@superinterface/react 3.12.0 → 3.13.1

package/dist/index.cjs

@@ -11005,7 +11005,7 @@ var useWebrtcAudioRuntime = function() {
     }
     function _initRealtimeSession() {
         _initRealtimeSession = _asyncToGenerator12(function() {
-            var peerConn, audioEl, ms, offer, searchParams, sdpResponse, answerSdp, answer, err;
+            var iceServers, peerConn, audioEl, ms, offer, searchParams, sdpResponse, reader, decoder, answerSdp, _ref, value, done, answer, err;
             return _ts_generator(this, function(_state) {
                 switch(_state.label){
                     case 0:
@@ -11016,7 +11016,30 @@ var useWebrtcAudioRuntime = function() {
                             8
                         ]);
                         setUserIsPending(true);
-                        peerConn = new RTCPeerConnection();
+                        iceServers = [
+                            {
+                                url: "stun:global.stun.twilio.com:3478",
+                                urls: "stun:global.stun.twilio.com:3478"
+                            },
+                            {
+                                urls: "stun:stun.l.google.com:19302"
+                            },
+                            {
+                                urls: "stun:stun1.l.google.com:19302"
+                            },
+                            {
+                                urls: "stun:stun2.l.google.com:19302"
+                            },
+                            {
+                                urls: "stun:stun4.l.google.com:19302"
+                            },
+                            {
+                                urls: "stun:stun.stunprotocol.org:3478"
+                            }
+                        ];
+                        peerConn = new RTCPeerConnection({
+                            iceServers: iceServers
+                        });
                         pcRef.current = peerConn;
                         audioEl = document.createElement("audio");
                         audioEl.autoplay = true;
@@ -11092,12 +11115,28 @@ var useWebrtcAudioRuntime = function() {
                         ];
                     case 4:
                         sdpResponse = _state.sent();
+                        if (!sdpResponse.ok) {
+                            throw new Error("Server responded with status ".concat(sdpResponse.status));
+                        }
+                        if (!sdpResponse.body) {
+                            throw new Error("ReadableStream not supported in this browser.");
+                        }
+                        reader = sdpResponse.body.getReader();
+                        decoder = new TextDecoder("utf-8");
+                        answerSdp = "";
                         return [
                             4,
-                            sdpResponse.text()
+                            reader.read()
                         ];
                     case 5:
-                        answerSdp = _state.sent();
+                        _ref = _state.sent(), value = _ref.value, done = _ref.done;
+                        if (done) {
+                            throw new Error("Stream closed before SDP was received");
+                        }
+                        answerSdp += decoder.decode(value, {
+                            stream: true
+                        });
+                        console.log("Received SDP Answer:", answerSdp);
                         answer = {
                             type: "answer",
                             sdp: answerSdp