcConnected) return false;
-		const videoTrack = this.incomingStream?.getTrack(
-			`video-${this.user_id}`,
-		);
-
-		if (videoTrack) return true;
-
-		return false;
-	}
-
-	public getIncomingStreamSSRCs(): SSRCs {
-		if (!this.webrtcConnected)
-			return { audio_ssrc: 0, video_ssrc: 0, rtx_ssrc: 0 };
-
-		const audioTrack = this.incomingStream?.getTrack(
-			`audio-${this.user_id}`,
-		);
-		const audio_ssrc =
-			audioTrack?.getSSRCs()[audioTrack.getDefaultEncoding().id];
-		const videoTrack = this.incomingStream?.getTrack(
-			`video-${this.user_id}`,
-		);
-		const video_ssrc =
-			videoTrack?.getSSRCs()[videoTrack.getDefaultEncoding().id];
-
-		return {
-			audio_ssrc: audio_ssrc?.media ?? 0,
-			video_ssrc: video_ssrc?.media ?? 0,
-			rtx_ssrc: video_ssrc?.rtx ?? 0,
-		};
-	}
-
-	public getOutgoingStreamSSRCsForUser(user_id: string): SSRCs {
-		const outgoingStream = this.outgoingStream;
-
-		const audioTrack = outgoingStream?.getTrack(`audio-${user_id}`);
-		const audio_ssrc = audioTrack?.getSSRCs();
-		const videoTrack = outgoingStream?.getTrack(`video-${user_id}`);
-		const video_ssrc = videoTrack?.getSSRCs();
-
-		return {
-			audio_ssrc: audio_ssrc?.media ?? 0,
-			video_ssrc: video_ssrc?.media ?? 0,
-			rtx_ssrc: video_ssrc?.rtx ?? 0,
-		};
-	}
-
-	public publishTrack(type: "audio" | "video", ssrc: SSRCs) {
-		if (!this.transport) return;
-
-		const id = `${type}-${this.user_id}`;
-		const existingTrack = this.incomingStream?.getTrack(id);
-
-		if (existingTrack) {
-			console.error(`error: attempted to create duplicate track ${id}`);
-			return;
-		}
-		let ssrcs;
-		if (type === "audio") {
-			ssrcs = { media: ssrc.audio_ssrc! };
-		} else {
-			ssrcs = { media: ssrc.video_ssrc!, rtx: ssrc.rtx_ssrc };
-		}
-		const track = this.transport?.createIncomingStreamTrack(
-			type,
-			{ id, ssrcs: ssrcs, media: type },
-			this.incomingStream,
-		);
-
-		//this.channel?.onClientPublishTrack(this, track, ssrcs);
-	}
-
-	public subscribeToTrack(user_id: string, type: "audio" | "video") {
-		if (!this.transport) return;
-
-		const id = `${type}-${user_id}`;
-
-		const otherClient = this.room?.getClientById(user_id);
-		const incomingStream = otherClient?.incomingStream;
-		const incomingTrack = incomingStream?.getTrack(id);
-
-		if (!incomingTrack) {
-			console.error(`error subscribing, not track found ${id}`);
-			return;
-		}
-
-		let ssrcs;
-		if (type === "audio") {
-			ssrcs = {
-				media: otherClient?.getIncomingStreamSSRCs().audio_ssrc!,
-			};
-		} else {
-			ssrcs = {
-				media: otherClient?.getIncomingStreamSSRCs().video_ssrc!,
-				rtx: otherClient?.getIncomingStreamSSRCs().rtx_ssrc,
-			};
-		}
-
-		const outgoingTrack = this.transport?.createOutgoingStreamTrack(
-			incomingTrack.media,
-			{ id, ssrcs, media: incomingTrack.media },
-			this.outgoingStream,
-		);
-
-		outgoingTrack?.attachTo(incomingTrack);
-	}
-}
diff --git a/src/webrtc/medooze/VoiceRoom.ts b/src/webrtc/medooze/VoiceRoom.ts
deleted file mode 100644
index 55eb6fea..00000000
--- a/src/webrtc/medooze/VoiceRoom.ts
+++ /dev/null
@@ -1,119 +0,0 @@
-import { MedoozeSignalingDelegate } from "./MedoozeSignalingDelegate";
-import {
-	IncomingStreamTrack,
-	SSRCs,
-	Transport,
-} from "@dank074/medooze-media-server";
-import { MedoozeWebRtcClient } from "./MedoozeWebRtcClient";
-import { StreamInfo } from "semantic-sdp";
-
-export class VoiceRoom {
-	private _clients: Map<string, MedoozeWebRtcClient>;
-	private _id: string;
-	private _sfu: MedoozeSignalingDelegate;
-	private _type: "guild-voice" | "dm-voice" | "stream";
-
-	constructor(
-		id: string,
-		type: "guild-voice" | "dm-voice" | "stream",
-		sfu: MedoozeSignalingDelegate,
-	) {
-		this._id = id;
-		this._type = type;
-		this._clients = new Map();
-		this._sfu = sfu;
-	}
-
-	onClientJoin = (client: MedoozeWebRtcClient) => {
-		// do shit here
-		this._clients.set(client.user_id, client);
-	};
-
-	onClientOffer = (client: MedoozeWebRtcClient, transport: Transport) => {
-		client.transport = transport;
-
-		client.transport.on("dtlsstate", (state, self) => {
-			if (state === "connected") {
-				client.webrtcConnected = true;
-				console.log("connected");
-			}
-		});
-
-		client.incomingStream = transport.createIncomingStream(
-			new StreamInfo(`in-${client.user_id}`),
-		);
-
-		client.outgoingStream = transport.createOutgoingStream(
-			new StreamInfo(`out-${client.user_id}`),
-		);
-
-		client.webrtcConnected = true;
-
-		// subscribe to all current streams from this channel
-		// for(const otherClient of this._clients.values()) {
-		//     const incomingStream = otherClient.incomingStream
-
-		//     if(!incomingStream) continue;
-
-		//     for(const track of (incomingStream.getTracks())) {
-		//         client.subscribeToTrack(otherClient.user_id, track.media)
-		//     }
-		// }
-	};
-
-	onClientLeave = (client: MedoozeWebRtcClient) => {
-		console.log("stopping client");
-		this._clients.delete(client.user_id);
-
-		// stop the client
-		if (!client.isStopped) {
-			client.isStopped = true;
-
-			for (const otherClient of this.clients.values()) {
-				//remove outgoing track for this user
-				otherClient.outgoingStream
-					?.getTrack(`audio-${client.user_id}`)
-					?.stop();
-				otherClient.outgoingStream
-					?.getTrack(`video-${client.user_id}`)
-					?.stop();
-			}
-
-			client.incomingStream?.stop();
-			client.outgoingStream?.stop();
-
-			client.transport?.stop();
-			client.room = undefined;
-			client.incomingStream = undefined;
-			client.outgoingStream = undefined;
-			client.transport = undefined;
-			client.websocket = undefined;
-		}
-	};
-
-	get clients(): Map<string, MedoozeWebRtcClient> {
-		return this._clients;
-	}
-
-	getClientById = (id: string) => {
-		return this._clients.get(id);
-	};
-
-	get id(): string {
-		return this._id;
-	}
-
-	get type(): "guild-voice" | "dm-voice" | "stream" {
-		return this._type;
-	}
-
-	public dispose(): void {
-		const clients = this._clients.values();
-		for (const client of clients) {
-			this.onClientLeave(client);
-		}
-		this._clients.clear();
-		this._sfu = undefined!;
-		this._clients = undefined!;
-	}
-}
diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index 61be1e60..1f21be83 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -20,10 +20,10 @@ import {
 	mediaServer,
 	VoiceOPCodes,
 	VoicePayload,
-	WebRtcClient,
 	WebRtcWebSocket,
 	Send,
 } from "@spacebar/webrtc";
+import type { WebRtcClient } from "spacebar-webrtc-types";
 
 export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 	if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
diff --git a/src/webrtc/util/MediaServer.ts b/src/webrtc/util/MediaServer.ts
index befacbce..9842efb7 100644
--- a/src/webrtc/util/MediaServer.ts
+++ b/src/webrtc/util/MediaServer.ts
@@ -16,25 +16,49 @@
 	along with this program.  If not, see <https://www.gnu.org/licenses/>.
 */
 
-//import { MedoozeSignalingDelegate } from "../medooze/MedoozeSignalingDelegate";
-import { SignalingDelegate } from "./SignalingDelegate";
+import type { SignalingDelegate } from "spacebar-webrtc-types";
 import { green, red } from "picocolors";
 
 export let mediaServer: SignalingDelegate;
 
+export const WRTC_PUBLIC_IP = process.env.WRTC_PUBLIC_IP ?? "127.0.0.1";
+export const WRTC_PORT_MIN = process.env.WRTC_PORT_MIN
+	? parseInt(process.env.WRTC_PORT_MIN)
+	: 2000;
+export const WRTC_PORT_MAX = process.env.WRTC_PORT_MAX
+	? parseInt(process.env.WRTC_PORT_MAX)
+	: 65000;
+
+const selectedWrtcLibrary = process.env.WRTC_LIBRARY;
+
+// could not find a way to hide stack trace from base Error object
+class NoConfiguredLibraryError implements Error {
+	name: string;
+	message: string;
+	stack?: string | undefined;
+	cause?: unknown;
+
+	constructor(message: string) {
+		this.name = "NoConfiguredLibraryError";
+		this.message = message;
+	}
+}
+
 (async () => {
 	try {
-		//mediaServer = require('../medooze/MedoozeSignalingDelegate');
-		mediaServer = new (
-			await import("../medooze/MedoozeSignalingDelegate")
-		).MedoozeSignalingDelegate();
+		//mediaServer = require('medooze-spacebar-wrtc');
+		if (!selectedWrtcLibrary)
+			throw new NoConfiguredLibraryError("No library configured in .env");
+
+		mediaServer = new // @ts-ignore
+		(await import(selectedWrtcLibrary)).default();
 
 		console.log(
-			`[WebRTC] ${green("Succesfully loaded MedoozeSignalingDelegate")}`,
+			`[WebRTC] ${green(`Succesfully loaded ${selectedWrtcLibrary}`)}`,
 		);
-	} catch (e) {
+	} catch (error) {
 		console.log(
-			`[WebRTC] ${red("Failed to import MedoozeSignalingDelegate")}`,
+			`[WebRTC] ${red(`Failed to import ${selectedWrtcLibrary}: ${error instanceof NoConfiguredLibraryError ? error.message : ""}`)}`,
 		);
 	}
 })();
diff --git a/src/webrtc/util/SignalingDelegate.ts b/src/webrtc/util/SignalingDelegate.ts
deleted file mode 100644
index e2f010d2..00000000
--- a/src/webrtc/util/SignalingDelegate.ts
+++ /dev/null
@@ -1,22 +0,0 @@
-import { Codec, WebRtcClient } from "./WebRtcClient";
-
-export interface SignalingDelegate {
-	start: () => Promise<void>;
-	stop: () => Promise<void>;
-	join<T>(
-		rtcServerId: string,
-		userId: string,
-		ws: T,
-		type: "guild-voice" | "dm-voice" | "stream",
-	): WebRtcClient<T>;
-	onOffer<T>(
-		client: WebRtcClient<T>,
-		offer: string,
-		codecs: Codec[],
-	): Promise<string>;
-	onClientClose<T>(client: WebRtcClient<T>): void;
-	updateSDP(offer: string): void;
-	getClientsForRtcServer<T>(rtcServerId: string): Set<WebRtcClient<T>>;
-	get ip(): string;
-	get port(): number;
-}
diff --git a/src/webrtc/util/WebRtcClient.ts b/src/webrtc/util/WebRtcClient.ts
deleted file mode 100644
index 3dcd0493..00000000
--- a/src/webrtc/util/WebRtcClient.ts
+++ /dev/null
@@ -1,31 +0,0 @@
-export interface WebRtcClient<T> {
-	websocket: T;
-	user_id: string;
-	rtc_server_id: string;
-	webrtcConnected: boolean;
-	getIncomingStreamSSRCs: () => SSRCs;
-	getOutgoingStreamSSRCsForUser: (user_id: string) => SSRCs;
-	isProducingAudio: () => boolean;
-	isProducingVideo: () => boolean;
-	publishTrack: (type: "audio" | "video", ssrc: SSRCs) => void;
-	subscribeToTrack: (user_id: string, type: "audio" | "video") => void;
-}
-
-export interface SSRCs {
-	audio_ssrc?: number;
-	video_ssrc?: number;
-	rtx_ssrc?: number;
-}
-
-export interface RtpHeader {
-	uri: string;
-	id: number;
-}
-
-export interface Codec {
-	name: "opus" | "VP8" | "VP9" | "H264";
-	type: "audio" | "video";
-	priority: number;
-	payload_type: number;
-	rtx_payload_type?: number;
-}
diff --git a/src/webrtc/util/WebRtcWebSocket.ts b/src/webrtc/util/WebRtcWebSocket.ts
index 086e1247..5bb2da46 100644
--- a/src/webrtc/util/WebRtcWebSocket.ts
+++ b/src/webrtc/util/WebRtcWebSocket.ts
@@ -1,5 +1,5 @@
 import { WebSocket } from "@spacebar/gateway";
-import { WebRtcClient } from "./WebRtcClient";
+import type { WebRtcClient } from "spacebar-webrtc-types";
 
 export interface WebRtcWebSocket extends WebSocket {
 	type: "guild-voice" | "dm-voice" | "stream";
diff --git a/src/webrtc/util/index.ts b/src/webrtc/util/index.ts
index aa33ad2d..264f1ecd 100644
--- a/src/webrtc/util/index.ts
+++ b/src/webrtc/util/index.ts
@@ -18,6 +18,5 @@
 
 export * from "./Constants";
 export * from "./MediaServer";
-export * from "./WebRtcClient";
 export * from "./WebRtcWebSocket";
 export * from "./Send";