Decouple gateway connection state from inner ws object github/dev/gwsep dev/gwsep

3 files changed, 13 insertions, 12 deletions

diff --git a/src/webrtc/events/Close.ts b/src/webrtc/events/Close.ts

index e88b2938..bf2b4e22 100644
--- a/src/webrtc/events/Close.ts
+++ b/src/webrtc/events/Close.ts
@@ -18,8 +18,8 @@
 
 import { WebSocket } from "@spacebar/gateway";
 
-export async function onClose(this: WebSocket, code: number, reason: string) {
+export async function onClose(socket: WebSocket, code: number, reason: Buffer) {
     console.log("[WebRTC] closed", code, reason.toString());
 
-    this.removeAllListeners();
+    socket.rawSocket.removeAllListeners();
 }
diff --git a/src/webrtc/events/Connection.ts b/src/webrtc/events/Connection.ts

index f1808399..2949ea75 100644
--- a/src/webrtc/events/Connection.ts
+++ b/src/webrtc/events/Connection.ts
@@ -28,10 +28,11 @@ import { onMessage } from "./Message";
 // TODO: specify rate limit in config
 // TODO: check msg max size
 
-export async function Connection(this: WS.Server, socket: WebRtcWebSocket, request: IncomingMessage) {
+export async function Connection(this: WS.Server, rawSocket: WS, request: IncomingMessage) {
+    const socket = new WebRtcWebSocket(rawSocket);
     try {
-        socket.on("close", onClose.bind(socket));
-        socket.on("message", onMessage.bind(socket));
+        socket.rawSocket.on("close", (code, reason) => onClose(socket, code, reason));
+        socket.rawSocket.on("message", (data, isBinary) => onMessage(socket, data as Buffer));
         console.log("[WebRTC] new connection", request.url);
 
         if (process.env.WS_LOGEVENTS) {
@@ -45,7 +46,7 @@ export async function Connection(this: WS.Server, socket: WebRtcWebSocket, reque
                 "pong",
                 "unexpected-response",
             ].forEach((x) => {
-                socket.on(x, (y) => console.log("[WebRTC]", x, y));
+                socket.rawSocket.on(x, (y) => console.log("[WebRTC]", x, y));
             });
         }
 
@@ -53,11 +54,11 @@ export async function Connection(this: WS.Server, socket: WebRtcWebSocket, reque
 
         socket.encoding = "json";
         socket.version = Number(searchParams.get("v")) || 5;
-        if (socket.version < 3) return socket.close(CLOSECODES.Unknown_error, "invalid version");
+        if (socket.version < 3) return socket.rawSocket.close(CLOSECODES.Unknown_error, "invalid version");
 
         setHeartbeat(socket);
 
-        socket.readyTimeout = setTimeout(() => socket.close(CLOSECODES.Session_timed_out), 1000 * 30);
+        socket.readyTimeout = setTimeout(() => socket.rawSocket.close(CLOSECODES.Session_timed_out), 1000 * 30);
 
         await Send(socket, {
             op: VoiceOPCodes.HELLO,
@@ -67,6 +68,6 @@ export async function Connection(this: WS.Server, socket: WebRtcWebSocket, reque
         });
     } catch (error) {
         console.error("[WebRTC]", error);
-        return socket.close(CLOSECODES.Unknown_error);
+        return socket.rawSocket.close(CLOSECODES.Unknown_error);
     }
 }
diff --git a/src/webrtc/events/Message.ts b/src/webrtc/events/Message.ts

index b5455d97..0ec2108a 100644
--- a/src/webrtc/events/Message.ts
+++ b/src/webrtc/events/Message.ts
@@ -20,10 +20,10 @@ import { CLOSECODES } from "@spacebar/gateway";
 import OPCodeHandlers from "../opcodes";
 import { VoiceOPCodes, VoicePayload, WebRtcWebSocket } from "../util";
 
-export async function onMessage(this: WebRtcWebSocket, buffer: Buffer) {
+export async function onMessage(socket: WebRtcWebSocket, buffer: Buffer) {
     try {
         const data: VoicePayload = JSON.parse(buffer.toString());
-        if (data.op !== VoiceOPCodes.IDENTIFY && !this.user_id) return this.close(CLOSECODES.Not_authenticated);
+        if (data.op !== VoiceOPCodes.IDENTIFY && !socket.user_id) return socket.rawSocket.close(CLOSECODES.Not_authenticated);
 
         const OPCodeHandler = OPCodeHandlers[data.op];
         if (!OPCodeHandler) {
@@ -37,7 +37,7 @@ export async function onMessage(this: WebRtcWebSocket, buffer: Buffer) {
             console.log("[WebRTC] Opcode " + VoiceOPCodes[data.op]);
         }
 
-        return await OPCodeHandler.call(this, data);
+        return await OPCodeHandler.call(socket, data);
     } catch (error) {
         console.error("[WebRTC] error", error);
         // if (!this.CLOSED && this.CLOSING) return this.close(CloseCodes.Unknown_error);