feat(proxy-engine): add Rust-based outbound calling, WebRTC bridging, and voicemail handling

rust/crates/proxy-engine/src/audio_player.rs

@@ -0,0 +1,173 @@
+//! Audio player — reads a WAV file and streams it as RTP packets.
+
+use crate::rtp::{build_rtp_header, rtp_clock_increment};
+use codec_lib::{codec_sample_rate, TranscodeState};
+use std::net::SocketAddr;
+use std::path::Path;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+use tokio::time::{self, Duration};
+
+/// Play a WAV file as RTP to a destination.
+/// Returns when playback is complete.
+pub async fn play_wav_file(
+    file_path: &str,
+    socket: Arc<UdpSocket>,
+    dest: SocketAddr,
+    codec_pt: u8,
+    ssrc: u32,
+) -> Result<u32, String> {
+    let path = Path::new(file_path);
+    if !path.exists() {
+        return Err(format!("WAV file not found: {file_path}"));
+    }
+
+    // Read WAV file.
+    let mut reader =
+        hound::WavReader::open(path).map_err(|e| format!("open WAV {file_path}: {e}"))?;
+    let spec = reader.spec();
+    let wav_rate = spec.sample_rate;
+
+    // Read all samples as i16.
+    let samples: Vec<i16> = if spec.bits_per_sample == 16 {
+        reader
+            .samples::<i16>()
+            .filter_map(|s| s.ok())
+            .collect()
+    } else if spec.bits_per_sample == 32 && spec.sample_format == hound::SampleFormat::Float {
+        reader
+            .samples::<f32>()
+            .filter_map(|s| s.ok())
+            .map(|s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
+            .collect()
+    } else {
+        return Err(format!(
+            "unsupported WAV format: {}bit {:?}",
+            spec.bits_per_sample, spec.sample_format
+        ));
+    };
+
+    if samples.is_empty() {
+        return Ok(0);
+    }
+
+    // Create codec state for encoding.
+    let mut transcoder = TranscodeState::new().map_err(|e| format!("codec init: {e}"))?;
+
+    // Resample to target codec rate.
+    let target_rate = codec_sample_rate(codec_pt);
+    let resampled = if wav_rate != target_rate {
+        transcoder
+            .resample(&samples, wav_rate, target_rate)
+            .map_err(|e| format!("resample: {e}"))?
+    } else {
+        samples
+    };
+
+    // Calculate frame size (20ms of audio at target rate).
+    let frame_samples = (target_rate as usize) / 50; // 20ms = 1/50 second
+
+    // Stream as RTP at 20ms intervals.
+    let mut seq: u16 = 0;
+    let mut ts: u32 = 0;
+    let mut offset = 0;
+    let mut interval = time::interval(Duration::from_millis(20));
+    let mut frames_sent = 0u32;
+
+    while offset < resampled.len() {
+        interval.tick().await;
+
+        let end = (offset + frame_samples).min(resampled.len());
+        let frame = &resampled[offset..end];
+
+        // Pad short final frame with silence.
+        let frame_data = if frame.len() < frame_samples {
+            let mut padded = frame.to_vec();
+            padded.resize(frame_samples, 0);
+            padded
+        } else {
+            frame.to_vec()
+        };
+
+        // Encode to target codec.
+        let encoded = match transcoder.encode_from_pcm(&frame_data, codec_pt) {
+            Ok(e) if !e.is_empty() => e,
+            _ => {
+                offset += frame_samples;
+                continue;
+            }
+        };
+
+        // Build RTP packet.
+        let header = build_rtp_header(codec_pt, seq, ts, ssrc);
+        let mut packet = header.to_vec();
+        packet.extend_from_slice(&encoded);
+
+        let _ = socket.send_to(&packet, dest).await;
+
+        seq = seq.wrapping_add(1);
+        ts = ts.wrapping_add(rtp_clock_increment(codec_pt));
+        offset += frame_samples;
+        frames_sent += 1;
+    }
+
+    Ok(frames_sent)
+}
+
+/// Generate and play a beep tone (sine wave) as RTP.
+pub async fn play_beep(
+    socket: Arc<UdpSocket>,
+    dest: SocketAddr,
+    codec_pt: u8,
+    ssrc: u32,
+    start_seq: u16,
+    start_ts: u32,
+    freq_hz: u32,
+    duration_ms: u32,
+) -> Result<(u16, u32), String> {
+    let mut transcoder = TranscodeState::new().map_err(|e| format!("codec init: {e}"))?;
+    let target_rate = codec_sample_rate(codec_pt);
+    let frame_samples = (target_rate as usize) / 50;
+    let total_samples = (target_rate as usize * duration_ms as usize) / 1000;
+
+    // Generate sine wave.
+    let amplitude = 16000i16;
+    let sine: Vec<i16> = (0..total_samples)
+        .map(|i| {
+            let t = i as f64 / target_rate as f64;
+            (amplitude as f64 * (2.0 * std::f64::consts::PI * freq_hz as f64 * t).sin()) as i16
+        })
+        .collect();
+
+    let mut seq = start_seq;
+    let mut ts = start_ts;
+    let mut offset = 0;
+    let mut interval = time::interval(Duration::from_millis(20));
+
+    while offset < sine.len() {
+        interval.tick().await;
+
+        let end = (offset + frame_samples).min(sine.len());
+        let mut frame = sine[offset..end].to_vec();
+        frame.resize(frame_samples, 0);
+
+        let encoded = match transcoder.encode_from_pcm(&frame, codec_pt) {
+            Ok(e) if !e.is_empty() => e,
+            _ => {
+                offset += frame_samples;
+                continue;
+            }
+        };
+
+        let header = build_rtp_header(codec_pt, seq, ts, ssrc);
+        let mut packet = header.to_vec();
+        packet.extend_from_slice(&encoded);
+        let _ = socket.send_to(&packet, dest).await;
+
+        seq = seq.wrapping_add(1);
+        ts = ts.wrapping_add(rtp_clock_increment(codec_pt));
+        offset += frame_samples;
+    }
+
+    Ok((seq, ts))
+}

rust/crates/proxy-engine/src/call_manager.rs

@@ -241,15 +241,25 @@ impl CallManager {
             .unwrap_or("")
             .to_string();
 
-        // Resolve target device (first registered device for now).
+        // Resolve target device (first registered device).
         let device_addr = match self.resolve_first_device(config, registrar) {
             Some(addr) => addr,
             None => {
-                // No device available — could route to voicemail
-                // For now, send 480 Temporarily Unavailable.
-                let resp = SipMessage::create_response(480, "Temporarily Unavailable", invite, None);
-                let _ = socket.send_to(&resp.serialize(), from_addr).await;
-                return None;
+                // No device registered — route to voicemail.
+                return self
+                    .route_to_voicemail(
+                        &call_id,
+                        invite,
+                        from_addr,
+                        &caller_number,
+                        provider_id,
+                        provider_config,
+                        config,
+                        rtp_pool,
+                        socket,
+                        public_ip,
+                    )
+                    .await;
             }
         };
 
@@ -487,6 +497,225 @@ impl CallManager {
         self.calls.contains_key(sip_call_id)
     }
 
+    // --- Dashboard outbound call (B2BUA) ---
+
+    /// Initiate an outbound call from the dashboard.
+    /// Builds an INVITE from scratch and sends it to the provider.
+    /// The browser connects separately via WebRTC and gets linked to this call.
+    pub async fn make_outbound_call(
+        &mut self,
+        number: &str,
+        provider_config: &ProviderConfig,
+        config: &AppConfig,
+        rtp_pool: &mut RtpPortPool,
+        socket: &UdpSocket,
+        public_ip: Option<&str>,
+        registered_aor: &str,
+    ) -> Option<String> {
+        let call_id = self.next_call_id();
+        let lan_ip = &config.proxy.lan_ip;
+        let lan_port = config.proxy.lan_port;
+        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());
+
+        let provider_dest: SocketAddr = match provider_config.outbound_proxy.to_socket_addr() {
+            Some(a) => a,
+            None => return None,
+        };
+
+        // Allocate RTP port for the provider leg.
+        let rtp_alloc = match rtp_pool.allocate().await {
+            Some(a) => a,
+            None => return None,
+        };
+
+        // Build the SIP Call-ID for this new dialog.
+        let sip_call_id = sip_proto::helpers::generate_call_id(None);
+
+        // Build SDP offer.
+        let sdp = sip_proto::helpers::build_sdp(&sip_proto::helpers::SdpOptions {
+            ip: pub_ip,
+            port: rtp_alloc.port,
+            payload_types: &provider_config.codecs,
+            ..Default::default()
+        });
+
+        // Build INVITE.
+        let to_uri = format!("sip:{number}@{}", provider_config.domain);
+        let invite = SipMessage::create_request(
+            "INVITE",
+            &to_uri,
+            sip_proto::message::RequestOptions {
+                via_host: pub_ip.to_string(),
+                via_port: lan_port,
+                via_transport: None,
+                via_branch: Some(sip_proto::helpers::generate_branch()),
+                from_uri: registered_aor.to_string(),
+                from_display_name: None,
+                from_tag: Some(sip_proto::helpers::generate_tag()),
+                to_uri: to_uri.clone(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some(sip_call_id.clone()),
+                cseq: Some(1),
+                contact: Some(format!("<sip:{pub_ip}:{lan_port}>")),
+                max_forwards: Some(70),
+                body: Some(sdp),
+                content_type: Some("application/sdp".to_string()),
+                extra_headers: Some(vec![
+                    ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
+                    ("Allow".to_string(), "INVITE, ACK, OPTIONS, CANCEL, BYE, INFO".to_string()),
+                ]),
+            },
+        );
+
+        // Send INVITE to provider.
+        let _ = socket.send_to(&invite.serialize(), provider_dest).await;
+
+        // Create call entry — device_addr is a dummy (WebRTC will be linked later).
+        let dummy_addr: SocketAddr = "127.0.0.1:0".parse().unwrap();
+        let call = PassthroughCall {
+            id: call_id.clone(),
+            sip_call_id: sip_call_id.clone(),
+            state: CallState::SettingUp,
+            direction: CallDirection::Outbound,
+            created_at: Instant::now(),
+            caller_number: Some(registered_aor.to_string()),
+            callee_number: Some(number.to_string()),
+            provider_id: provider_config.id.clone(),
+            provider_addr: provider_dest,
+            provider_media: None,
+            device_addr: dummy_addr,
+            device_media: None,
+            rtp_port: rtp_alloc.port,
+            rtp_socket: rtp_alloc.socket.clone(),
+            pkt_from_device: 0,
+            pkt_from_provider: 0,
+        };
+        self.calls.insert(sip_call_id, call);
+
+        Some(call_id)
+    }
+
+    // --- Voicemail ---
+
+    /// Route a call to voicemail: answer the INVITE, play greeting, record message.
+    async fn route_to_voicemail(
+        &mut self,
+        call_id: &str,
+        invite: &SipMessage,
+        from_addr: SocketAddr,
+        caller_number: &str,
+        provider_id: &str,
+        provider_config: &ProviderConfig,
+        config: &AppConfig,
+        rtp_pool: &mut RtpPortPool,
+        socket: &UdpSocket,
+        public_ip: Option<&str>,
+    ) -> Option<String> {
+        let lan_ip = &config.proxy.lan_ip;
+        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());
+
+        // Allocate RTP port for the voicemail session.
+        let rtp_alloc = match rtp_pool.allocate().await {
+            Some(a) => a,
+            None => {
+                let resp =
+                    SipMessage::create_response(503, "Service Unavailable", invite, None);
+                let _ = socket.send_to(&resp.serialize(), from_addr).await;
+                return None;
+            }
+        };
+
+        // Determine provider's preferred codec.
+        let codec_pt = provider_config.codecs.first().copied().unwrap_or(9); // default G.722
+
+        // Build SDP with our RTP port.
+        let sdp = sip_proto::helpers::build_sdp(&sip_proto::helpers::SdpOptions {
+            ip: pub_ip,
+            port: rtp_alloc.port,
+            payload_types: &provider_config.codecs,
+            ..Default::default()
+        });
+
+        // Answer the INVITE with 200 OK.
+        let response = SipMessage::create_response(
+            200,
+            "OK",
+            invite,
+            Some(sip_proto::message::ResponseOptions {
+                to_tag: Some(sip_proto::helpers::generate_tag()),
+                contact: Some(format!("<sip:{}:{}>", lan_ip, config.proxy.lan_port)),
+                body: Some(sdp),
+                content_type: Some("application/sdp".to_string()),
+                ..Default::default()
+            }),
+        );
+        let _ = socket.send_to(&response.serialize(), from_addr).await;
+
+        // Extract provider media from original SDP.
+        let provider_media = if invite.has_sdp_body() {
+            sip_proto::helpers::parse_sdp_endpoint(&invite.body)
+                .and_then(|ep| format!("{}:{}", ep.address, ep.port).parse().ok())
+        } else {
+            Some(from_addr) // fallback to signaling address
+        };
+        let provider_media = provider_media.unwrap_or(from_addr);
+
+        // Create a voicemail call entry for BYE routing.
+        let call = PassthroughCall {
+            id: call_id.to_string(),
+            sip_call_id: invite.call_id().to_string(),
+            state: CallState::Voicemail,
+            direction: CallDirection::Inbound,
+            created_at: std::time::Instant::now(),
+            caller_number: Some(caller_number.to_string()),
+            callee_number: None,
+            provider_id: provider_id.to_string(),
+            provider_addr: from_addr,
+            provider_media: Some(provider_media),
+            device_addr: from_addr, // no device — just use provider addr as placeholder
+            device_media: None,
+            rtp_port: rtp_alloc.port,
+            rtp_socket: rtp_alloc.socket.clone(),
+            pkt_from_device: 0,
+            pkt_from_provider: 0,
+        };
+        self.calls.insert(invite.call_id().to_string(), call);
+
+        // Build recording file path.
+        let timestamp = std::time::SystemTime::now()
+            .duration_since(std::time::UNIX_EPOCH)
+            .unwrap_or_default()
+            .as_millis();
+        let recording_dir = format!(".nogit/voicemail/default");
+        let recording_path = format!("{recording_dir}/msg-{timestamp}.wav");
+
+        // Look for a greeting WAV file.
+        let greeting_wav = find_greeting_wav();
+
+        // Spawn the voicemail session.
+        let out_tx = self.out_tx.clone();
+        let call_id_owned = call_id.to_string();
+        let caller_owned = caller_number.to_string();
+        let rtp_socket = rtp_alloc.socket;
+        tokio::spawn(async move {
+            crate::voicemail::run_voicemail_session(
+                rtp_socket,
+                provider_media,
+                codec_pt,
+                greeting_wav,
+                recording_path,
+                120_000, // max 120 seconds
+                call_id_owned,
+                caller_owned,
+                out_tx,
+            )
+            .await;
+        });
+
+        Some(call_id.to_string())
+    }
+
     // --- Internal helpers ---
 
     fn resolve_first_device(&self, config: &AppConfig, registrar: &Registrar) -> Option<SocketAddr> {
@@ -495,10 +724,25 @@ impl CallManager {
                 return Some(addr);
             }
         }
-        None
+        None // No device registered — caller goes to voicemail.
     }
 }
 
+/// Find a voicemail greeting WAV file.
+fn find_greeting_wav() -> Option<String> {
+    // Check common locations for a pre-generated greeting.
+    let candidates = [
+        ".nogit/voicemail/default/greeting.wav",
+        ".nogit/voicemail/greeting.wav",
+    ];
+    for path in &candidates {
+        if std::path::Path::new(path).exists() {
+            return Some(path.to_string());
+        }
+    }
+    None // No greeting found — voicemail will just play the beep.
+}
+
 /// Rewrite SDP for provider→device direction (use LAN IP).
 fn rewrite_sdp_for_device(msg: &mut SipMessage, lan_ip: &str, rtp_port: u16) {
     if msg.has_sdp_body() {

rust/crates/proxy-engine/src/config.rs

@@ -14,8 +14,18 @@ pub struct Endpoint {
 }
 
 impl Endpoint {
+    /// Resolve to a SocketAddr. Handles both IP addresses and hostnames.
     pub fn to_socket_addr(&self) -> Option<SocketAddr> {
-        format!("{}:{}", self.address, self.port).parse().ok()
+        // Try direct parse first (IP address).
+        if let Ok(addr) = format!("{}:{}", self.address, self.port).parse() {
+            return Some(addr);
+        }
+        // DNS resolution for hostnames.
+        use std::net::ToSocketAddrs;
+        format!("{}:{}", self.address, self.port)
+            .to_socket_addrs()
+            .ok()
+            .and_then(|mut addrs| addrs.next())
     }
 }
 

rust/crates/proxy-engine/src/main.rs

@@ -6,15 +6,19 @@
 ///
 /// No raw SIP ever touches TypeScript.
 
+mod audio_player;
 mod call;
 mod call_manager;
 mod config;
 mod dtmf;
 mod ipc;
 mod provider;
+mod recorder;
 mod registrar;
 mod rtp;
 mod sip_transport;
+mod voicemail;
+mod webrtc_engine;
 
 use crate::call_manager::CallManager;
 use crate::config::AppConfig;
@@ -23,6 +27,7 @@ use crate::provider::ProviderManager;
 use crate::registrar::Registrar;
 use crate::rtp::RtpPortPool;
 use crate::sip_transport::SipTransport;
+use crate::webrtc_engine::WebRtcEngine;
 use sip_proto::message::SipMessage;
 use std::net::SocketAddr;
 use std::sync::Arc;
@@ -37,6 +42,7 @@ struct ProxyEngine {
     provider_mgr: ProviderManager,
     registrar: Registrar,
     call_mgr: CallManager,
+    webrtc: WebRtcEngine,
     rtp_pool: Option<RtpPortPool>,
     out_tx: OutTx,
 }
@@ -49,6 +55,7 @@ impl ProxyEngine {
             provider_mgr: ProviderManager::new(out_tx.clone()),
             registrar: Registrar::new(out_tx.clone()),
             call_mgr: CallManager::new(out_tx.clone()),
+            webrtc: WebRtcEngine::new(out_tx.clone()),
             rtp_pool: None,
             out_tx,
         }
@@ -111,7 +118,12 @@ async fn handle_command(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: Co
     match cmd.method.as_str() {
         "configure" => handle_configure(engine, out_tx, &cmd).await,
         "hangup" => handle_hangup(engine, out_tx, &cmd).await,
+        "make_call" => handle_make_call(engine, out_tx, &cmd).await,
         "get_status" => handle_get_status(engine, out_tx, &cmd).await,
+        "webrtc_offer" => handle_webrtc_offer(engine, out_tx, &cmd).await,
+        "webrtc_ice" => handle_webrtc_ice(engine, out_tx, &cmd).await,
+        "webrtc_link" => handle_webrtc_link(engine, out_tx, &cmd).await,
+        "webrtc_close" => handle_webrtc_close(engine, out_tx, &cmd).await,
         _ => respond_err(out_tx, &cmd.id, &format!("unknown command: {}", cmd.method)),
     }
 }
@@ -413,6 +425,78 @@ async fn handle_get_status(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd:
     respond_ok(out_tx, &cmd.id, serde_json::json!({ "calls": calls }));
 }
 
+/// Handle `make_call` — initiate an outbound call to a number via a provider.
+async fn handle_make_call(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let number = match cmd.params.get("number").and_then(|v| v.as_str()) {
+        Some(n) => n.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing number"); return; }
+    };
+    let provider_id = cmd.params.get("provider_id").and_then(|v| v.as_str());
+
+    let mut eng = engine.lock().await;
+    let config_ref = match &eng.config {
+        Some(c) => c.clone(),
+        None => { respond_err(out_tx, &cmd.id, "not configured"); return; }
+    };
+
+    // Resolve provider.
+    let provider_config = if let Some(pid) = provider_id {
+        config_ref.providers.iter().find(|p| p.id == pid).cloned()
+    } else {
+        // Use route resolution or first provider.
+        let route = config_ref.resolve_outbound_route(&number, None, &|_| true);
+        route.map(|r| r.provider)
+    };
+
+    let provider_config = match provider_config {
+        Some(p) => p,
+        None => { respond_err(out_tx, &cmd.id, "no provider available"); return; }
+    };
+
+    // Get public IP and registered AOR from provider state.
+    let (public_ip, registered_aor) = if let Some(ps_arc) = eng.provider_mgr.find_by_address(
+        &provider_config.outbound_proxy.to_socket_addr().unwrap_or_else(|| "0.0.0.0:0".parse().unwrap())
+    ).await {
+        let ps = ps_arc.lock().await;
+        (ps.public_ip.clone(), ps.registered_aor.clone())
+    } else {
+        // Fallback — construct AOR from config.
+        (None, format!("sip:{}@{}", provider_config.username, provider_config.domain))
+    };
+
+    let socket = match &eng.transport {
+        Some(t) => t.socket(),
+        None => { respond_err(out_tx, &cmd.id, "not initialized"); return; }
+    };
+
+    let ProxyEngine { ref mut call_mgr, ref mut rtp_pool, .. } = *eng;
+    let rtp_pool = rtp_pool.as_mut().unwrap();
+
+    let call_id = call_mgr.make_outbound_call(
+        &number,
+        &provider_config,
+        &config_ref,
+        rtp_pool,
+        &socket,
+        public_ip.as_deref(),
+        &registered_aor,
+    ).await;
+
+    match call_id {
+        Some(id) => {
+            emit_event(out_tx, "outbound_call_started", serde_json::json!({
+                "call_id": id,
+                "number": number,
+                "provider_id": provider_config.id,
+            }));
+            respond_ok(out_tx, &cmd.id, serde_json::json!({ "call_id": id }));
+        }
+        None => {
+            respond_err(out_tx, &cmd.id, "call origination failed — provider not registered or no ports available");
+        }
+    }
+}
+
 /// Handle the `hangup` command.
 async fn handle_hangup(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
     let call_id = match cmd.params.get("call_id").and_then(|v| v.as_str()) {
@@ -438,3 +522,105 @@ async fn handle_hangup(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Co
         respond_err(out_tx, &cmd.id, &format!("call {call_id} not found"));
     }
 }
+
+/// Handle `webrtc_offer` — browser sends SDP offer, we create PeerConnection and return answer.
+async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+    let offer_sdp = match cmd.params.get("sdp").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing sdp"); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    match eng.webrtc.handle_offer(&session_id, &offer_sdp).await {
+        Ok(answer_sdp) => {
+            respond_ok(out_tx, &cmd.id, serde_json::json!({
+                "session_id": session_id,
+                "sdp": answer_sdp,
+            }));
+        }
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}
+
+/// Handle `webrtc_ice` — forward ICE candidate from browser to Rust PeerConnection.
+async fn handle_webrtc_ice(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+    let candidate = cmd.params.get("candidate").and_then(|v| v.as_str()).unwrap_or("");
+    let sdp_mid = cmd.params.get("sdp_mid").and_then(|v| v.as_str());
+    let sdp_mline_index = cmd.params.get("sdp_mline_index").and_then(|v| v.as_u64()).map(|v| v as u16);
+
+    let eng = engine.lock().await;
+    match eng.webrtc.add_ice_candidate(&session_id, candidate, sdp_mid, sdp_mline_index).await {
+        Ok(()) => respond_ok(out_tx, &cmd.id, serde_json::json!({})),
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}
+
+/// Handle `webrtc_link` — link a WebRTC session to a SIP call for audio bridging.
+async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+    let call_id = match cmd.params.get("call_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing call_id"); return; }
+    };
+    let provider_addr = match cmd.params.get("provider_media_addr").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing provider_media_addr"); return; }
+    };
+    let provider_port = match cmd.params.get("provider_media_port").and_then(|v| v.as_u64()) {
+        Some(p) => p as u16,
+        None => { respond_err(out_tx, &cmd.id, "missing provider_media_port"); return; }
+    };
+    let sip_pt = cmd.params.get("sip_pt").and_then(|v| v.as_u64()).unwrap_or(9) as u8;
+
+    let provider_media: SocketAddr = match format!("{provider_addr}:{provider_port}").parse() {
+        Ok(a) => a,
+        Err(e) => { respond_err(out_tx, &cmd.id, &format!("bad address: {e}")); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    let sip_socket = match &eng.transport {
+        Some(t) => t.socket(),
+        None => { respond_err(out_tx, &cmd.id, "not initialized"); return; }
+    };
+
+    let bridge_info = crate::webrtc_engine::SipBridgeInfo {
+        provider_media,
+        sip_pt,
+        sip_socket,
+    };
+
+    if eng.webrtc.link_to_sip(&session_id, &call_id, bridge_info).await {
+        respond_ok(out_tx, &cmd.id, serde_json::json!({
+            "session_id": session_id,
+            "call_id": call_id,
+            "bridged": true,
+        }));
+    } else {
+        respond_err(out_tx, &cmd.id, &format!("session {session_id} not found"));
+    }
+}
+
+/// Handle `webrtc_close` — close a WebRTC session.
+async fn handle_webrtc_close(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    match eng.webrtc.close_session(&session_id).await {
+        Ok(()) => respond_ok(out_tx, &cmd.id, serde_json::json!({})),
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}

rust/crates/proxy-engine/src/recorder.rs

@@ -0,0 +1,132 @@
+//! Audio recorder — receives RTP packets and writes a WAV file.
+
+use codec_lib::TranscodeState;
+use std::path::Path;
+
+/// Active recording session.
+pub struct Recorder {
+    writer: hound::WavWriter<std::io::BufWriter<std::fs::File>>,
+    transcoder: TranscodeState,
+    source_pt: u8,
+    total_samples: u64,
+    sample_rate: u32,
+    max_samples: Option<u64>,
+    file_path: String,
+}
+
+impl Recorder {
+    /// Create a new recorder that writes to a WAV file.
+    /// `source_pt` is the RTP payload type of the incoming audio.
+    /// `max_duration_ms` optionally limits the recording length.
+    pub fn new(
+        file_path: &str,
+        source_pt: u8,
+        max_duration_ms: Option<u64>,
+    ) -> Result<Self, String> {
+        // Ensure parent directory exists.
+        if let Some(parent) = Path::new(file_path).parent() {
+            std::fs::create_dir_all(parent)
+                .map_err(|e| format!("create dir: {e}"))?;
+        }
+
+        let sample_rate = 8000u32; // Record at 8kHz (standard telephony)
+        let spec = hound::WavSpec {
+            channels: 1,
+            sample_rate,
+            bits_per_sample: 16,
+            sample_format: hound::SampleFormat::Int,
+        };
+
+        let writer = hound::WavWriter::create(file_path, spec)
+            .map_err(|e| format!("create WAV {file_path}: {e}"))?;
+
+        let transcoder = TranscodeState::new().map_err(|e| format!("codec init: {e}"))?;
+
+        let max_samples = max_duration_ms.map(|ms| (sample_rate as u64 * ms) / 1000);
+
+        Ok(Self {
+            writer,
+            transcoder,
+            source_pt,
+            total_samples: 0,
+            sample_rate,
+            max_samples,
+            file_path: file_path.to_string(),
+        })
+    }
+
+    /// Process an incoming RTP packet (full packet with header).
+    /// Returns true if recording should continue, false if max duration reached.
+    pub fn process_rtp(&mut self, data: &[u8]) -> bool {
+        if data.len() <= 12 {
+            return true; // Too short, skip.
+        }
+
+        let pt = data[1] & 0x7F;
+        // Skip telephone-event (DTMF) packets.
+        if pt == 101 {
+            return true;
+        }
+
+        let payload = &data[12..];
+        if payload.is_empty() {
+            return true;
+        }
+
+        // Decode to PCM.
+        let (pcm, rate) = match self.transcoder.decode_to_pcm(payload, self.source_pt) {
+            Ok(result) => result,
+            Err(_) => return true, // Decode failed, skip packet.
+        };
+
+        // Resample to 8kHz if needed.
+        let pcm_8k = if rate != self.sample_rate {
+            match self.transcoder.resample(&pcm, rate, self.sample_rate) {
+                Ok(r) => r,
+                Err(_) => return true,
+            }
+        } else {
+            pcm
+        };
+
+        // Write samples.
+        for &sample in &pcm_8k {
+            if let Err(_) = self.writer.write_sample(sample) {
+                return false;
+            }
+            self.total_samples += 1;
+
+            if let Some(max) = self.max_samples {
+                if self.total_samples >= max {
+                    return false; // Max duration reached.
+                }
+            }
+        }
+
+        true
+    }
+
+    /// Stop recording and finalize the WAV file.
+    pub fn stop(self) -> RecordingResult {
+        let duration_ms = if self.sample_rate > 0 {
+            (self.total_samples * 1000) / self.sample_rate as u64
+        } else {
+            0
+        };
+
+        // Writer is finalized on drop (writes RIFF header sizes).
+        drop(self.writer);
+
+        RecordingResult {
+            file_path: self.file_path,
+            duration_ms,
+            total_samples: self.total_samples,
+        }
+    }
+}
+
+pub struct RecordingResult {
+    pub file_path: String,
+    pub duration_ms: u64,
+    pub total_samples: u64,
+}

rust/crates/proxy-engine/src/voicemail.rs

@@ -0,0 +1,137 @@
+//! Voicemail session — answer → play greeting → beep → record → done.
+
+use crate::audio_player::{play_beep, play_wav_file};
+use crate::ipc::{emit_event, OutTx};
+use crate::recorder::Recorder;
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+
+/// Run a voicemail session on an RTP port.
+///
+/// 1. Plays the greeting WAV file to the caller
+/// 2. Plays a beep tone
+/// 3. Records the caller's message until BYE or max duration
+///
+/// The RTP receive loop is separate — it feeds packets to the recorder
+/// via the returned channel.
+pub async fn run_voicemail_session(
+    rtp_socket: Arc<UdpSocket>,
+    provider_media: SocketAddr,
+    codec_pt: u8,
+    greeting_wav: Option<String>,
+    recording_path: String,
+    max_recording_ms: u64,
+    call_id: String,
+    caller_number: String,
+    out_tx: OutTx,
+) {
+    let ssrc: u32 = rand::random();
+
+    emit_event(
+        &out_tx,
+        "voicemail_started",
+        serde_json::json!({
+            "call_id": call_id,
+            "caller_number": caller_number,
+        }),
+    );
+
+    // Step 1: Play greeting.
+    let mut next_seq: u16 = 0;
+    let mut next_ts: u32 = 0;
+
+    if let Some(wav_path) = &greeting_wav {
+        match play_wav_file(wav_path, rtp_socket.clone(), provider_media, codec_pt, ssrc).await {
+            Ok(frames) => {
+                next_seq = frames as u16;
+                next_ts = frames * crate::rtp::rtp_clock_increment(codec_pt);
+            }
+            Err(e) => {
+                emit_event(
+                    &out_tx,
+                    "voicemail_error",
+                    serde_json::json!({ "call_id": call_id, "error": format!("greeting: {e}") }),
+                );
+            }
+        }
+    }
+
+    // Step 2: Play beep (1kHz, 500ms).
+    match play_beep(
+        rtp_socket.clone(),
+        provider_media,
+        codec_pt,
+        ssrc,
+        next_seq,
+        next_ts,
+        1000,
+        500,
+    )
+    .await
+    {
+        Ok((_seq, _ts)) => {}
+        Err(e) => {
+            emit_event(
+                &out_tx,
+                "voicemail_error",
+                serde_json::json!({ "call_id": call_id, "error": format!("beep: {e}") }),
+            );
+        }
+    }
+
+    // Step 3: Record incoming audio.
+    let recorder = match Recorder::new(&recording_path, codec_pt, Some(max_recording_ms)) {
+        Ok(r) => r,
+        Err(e) => {
+            emit_event(
+                &out_tx,
+                "voicemail_error",
+                serde_json::json!({ "call_id": call_id, "error": format!("recorder: {e}") }),
+            );
+            return;
+        }
+    };
+
+    // Receive RTP and feed to recorder.
+    let result = record_from_socket(rtp_socket, recorder, max_recording_ms).await;
+
+    // Step 4: Done — emit recording result.
+    emit_event(
+        &out_tx,
+        "recording_done",
+        serde_json::json!({
+            "call_id": call_id,
+            "file_path": result.file_path,
+            "duration_ms": result.duration_ms,
+            "caller_number": caller_number,
+        }),
+    );
+}
+
+/// Read RTP packets from the socket and feed them to the recorder.
+/// Returns when the socket errors out (BYE closes the call/socket)
+/// or max duration is reached.
+async fn record_from_socket(
+    socket: Arc<UdpSocket>,
+    mut recorder: Recorder,
+    max_ms: u64,
+) -> crate::recorder::RecordingResult {
+    let mut buf = vec![0u8; 65535];
+    let deadline = tokio::time::Instant::now() + tokio::time::Duration::from_millis(max_ms + 2000);
+
+    loop {
+        let timeout = tokio::time::timeout_at(deadline, socket.recv_from(&mut buf));
+        match timeout.await {
+            Ok(Ok((n, _addr))) => {
+                if !recorder.process_rtp(&buf[..n]) {
+                    break; // Max duration reached.
+                }
+            }
+            Ok(Err(_)) => break,     // Socket error (closed).
+            Err(_) => break,          // Timeout (max duration + grace).
+        }
+    }
+
+    recorder.stop()
+}

rust/crates/proxy-engine/src/webrtc_engine.rs

@@ -0,0 +1,389 @@
+//! WebRTC engine — manages browser PeerConnections with SIP audio bridging.
+//!
+//! Browser Opus audio → Rust PeerConnection → transcode via codec-lib → SIP RTP
+//! SIP RTP → transcode via codec-lib → Rust PeerConnection → Browser Opus
+
+use crate::ipc::{emit_event, OutTx};
+use crate::rtp::{build_rtp_header, rtp_clock_increment};
+use codec_lib::{TranscodeState, PT_G722, PT_OPUS};
+use std::collections::HashMap;
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+use tokio::sync::Mutex;
+use webrtc::api::media_engine::MediaEngine;
+use webrtc::api::APIBuilder;
+use webrtc::ice_transport::ice_candidate::RTCIceCandidateInit;
+use webrtc::peer_connection::configuration::RTCConfiguration;
+use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
+use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
+use webrtc::peer_connection::RTCPeerConnection;
+use webrtc::rtp_transceiver::rtp_codec::RTCRtpCodecCapability;
+use webrtc::track::track_local::track_local_static_rtp::TrackLocalStaticRTP;
+use webrtc::track::track_local::{TrackLocal, TrackLocalWriter};
+
+/// SIP-side bridge info for a WebRTC session.
+#[derive(Clone)]
+pub struct SipBridgeInfo {
+    /// Provider's media endpoint (RTP destination).
+    pub provider_media: SocketAddr,
+    /// Provider's codec payload type (e.g. 9 for G.722).
+    pub sip_pt: u8,
+    /// The SIP UDP socket for sending RTP to the provider.
+    pub sip_socket: Arc<UdpSocket>,
+}
+
+/// A managed WebRTC session.
+struct WebRtcSession {
+    pc: Arc<RTCPeerConnection>,
+    local_track: Arc<TrackLocalStaticRTP>,
+    call_id: Option<String>,
+    /// SIP bridge — set when the session is linked to a call.
+    sip_bridge: Arc<Mutex<Option<SipBridgeInfo>>>,
+}
+
+/// Manages all WebRTC sessions.
+pub struct WebRtcEngine {
+    sessions: HashMap<String, WebRtcSession>,
+    out_tx: OutTx,
+}
+
+impl WebRtcEngine {
+    pub fn new(out_tx: OutTx) -> Self {
+        Self {
+            sessions: HashMap::new(),
+            out_tx,
+        }
+    }
+
+    /// Handle a WebRTC offer from a browser.
+    pub async fn handle_offer(
+        &mut self,
+        session_id: &str,
+        offer_sdp: &str,
+    ) -> Result<String, String> {
+        let mut media_engine = MediaEngine::default();
+        media_engine
+            .register_default_codecs()
+            .map_err(|e| format!("register codecs: {e}"))?;
+
+        let api = APIBuilder::new()
+            .with_media_engine(media_engine)
+            .build();
+
+        let config = RTCConfiguration {
+            ice_servers: vec![],
+            ..Default::default()
+        };
+
+        let pc = api
+            .new_peer_connection(config)
+            .await
+            .map_err(|e| format!("create peer connection: {e}"))?;
+        let pc = Arc::new(pc);
+
+        // Local audio track for sending audio to browser (Opus).
+        let local_track = Arc::new(TrackLocalStaticRTP::new(
+            RTCRtpCodecCapability {
+                mime_type: "audio/opus".to_string(),
+                clock_rate: 48000,
+                channels: 1,
+                ..Default::default()
+            },
+            "audio".to_string(),
+            "siprouter".to_string(),
+        ));
+
+        let _sender = pc
+            .add_track(local_track.clone() as Arc<dyn TrackLocal + Send + Sync>)
+            .await
+            .map_err(|e| format!("add track: {e}"))?;
+
+        // Shared SIP bridge info (populated when linked to a call).
+        let sip_bridge: Arc<Mutex<Option<SipBridgeInfo>>> = Arc::new(Mutex::new(None));
+
+        // ICE candidate handler.
+        let out_tx_ice = self.out_tx.clone();
+        let sid_ice = session_id.to_string();
+        pc.on_ice_candidate(Box::new(move |candidate| {
+            let out_tx = out_tx_ice.clone();
+            let sid = sid_ice.clone();
+            Box::pin(async move {
+                if let Some(c) = candidate {
+                    if let Ok(json) = c.to_json() {
+                        emit_event(
+                            &out_tx,
+                            "webrtc_ice_candidate",
+                            serde_json::json!({
+                                "session_id": sid,
+                                "candidate": json.candidate,
+                                "sdp_mid": json.sdp_mid,
+                                "sdp_mline_index": json.sdp_mline_index,
+                            }),
+                        );
+                    }
+                }
+            })
+        }));
+
+        // Connection state handler.
+        let out_tx_state = self.out_tx.clone();
+        let sid_state = session_id.to_string();
+        pc.on_peer_connection_state_change(Box::new(move |state| {
+            let out_tx = out_tx_state.clone();
+            let sid = sid_state.clone();
+            Box::pin(async move {
+                let state_str = match state {
+                    RTCPeerConnectionState::Connected => "connected",
+                    RTCPeerConnectionState::Disconnected => "disconnected",
+                    RTCPeerConnectionState::Failed => "failed",
+                    RTCPeerConnectionState::Closed => "closed",
+                    RTCPeerConnectionState::New => "new",
+                    RTCPeerConnectionState::Connecting => "connecting",
+                    _ => "unknown",
+                };
+                emit_event(
+                    &out_tx,
+                    "webrtc_state",
+                    serde_json::json!({ "session_id": sid, "state": state_str }),
+                );
+            })
+        }));
+
+        // Track handler — receives Opus audio from the browser.
+        // When SIP bridge is set, transcodes and forwards to provider.
+        let out_tx_track = self.out_tx.clone();
+        let sid_track = session_id.to_string();
+        let sip_bridge_for_track = sip_bridge.clone();
+        pc.on_track(Box::new(move |track, _receiver, _transceiver| {
+            let out_tx = out_tx_track.clone();
+            let sid = sid_track.clone();
+            let bridge = sip_bridge_for_track.clone();
+            Box::pin(async move {
+                let codec_info = track.codec();
+                emit_event(
+                    &out_tx,
+                    "webrtc_track",
+                    serde_json::json!({
+                        "session_id": sid,
+                        "kind": track.kind().to_string(),
+                        "codec": codec_info.capability.mime_type,
+                    }),
+                );
+
+                // Spawn the browser→SIP audio forwarding task.
+                tokio::spawn(browser_to_sip_loop(track, bridge, out_tx, sid));
+            })
+        }));
+
+        // Set remote offer.
+        let offer = RTCSessionDescription::offer(offer_sdp.to_string())
+            .map_err(|e| format!("parse offer: {e}"))?;
+        pc.set_remote_description(offer)
+            .await
+            .map_err(|e| format!("set remote description: {e}"))?;
+
+        // Create answer.
+        let answer = pc
+            .create_answer(None)
+            .await
+            .map_err(|e| format!("create answer: {e}"))?;
+        let answer_sdp = answer.sdp.clone();
+        pc.set_local_description(answer)
+            .await
+            .map_err(|e| format!("set local description: {e}"))?;
+
+        self.sessions.insert(
+            session_id.to_string(),
+            WebRtcSession {
+                pc,
+                local_track,
+                call_id: None,
+                sip_bridge,
+            },
+        );
+
+        Ok(answer_sdp)
+    }
+
+    /// Link a WebRTC session to a SIP call — sets up the audio bridge.
+    pub async fn link_to_sip(
+        &mut self,
+        session_id: &str,
+        call_id: &str,
+        bridge_info: SipBridgeInfo,
+    ) -> bool {
+        if let Some(session) = self.sessions.get_mut(session_id) {
+            session.call_id = Some(call_id.to_string());
+            let mut bridge = session.sip_bridge.lock().await;
+            *bridge = Some(bridge_info);
+            true
+        } else {
+            false
+        }
+    }
+
+    /// Send transcoded audio from the SIP side to the browser.
+    /// Called by the RTP relay when it receives a packet from the provider.
+    pub async fn forward_sip_to_browser(
+        &self,
+        session_id: &str,
+        sip_rtp_payload: &[u8],
+        sip_pt: u8,
+    ) -> Result<(), String> {
+        let session = self
+            .sessions
+            .get(session_id)
+            .ok_or_else(|| format!("session {session_id} not found"))?;
+
+        // Transcode SIP codec → Opus.
+        // We create a temporary TranscodeState per packet for simplicity.
+        // TODO: Use a per-session persistent state for proper codec continuity.
+        let mut transcoder = TranscodeState::new().map_err(|e| format!("codec: {e}"))?;
+        let opus_payload = transcoder
+            .transcode(sip_rtp_payload, sip_pt, PT_OPUS, Some("to_browser"))
+            .map_err(|e| format!("transcode: {e}"))?;
+
+        if opus_payload.is_empty() {
+            return Ok(());
+        }
+
+        // Build RTP header for Opus.
+        // TODO: Track seq/ts/ssrc per session for proper continuity.
+        let header = build_rtp_header(PT_OPUS, 0, 0, 0);
+        let mut packet = header.to_vec();
+        packet.extend_from_slice(&opus_payload);
+
+        session
+            .local_track
+            .write(&packet)
+            .await
+            .map(|_| ())
+            .map_err(|e| format!("write: {e}"))
+    }
+
+    pub async fn add_ice_candidate(
+        &self,
+        session_id: &str,
+        candidate: &str,
+        sdp_mid: Option<&str>,
+        sdp_mline_index: Option<u16>,
+    ) -> Result<(), String> {
+        let session = self
+            .sessions
+            .get(session_id)
+            .ok_or_else(|| format!("session {session_id} not found"))?;
+
+        let init = RTCIceCandidateInit {
+            candidate: candidate.to_string(),
+            sdp_mid: sdp_mid.map(|s| s.to_string()),
+            sdp_mline_index,
+            ..Default::default()
+        };
+
+        session
+            .pc
+            .add_ice_candidate(init)
+            .await
+            .map_err(|e| format!("add ICE: {e}"))
+    }
+
+    pub async fn close_session(&mut self, session_id: &str) -> Result<(), String> {
+        if let Some(session) = self.sessions.remove(session_id) {
+            session.pc.close().await.map_err(|e| format!("close: {e}"))?;
+        }
+        Ok(())
+    }
+
+    pub fn has_session(&self, session_id: &str) -> bool {
+        self.sessions.contains_key(session_id)
+    }
+}
+
+/// Browser → SIP audio forwarding loop.
+/// Reads Opus RTP from the browser, transcodes to the SIP codec, sends to provider.
+async fn browser_to_sip_loop(
+    track: Arc<webrtc::track::track_remote::TrackRemote>,
+    sip_bridge: Arc<Mutex<Option<SipBridgeInfo>>>,
+    out_tx: OutTx,
+    session_id: String,
+) {
+    // Create a persistent codec state for this direction.
+    let mut transcoder = match TranscodeState::new() {
+        Ok(t) => t,
+        Err(e) => {
+            emit_event(
+                &out_tx,
+                "webrtc_error",
+                serde_json::json!({ "session_id": session_id, "error": format!("codec init: {e}") }),
+            );
+            return;
+        }
+    };
+
+    let mut buf = vec![0u8; 1500];
+    let mut count = 0u64;
+    let mut to_sip_seq: u16 = 0;
+    let mut to_sip_ts: u32 = 0;
+    let to_sip_ssrc: u32 = rand::random();
+
+    loop {
+        match track.read(&mut buf).await {
+            Ok((rtp_packet, _attributes)) => {
+                count += 1;
+
+                // Get the SIP bridge info (may not be set yet if call isn't linked).
+                let bridge = sip_bridge.lock().await;
+                let bridge_info = match bridge.as_ref() {
+                    Some(b) => b.clone(),
+                    None => continue, // Not linked to a SIP call yet — drop the packet.
+                };
+                drop(bridge); // Release lock before doing I/O.
+
+                // Extract Opus payload from the RTP packet (skip 12-byte header).
+                let payload = &rtp_packet.payload;
+                if payload.is_empty() {
+                    continue;
+                }
+
+                // Transcode Opus → SIP codec (e.g. G.722).
+                let sip_payload = match transcoder.transcode(
+                    payload,
+                    PT_OPUS,
+                    bridge_info.sip_pt,
+                    Some("to_sip"),
+                ) {
+                    Ok(p) if !p.is_empty() => p,
+                    _ => continue,
+                };
+
+                // Build SIP RTP packet.
+                let header = build_rtp_header(bridge_info.sip_pt, to_sip_seq, to_sip_ts, to_sip_ssrc);
+                let mut sip_rtp = header.to_vec();
+                sip_rtp.extend_from_slice(&sip_payload);
+
+                to_sip_seq = to_sip_seq.wrapping_add(1);
+                to_sip_ts = to_sip_ts.wrapping_add(rtp_clock_increment(bridge_info.sip_pt));
+
+                // Send to provider.
+                let _ = bridge_info
+                    .sip_socket
+                    .send_to(&sip_rtp, bridge_info.provider_media)
+                    .await;
+
+                if count == 1 || count == 50 || count % 500 == 0 {
+                    emit_event(
+                        &out_tx,
+                        "webrtc_audio_tx",
+                        serde_json::json!({
+                            "session_id": session_id,
+                            "direction": "browser_to_sip",
+                            "packet_count": count,
+                        }),
+                    );
+                }
+            }
+            Err(_) => break, // Track ended.
+        }
+    }
+}