@superinterface/react 3.13.0 → 3.13.2

package/dist/index.cjs

@@ -11005,18 +11005,41 @@ var useWebrtcAudioRuntime = function() {
     }
     function _initRealtimeSession() {
         _initRealtimeSession = _asyncToGenerator12(function() {
-            var peerConn, audioEl, ms, offer, searchParams, sdpResponse, reader, decoder, answerSdp, _ref, value, done, answer, err;
+            var searchParams, iceServersResponse, iceServersData, peerConn, audioEl, ms, offer, sdpResponse, reader, decoder, answerSdp, _ref, value, done, answer, err;
             return _ts_generator(this, function(_state) {
                 switch(_state.label){
                     case 0:
                         _state.trys.push([
                             0,
-                            7,
+                            9,
                             ,
-                            8
+                            10
                         ]);
                         setUserIsPending(true);
-                        peerConn = new RTCPeerConnection();
+                        searchParams = new URLSearchParams(variableParams({
+                            variables: superinterfaceContext.variables,
+                            superinterfaceContext: superinterfaceContext
+                        }));
+                        return [
+                            4,
+                            fetch("".concat(superinterfaceContext.baseUrl, "/audio-runtimes/webrtc/ice-servers?").concat(searchParams), {
+                                method: "GET",
+                                headers: {
+                                    "Content-Type": "application/json"
+                                }
+                            })
+                        ];
+                    case 1:
+                        iceServersResponse = _state.sent();
+                        return [
+                            4,
+                            iceServersResponse.json()
+                        ];
+                    case 2:
+                        iceServersData = _state.sent();
+                        peerConn = new RTCPeerConnection({
+                            iceServers: iceServersData.iceServers
+                        });
                         pcRef.current = peerConn;
                         audioEl = document.createElement("audio");
                         audioEl.autoplay = true;
@@ -11057,7 +11080,7 @@ var useWebrtcAudioRuntime = function() {
                                 audio: true
                             })
                         ];
-                    case 1:
+                    case 3:
                         ms = _state.sent();
                         localStreamRef.current = ms;
                         ms.getTracks().forEach(function(t) {
@@ -11068,18 +11091,14 @@ var useWebrtcAudioRuntime = function() {
                             4,
                             peerConn.createOffer()
                         ];
-                    case 2:
+                    case 4:
                         offer = _state.sent();
                         return [
                             4,
                             peerConn.setLocalDescription(offer)
                         ];
-                    case 3:
+                    case 5:
                         _state.sent();
-                        searchParams = new URLSearchParams(variableParams({
-                            variables: superinterfaceContext.variables,
-                            superinterfaceContext: superinterfaceContext
-                        }));
                         return [
                             4,
                             fetch("".concat(superinterfaceContext.baseUrl, "/audio-runtimes/webrtc?").concat(searchParams), {
@@ -11090,7 +11109,7 @@ var useWebrtcAudioRuntime = function() {
                                 }
                             })
                         ];
-                    case 4:
+                    case 6:
                         sdpResponse = _state.sent();
                         if (!sdpResponse.ok) {
                             throw new Error("Server responded with status ".concat(sdpResponse.status));
@@ -11105,7 +11124,7 @@ var useWebrtcAudioRuntime = function() {
                             4,
                             reader.read()
                         ];
-                    case 5:
+                    case 7:
                         _ref = _state.sent(), value = _ref.value, done = _ref.done;
                         if (done) {
                             throw new Error("Stream closed before SDP was received");
@@ -11122,7 +11141,7 @@ var useWebrtcAudioRuntime = function() {
                             4,
                             peerConn.setRemoteDescription(answer)
                         ];
-                    case 6:
+                    case 8:
                         _state.sent();
                         buildAnalyzers(ms, audioEl);
                         setUserIsPending(false);
@@ -11131,9 +11150,9 @@ var useWebrtcAudioRuntime = function() {
                         setAssistantPlaying(true);
                         return [
                             3,
-                            8
+                            10
                         ];
-                    case 7:
+                    case 9:
                         err = _state.sent();
                         console.error("Error initRealtimeSession:", err);
                         setUserIsPending(false);
@@ -11145,9 +11164,9 @@ var useWebrtcAudioRuntime = function() {
                         setAssistantAudioPlayed(false);
                         return [
                             3,
-                            8
+                            10
                         ];
-                    case 8:
+                    case 10:
                         return [
                             2
                         ];