v1.14.0

changelog.md

@@ -1,5 +1,22 @@
 # Changelog
 
+## 2026-04-10 - 1.14.0 - feat(proxy-engine)
+add multiparty call mixing with dynamic SIP and WebRTC leg management
+
+- replace passthrough call handling with a mixer-backed call model that tracks multiple legs and exposes leg status in call state output
+- add mixer and leg I/O infrastructure to bridge SIP RTP and WebRTC audio through channel-based mix-minus processing
+- introduce add_leg and remove_leg proxy commands and wire frontend bridge APIs to manage external call legs
+- emit leg lifecycle events for observability and mark unimplemented device-leg and transfer HTTP endpoints with 501 responses
+
+## 2026-04-10 - 1.13.0 - feat(proxy-engine,webrtc)
+add B2BUA SIP leg handling and WebRTC call bridging for outbound calls
+
+- introduce a new SipLeg module to manage outbound provider dialogs, including INVITE lifecycle, digest auth retries, ACK handling, media endpoint tracking, and termination
+- store outbound dashboard calls as B2BUA calls in the call manager and emit provider media details on call_answered for bridge setup
+- separate SIP and WebRTC engine locking to avoid contention and deadlocks while linking sessions to call RTP sockets
+- add bidirectional RTP bridging between provider SIP media and browser WebRTC audio using the allocated RTP socket
+- wire browser webrtc-accept events in the frontend and sipproxy so session-to-call linking can occur when media and acceptance arrive in either order
+
 ## 2026-04-10 - 1.12.0 - feat(proxy-engine)
 add Rust-based outbound calling, WebRTC bridging, and voicemail handling
 

package.json

@@ -1,6 +1,6 @@
 {
   "name": "siprouter",
-  "version": "1.12.0",
+  "version": "1.14.0",
   "private": true,
   "type": "module",
   "scripts": {

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

@@ -1,12 +1,19 @@
-//! Call hub — owns legs and bridges media.
+//! Call hub — owns N legs and a mixer task.
 //!
-//! Each Call has a unique ID and tracks its state, direction, and associated
-//! SIP Call-IDs for message routing.
+//! Every call has a central mixer that provides mix-minus audio to all
+//! participants. Legs can be added and removed dynamically mid-call.
 
+use crate::mixer::{MixerCommand, RtpPacket};
+use crate::sip_leg::SipLeg;
+use std::collections::HashMap;
 use std::net::SocketAddr;
 use std::sync::Arc;
 use std::time::Instant;
 use tokio::net::UdpSocket;
+use tokio::sync::mpsc;
+use tokio::task::JoinHandle;
+
+pub type LegId = String;
 
 /// Call state machine.
 #[derive(Debug, Clone, Copy, PartialEq, Eq)]
@@ -15,8 +22,6 @@ pub enum CallState {
     Ringing,
     Connected,
     Voicemail,
-    Ivr,
-    Terminating,
     Terminated,
 }
 
@@ -27,8 +32,6 @@ impl CallState {
             Self::Ringing => "ringing",
             Self::Connected => "connected",
             Self::Voicemail => "voicemail",
-            Self::Ivr => "ivr",
-            Self::Terminating => "terminating",
             Self::Terminated => "terminated",
         }
     }
@@ -49,43 +52,172 @@ impl CallDirection {
     }
 }
 
-/// A passthrough call — both sides share the same SIP Call-ID.
-/// The proxy rewrites SDP/Contact/Request-URI and relays RTP.
-pub struct PassthroughCall {
+/// The type of a call leg.
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub enum LegKind {
+    SipProvider,
+    SipDevice,
+    WebRtc,
+    Media, // voicemail playback, IVR, recording
+}
+
+impl LegKind {
+    pub fn as_str(&self) -> &'static str {
+        match self {
+            Self::SipProvider => "sip-provider",
+            Self::SipDevice => "sip-device",
+            Self::WebRtc => "webrtc",
+            Self::Media => "media",
+        }
+    }
+}
+
+/// Per-leg state.
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub enum LegState {
+    Inviting,
+    Ringing,
+    Connected,
+    Terminated,
+}
+
+impl LegState {
+    pub fn as_str(&self) -> &'static str {
+        match self {
+            Self::Inviting => "inviting",
+            Self::Ringing => "ringing",
+            Self::Connected => "connected",
+            Self::Terminated => "terminated",
+        }
+    }
+}
+
+/// Information about a single leg in a call.
+pub struct LegInfo {
+    pub id: LegId,
+    pub kind: LegKind,
+    pub state: LegState,
+    pub codec_pt: u8,
+
+    /// For SIP legs: the SIP dialog manager (handles 407 auth, BYE, etc).
+    pub sip_leg: Option<SipLeg>,
+    /// For SIP legs: the SIP Call-ID for message routing.
+    pub sip_call_id: Option<String>,
+    /// For WebRTC legs: the session ID in WebRtcEngine.
+    pub webrtc_session_id: Option<String>,
+    /// The RTP socket allocated for this leg.
+    pub rtp_socket: Option<Arc<UdpSocket>>,
+    /// The RTP port number.
+    pub rtp_port: u16,
+    /// The remote media endpoint (learned from SDP or address learning).
+    pub remote_media: Option<SocketAddr>,
+    /// SIP signaling address (provider or device).
+    pub signaling_addr: Option<SocketAddr>,
+}
+
+/// A multiparty call with N legs and a central mixer.
+pub struct Call {
     pub id: String,
-    pub sip_call_id: String,
     pub state: CallState,
     pub direction: CallDirection,
     pub created_at: Instant,
 
-    // Call metadata.
+    // Metadata.
     pub caller_number: Option<String>,
     pub callee_number: Option<String>,
     pub provider_id: String,
 
-    // Provider side.
-    pub provider_addr: SocketAddr,
-    pub provider_media: Option<SocketAddr>,
+    /// All legs in this call, keyed by leg ID.
+    pub legs: HashMap<LegId, LegInfo>,
 
-    // Device side.
-    pub device_addr: SocketAddr,
-    pub device_media: Option<SocketAddr>,
+    /// Channel to send commands to the mixer task.
+    pub mixer_cmd_tx: mpsc::Sender<MixerCommand>,
 
-    // RTP relay.
-    pub rtp_port: u16,
-    pub rtp_socket: Arc<UdpSocket>,
-
-    // Packet counters.
-    pub pkt_from_device: u64,
-    pub pkt_from_provider: u64,
+    /// Handle to the mixer task (aborted on call teardown).
+    mixer_task: Option<JoinHandle<()>>,
 }
 
-impl PassthroughCall {
+impl Call {
+    pub fn new(
+        id: String,
+        direction: CallDirection,
+        provider_id: String,
+        mixer_cmd_tx: mpsc::Sender<MixerCommand>,
+        mixer_task: JoinHandle<()>,
+    ) -> Self {
+        Self {
+            id,
+            state: CallState::SettingUp,
+            direction,
+            created_at: Instant::now(),
+            caller_number: None,
+            callee_number: None,
+            provider_id,
+            legs: HashMap::new(),
+            mixer_cmd_tx,
+            mixer_task: Some(mixer_task),
+        }
+    }
+
+    /// Add a leg to the mixer. Sends the AddLeg command with channel endpoints.
+    pub async fn add_leg_to_mixer(
+        &self,
+        leg_id: &str,
+        codec_pt: u8,
+        inbound_rx: mpsc::Receiver<RtpPacket>,
+        outbound_tx: mpsc::Sender<Vec<u8>>,
+    ) {
+        let _ = self
+            .mixer_cmd_tx
+            .send(MixerCommand::AddLeg {
+                leg_id: leg_id.to_string(),
+                codec_pt,
+                inbound_rx,
+                outbound_tx,
+            })
+            .await;
+    }
+
+    /// Remove a leg from the mixer.
+    pub async fn remove_leg_from_mixer(&self, leg_id: &str) {
+        let _ = self
+            .mixer_cmd_tx
+            .send(MixerCommand::RemoveLeg {
+                leg_id: leg_id.to_string(),
+            })
+            .await;
+    }
+
     pub fn duration_secs(&self) -> u64 {
         self.created_at.elapsed().as_secs()
     }
 
+    /// Shut down the mixer and abort its task.
+    pub async fn shutdown_mixer(&mut self) {
+        let _ = self.mixer_cmd_tx.send(MixerCommand::Shutdown).await;
+        if let Some(handle) = self.mixer_task.take() {
+            handle.abort();
+        }
+    }
+
+    /// Produce a JSON status snapshot for the dashboard.
     pub fn to_status_json(&self) -> serde_json::Value {
+        let legs: Vec<serde_json::Value> = self
+            .legs
+            .values()
+            .filter(|l| l.state != LegState::Terminated)
+            .map(|l| {
+                serde_json::json!({
+                    "id": l.id,
+                    "type": l.kind.as_str(),
+                    "state": l.state.as_str(),
+                    "codec": sip_proto::helpers::codec_name(l.codec_pt),
+                    "rtpPort": l.rtp_port,
+                    "remoteMedia": l.remote_media.map(|a| format!("{}:{}", a.ip(), a.port())),
+                })
+            })
+            .collect();
+
         serde_json::json!({
             "id": self.id,
             "state": self.state.as_str(),
@@ -93,11 +225,8 @@ impl PassthroughCall {
             "callerNumber": self.caller_number,
             "calleeNumber": self.callee_number,
             "providerUsed": self.provider_id,
-            "createdAt": self.created_at.elapsed().as_millis(),
             "duration": self.duration_secs(),
-            "rtpPort": self.rtp_port,
-            "pktFromDevice": self.pkt_from_device,
-            "pktFromProvider": self.pkt_from_provider,
+            "legs": legs,
         })
     }
 }

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

@@ -0,0 +1,80 @@
+//! Leg I/O task spawners.
+//!
+//! Each SIP leg gets two tasks:
+//! - Inbound: recv_from on RTP socket → strip header → send RtpPacket to mixer channel
+//! - Outbound: recv encoded RTP from mixer channel → send_to remote media endpoint
+//!
+//! WebRTC leg I/O is handled inside webrtc_engine.rs (on_track + track.write).
+
+use crate::mixer::RtpPacket;
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+use tokio::sync::mpsc;
+
+/// Channel pair for connecting a leg to the mixer.
+pub struct LegChannels {
+    /// Mixer receives decoded packets from this leg.
+    pub inbound_tx: mpsc::Sender<RtpPacket>,
+    pub inbound_rx: mpsc::Receiver<RtpPacket>,
+    /// Mixer sends encoded RTP to this leg.
+    pub outbound_tx: mpsc::Sender<Vec<u8>>,
+    pub outbound_rx: mpsc::Receiver<Vec<u8>>,
+}
+
+/// Create a channel pair for a leg.
+pub fn create_leg_channels() -> LegChannels {
+    let (inbound_tx, inbound_rx) = mpsc::channel::<RtpPacket>(64);
+    let (outbound_tx, outbound_rx) = mpsc::channel::<Vec<u8>>(8);
+    LegChannels {
+        inbound_tx,
+        inbound_rx,
+        outbound_tx,
+        outbound_rx,
+    }
+}
+
+/// Spawn the inbound I/O task for a SIP leg.
+/// Reads RTP from the socket, strips the 12-byte header, sends payload to the mixer.
+/// Returns the JoinHandle (exits when the inbound_tx channel is dropped).
+pub fn spawn_sip_inbound(
+    rtp_socket: Arc<UdpSocket>,
+    inbound_tx: mpsc::Sender<RtpPacket>,
+) -> tokio::task::JoinHandle<()> {
+    tokio::spawn(async move {
+        let mut buf = vec![0u8; 1500];
+        loop {
+            match rtp_socket.recv_from(&mut buf).await {
+                Ok((n, _from)) => {
+                    if n < 12 {
+                        continue; // Too small for RTP header.
+                    }
+                    let pt = buf[1] & 0x7F;
+                    let payload = buf[12..n].to_vec();
+                    if payload.is_empty() {
+                        continue;
+                    }
+                    if inbound_tx.send(RtpPacket { payload, payload_type: pt }).await.is_err() {
+                        break; // Channel closed — leg removed.
+                    }
+                }
+                Err(_) => break, // Socket error.
+            }
+        }
+    })
+}
+
+/// Spawn the outbound I/O task for a SIP leg.
+/// Reads encoded RTP packets from the mixer and sends them to the remote media endpoint.
+/// Returns the JoinHandle (exits when the outbound_rx channel is closed).
+pub fn spawn_sip_outbound(
+    rtp_socket: Arc<UdpSocket>,
+    remote_media: SocketAddr,
+    mut outbound_rx: mpsc::Receiver<Vec<u8>>,
+) -> tokio::task::JoinHandle<()> {
+    tokio::spawn(async move {
+        while let Some(rtp_data) = outbound_rx.recv().await {
+            let _ = rtp_socket.send_to(&rtp_data, remote_media).await;
+        }
+    })
+}

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

@@ -12,10 +12,13 @@ mod call_manager;
 mod config;
 mod dtmf;
 mod ipc;
+mod leg_io;
+mod mixer;
 mod provider;
 mod recorder;
 mod registrar;
 mod rtp;
+mod sip_leg;
 mod sip_transport;
 mod voicemail;
 mod webrtc_engine;
@@ -35,14 +38,15 @@ use tokio::io::{AsyncBufReadExt, AsyncWriteExt, BufReader};
 use tokio::net::UdpSocket;
 use tokio::sync::{mpsc, Mutex};
 
-/// Shared mutable state for the proxy engine.
+/// Shared mutable state for the proxy engine (SIP side).
+/// WebRTC is intentionally kept in a separate lock to avoid contention
+/// between SIP packet handlers and WebRTC command handlers.
 struct ProxyEngine {
     config: Option<AppConfig>,
     transport: Option<SipTransport>,
     provider_mgr: ProviderManager,
     registrar: Registrar,
     call_mgr: CallManager,
-    webrtc: WebRtcEngine,
     rtp_pool: Option<RtpPortPool>,
     out_tx: OutTx,
 }
@@ -55,7 +59,6 @@ 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,
         }
@@ -83,9 +86,12 @@ async fn main() {
     // Emit ready event.
     emit_event(&out_tx, "ready", serde_json::json!({}));
 
-    // Shared engine state.
+    // Shared engine state (SIP side).
     let engine = Arc::new(Mutex::new(ProxyEngine::new(out_tx.clone())));
 
+    // WebRTC engine — separate lock to avoid deadlock with SIP handlers.
+    let webrtc = Arc::new(Mutex::new(WebRtcEngine::new(out_tx.clone())));
+
     // Read commands from stdin.
     let stdin = tokio::io::stdin();
     let reader = BufReader::new(stdin);
@@ -105,25 +111,36 @@ async fn main() {
         };
 
         let engine = engine.clone();
+        let webrtc = webrtc.clone();
         let out_tx = out_tx.clone();
 
         // Handle commands — some are async, so we spawn.
         tokio::spawn(async move {
-            handle_command(engine, &out_tx, cmd).await;
+            handle_command(engine, webrtc, &out_tx, cmd).await;
         });
     }
 }
 
-async fn handle_command(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: Command) {
+async fn handle_command(
+    engine: Arc<Mutex<ProxyEngine>>,
+    webrtc: Arc<Mutex<WebRtcEngine>>,
+    out_tx: &OutTx,
+    cmd: Command,
+) {
     match cmd.method.as_str() {
+        // SIP commands — lock engine only.
         "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,
+        "add_leg" => handle_add_leg(engine, out_tx, &cmd).await,
+        "remove_leg" => handle_remove_leg(engine, out_tx, &cmd).await,
+        // WebRTC commands — lock webrtc only (no engine contention).
+        "webrtc_offer" => handle_webrtc_offer(webrtc, out_tx, &cmd).await,
+        "webrtc_ice" => handle_webrtc_ice(webrtc, out_tx, &cmd).await,
+        "webrtc_close" => handle_webrtc_close(webrtc, out_tx, &cmd).await,
+        // webrtc_link needs both: engine (for mixer channels) and webrtc (for session).
+        "webrtc_link" => handle_webrtc_link(engine, webrtc, out_tx, &cmd).await,
         _ => respond_err(out_tx, &cmd.id, &format!("unknown command: {}", cmd.method)),
     }
 }
@@ -246,14 +263,11 @@ async fn handle_sip_packet(
     }
 
     // 3. Route to existing call by SIP Call-ID.
-    // Check if this Call-ID belongs to an active call (avoids borrow conflict).
     if eng.call_mgr.has_call(msg.call_id()) {
         let config_ref = eng.config.as_ref().unwrap().clone();
-        // Temporarily take registrar to avoid overlapping borrows.
-        let registrar_dummy = Registrar::new(eng.out_tx.clone());
         if eng
             .call_mgr
-            .route_sip_message(&msg, from_addr, socket, &config_ref, &registrar_dummy)
+            .route_sip_message(&msg, from_addr, socket, &config_ref)
             .await
         {
             return;
@@ -524,7 +538,8 @@ async fn handle_hangup(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Co
 }
 
 /// 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) {
+/// Uses only the WebRTC lock — no contention with SIP handlers.
+async fn handle_webrtc_offer(webrtc: Arc<Mutex<WebRtcEngine>>, 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; }
@@ -534,8 +549,8 @@ async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cm
         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 {
+    let mut wrtc = webrtc.lock().await;
+    match wrtc.handle_offer(&session_id, &offer_sdp).await {
         Ok(answer_sdp) => {
             respond_ok(out_tx, &cmd.id, serde_json::json!({
                 "session_id": session_id,
@@ -547,7 +562,8 @@ async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cm
 }
 
 /// 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) {
+/// Uses only the WebRTC lock.
+async fn handle_webrtc_ice(webrtc: Arc<Mutex<WebRtcEngine>>, 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; }
@@ -556,15 +572,22 @@ async fn handle_webrtc_ice(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd:
     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 {
+    let wrtc = webrtc.lock().await;
+    match wrtc.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) {
+/// Handle `webrtc_link` — link a WebRTC session to a call's mixer for audio bridging.
+/// Creates channels, adds WebRTC leg to the call, wires the WebRTC engine.
+/// Locks are never held simultaneously — no deadlock possible.
+async fn handle_webrtc_link(
+    engine: Arc<Mutex<ProxyEngine>>,
+    webrtc: Arc<Mutex<WebRtcEngine>>,
+    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; }
@@ -573,34 +596,67 @@ async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd
         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; }
-    };
+    // Create channels for the WebRTC leg.
+    let channels = crate::leg_io::create_leg_channels();
 
-    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; }
-    };
+    // Briefly lock engine to add the WebRTC leg to the call's mixer.
+    {
+        let eng = engine.lock().await;
+        let call = match eng.call_mgr.calls.get(&call_id) {
+            Some(c) => c,
+            None => {
+                respond_err(out_tx, &cmd.id, &format!("call {call_id} not found"));
+                return;
+            }
+        };
+        // Add to mixer via channel.
+        call.add_leg_to_mixer(
+            &session_id,
+            codec_lib::PT_OPUS,
+            channels.inbound_rx,
+            channels.outbound_tx,
+        )
+        .await;
+    } // engine lock released
 
-    let bridge_info = crate::webrtc_engine::SipBridgeInfo {
-        provider_media,
-        sip_pt,
-        sip_socket,
-    };
+    // Lock webrtc to wire the channels.
+    let mut wrtc = webrtc.lock().await;
+    if wrtc
+        .link_to_mixer(&session_id, &call_id, channels.inbound_tx, channels.outbound_rx)
+        .await
+    {
+        // Also store the WebRTC leg info in the call.
+        drop(wrtc); // Release webrtc lock before re-acquiring engine.
+        {
+            let mut eng = engine.lock().await;
+            if let Some(call) = eng.call_mgr.calls.get_mut(&call_id) {
+                call.legs.insert(
+                    session_id.clone(),
+                    crate::call::LegInfo {
+                        id: session_id.clone(),
+                        kind: crate::call::LegKind::WebRtc,
+                        state: crate::call::LegState::Connected,
+                        codec_pt: codec_lib::PT_OPUS,
+                        sip_leg: None,
+                        sip_call_id: None,
+                        webrtc_session_id: Some(session_id.clone()),
+                        rtp_socket: None,
+                        rtp_port: 0,
+                        remote_media: None,
+                        signaling_addr: None,
+                    },
+                );
+            }
+        }
+
+        emit_event(out_tx, "leg_added", serde_json::json!({
+            "call_id": call_id,
+            "leg_id": session_id,
+            "kind": "webrtc",
+            "state": "connected",
+        }));
 
-    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,
@@ -611,15 +667,108 @@ async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd
     }
 }
 
+/// Handle `add_leg` — add a new SIP leg to an existing call.
+async fn handle_add_leg(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    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 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 {
+        config_ref.resolve_outbound_route(&number, None, &|_| true).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 registered AOR.
+    let 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.registered_aor.clone()
+    } else {
+        format!("sip:{}@{}", provider_config.username, provider_config.domain)
+    };
+
+    let public_ip = 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()
+    } else {
+        None
+    };
+
+    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 leg_id = call_mgr.add_external_leg(
+        &call_id, &number, &provider_config, &config_ref,
+        rtp_pool, &socket, public_ip.as_deref(), &registered_aor,
+    ).await;
+
+    match leg_id {
+        Some(lid) => respond_ok(out_tx, &cmd.id, serde_json::json!({ "leg_id": lid })),
+        None => respond_err(out_tx, &cmd.id, "failed to add leg"),
+    }
+}
+
+/// Handle `remove_leg` — remove a leg from a call.
+async fn handle_remove_leg(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    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 leg_id = match cmd.params.get("leg_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing leg_id"); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    let socket = match &eng.transport {
+        Some(t) => t.socket(),
+        None => { respond_err(out_tx, &cmd.id, "not initialized"); return; }
+    };
+
+    if eng.call_mgr.remove_leg(&call_id, &leg_id, &socket).await {
+        respond_ok(out_tx, &cmd.id, serde_json::json!({}));
+    } else {
+        respond_err(out_tx, &cmd.id, &format!("call/leg not found"));
+    }
+}
+
 /// Handle `webrtc_close` — close a WebRTC session.
-async fn handle_webrtc_close(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+/// Uses only the WebRTC lock.
+async fn handle_webrtc_close(webrtc: Arc<Mutex<WebRtcEngine>>, 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 {
+    let mut wrtc = webrtc.lock().await;
+    match wrtc.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/mixer.rs

@@ -0,0 +1,232 @@
+//! Audio mixer — mix-minus engine for multiparty calls.
+//!
+//! Each Call spawns one mixer task. Legs communicate with the mixer via
+//! tokio mpsc channels — no shared mutable state, no lock contention.
+//!
+//! The mixer runs a 20ms tick loop:
+//! 1. Drain inbound channels, decode to PCM, resample to 16kHz
+//! 2. Compute total mix (sum of all legs' PCM as i32)
+//! 3. For each leg: mix-minus = total - own, resample to leg codec rate, encode, send
+
+use crate::ipc::{emit_event, OutTx};
+use crate::rtp::{build_rtp_header, rtp_clock_increment};
+use codec_lib::{codec_sample_rate, TranscodeState};
+use std::collections::HashMap;
+use tokio::sync::mpsc;
+use tokio::task::JoinHandle;
+use tokio::time::{self, Duration, MissedTickBehavior};
+
+/// Mixing sample rate — 16kHz. G.722 is native, G.711 needs 2× upsample, Opus needs 3× downsample.
+const MIX_RATE: u32 = 16000;
+/// Samples per 20ms frame at the mixing rate.
+const MIX_FRAME_SIZE: usize = 320; // 16000 * 0.020
+
+/// A raw RTP payload received from a leg (no RTP header).
+pub struct RtpPacket {
+    pub payload: Vec<u8>,
+    pub payload_type: u8,
+}
+
+/// Commands sent to the mixer task via a control channel.
+pub enum MixerCommand {
+    /// Add a new leg to the mix.
+    AddLeg {
+        leg_id: String,
+        codec_pt: u8,
+        inbound_rx: mpsc::Receiver<RtpPacket>,
+        outbound_tx: mpsc::Sender<Vec<u8>>,
+    },
+    /// Remove a leg from the mix (channels are dropped, I/O tasks exit).
+    RemoveLeg { leg_id: String },
+    /// Shut down the mixer.
+    Shutdown,
+}
+
+/// Internal per-leg state inside the mixer.
+struct MixerLegSlot {
+    codec_pt: u8,
+    transcoder: TranscodeState,
+    inbound_rx: mpsc::Receiver<RtpPacket>,
+    outbound_tx: mpsc::Sender<Vec<u8>>,
+    /// Last decoded PCM frame at MIX_RATE (320 samples). Used for mix-minus.
+    last_pcm_frame: Vec<i16>,
+    /// Number of consecutive ticks with no inbound packet.
+    silent_ticks: u32,
+    // RTP output state.
+    rtp_seq: u16,
+    rtp_ts: u32,
+    rtp_ssrc: u32,
+}
+
+/// Spawn the mixer task for a call. Returns the command sender and task handle.
+pub fn spawn_mixer(
+    call_id: String,
+    out_tx: OutTx,
+) -> (mpsc::Sender<MixerCommand>, JoinHandle<()>) {
+    let (cmd_tx, cmd_rx) = mpsc::channel::<MixerCommand>(32);
+
+    let handle = tokio::spawn(async move {
+        mixer_loop(call_id, cmd_rx, out_tx).await;
+    });
+
+    (cmd_tx, handle)
+}
+
+/// The 20ms mixing loop.
+async fn mixer_loop(
+    call_id: String,
+    mut cmd_rx: mpsc::Receiver<MixerCommand>,
+    out_tx: OutTx,
+) {
+    let mut legs: HashMap<String, MixerLegSlot> = HashMap::new();
+    let mut interval = time::interval(Duration::from_millis(20));
+    interval.set_missed_tick_behavior(MissedTickBehavior::Skip);
+
+    loop {
+        interval.tick().await;
+
+        // 1. Process control commands (non-blocking).
+        loop {
+            match cmd_rx.try_recv() {
+                Ok(MixerCommand::AddLeg {
+                    leg_id,
+                    codec_pt,
+                    inbound_rx,
+                    outbound_tx,
+                }) => {
+                    let transcoder = match TranscodeState::new() {
+                        Ok(t) => t,
+                        Err(e) => {
+                            emit_event(
+                                &out_tx,
+                                "mixer_error",
+                                serde_json::json!({
+                                    "call_id": call_id,
+                                    "leg_id": leg_id,
+                                    "error": format!("codec init: {e}"),
+                                }),
+                            );
+                            continue;
+                        }
+                    };
+                    legs.insert(
+                        leg_id,
+                        MixerLegSlot {
+                            codec_pt,
+                            transcoder,
+                            inbound_rx,
+                            outbound_tx,
+                            last_pcm_frame: vec![0i16; MIX_FRAME_SIZE],
+                            silent_ticks: 0,
+                            rtp_seq: 0,
+                            rtp_ts: 0,
+                            rtp_ssrc: rand::random(),
+                        },
+                    );
+                }
+                Ok(MixerCommand::RemoveLeg { leg_id }) => {
+                    legs.remove(&leg_id);
+                    // Channels drop → I/O tasks exit cleanly.
+                }
+                Ok(MixerCommand::Shutdown) => return,
+                Err(mpsc::error::TryRecvError::Empty) => break,
+                Err(mpsc::error::TryRecvError::Disconnected) => return,
+            }
+        }
+
+        if legs.is_empty() {
+            continue;
+        }
+
+        // 2. Drain inbound packets, decode to 16kHz PCM.
+        let leg_ids: Vec<String> = legs.keys().cloned().collect();
+        for lid in &leg_ids {
+            let slot = legs.get_mut(lid).unwrap();
+
+            // Drain channel, keep only the latest packet (simple jitter handling).
+            let mut latest: Option<RtpPacket> = None;
+            loop {
+                match slot.inbound_rx.try_recv() {
+                    Ok(pkt) => latest = Some(pkt),
+                    Err(_) => break,
+                }
+            }
+
+            if let Some(pkt) = latest {
+                slot.silent_ticks = 0;
+                match slot.transcoder.decode_to_pcm(&pkt.payload, pkt.payload_type) {
+                    Ok((pcm, rate)) => {
+                        // Resample to mixing rate if needed.
+                        let pcm_mix = if rate == MIX_RATE {
+                            pcm
+                        } else {
+                            slot.transcoder
+                                .resample(&pcm, rate, MIX_RATE)
+                                .unwrap_or_else(|_| vec![0i16; MIX_FRAME_SIZE])
+                        };
+                        // Pad or truncate to exactly MIX_FRAME_SIZE.
+                        let mut frame = pcm_mix;
+                        frame.resize(MIX_FRAME_SIZE, 0);
+                        slot.last_pcm_frame = frame;
+                    }
+                    Err(_) => {
+                        // Decode failed — use silence.
+                        slot.last_pcm_frame = vec![0i16; MIX_FRAME_SIZE];
+                    }
+                }
+            } else {
+                slot.silent_ticks += 1;
+                // After 150 ticks (3 seconds) of silence, zero out to avoid stale audio.
+                if slot.silent_ticks > 150 {
+                    slot.last_pcm_frame = vec![0i16; MIX_FRAME_SIZE];
+                }
+            }
+        }
+
+        // 3. Compute total mix (sum of all legs as i32 to avoid overflow).
+        let mut total_mix = vec![0i32; MIX_FRAME_SIZE];
+        for slot in legs.values() {
+            for (i, &s) in slot.last_pcm_frame.iter().enumerate().take(MIX_FRAME_SIZE) {
+                total_mix[i] += s as i32;
+            }
+        }
+
+        // 4. For each leg: mix-minus, resample, encode, send.
+        for slot in legs.values_mut() {
+            // Mix-minus: total minus this leg's own contribution.
+            let mut mix_minus = Vec::with_capacity(MIX_FRAME_SIZE);
+            for i in 0..MIX_FRAME_SIZE {
+                let sample =
+                    (total_mix[i] - slot.last_pcm_frame[i] as i32).clamp(-32768, 32767) as i16;
+                mix_minus.push(sample);
+            }
+
+            // Resample from 16kHz to the leg's codec native rate.
+            let target_rate = codec_sample_rate(slot.codec_pt);
+            let resampled = if target_rate == MIX_RATE {
+                mix_minus
+            } else {
+                slot.transcoder
+                    .resample(&mix_minus, MIX_RATE, target_rate)
+                    .unwrap_or_default()
+            };
+
+            // Encode to the leg's codec.
+            let encoded = match slot.transcoder.encode_from_pcm(&resampled, slot.codec_pt) {
+                Ok(e) if !e.is_empty() => e,
+                _ => continue,
+            };
+
+            // Build RTP packet with header.
+            let header = build_rtp_header(slot.codec_pt, slot.rtp_seq, slot.rtp_ts, slot.rtp_ssrc);
+            let mut rtp = header.to_vec();
+            rtp.extend_from_slice(&encoded);
+
+            slot.rtp_seq = slot.rtp_seq.wrapping_add(1);
+            slot.rtp_ts = slot.rtp_ts.wrapping_add(rtp_clock_increment(slot.codec_pt));
+
+            // Non-blocking send — drop frame if channel is full.
+            let _ = slot.outbound_tx.try_send(rtp);
+        }
+    }
+}

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

@@ -0,0 +1,475 @@
+//! SipLeg — manages one side of a B2BUA call.
+//!
+//! Handles the full INVITE lifecycle:
+//! - Send INVITE with SDP
+//! - Handle 407 Proxy Authentication (digest auth retry)
+//! - Handle 200 OK (ACK, learn media endpoint)
+//! - Handle BYE/CANCEL (teardown)
+//! - Track SIP dialog state (early → confirmed → terminated)
+//!
+//! Ported from ts/call/sip-leg.ts.
+
+use sip_proto::dialog::{DialogState, SipDialog};
+use sip_proto::helpers::{
+    build_sdp, compute_digest_auth, generate_branch, generate_tag, parse_digest_challenge,
+    parse_sdp_endpoint, SdpOptions,
+};
+use sip_proto::message::{RequestOptions, SipMessage};
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+
+/// State of a SIP leg.
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub enum LegState {
+    Inviting,
+    Ringing,
+    Connected,
+    Terminating,
+    Terminated,
+}
+
+/// Configuration for creating a SIP leg.
+pub struct SipLegConfig {
+    /// Proxy LAN IP (for Via, Contact, SDP).
+    pub lan_ip: String,
+    /// Proxy LAN port.
+    pub lan_port: u16,
+    /// Public IP (for provider-facing legs).
+    pub public_ip: Option<String>,
+    /// SIP target endpoint (provider outbound proxy or device address).
+    pub sip_target: SocketAddr,
+    /// Provider credentials (for 407 auth).
+    pub username: Option<String>,
+    pub password: Option<String>,
+    pub registered_aor: Option<String>,
+    /// Codec payload types to offer.
+    pub codecs: Vec<u8>,
+    /// Our RTP port for SDP.
+    pub rtp_port: u16,
+}
+
+/// A SIP leg with full dialog management.
+pub struct SipLeg {
+    pub id: String,
+    pub state: LegState,
+    pub config: SipLegConfig,
+    pub dialog: Option<SipDialog>,
+
+    /// The INVITE we sent (needed for CANCEL and 407 ACK).
+    invite: Option<SipMessage>,
+    /// Original unauthenticated INVITE (for re-ACKing retransmitted 407s).
+    orig_invite: Option<SipMessage>,
+    /// Whether we've attempted digest auth.
+    auth_attempted: bool,
+
+    /// Remote media endpoint (learned from SDP in 200 OK).
+    pub remote_media: Option<SocketAddr>,
+}
+
+impl SipLeg {
+    pub fn new(id: String, config: SipLegConfig) -> Self {
+        Self {
+            id,
+            state: LegState::Inviting,
+            config,
+            dialog: None,
+            invite: None,
+            orig_invite: None,
+            auth_attempted: false,
+            remote_media: None,
+        }
+    }
+
+    /// Build and send an INVITE to establish this leg.
+    pub async fn send_invite(
+        &mut self,
+        from_uri: &str,
+        to_uri: &str,
+        sip_call_id: &str,
+        socket: &UdpSocket,
+    ) {
+        let ip = self
+            .config
+            .public_ip
+            .as_deref()
+            .unwrap_or(&self.config.lan_ip);
+
+        let sdp = build_sdp(&SdpOptions {
+            ip,
+            port: self.config.rtp_port,
+            payload_types: &self.config.codecs,
+            ..Default::default()
+        });
+
+        let invite = SipMessage::create_request(
+            "INVITE",
+            to_uri,
+            RequestOptions {
+                via_host: ip.to_string(),
+                via_port: self.config.lan_port,
+                via_transport: None,
+                via_branch: Some(generate_branch()),
+                from_uri: from_uri.to_string(),
+                from_display_name: None,
+                from_tag: Some(generate_tag()),
+                to_uri: to_uri.to_string(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some(sip_call_id.to_string()),
+                cseq: Some(1),
+                contact: Some(format!("<sip:{ip}:{}>", self.config.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()),
+                ]),
+            },
+        );
+
+        self.dialog = Some(SipDialog::from_uac_invite(&invite, ip, self.config.lan_port));
+        self.invite = Some(invite.clone());
+        self.state = LegState::Inviting;
+
+        let _ = socket.send_to(&invite.serialize(), self.config.sip_target).await;
+    }
+
+    /// Handle an incoming SIP message routed to this leg.
+    /// Returns an optional reply to send (e.g. ACK, auth retry INVITE).
+    pub fn handle_message(&mut self, msg: &SipMessage) -> SipLegAction {
+        if msg.is_response() {
+            self.handle_response(msg)
+        } else {
+            self.handle_request(msg)
+        }
+    }
+
+    fn handle_response(&mut self, msg: &SipMessage) -> SipLegAction {
+        let code = msg.status_code().unwrap_or(0);
+        let cseq_method = msg.cseq_method().unwrap_or("").to_uppercase();
+
+        if cseq_method != "INVITE" {
+            return SipLegAction::None;
+        }
+
+        // Handle retransmitted 407 for the original unauthenticated INVITE.
+        if self.auth_attempted {
+            if let Some(dialog) = &self.dialog {
+                let response_cseq: u32 = msg
+                    .get_header("CSeq")
+                    .and_then(|s| s.split_whitespace().next())
+                    .and_then(|s| s.parse().ok())
+                    .unwrap_or(0);
+                if response_cseq < dialog.local_cseq && code >= 400 {
+                    // ACK the retransmitted error response.
+                    if let Some(orig) = &self.orig_invite {
+                        let ack = build_non_2xx_ack(orig, msg);
+                        return SipLegAction::Send(ack.serialize());
+                    }
+                    return SipLegAction::None;
+                }
+            }
+        }
+
+        // Handle 407 Proxy Authentication Required.
+        if code == 407 {
+            return self.handle_auth_challenge(msg);
+        }
+
+        // Update dialog state.
+        if let Some(dialog) = &mut self.dialog {
+            dialog.process_response(msg);
+        }
+
+        if code == 180 || code == 183 {
+            self.state = LegState::Ringing;
+            SipLegAction::StateChange(LegState::Ringing)
+        } else if code >= 200 && code < 300 {
+            // ACK the 200 OK.
+            let ack_buf = if let Some(dialog) = &self.dialog {
+                let ack = dialog.create_ack();
+                Some(ack.serialize())
+            } else {
+                None
+            };
+
+            // If already connected (200 retransmit), just re-ACK.
+            if self.state == LegState::Connected {
+                return match ack_buf {
+                    Some(buf) => SipLegAction::Send(buf),
+                    None => SipLegAction::None,
+                };
+            }
+
+            // Learn media endpoint from SDP.
+            if msg.has_sdp_body() {
+                if let Some(ep) = parse_sdp_endpoint(&msg.body) {
+                    if let Ok(addr) = format!("{}:{}", ep.address, ep.port).parse() {
+                        self.remote_media = Some(addr);
+                    }
+                }
+            }
+
+            self.state = LegState::Connected;
+
+            match ack_buf {
+                Some(buf) => SipLegAction::ConnectedWithAck(buf),
+                None => SipLegAction::StateChange(LegState::Connected),
+            }
+        } else if code >= 300 {
+            self.state = LegState::Terminated;
+            if let Some(dialog) = &mut self.dialog {
+                dialog.terminate();
+            }
+            SipLegAction::Terminated(format!("rejected_{code}"))
+        } else {
+            SipLegAction::None // 1xx provisional
+        }
+    }
+
+    fn handle_auth_challenge(&mut self, msg: &SipMessage) -> SipLegAction {
+        if self.auth_attempted {
+            self.state = LegState::Terminated;
+            if let Some(dialog) = &mut self.dialog {
+                dialog.terminate();
+            }
+            return SipLegAction::Terminated("auth_rejected".to_string());
+        }
+        self.auth_attempted = true;
+
+        let challenge_header = match msg.get_header("Proxy-Authenticate") {
+            Some(h) => h,
+            None => {
+                self.state = LegState::Terminated;
+                return SipLegAction::Terminated("407_no_challenge".to_string());
+            }
+        };
+
+        let challenge = match parse_digest_challenge(challenge_header) {
+            Some(c) => c,
+            None => {
+                self.state = LegState::Terminated;
+                return SipLegAction::Terminated("407_bad_challenge".to_string());
+            }
+        };
+
+        let password = match &self.config.password {
+            Some(p) => p.clone(),
+            None => {
+                self.state = LegState::Terminated;
+                return SipLegAction::Terminated("407_no_password".to_string());
+            }
+        };
+
+        let aor = match &self.config.registered_aor {
+            Some(a) => a.clone(),
+            None => {
+                self.state = LegState::Terminated;
+                return SipLegAction::Terminated("407_no_aor".to_string());
+            }
+        };
+
+        let username = aor
+            .trim_start_matches("sip:")
+            .trim_start_matches("sips:")
+            .split('@')
+            .next()
+            .unwrap_or("")
+            .to_string();
+
+        let dest_uri = self
+            .invite
+            .as_ref()
+            .and_then(|i| i.request_uri())
+            .unwrap_or("")
+            .to_string();
+
+        let auth_value = compute_digest_auth(
+            &username,
+            &password,
+            &challenge.realm,
+            &challenge.nonce,
+            "INVITE",
+            &dest_uri,
+            challenge.algorithm.as_deref(),
+            challenge.opaque.as_deref(),
+        );
+
+        // ACK the 407.
+        let mut ack_buf = None;
+        if let Some(invite) = &self.invite {
+            let ack = build_non_2xx_ack(invite, msg);
+            ack_buf = Some(ack.serialize());
+        }
+
+        // Save original INVITE for retransmission handling.
+        self.orig_invite = self.invite.clone();
+
+        // Build authenticated INVITE with same From tag, CSeq=2.
+        let ip = self
+            .config
+            .public_ip
+            .as_deref()
+            .unwrap_or(&self.config.lan_ip);
+        let from_tag = self
+            .dialog
+            .as_ref()
+            .map(|d| d.local_tag.clone())
+            .unwrap_or_else(generate_tag);
+
+        let sdp = build_sdp(&SdpOptions {
+            ip,
+            port: self.config.rtp_port,
+            payload_types: &self.config.codecs,
+            ..Default::default()
+        });
+
+        let call_id = self
+            .dialog
+            .as_ref()
+            .map(|d| d.call_id.clone())
+            .unwrap_or_default();
+
+        let invite_auth = SipMessage::create_request(
+            "INVITE",
+            &dest_uri,
+            RequestOptions {
+                via_host: ip.to_string(),
+                via_port: self.config.lan_port,
+                via_transport: None,
+                via_branch: Some(generate_branch()),
+                from_uri: aor,
+                from_display_name: None,
+                from_tag: Some(from_tag),
+                to_uri: dest_uri.clone(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some(call_id),
+                cseq: Some(2),
+                contact: Some(format!("<sip:{ip}:{}>", self.config.lan_port)),
+                max_forwards: Some(70),
+                body: Some(sdp),
+                content_type: Some("application/sdp".to_string()),
+                extra_headers: Some(vec![
+                    ("Proxy-Authorization".to_string(), auth_value),
+                    ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
+                ]),
+            },
+        );
+
+        self.invite = Some(invite_auth.clone());
+        if let Some(dialog) = &mut self.dialog {
+            dialog.local_cseq = 2;
+        }
+
+        // Return both the ACK for the 407 and the new authenticated INVITE.
+        let invite_buf = invite_auth.serialize();
+        SipLegAction::AuthRetry {
+            ack_407: ack_buf,
+            invite_with_auth: invite_buf,
+        }
+    }
+
+    fn handle_request(&mut self, msg: &SipMessage) -> SipLegAction {
+        let method = msg.method().unwrap_or("");
+
+        if method == "BYE" {
+            let ok = SipMessage::create_response(200, "OK", msg, None);
+            self.state = LegState::Terminated;
+            if let Some(dialog) = &mut self.dialog {
+                dialog.terminate();
+            }
+            return SipLegAction::SendAndTerminate(ok.serialize(), "bye".to_string());
+        }
+
+        if method == "INFO" {
+            let ok = SipMessage::create_response(200, "OK", msg, None);
+            return SipLegAction::Send(ok.serialize());
+        }
+
+        SipLegAction::None
+    }
+
+    /// Build a BYE or CANCEL to tear down this leg.
+    pub fn build_hangup(&mut self) -> Option<Vec<u8>> {
+        let dialog = self.dialog.as_mut()?;
+
+        let msg = if dialog.state == DialogState::Confirmed {
+            dialog.create_request("BYE", None, None, None)
+        } else if dialog.state == DialogState::Early {
+            if let Some(invite) = &self.invite {
+                dialog.create_cancel(invite)
+            } else {
+                return None;
+            }
+        } else {
+            return None;
+        };
+
+        self.state = LegState::Terminating;
+        dialog.terminate();
+        Some(msg.serialize())
+    }
+
+    /// Get the SIP Call-ID for routing.
+    pub fn sip_call_id(&self) -> Option<&str> {
+        self.dialog.as_ref().map(|d| d.call_id.as_str())
+    }
+}
+
+/// Actions produced by the SipLeg message handler.
+pub enum SipLegAction {
+    /// No action needed.
+    None,
+    /// Send a SIP message (ACK, 200 OK to INFO, etc.).
+    Send(Vec<u8>),
+    /// Leg state changed.
+    StateChange(LegState),
+    /// Connected — send this ACK.
+    ConnectedWithAck(Vec<u8>),
+    /// Terminated with a reason.
+    Terminated(String),
+    /// Send 200 OK and terminate.
+    SendAndTerminate(Vec<u8>, String),
+    /// 407 auth retry — send ACK for 407, then send new INVITE with auth.
+    AuthRetry {
+        ack_407: Option<Vec<u8>>,
+        invite_with_auth: Vec<u8>,
+    },
+}
+
+/// Build an ACK for a non-2xx response (same transaction as the INVITE).
+fn build_non_2xx_ack(original_invite: &SipMessage, response: &SipMessage) -> SipMessage {
+    let via = original_invite.get_header("Via").unwrap_or("").to_string();
+    let from = original_invite
+        .get_header("From")
+        .unwrap_or("")
+        .to_string();
+    let to = response.get_header("To").unwrap_or("").to_string();
+    let call_id = original_invite.call_id().to_string();
+    let cseq_num: u32 = original_invite
+        .get_header("CSeq")
+        .and_then(|s| s.split_whitespace().next())
+        .and_then(|s| s.parse().ok())
+        .unwrap_or(1);
+
+    let ruri = original_invite
+        .request_uri()
+        .unwrap_or("sip:unknown")
+        .to_string();
+
+    SipMessage::new(
+        format!("ACK {ruri} SIP/2.0"),
+        vec![
+            ("Via".to_string(), via),
+            ("From".to_string(), from),
+            ("To".to_string(), to),
+            ("Call-ID".to_string(), call_id),
+            ("CSeq".to_string(), format!("{cseq_num} ACK")),
+            ("Max-Forwards".to_string(), "70".to_string()),
+            ("Content-Length".to_string(), "0".to_string()),
+        ],
+        String::new(),
+    )
+}

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

@@ -1,16 +1,17 @@
-//! WebRTC engine — manages browser PeerConnections with SIP audio bridging.
+//! WebRTC engine — manages browser PeerConnections.
 //!
-//! Browser Opus audio → Rust PeerConnection → transcode via codec-lib → SIP RTP
-//! SIP RTP → transcode via codec-lib → Rust PeerConnection → Browser Opus
+//! Audio bridging is now channel-based:
+//! - Browser Opus audio → on_track → mixer inbound channel
+//! - Mixer outbound channel → Opus RTP → TrackLocalStaticRTP → browser
+//!
+//! The mixer handles all transcoding. The WebRTC engine just shuttles raw 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 crate::mixer::RtpPacket;
+use codec_lib::PT_OPUS;
 use std::collections::HashMap;
-use std::net::SocketAddr;
 use std::sync::Arc;
-use tokio::net::UdpSocket;
-use tokio::sync::Mutex;
+use tokio::sync::{mpsc, Mutex};
 use webrtc::api::media_engine::MediaEngine;
 use webrtc::api::APIBuilder;
 use webrtc::ice_transport::ice_candidate::RTCIceCandidateInit;
@@ -22,24 +23,14 @@ 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>>>,
+    /// Channel sender for forwarding browser Opus audio to the mixer.
+    /// Set when the session is linked to a call via link_to_mixer().
+    mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>>,
 }
 
 /// Manages all WebRTC sessions.
@@ -56,7 +47,7 @@ impl WebRtcEngine {
         }
     }
 
-    /// Handle a WebRTC offer from a browser.
+    /// Handle a WebRTC offer from a browser — create PeerConnection, return SDP answer.
     pub async fn handle_offer(
         &mut self,
         session_id: &str,
@@ -99,8 +90,9 @@ impl WebRtcEngine {
             .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));
+        // Shared mixer channel sender (populated when linked to a call).
+        let mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>> =
+            Arc::new(Mutex::new(None));
 
         // ICE candidate handler.
         let out_tx_ice = self.out_tx.clone();
@@ -151,14 +143,14 @@ impl WebRtcEngine {
         }));
 
         // Track handler — receives Opus audio from the browser.
-        // When SIP bridge is set, transcodes and forwards to provider.
+        // Forwards raw Opus payload to the mixer channel (when linked).
         let out_tx_track = self.out_tx.clone();
         let sid_track = session_id.to_string();
-        let sip_bridge_for_track = sip_bridge.clone();
+        let mixer_tx_for_track = mixer_tx.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();
+            let mixer_tx = mixer_tx_for_track.clone();
             Box::pin(async move {
                 let codec_info = track.codec();
                 emit_event(
@@ -171,8 +163,8 @@ impl WebRtcEngine {
                     }),
                 );
 
-                // Spawn the browser→SIP audio forwarding task.
-                tokio::spawn(browser_to_sip_loop(track, bridge, out_tx, sid));
+                // Spawn browser→mixer forwarding task.
+                tokio::spawn(browser_to_mixer_loop(track, mixer_tx, out_tx, sid));
             })
         }));
 
@@ -199,67 +191,41 @@ impl WebRtcEngine {
                 pc,
                 local_track,
                 call_id: None,
-                sip_bridge,
+                mixer_tx,
             },
         );
 
         Ok(answer_sdp)
     }
 
-    /// Link a WebRTC session to a SIP call — sets up the audio bridge.
-    pub async fn link_to_sip(
+    /// Link a WebRTC session to a call's mixer via channels.
+    /// - `inbound_tx`: browser audio goes TO the mixer through this channel
+    /// - `outbound_rx`: mixed audio comes FROM the mixer through this channel
+    pub async fn link_to_mixer(
         &mut self,
         session_id: &str,
         call_id: &str,
-        bridge_info: SipBridgeInfo,
+        inbound_tx: mpsc::Sender<RtpPacket>,
+        outbound_rx: mpsc::Receiver<Vec<u8>>,
     ) -> 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
-        }
-    }
+        let session = match self.sessions.get_mut(session_id) {
+            Some(s) => s,
+            None => return 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"))?;
+        session.call_id = Some(call_id.to_string());
 
-        // 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(());
+        // Set the mixer sender so the on_track loop starts forwarding.
+        {
+            let mut tx = session.mixer_tx.lock().await;
+            *tx = Some(inbound_tx);
         }
 
-        // 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);
+        // Spawn mixer→browser outbound task.
+        let local_track = session.local_track.clone();
+        tokio::spawn(mixer_to_browser_loop(outbound_rx, local_track));
 
-        session
-            .local_track
-            .write(&packet)
-            .await
-            .map(|_| ())
-            .map_err(|e| format!("write: {e}"))
+        true
     }
 
     pub async fn add_ice_candidate(
@@ -294,90 +260,48 @@ impl WebRtcEngine {
         }
         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(
+/// Browser → Mixer audio forwarding loop.
+/// Reads Opus RTP from the browser track, sends raw Opus payload to the mixer channel.
+async fn browser_to_mixer_loop(
     track: Arc<webrtc::track::track_remote::TrackRemote>,
-    sip_bridge: Arc<Mutex<Option<SipBridgeInfo>>>,
+    mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>>,
     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;
+                // Send raw Opus payload to mixer (if linked).
+                let tx = mixer_tx.lock().await;
+                if let Some(ref tx) = *tx {
+                    let _ = tx
+                        .send(RtpPacket {
+                            payload: payload.to_vec(),
+                            payload_type: PT_OPUS,
+                        })
+                        .await;
+                }
+                drop(tx);
 
                 if count == 1 || count == 50 || count % 500 == 0 {
                     emit_event(
                         &out_tx,
-                        "webrtc_audio_tx",
+                        "webrtc_audio_rx",
                         serde_json::json!({
                             "session_id": session_id,
-                            "direction": "browser_to_sip",
+                            "direction": "browser_to_mixer",
                             "packet_count": count,
                         }),
                     );
@@ -387,3 +311,14 @@ async fn browser_to_sip_loop(
         }
     }
 }
+
+/// Mixer → Browser audio forwarding loop.
+/// Reads Opus-encoded RTP packets from the mixer and writes to the WebRTC track.
+async fn mixer_to_browser_loop(
+    mut outbound_rx: mpsc::Receiver<Vec<u8>>,
+    local_track: Arc<TrackLocalStaticRTP>,
+) {
+    while let Some(rtp_data) = outbound_rx.recv().await {
+        let _ = local_track.write(&rtp_data).await;
+    }
+}

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.12.0',
+  version: '1.14.0',
   description: 'undefined'
 }

ts/frontend.ts

@@ -128,14 +128,14 @@ async function handleRequest(
     }
   }
 
-  // API: add leg to call.
+  // API: add leg to call (device — not yet implemented, needs device-to-call routing).
   if (url.pathname.startsWith('/api/call/') && url.pathname.endsWith('/addleg') && method === 'POST') {
     try {
       const callId = url.pathname.split('/')[3];
       const body = await readJsonBody(req);
       if (!body?.deviceId) return sendJson(res, { ok: false, error: 'missing deviceId' }, 400);
-      const ok = callManager?.addDeviceToCall(callId, body.deviceId) ?? false;
-      return sendJson(res, { ok });
+      // TODO: implement device leg addition (needs SIP INVITE to device).
+      return sendJson(res, { ok: false, error: 'not yet implemented' }, 501);
     } catch (e: any) {
       return sendJson(res, { ok: false, error: e.message }, 400);
     }
@@ -147,8 +147,9 @@ async function handleRequest(
       const callId = url.pathname.split('/')[3];
       const body = await readJsonBody(req);
       if (!body?.number) return sendJson(res, { ok: false, error: 'missing number' }, 400);
-      const ok = callManager?.addExternalToCall(callId, body.number, body.providerId) ?? false;
-      return sendJson(res, { ok });
+      const { addLeg: addLegFn } = await import('./proxybridge.ts');
+      const legId = await addLegFn(callId, body.number, body.providerId);
+      return sendJson(res, { ok: !!legId, legId });
     } catch (e: any) {
       return sendJson(res, { ok: false, error: e.message }, 400);
     }
@@ -160,22 +161,22 @@ async function handleRequest(
       const callId = url.pathname.split('/')[3];
       const body = await readJsonBody(req);
       if (!body?.legId) return sendJson(res, { ok: false, error: 'missing legId' }, 400);
-      const ok = callManager?.removeLegFromCall(callId, body.legId) ?? false;
+      const { removeLeg: removeLegFn } = await import('./proxybridge.ts');
+      const ok = await removeLegFn(callId, body.legId);
       return sendJson(res, { ok });
     } catch (e: any) {
       return sendJson(res, { ok: false, error: e.message }, 400);
     }
   }
 
-  // API: transfer leg.
+  // API: transfer leg (not yet implemented).
   if (url.pathname === '/api/transfer' && method === 'POST') {
     try {
       const body = await readJsonBody(req);
       if (!body?.sourceCallId || !body?.legId || !body?.targetCallId) {
         return sendJson(res, { ok: false, error: 'missing sourceCallId, legId, or targetCallId' }, 400);
       }
-      const ok = callManager?.transferLeg(body.sourceCallId, body.legId, body.targetCallId) ?? false;
-      return sendJson(res, { ok });
+      return sendJson(res, { ok: false, error: 'not yet implemented' }, 501);
     } catch (e: any) {
       return sendJson(res, { ok: false, error: e.message }, 400);
     }
@@ -339,11 +340,13 @@ export function initWebUi(
   onHangupCall: (callId: string) => boolean,
   onConfigSaved?: () => void,
   callManager?: CallManager,
+  voiceboxManager?: VoiceboxManager,
   /** WebRTC signaling handlers — forwarded to Rust proxy-engine. */
   onWebRtcOffer?: (sessionId: string, sdp: string, ws: WebSocket) => Promise<void>,
   onWebRtcIce?: (sessionId: string, candidate: any) => Promise<void>,
   onWebRtcClose?: (sessionId: string) => Promise<void>,
-  voiceboxManager?: VoiceboxManager,
+  /** Called when browser sends webrtc-accept (callId + sessionId linking). */
+  onWebRtcAccept?: (callId: string, sessionId: string) => void,
 ): void {
   const WEB_PORT = 3060;
 
@@ -382,6 +385,7 @@ export function initWebUi(
         if (msg.type === 'webrtc-offer' && msg.sessionId) {
           // Forward to Rust proxy-engine for WebRTC handling.
           if (onWebRtcOffer) {
+            log(`[webrtc-ws] offer msg keys: ${Object.keys(msg).join(',')}, sdp type: ${typeof msg.sdp}, sdp len: ${msg.sdp?.length || 0}`);
             onWebRtcOffer(msg.sessionId, msg.sdp, socket as any).catch((e: any) =>
               log(`[webrtc] offer error: ${e.message}`));
           }
@@ -394,8 +398,10 @@ export function initWebUi(
             onWebRtcClose(msg.sessionId).catch(() => {});
           }
         } else if (msg.type === 'webrtc-accept' && msg.callId) {
-          // TODO: Wire to Rust call linking.
           log(`[webrtc] accept: call=${msg.callId} session=${msg.sessionId || 'none'}`);
+          if (onWebRtcAccept && msg.sessionId) {
+            onWebRtcAccept(msg.callId, msg.sessionId);
+          }
         } else if (msg.type?.startsWith('webrtc-')) {
           msg._remoteIp = remoteIp;
           handleWebRtcSignaling(socket as any, msg);

ts/proxybridge.ts

@@ -238,6 +238,38 @@ export async function webrtcLink(sessionId: string, callId: string, providerMedi
   }
 }
 
+/**
+ * Add an external SIP leg to an existing call (multiparty).
+ */
+export async function addLeg(callId: string, number: string, providerId?: string): Promise<string | null> {
+  if (!bridge || !initialized) return null;
+  try {
+    const result = await bridge.sendCommand('add_leg', {
+      call_id: callId,
+      number,
+      provider_id: providerId,
+    } as any);
+    return (result as any)?.leg_id || null;
+  } catch (e: any) {
+    logFn?.(`[proxy-engine] add_leg error: ${e?.message || e}`);
+    return null;
+  }
+}
+
+/**
+ * Remove a leg from a call.
+ */
+export async function removeLeg(callId: string, legId: string): Promise<boolean> {
+  if (!bridge || !initialized) return false;
+  try {
+    await bridge.sendCommand('remove_leg', { call_id: callId, leg_id: legId } as any);
+    return true;
+  } catch (e: any) {
+    logFn?.(`[proxy-engine] remove_leg error: ${e?.message || e}`);
+    return false;
+  }
+}
+
 /**
  * Close a WebRTC session.
  */

ts/sipproxy.ts

@@ -37,7 +37,10 @@ import {
   shutdownProxyEngine,
   webrtcOffer,
   webrtcIce,
+  webrtcLink,
   webrtcClose,
+  addLeg,
+  removeLeg,
 } from './proxybridge.ts';
 import type {
   IIncomingCallEvent,
@@ -118,6 +121,12 @@ const activeCalls = new Map<string, IActiveCall>();
 const callHistory: ICallHistoryEntry[] = [];
 const MAX_HISTORY = 100;
 
+// WebRTC session ↔ call linking state.
+// Both pieces (session accept + call media info) can arrive in any order.
+const webrtcSessionToCall = new Map<string, string>(); // sessionId → callId
+const webrtcCallToSession = new Map<string, string>(); // callId → sessionId
+const pendingCallMedia = new Map<string, { addr: string; port: number; sipPt: number }>(); // callId → provider media info
+
 // Initialize provider statuses from config (all start as unregistered).
 for (const p of appConfig.providers) {
   providerStatuses.set(p.id, {
@@ -271,6 +280,17 @@ async function startProxyEngine(): Promise<void> {
       state: 'setting-up',
       startedAt: Date.now(),
     });
+
+    // Notify all browser devices — they can connect via WebRTC to listen/talk.
+    const browserIds = getAllBrowserDeviceIds();
+    for (const bid of browserIds) {
+      sendToBrowserDevice(bid, {
+        type: 'webrtc-incoming',
+        callId: data.call_id,
+        from: data.number,
+        deviceId: bid,
+      });
+    }
   });
 
   onProxyEvent('call_ringing', (data: { call_id: string }) => {
@@ -278,12 +298,33 @@ async function startProxyEngine(): Promise<void> {
     if (call) call.state = 'ringing';
   });
 
-  onProxyEvent('call_answered', (data: { call_id: string }) => {
+  onProxyEvent('call_answered', (data: { call_id: string; provider_media_addr?: string; provider_media_port?: number; sip_pt?: number }) => {
     const call = activeCalls.get(data.call_id);
     if (call) {
       call.state = 'connected';
       log(`[call] ${data.call_id} connected`);
     }
+
+    // Try to link WebRTC session to this call for audio bridging.
+    if (data.provider_media_addr && data.provider_media_port) {
+      const sessionId = webrtcCallToSession.get(data.call_id);
+      if (sessionId) {
+        // Both session and media info available — link now.
+        const sipPt = data.sip_pt ?? 9;
+        log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${data.call_id} media=${data.provider_media_addr}:${data.provider_media_port} pt=${sipPt}`);
+        webrtcLink(sessionId, data.call_id, data.provider_media_addr, data.provider_media_port, sipPt).then((ok) => {
+          log(`[webrtc] link result: ${ok}`);
+        });
+      } else {
+        // Session not yet accepted — store media info for when it arrives.
+        pendingCallMedia.set(data.call_id, {
+          addr: data.provider_media_addr,
+          port: data.provider_media_port,
+          sipPt: data.sip_pt ?? 9,
+        });
+        log(`[webrtc] media info cached for call=${data.call_id}, waiting for session accept`);
+      }
+    }
   });
 
   onProxyEvent('call_ended', (data: ICallEndedEvent) => {
@@ -301,6 +342,18 @@ async function startProxyEngine(): Promise<void> {
       });
       if (callHistory.length > MAX_HISTORY) callHistory.pop();
       activeCalls.delete(data.call_id);
+
+      // Notify browser(s) that the call ended.
+      broadcastWs('webrtc-call-ended', { callId: data.call_id });
+
+      // Clean up WebRTC session mappings.
+      const sessionId = webrtcCallToSession.get(data.call_id);
+      if (sessionId) {
+        webrtcCallToSession.delete(data.call_id);
+        webrtcSessionToCall.delete(sessionId);
+        webrtcClose(sessionId).catch(() => {});
+      }
+      pendingCallMedia.delete(data.call_id);
     }
   });
 
@@ -308,6 +361,19 @@ async function startProxyEngine(): Promise<void> {
     log(`[sip] unhandled ${data.method_or_status} Call-ID=${data.call_id?.slice(0, 20)} from=${data.from_addr}:${data.from_port}`);
   });
 
+  // Leg events (multiparty).
+  onProxyEvent('leg_added', (data: any) => {
+    log(`[leg] added: call=${data.call_id} leg=${data.leg_id} kind=${data.kind} state=${data.state}`);
+  });
+
+  onProxyEvent('leg_removed', (data: any) => {
+    log(`[leg] removed: call=${data.call_id} leg=${data.leg_id}`);
+  });
+
+  onProxyEvent('leg_state_changed', (data: any) => {
+    log(`[leg] state: call=${data.call_id} leg=${data.leg_id} → ${data.state}`);
+  });
+
   // WebRTC events from Rust — forward ICE candidates to browser via WebSocket.
   onProxyEvent('webrtc_ice_candidate', (data: any) => {
     // Find the browser's WebSocket by session ID and send the ICE candidate.
@@ -467,14 +533,22 @@ initWebUi(
     }
   },
   undefined, // callManager — legacy, replaced by Rust proxy-engine
-  voiceboxManager,
+  voiceboxManager, // voiceboxManager
   // WebRTC signaling → forwarded to Rust proxy-engine.
   async (sessionId, sdp, ws) => {
-    log(`[webrtc] offer from browser session=${sessionId.slice(0, 8)}`);
+    log(`[webrtc] offer from browser session=${sessionId.slice(0, 8)} sdp_type=${typeof sdp} sdp_len=${sdp?.length || 0}`);
+    if (!sdp || typeof sdp !== 'string' || sdp.length < 10) {
+      log(`[webrtc] WARNING: invalid SDP (type=${typeof sdp}), skipping offer`);
+      return;
+    }
+    log(`[webrtc] sending offer to Rust (${sdp.length}b)...`);
     const result = await webrtcOffer(sessionId, sdp);
+    log(`[webrtc] Rust result: ${JSON.stringify(result)?.slice(0, 200)}`);
     if (result?.sdp) {
       ws.send(JSON.stringify({ type: 'webrtc-answer', sessionId, sdp: result.sdp }));
       log(`[webrtc] answer sent to browser session=${sessionId.slice(0, 8)}`);
+    } else {
+      log(`[webrtc] ERROR: no answer SDP from Rust`);
     }
   },
   async (sessionId, candidate) => {
@@ -483,6 +557,26 @@ initWebUi(
   async (sessionId) => {
     await webrtcClose(sessionId);
   },
+  // onWebRtcAccept — browser has accepted a call, linking session to call.
+  (callId: string, sessionId: string) => {
+    log(`[webrtc] accept: callId=${callId} sessionId=${sessionId.slice(0, 8)}`);
+
+    // Store bidirectional mapping.
+    webrtcSessionToCall.set(sessionId, callId);
+    webrtcCallToSession.set(callId, sessionId);
+
+    // Check if we already have media info for this call (provider answered first).
+    const media = pendingCallMedia.get(callId);
+    if (media) {
+      pendingCallMedia.delete(callId);
+      log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${callId} media=${media.addr}:${media.port} pt=${media.sipPt}`);
+      webrtcLink(sessionId, callId, media.addr, media.port, media.sipPt).then((ok) => {
+        log(`[webrtc] link result: ${ok}`);
+      });
+    } else {
+      log(`[webrtc] session ${sessionId.slice(0, 8)} accepted, waiting for call_answered media info`);
+    }
+  },
 );
 
 // ---------------------------------------------------------------------------

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.12.0',
+  version: '1.14.0',
   description: 'undefined'
 }