diff --git a/src/webrtc/opcodes/Identify.ts b/src/webrtc/opcodes/Identify.ts
index f39958d4..065813fb 100644
--- a/src/webrtc/opcodes/Identify.ts
+++ b/src/webrtc/opcodes/Identify.ts
@@ -56,9 +56,6 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 	if (voiceState) {
 		type = voiceState.guild_id === server_id ? "guild-voice" : "dm-voice";
 		authenticated = true;
-		this.guild_id =
-			type === "guild-voice" ? voiceState.guild_id : undefined;
-		this.channel_id = voiceState.channel_id;
 	} else {
 		// if its not a guild/dm voice connection, check if it is a go live stream
 		const streamSession = await StreamSession.findOne({
@@ -78,8 +75,6 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 			streamSession.used = true;
 			await streamSession.save();
 
-			this.channel_id = streamSession.stream.channel_id;
-
 			this.once("close", async () => {
 				await streamSession.remove();
 			});
diff --git a/src/webrtc/util/WebRtcWebSocket.ts b/src/webrtc/util/WebRtcWebSocket.ts
index bca8b994..5bb2da46 100644
--- a/src/webrtc/util/WebRtcWebSocket.ts
+++ b/src/webrtc/util/WebRtcWebSocket.ts
@@ -3,7 +3,5 @@ import type { WebRtcClient } from "spacebar-webrtc-types";
 
 export interface WebRtcWebSocket extends WebSocket {
 	type: "guild-voice" | "dm-voice" | "stream";
-	guild_id?: string;
-	channel_id: string;
 	webRtcClient?: WebRtcClient<WebRtcWebSocket>;
 }