use new webrtc server --- support iphone

Author: dgruss

README.md

@@ -16,9 +16,6 @@ SmartMicrophone is a modern, open-source web-based microphone and remote control
 
 ---
 
-## Bugs
-- **No iPhone Support:** for some reason webrtc-cli does not like webrtc streams from iphones. Currently Android and Linux clients work.
-
 ## Installation
 git clone https://github.com/dgruss/SmartMicrophone.git
 ### Supported OS: Ubuntu/Debian (recommended)
@@ -27,18 +24,14 @@ git clone https://github.com/dgruss/SmartMicrophone.git
 ```sh
 sudo add-apt-repository ppa:longsleep/golang-backports
 sudo apt update
-sudo apt update
-sudo apt install python3 python3-pip git make gcc make pkg-config libopus-dev libopusfile-dev libpulse-dev golang-go libsdl2-image-dev python3-flask pipewire pipewire-pulse
+sudo apt install python3 python3-pip git make gcc make pkg-config libopus-dev libopusfile-dev libpulse-dev golang-go libsdl2-image-dev python3-flask pipewire pipewire-pulse libpulse-dev libopus-dev
 ```
 
-#### 2. Clone the Repository, Submodules, and build webrtc-cli
+#### 2. Clone the Repository, Submodules, and build pulse-receive
 ```sh
 git clone https://github.com/dgruss/SmartMicrophone.git
-cd SmartMicrophone
-git submodule init
-git submodule update
-cd webrtc-cli
-make
+cd SmartMicrophone/pulse-receive
+go build .
 cd ..
 ```
 
@@ -47,6 +40,17 @@ cd ..
 - Some features require using [the dgruss beta3](https://github.com/dgruss/USDX/tree/beta3) version as they are not upstreamed yet
 - Make sure you know the path to your usdx directory (e.g., `/home/user/usdx`)
 
+```
+cd ~
+git clone https://github.com/dgruss/USDX.git usdx
+cd usdx
+git checkout beta3
+sudo apt install git automake make gcc fpc libsdl2-image-dev libavformat-dev libavcodec-dev libavutil-dev libswresample-dev libswscale-dev libsqlite3-dev libfreetype6-dev portaudio19-dev libportmidi-dev liblua5.3-dev libopencv-videoio-dev fonts-dejavu
+./autogen.sh
+./configure --without-portaudio --without-portmixer
+make
+```
+
 #### 4. (Optional) Configure SSL, Set up Wi-Fi Hotspot and Internet Forwarding
 - Place your SSL certificate and key files in the project directory if you want HTTPS --- this might be required to convince your phone to use WebRTC. Self-signed certificates work. Otherwise you can use a domain or subdomain you have on the Internet (give the DNS entry a short lifetime on your server!) and configure SmartMicrophone to use domain and certificates
 - See Advanced Networking below for details

server.py

@@ -119,7 +119,6 @@ def gen():
 
     return Response(stream_with_context(gen()), mimetype='text/event-stream')
 
-
 # Endpoint that merges rooms and control status and records a heartbeat
 @app.route('/status', methods=['GET'])
 def status():
@@ -138,6 +137,9 @@ def status():
 @app.before_request
 def log_incoming_request():
     try:
+        sid = session.get('session_id')
+        if sid:
+            LAST_SEEN[sid] = time.time()
         if not request.path in ['/rooms', '/status', '/control/status']:
             logger.info('Incoming request: %s %s args=%s', request.method, request.path, dict(request.args))
     except Exception:
@@ -157,7 +159,7 @@ def index():
     # maybe this person just reloaded but still has a microphone running
     if 'session_id' in session and is_youngest_session():
         mic_idx = session.get('microphone_index')
-        logger.info("Session %s is still active, automatically reconnecting microphone %s.", session.get('session_id'), mic_idx)
+        logger.debug("Session %s is still active, automatically reconnecting microphone %s.", session.get('session_id'), mic_idx)
         # only reconnect automatically if we actually have a microphone index in the session
         if mic_idx is not None:
             automatically_reconnect = True
@@ -183,8 +185,9 @@ def api():
     action = request.form.get('action')
     logger.info(f'Received action: {action}')
 
+
     if action == 'start_webrtc':
-        # Start per-player webrtc-cli using the provided SDP offer
+        # Start per-player pulse-receive using the provided SDP offer
         offer = request.form.get('offer')
         if not offer:
             return jsonify({'success': False, 'error': 'Missing offer'})
@@ -213,9 +216,6 @@ def api():
             'microphone_start_timestamp': session['microphone_start_timestamp']
         }
 
-        return jsonify({'success': True, 'answer': start_res.get('answer'), 'player_id': player_id})
-
-
     if action == 'get_assignments':
         return jsonify({'success': True, 'assignments': get_mic_assignments()})
 
@@ -301,7 +301,7 @@ def run_xdotool_command(args):
                 ids = [l.strip() for l in ws.stdout.splitlines() if l.strip()]
                 if ids:
                     ULTRASTAR_WINDOW_ID = ids[0]
-                    logger.info('Cached UltraStar window id: %s', ULTRASTAR_WINDOW_ID)
+                    logger.debug('Cached UltraStar window id: %s', ULTRASTAR_WINDOW_ID)
                 else:
                     logger.warning('No UltraStar window found via `xdotool search UltraStar`')
                     return False, 'window not found'
@@ -385,7 +385,7 @@ def control_release():
     CONTROL_OWNER = None
     CONTROL_OWNER_NAME = None
     CONTROL_TIMESTAMP = 0
-    logger.info('Control released by session %s', sid)
+    logger.debug('Control released by session %s', sid)
     return jsonify({'success': True})
 
 
@@ -1227,6 +1227,20 @@ def stale_cleanup_loop():
                 stale = []
                 for sid, last in list(LAST_SEEN.items()):
                     if now - last > 10.0:
+                        # If this session has an associated microphone process that is still alive,
+                        # treat the session as active and skip stale removal.
+                        try:
+                            mic = mgr.microphones.get(sid)
+                            if mic:
+                                try:
+                                    if mic.is_process_alive():
+                                        logger.debug('Session %s has active microphone; skipping stale removal', sid)
+                                        continue
+                                except Exception:
+                                    # If mic liveness check fails, fall back to treating as stale
+                                    logger.exception('Error checking mic liveness for session %s', sid)
+                        except Exception:
+                            logger.exception('Error inspecting microphones for session %s', sid)
                         stale.append(sid)
                 for sid in stale:
                     try:

static/script.js

@@ -735,9 +735,9 @@ document.addEventListener('DOMContentLoaded', function() {
         const pc = new RTCPeerConnection();
         window.smartMicPC = pc;
 
-        // Send local ICE candidates to server if needed (server-side webrtc-cli may not need them)
+        // Send local ICE candidates to server if needed (server-side pulse-receive may not need them)
         pc.onicecandidate = (ev) => {
-            // we do not currently send candidates to server; server's webrtc-cli is expected to be
+            // we do not currently send candidates to server; server's pulse-receive is expected to be
             // the offer/answer terminator. If needed, implement trickle ICE here.
             if (!ev.candidate) return;
             printLog('Local ICE candidate: ' + JSON.stringify(ev.candidate));