diff --git a/src/auracast/multicast.py b/src/auracast/multicast.py
index 55a82fa..ef0303e 100644
--- a/src/auracast/multicast.py
+++ b/src/auracast/multicast.py
@@ -562,11 +562,11 @@ if __name__ == "__main__":
         #big.code = 'ff'*16 # returns hci/HCI_ENCRYPTION_MODE_NOT_ACCEPTABLE_ERROR
         #big.code = '78 e5 dc f1 34 ab 42 bf c1 92 ef dd 3a fd 67 ae'
         big.precode_wav = True
-        #big.audio_source = big.audio_source.replace('.wav', '_10_16_32.lc3') #lc3 precoded files
-        #big.audio_source = read_lc3_file(big.audio_source)  # load files in advance
+        big.audio_source = big.audio_source.replace('.wav', '_10_16_32.lc3') #lc3 precoded files
+        big.audio_source = read_lc3_file(big.audio_source)  # load files in advance
 
         # --- Network_uncoded mode using NetworkAudioReceiver ---
-        big.audio_source = NetworkAudioReceiverUncoded(port=50007, samplerate=16000, channels=1, chunk_size=1024)
+        #big.audio_source = NetworkAudioReceiverUncoded(port=50007, samplerate=16000, channels=1, chunk_size=1024)
     
     # 16kHz works reliably with 3 streams
     # 24kHz is only working with 2 streams - probably airtime constraint
diff --git a/src/auracast/server/multicast_frontend.py b/src/auracast/server/multicast_frontend.py
index 0e45bb3..ed3af61 100644
--- a/src/auracast/server/multicast_frontend.py
+++ b/src/auracast/server/multicast_frontend.py
@@ -1,52 +1,110 @@
 # frontend/app.py
+from itertools import filterfalse
 import streamlit as st
 import requests
+from auracast import auracast_config
 
 # Global: desired packetization time in ms for Opus (should match backend)
 PTIME = 40
 BACKEND_URL = "http://localhost:5000"
 
 
-st.title("🎙️ WebRTC mic → backend demo")
-st.markdown("Click start and speak; watch your backend logs to see incoming RTP.")
+st.title("🎙️ Auracast Audio Mode Control")
 
-component = f"""
-<button id='go'>Start microphone</button>
-<script>
-    const go = document.getElementById('go');
-    go.onclick = async () => {{
-    go.disabled = true;
-    const pc = new RTCPeerConnection(); // No STUN needed for localhost
-    const stream = await navigator.mediaDevices.getUserMedia({{audio:true}});
-    stream.getTracks().forEach(t => pc.addTrack(t, stream));
+# Audio mode selection
+audio_mode = st.selectbox(
+    "Audio Mode",
+    ["Webapp", "Network", "Cloud Announcements"]
+)
 
-    // --- WebRTC offer/answer exchange ---
-    const offer = await pc.createOffer()
-    // Patch SDP offer to include a=ptime using global PTIME
-    let sdp = offer.sdp;
-    const ptime_line = 'a=ptime:{PTIME}';
-    const maxptime_line = 'a=maxptime:{PTIME}';
-    if (sdp.includes('a=sendrecv')) {{
-        sdp = sdp.replace('a=sendrecv', 'a=sendrecv\\n' + ptime_line + '\\n' + maxptime_line);
-    }} else {{
-        sdp += '\\n' + ptime_line + '\\n' + maxptime_line;
-    }}
-    const patched_offer = new RTCSessionDescription({{sdp, type: offer.type}})
-    await pc.setLocalDescription(patched_offer)
+if audio_mode in ["Webapp", "Network"]:
+    # Stream quality selection
+    quality = st.selectbox("Stream Quality", ["High (48kHz)", "Mid (24kHz)", "Fair (16kHz)"])
+    quality_map = {
+        "High (48kHz)": {"rate": 48000, "octets": 120},
+        "Mid (24kHz)": {"rate": 24000, "octets": 60},
+        "Fair (16kHz)": {"rate": 16000, "octets": 40},
+    }
+    stream_name = st.text_input("Channel Name", value="Broadcast0")
+    language = st.text_input("Language (ISO 639-3)", value="deu")
+    start_stream = st.button("Setup Auracast")
 
-    // Send offer to backend
-    const response = await fetch(
-        "{BACKEND_URL}/offer",
-        {{
-            method: 'POST',
-            headers: {{'Content-Type':'application/json'}},
-            body: JSON.stringify({{sdp: pc.localDescription.sdp, type: pc.localDescription.type}})
-        }}
+    if start_stream:
+        # Prepare config using the model (do NOT send qos_config, only relevant fields)
+        q = quality_map[quality]
+        
+        config = auracast_config.AuracastConfigGroup(
+            transport="auto",
+            bigs = [
+                auracast_config.AuracastBigConfig(
+                    name=stream_name,
+                    program_info=f"{stream_name} {quality}",
+                    language=language,
+                    audio_source="network" if audio_mode=="Webapp" else "network",
+                    input_format="auto",
+                    iso_que_len=64, # TODO: this should be way less to decrease delay
+                    sampling_frequency=q['rate'],
+                    octets_per_frame=q['octets'],
+                ),
+            ]
     )
-    const answer = await response.json()
-    await pc.setRemoteDescription(new RTCSessionDescription({{sdp: answer.sdp, type: answer.type}}))
-    }};
-</script>
-"""
-st.components.v1.html(component, height=80)
+        try:
+            r = requests.post(f"{BACKEND_URL}/init", json=config.model_dump())
+            if r.status_code == 200:
+                st.success("Stream initialized!")
+            else:
+                st.error(f"Failed to initialize: {r.text}")
+        except Exception as e:
+            st.error(f"Error: {e}")
+
+    if audio_mode == "Webapp":
+        st.markdown("Click start and speak; watch your backend logs to see incoming RTP.")
+        component = f"""
+        <button id='go'>Start microphone</button>
+        <script>
+            const go = document.getElementById('go');
+            go.onclick = async () => {{
+            go.disabled = true;
+            const pc = new RTCPeerConnection(); // No STUN needed for localhost
+            const stream = await navigator.mediaDevices.getUserMedia({{audio:true}});
+            stream.getTracks().forEach(t => pc.addTrack(t, stream));
+            // --- WebRTC offer/answer exchange ---
+            const offer = await pc.createOffer()
+            // Patch SDP offer to include a=ptime using global PTIME
+            let sdp = offer.sdp;
+            const ptime_line = 'a=ptime:{PTIME}';
+            const maxptime_line = 'a=maxptime:{PTIME}';
+            if (sdp.includes('a=sendrecv')) {{
+                sdp = sdp.replace('a=sendrecv', 'a=sendrecv\\n' + ptime_line + '\\n' + maxptime_line);
+            }} else {{
+                sdp += '\\n' + ptime_line + '\\n' + maxptime_line;
+            }}
+            const patched_offer = new RTCSessionDescription({{sdp, type: offer.type}})
+            await pc.setLocalDescription(patched_offer)
+            // Send offer to backend
+            const response = await fetch(
+                "{BACKEND_URL}/offer",
+                {{
+                    method: 'POST',
+                    headers: {{'Content-Type':'application/json'}},
+                    body: JSON.stringify({{sdp: pc.localDescription.sdp, type: pc.localDescription.type}})
+                }}
+            )
+            const answer = await response.json()
+            await pc.setRemoteDescription(new RTCSessionDescription({{sdp: answer.sdp, type: answer.type}}))
+            }};
+        </script>
+        """
+        st.components.v1.html(component, height=80)
+else:
+    st.header("Advertised Streams (Cloud Announcements)")
+    st.info("This feature requires backend support to list advertised streams.")
+    # Placeholder for future implementation
+    # Example: r = requests.get(f"{BACKEND_URL}/advertised_streams")
+    # if r.status_code == 200:
+    #     streams = r.json()
+    #     for s in streams:
+    #         st.write(s)
+    # else:
+    #     st.error("Could not fetch advertised streams.")
 
diff --git a/src/auracast/server/multicast_server.py b/src/auracast/server/multicast_server.py
index be916d9..e11b727 100644
--- a/src/auracast/server/multicast_server.py
+++ b/src/auracast/server/multicast_server.py
@@ -9,7 +9,7 @@ import av
 import av.audio.layout
 from typing import List, Set
 from pydantic import BaseModel
-
+import traceback
 
 app = FastAPI()
 
@@ -59,6 +59,7 @@ async def initialize(conf: auracast_config.AuracastConfigGroup):
         )
         await multicaster.init_broadcast()
     except Exception as e:
+        log.error("Exception in /init: %s", traceback.format_exc())
         raise HTTPException(status_code=500, detail=str(e))
 
 
@@ -102,7 +103,7 @@ async def get_status():
         }
 
 
-PTIME = 40
+PTIME = 160 # TODO: seems to have no effect at all
 pcs: Set[RTCPeerConnection] = set()      # keep refs so they don’t GC early
 
 class Offer(BaseModel):
@@ -174,7 +175,7 @@ async def shutdown():
 if __name__ == '__main__':
     import uvicorn
     log.basicConfig(
-        level=log.INFO,
+        level=log.DEBUG,
         format='%(module)s.py:%(lineno)d %(levelname)s: %(message)s'
     )
     uvicorn.run(app, host="0.0.0.0", port=5000)
\ No newline at end of file