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

3 files changed, 7 insertions, 6 deletions

diff --git a/src/webrtc/opcodes/Heartbeat.ts b/src/webrtc/opcodes/Heartbeat.ts

index ee0e72b3..8e38a11a 100644
--- a/src/webrtc/opcodes/Heartbeat.ts
+++ b/src/webrtc/opcodes/Heartbeat.ts
@@ -21,7 +21,7 @@ import { VoiceOPCodes, VoicePayload, WebRtcWebSocket, Send } from "../util";
 
 export async function onHeartbeat(this: WebRtcWebSocket, data: VoicePayload) {
     setHeartbeat(this);
-    if (isNaN(data.d)) return this.close(CLOSECODES.Decode_error);
+    if (isNaN(data.d)) return this.rawSocket.close(CLOSECODES.Decode_error);
 
     await Send(this, { op: VoiceOPCodes.HEARTBEAT_ACK, d: data.d });
 }
diff --git a/src/webrtc/opcodes/Identify.ts b/src/webrtc/opcodes/Identify.ts

index 7d776d9c..fdf5f449 100644
--- a/src/webrtc/opcodes/Identify.ts
+++ b/src/webrtc/opcodes/Identify.ts
@@ -64,14 +64,14 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
             streamSession.used = true;
             await streamSession.save();
 
-            this.once("close", async () => {
+            this.rawSocket.once("close", async () => {
                 await streamSession.remove();
             });
         }
     }
 
     // if it doesnt match any then not valid token
-    if (!authenticated) return this.close(CLOSECODES.Authentication_failed);
+    if (!authenticated) return this.rawSocket.close(CLOSECODES.Authentication_failed);
 
     this.user_id = user_id;
     this.session_id = session_id;
@@ -82,10 +82,10 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
     try {
         this.webRtcClient = await mediaServer.join(voiceRoomId, this.user_id, this, type!);
     } catch (e) {
-        return this.close(4013);
+        return this.rawSocket.close(4013);
     }
 
-    this.on("close", () => {
+    this.rawSocket.on("close", () => {
         // ice-lite media server relies on this to know when the peer went away
         mediaServer.onClientClose(this.webRtcClient!);
     });
diff --git a/src/webrtc/opcodes/SelectProtocol.ts b/src/webrtc/opcodes/SelectProtocol.ts

index 6a733932..c3b61554 100644
--- a/src/webrtc/opcodes/SelectProtocol.ts
+++ b/src/webrtc/opcodes/SelectProtocol.ts
@@ -15,6 +15,7 @@
 	You should have received a copy of the GNU Affero General Public License
 	along with this program.  If not, see <https://www.gnu.org/licenses/>.
 */
+
 import { SelectProtocolSchema, validateSchema } from "@spacebar/schemas";
 import { VoiceOPCodes, VoicePayload, WebRtcWebSocket, mediaServer, Send } from "@spacebar/webrtc";
 
@@ -24,7 +25,7 @@ export async function onSelectProtocol(this: WebRtcWebSocket, payload: VoicePayl
     const data = validateSchema("SelectProtocolSchema", payload.d) as SelectProtocolSchema;
 
     // UDP protocol not currently supported. Maybe in the future?
-    if (data.protocol !== "webrtc") return this.close(4000, "only webrtc protocol supported currently");
+    if (data.protocol !== "webrtc") return this.rawSocket.close(4000, "only webrtc protocol supported currently");
 
     const response = await mediaServer.onOffer(this.webRtcClient, data.sdp!, data.codecs ?? []);