diff --git a/examples/foundational/websocket-server/frames.proto b/examples/foundational/websocket-server/frames.proto
new file mode 100644
index 000000000..7ecea6d25
--- /dev/null
+++ b/examples/foundational/websocket-server/frames.proto
@@ -0,0 +1,25 @@
+syntax = "proto3";
+
+package dailyai_proto;
+
+message TextFrame {
+    string text = 1;
+}
+
+message AudioFrame {
+    bytes audio = 1;
+}
+
+message TranscriptionFrame {
+    string text = 1;
+    string participant_id = 2;
+    string timestamp = 3;
+}
+
+message Frame {
+    oneof frame {
+        TextFrame text = 1;
+        AudioFrame audio = 2;
+        TranscriptionFrame transcription = 3;
+    }
+}
diff --git a/examples/foundational/websocket-server/index.html b/examples/foundational/websocket-server/index.html
index 51b409573..130faeb9e 100644
--- a/examples/foundational/websocket-server/index.html
+++ b/examples/foundational/websocket-server/index.html
@@ -15,8 +15,12 @@ <h1>WebSocket Audio Stream</h1>
     <script>
         let audioContext;
         let microphoneStream;
+        let scriptProcessor;
+        let source;
         let audioCtx;
         let frame;
+        let echo_interval;
+        let b2p_interval;
 
         proto = protobuf.load("frames.proto", function (err, root) {
             if (err) {
@@ -25,7 +29,6 @@ <h1>WebSocket Audio Stream</h1>
             frame = root.lookupType("dailyai_proto.Frame");
         });
 
-
         async function startAudio() {
             audioCtx = new (window.AudioContext || window.webkitAudioContext)();
 
@@ -41,33 +44,107 @@ <h1>WebSocket Audio Stream</h1>
             const context = new AudioContext();
 
             // Create a ScriptProcessorNode.
-            const scriptProcessor = context.createScriptProcessor(8192, 1, 1);
+            scriptProcessor = context.createScriptProcessor(8192, 1, 1);
 
             // Connect the microphone stream to the ScriptProcessorNode.
-            const source = context.createMediaStreamSource(microphoneStream);
+            source = context.createMediaStreamSource(microphoneStream);
             source.connect(scriptProcessor);
 
             // Connect the ScriptProcessorNode to the destination.
             scriptProcessor.connect(context.destination);
 
+            var audio_buffer = []
+            function buffer_to_proto() {
+                MIN_AUDIO_SIZE = 6400 // send an audio frame every 1/4 of a second
+                if (audio_buffer.length > MIN_AUDIO_SIZE) {
+                    var audioFrame = frame.create({ audio: { audio: audio_buffer.slice(0, MIN_AUDIO_SIZE) } });
+                    var encodedFrame = new Uint8Array(frame.encode(audioFrame).finish())
+                    console.log("sending audio frame")
+                    ws.send(encodedFrame)
+                    audio_buffer = audio_buffer.slice(MIN_AUDIO_SIZE)
+                }
+            }
+
+
             scriptProcessor.onaudioprocess = (event) => {
                 const rawLeftChannelData = event.inputBuffer.getChannelData(0);
-                scaledToInt = []
-                for (var i = 0; i < rawLeftChannelData.length; i++) {
-                    // Convert each item from -1.0-1.0 to a 16-bit signed integer
-                    scaledToInt[i] = (rawLeftChannelData[i] * 32767 + 32767) % 65535;
+                const skipRatio = Math.floor(audioCtx.sampleRate / (16000 * 2));
+                var scaledToInt = []
+                var sum = 0
+                var mx = 0
+                for (var i = 0; i < rawLeftChannelData.length; i += skipRatio) {
+                    // Convert each item from -1.0-1.0 to a 16-bit unsigned integer
+                    var normalized = ((rawLeftChannelData[i] * 32768.0) + 32768) % 65536 - 32768;
+                    // swap the bytes
+                    normalized = ((normalized & 0xff) << 8) | ((normalized >> 8) & 0xff);
+
+                    scaledToInt.push(normalized);
+                    // swap the bytes maybe
+                    sum += normalized
+                    max = Math.max(mx, Math.abs(normalized))
                 }
-                //ws.send(scaledToInt);
+                audio_buffer = audio_buffer.concat(scaledToInt)
             };
 
-
             initWebSocket();
+            b2p_interval = setInterval(buffer_to_proto, 1)
         }
 
         function stopAudio() {
-            microphoneStream.disconnect();
-            ws.close();
-            ws = undefined;
+            if (ws) {
+                ws.close();
+                scriptProcessor.disconnect();
+                source.disconnect();
+                echo_interval = clearInterval(echo_interval)
+                b2p_interval = clearInterval(b2p_interval)
+                ws = undefined;
+
+            }
+        }
+
+        var audioChunks = [];
+        var isPlaying = false;
+
+        function playNextChunk() {
+            if (audioChunks.length === 0) {
+                isPlaying = false;
+                return;
+            }
+
+            isPlaying = true;
+            var audioOutBuffer = audioChunks.shift(); // Get the next audio chunk
+            var source = audioCtx.createBufferSource();
+            source.buffer = audioOutBuffer;
+            source.connect(audioCtx.destination);
+            source.onended = playNextChunk; // Set the onended event handler
+            source.start();
+        }
+
+        function enqueueAudioFromProto(arrayBuffer) {
+            var parsedFrame = frame.decode(new Uint8Array(arrayBuffer));
+            if (!parsedFrame?.audio) {
+                console.log("no audio in frame")
+                return false;
+            }
+            var frameCount = parsedFrame.audio.audio.length / 2;
+            var audioOutBuffer = audioCtx.createBuffer(1, frameCount, 16000);
+            var nowBuffering = audioOutBuffer.getChannelData(0);
+            var buffer = parsedFrame.audio.audio.buffer;
+            var view = new Int16Array(buffer);
+            var sum = 0;
+            var mx = 0
+            for (var i = 0; i < frameCount; i++) {
+                var word = view[i];
+                //nowBuffering[i] = word;
+                nowBuffering[i] = ((word + 32768) % 65536 - 32768) / 32768.0;
+                sum += nowBuffering[i];
+                mx = Math.max(mx, Math.abs(nowBuffering[i]))
+            }
+            audioChunks.push(audioOutBuffer); // Add the new audio chunk to the array
+
+            if (!isPlaying) {
+                playNextChunk(); // Start playing if not already playing
+            }
         }
 
         function initWebSocket() {
@@ -77,48 +154,13 @@ <h1>WebSocket Audio Stream</h1>
                 console.log('WebSocket connection established.');
             });
 
-            var audioChunks = [];
-            var isPlaying = false;
-
-            function playNextChunk() {
-                if (audioChunks.length === 0) {
-                    isPlaying = false;
-                    return;
-                }
-
-                isPlaying = true;
-                var audioOutBuffer = audioChunks.shift(); // Get the next audio chunk
-                var source = audioCtx.createBufferSource();
-                source.buffer = audioOutBuffer;
-                source.connect(audioCtx.destination);
-                source.onended = playNextChunk; // Set the onended event handler
-                source.start();
-            }
-
             ws.addEventListener('message', async function (event) {
                 if (!event.data?.size) {
                     return false;
                 }
 
                 try {
-                    var parsedFrame = frame.decode(new Uint8Array(await event.data.arrayBuffer()));
-                    if (!parsedFrame?.audio) {
-                        return false;
-                    }
-                    var frameCount = parsedFrame.audio.audio.length / 2;
-                    var audioOutBuffer = audioCtx.createBuffer(1, frameCount, 16000);
-                    var nowBuffering = audioOutBuffer.getChannelData(0);
-                    var buffer = parsedFrame.audio.audio.buffer;
-                    var view = new Int16Array(buffer);
-                    for (var i = 0; i < frameCount; i++) {
-                        var word = view[i];
-                        nowBuffering[i] = ((word + 32768) % 65536 - 32768) / 32768.0;
-                    }
-                    audioChunks.push(audioOutBuffer); // Add the new audio chunk to the array
-
-                    if (!isPlaying) {
-                        playNextChunk(); // Start playing if not already playing
-                    }
+                    enqueueAudioFromProto(await event.data.arrayBuffer())
                     return false;
                 } catch (error) {
                     console.error('Error playing audio:', error);
diff --git a/examples/foundational/websocket-server/sample.py b/examples/foundational/websocket-server/sample.py
index a8d10d996..70ad30b1d 100644
--- a/examples/foundational/websocket-server/sample.py
+++ b/examples/foundational/websocket-server/sample.py
@@ -5,7 +5,7 @@
 import wave
 
 from dailyai.pipeline.frame_processor import FrameProcessor
-from dailyai.pipeline.frames import AudioFrame, EndFrame, EndPipeFrame, TextFrame
+from dailyai.pipeline.frames import AudioFrame, EndFrame, EndPipeFrame, TextFrame, TranscriptionQueueFrame
 from dailyai.pipeline.pipeline import Pipeline
 from dailyai.services.elevenlabs_ai_service import ElevenLabsTTSService
 from dailyai.services.websocket_transport_service import WebsocketTransport
@@ -19,7 +19,7 @@
 async def main():
     async with aiohttp.ClientSession() as session:
         transport = WebsocketTransport(
-            mic_enabled=True, speaker_enabled=True)
+            mic_enabled=True, speaker_enabled=True, duration_minutes=120)
 
         tts = ElevenLabsTTSService(
             aiohttp_session=session,
@@ -27,36 +27,53 @@ async def main():
             voice_id=os.getenv("ELEVENLABS_VOICE_ID"),
         )
 
-        class WriteToWav(FrameProcessor):
+        class AudioWriter(FrameProcessor):
+            SIZE = 160000
+
+            def __init__(self, *args, **kwargs):
+                super().__init__(*args, **kwargs)
+                self._buffer = bytes()
+                self._counter = 0
+
             async def process_frame(self, frame):
                 if isinstance(frame, AudioFrame):
-                    with wave.open("output.wav", "wb") as f:
-                        f.setnchannels(1)
-                        f.setsampwidth(2)
-                        f.setframerate(16000)
-                        f.writeframes(frame.data)
-                yield frame
+                    self._buffer += frame.data
+                    if len(self._buffer) >= AudioWriter.SIZE:
+                        with wave.open(f"output-{self._counter}.wav", "wb") as f:
+                            f.setnchannels(1)
+                            f.setsampwidth(2)
+                            f.setframerate(16000)
+                            f.writeframes(self._buffer)
+                            self._counter += 1
+                        self._buffer = self._buffer[AudioWriter.SIZE:]
+                    yield frame
+                else:
+                    yield frame
 
         class AudioChunker(FrameProcessor):
-            def __init__(self, *args, **kwargs):
-                super().__init__(*args, **kwargs)
+            def __init__(self):
+                super().__init__()
                 self._buffer = bytes()
 
             async def process_frame(self, frame):
                 if isinstance(frame, AudioFrame):
                     self._buffer += frame.data
-                    if len(self._buffer) >= 1600:
-                        yield AudioFrame(self._buffer[:1600])
-                        self._buffer = self._buffer[1600:]
-                elif isinstance(frame, EndFrame) or isinstance(frame, EndPipeFrame):
-                    if self._buffer:
-                        yield AudioFrame(self._buffer)
-                        self._buffer = bytes()
+                    while len(self._buffer) >= 16000:
+                        yield AudioFrame(self._buffer[:16000])
+                        self._buffer = self._buffer[16000:]
+                else:
+                    yield frame
+
+        class WhisperTranscriber(FrameProcessor):
+            async def process_frame(self, frame):
+                if isinstance(frame, TranscriptionQueueFrame):
+                    print(f"Transcribed: {frame.text}")
                 else:
                     yield frame
 
         # pipeline = Pipeline([WriteToWav(), WhisperSTTService(), tts])
-        pipeline = Pipeline([tts, AudioChunker()])
+        pipeline = Pipeline(
+            [AudioWriter(), WhisperSTTService(), WhisperTranscriber(), tts, AudioChunker()])
 
         @transport.on_connection
         async def queue_frame():
diff --git a/output-0.wav b/output-0.wav
new file mode 100644
index 000000000..8ee274a7c
Binary files /dev/null and b/output-0.wav differ
diff --git a/output-1.wav b/output-1.wav
new file mode 100644
index 000000000..390395e6a
Binary files /dev/null and b/output-1.wav differ
diff --git a/output.wav b/output.wav
new file mode 100644
index 000000000..dddec143c
Binary files /dev/null and b/output.wav differ
diff --git a/src/dailyai/pipeline/frames.py b/src/dailyai/pipeline/frames.py
index 893a50aea..511025d9b 100644
--- a/src/dailyai/pipeline/frames.py
+++ b/src/dailyai/pipeline/frames.py
@@ -298,6 +298,9 @@ def FrameFromProto(proto: frame_protos.Frame) -> Frame:
     text_frame = TextFrame(text="Hello there!")
     print(text_frame)
     print(text_frame.to_proto().SerializeToString())
+    serialized = text_frame.to_proto().SerializeToString()
+    print(type(serialized))
+    print(frame_protos.Frame.FromString(serialized))
     print(TextFrame.from_proto(text_frame.to_proto()))
 
     transcripton_frame = TranscriptionQueueFrame(
diff --git a/src/dailyai/services/websocket_transport_service.py b/src/dailyai/services/websocket_transport_service.py
index 53bd40958..b4273ce0b 100644
--- a/src/dailyai/services/websocket_transport_service.py
+++ b/src/dailyai/services/websocket_transport_service.py
@@ -55,7 +55,7 @@ async def start_server() -> None:
                 print("Server started")
                 await self._stop_server_event.wait()
                 print("Server stopped")
-            await self.receive_queue.put(EndFrame)
+            await self.receive_queue.put(EndFrame())
 
         await asyncio.gather(start_server(), timeout_task(), send_task(), pipeline.run_pipeline())
 
@@ -73,15 +73,6 @@ async def _websocket_handler(self, websocket: websockets.WebSocketServerProtocol
 
         self._websocket = websocket
         async for message in websocket:
-            """
-            generic_frame = frame_protos.Frame.ParseFromString(message)
+            generic_frame = frame_protos.Frame.FromString(message)
             frame = FrameFromProto(generic_frame)
-            if isinstance(frame, AudioFrame):
-                with wave.open("output.wav", "wb") as f:
-                    f.setnchannels(1)
-                    f.setsampwidth(2)
-                    f.setframerate(16000)
-                    f.writeframes(frame.data)
             await self.receive_queue.put(frame)
-            """
-            pass
diff --git a/src/dailyai/services/whisper_ai_services.py b/src/dailyai/services/whisper_ai_services.py
index 2ed1c3563..cc657e6cf 100644
--- a/src/dailyai/services/whisper_ai_services.py
+++ b/src/dailyai/services/whisper_ai_services.py
@@ -52,5 +52,4 @@ async def run_stt(self, audio: BinaryIO) -> str:
         res: str = ""
         for segment in segments:
             res += f"{segment.text} "
-            print("Transcription: ", segment.text)
         return res