feat(proxy-engine,webrtc): add B2BUA SIP leg handling and WebRTC call bridging for outbound calls
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@@ -16,6 +16,7 @@ mod provider;
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mod recorder;
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mod registrar;
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mod rtp;
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mod sip_leg;
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mod sip_transport;
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mod voicemail;
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mod webrtc_engine;
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@@ -35,14 +36,15 @@ use tokio::io::{AsyncBufReadExt, AsyncWriteExt, BufReader};
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use tokio::net::UdpSocket;
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use tokio::sync::{mpsc, Mutex};
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/// Shared mutable state for the proxy engine.
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/// Shared mutable state for the proxy engine (SIP side).
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/// WebRTC is intentionally kept in a separate lock to avoid contention
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/// between SIP packet handlers and WebRTC command handlers.
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struct ProxyEngine {
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config: Option<AppConfig>,
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transport: Option<SipTransport>,
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provider_mgr: ProviderManager,
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registrar: Registrar,
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call_mgr: CallManager,
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webrtc: WebRtcEngine,
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rtp_pool: Option<RtpPortPool>,
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out_tx: OutTx,
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}
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@@ -55,7 +57,6 @@ impl ProxyEngine {
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provider_mgr: ProviderManager::new(out_tx.clone()),
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registrar: Registrar::new(out_tx.clone()),
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call_mgr: CallManager::new(out_tx.clone()),
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webrtc: WebRtcEngine::new(out_tx.clone()),
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rtp_pool: None,
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out_tx,
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}
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@@ -83,9 +84,12 @@ async fn main() {
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// Emit ready event.
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emit_event(&out_tx, "ready", serde_json::json!({}));
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// Shared engine state.
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// Shared engine state (SIP side).
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let engine = Arc::new(Mutex::new(ProxyEngine::new(out_tx.clone())));
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// WebRTC engine — separate lock to avoid deadlock with SIP handlers.
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let webrtc = Arc::new(Mutex::new(WebRtcEngine::new(out_tx.clone())));
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// Read commands from stdin.
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let stdin = tokio::io::stdin();
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let reader = BufReader::new(stdin);
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@@ -105,25 +109,34 @@ async fn main() {
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};
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let engine = engine.clone();
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let webrtc = webrtc.clone();
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let out_tx = out_tx.clone();
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// Handle commands — some are async, so we spawn.
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tokio::spawn(async move {
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handle_command(engine, &out_tx, cmd).await;
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handle_command(engine, webrtc, &out_tx, cmd).await;
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});
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}
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}
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async fn handle_command(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: Command) {
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async fn handle_command(
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engine: Arc<Mutex<ProxyEngine>>,
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webrtc: Arc<Mutex<WebRtcEngine>>,
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out_tx: &OutTx,
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cmd: Command,
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) {
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match cmd.method.as_str() {
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// SIP commands — lock engine only.
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"configure" => handle_configure(engine, out_tx, &cmd).await,
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"hangup" => handle_hangup(engine, out_tx, &cmd).await,
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"make_call" => handle_make_call(engine, out_tx, &cmd).await,
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"get_status" => handle_get_status(engine, out_tx, &cmd).await,
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"webrtc_offer" => handle_webrtc_offer(engine, out_tx, &cmd).await,
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"webrtc_ice" => handle_webrtc_ice(engine, out_tx, &cmd).await,
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"webrtc_link" => handle_webrtc_link(engine, out_tx, &cmd).await,
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"webrtc_close" => handle_webrtc_close(engine, out_tx, &cmd).await,
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// WebRTC commands — lock webrtc only (no engine contention).
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"webrtc_offer" => handle_webrtc_offer(webrtc, out_tx, &cmd).await,
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"webrtc_ice" => handle_webrtc_ice(webrtc, out_tx, &cmd).await,
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"webrtc_close" => handle_webrtc_close(webrtc, out_tx, &cmd).await,
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// webrtc_link needs both: engine (for RTP socket) and webrtc (for session).
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"webrtc_link" => handle_webrtc_link(engine, webrtc, out_tx, &cmd).await,
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_ => respond_err(out_tx, &cmd.id, &format!("unknown command: {}", cmd.method)),
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}
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}
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@@ -524,7 +537,8 @@ async fn handle_hangup(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Co
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}
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/// Handle `webrtc_offer` — browser sends SDP offer, we create PeerConnection and return answer.
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async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
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/// Uses only the WebRTC lock — no contention with SIP handlers.
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async fn handle_webrtc_offer(webrtc: Arc<Mutex<WebRtcEngine>>, out_tx: &OutTx, cmd: &Command) {
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let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
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Some(s) => s.to_string(),
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None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
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@@ -534,8 +548,8 @@ async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cm
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None => { respond_err(out_tx, &cmd.id, "missing sdp"); return; }
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};
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let mut eng = engine.lock().await;
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match eng.webrtc.handle_offer(&session_id, &offer_sdp).await {
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let mut wrtc = webrtc.lock().await;
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match wrtc.handle_offer(&session_id, &offer_sdp).await {
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Ok(answer_sdp) => {
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respond_ok(out_tx, &cmd.id, serde_json::json!({
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"session_id": session_id,
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@@ -547,7 +561,8 @@ async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cm
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}
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/// Handle `webrtc_ice` — forward ICE candidate from browser to Rust PeerConnection.
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async fn handle_webrtc_ice(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
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/// Uses only the WebRTC lock.
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async fn handle_webrtc_ice(webrtc: Arc<Mutex<WebRtcEngine>>, out_tx: &OutTx, cmd: &Command) {
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let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
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Some(s) => s.to_string(),
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None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
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@@ -556,15 +571,22 @@ async fn handle_webrtc_ice(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd:
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let sdp_mid = cmd.params.get("sdp_mid").and_then(|v| v.as_str());
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let sdp_mline_index = cmd.params.get("sdp_mline_index").and_then(|v| v.as_u64()).map(|v| v as u16);
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let eng = engine.lock().await;
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match eng.webrtc.add_ice_candidate(&session_id, candidate, sdp_mid, sdp_mline_index).await {
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let wrtc = webrtc.lock().await;
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match wrtc.add_ice_candidate(&session_id, candidate, sdp_mid, sdp_mline_index).await {
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Ok(()) => respond_ok(out_tx, &cmd.id, serde_json::json!({})),
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Err(e) => respond_err(out_tx, &cmd.id, &e),
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}
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}
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/// Handle `webrtc_link` — link a WebRTC session to a SIP call for audio bridging.
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async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
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/// Briefly locks engine to get the RTP socket, then locks webrtc to set up the bridge.
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/// Locks are never held simultaneously — no deadlock possible.
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async fn handle_webrtc_link(
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engine: Arc<Mutex<ProxyEngine>>,
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webrtc: Arc<Mutex<WebRtcEngine>>,
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out_tx: &OutTx,
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cmd: &Command,
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) {
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let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
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Some(s) => s.to_string(),
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None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
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@@ -588,19 +610,29 @@ async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd
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Err(e) => { respond_err(out_tx, &cmd.id, &format!("bad address: {e}")); return; }
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};
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let mut eng = engine.lock().await;
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let sip_socket = match &eng.transport {
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Some(t) => t.socket(),
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None => { respond_err(out_tx, &cmd.id, "not initialized"); return; }
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// Briefly lock engine to get the B2BUA call's RTP socket.
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let rtp_socket = {
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let eng = engine.lock().await;
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eng.call_mgr.get_b2bua_rtp_socket(&call_id)
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}; // engine lock released here
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let rtp_socket = match rtp_socket {
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Some(s) => s,
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None => {
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respond_err(out_tx, &cmd.id, &format!("call {call_id} not found or no RTP socket"));
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return;
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}
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};
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let bridge_info = crate::webrtc_engine::SipBridgeInfo {
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provider_media,
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sip_pt,
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sip_socket,
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rtp_socket,
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};
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if eng.webrtc.link_to_sip(&session_id, &call_id, bridge_info).await {
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// Lock webrtc to set up the audio bridge.
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let mut wrtc = webrtc.lock().await;
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if wrtc.link_to_sip(&session_id, &call_id, bridge_info).await {
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respond_ok(out_tx, &cmd.id, serde_json::json!({
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"session_id": session_id,
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"call_id": call_id,
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@@ -612,14 +644,15 @@ async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd
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}
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/// Handle `webrtc_close` — close a WebRTC session.
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async fn handle_webrtc_close(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
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/// Uses only the WebRTC lock.
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async fn handle_webrtc_close(webrtc: Arc<Mutex<WebRtcEngine>>, out_tx: &OutTx, cmd: &Command) {
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let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
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Some(s) => s.to_string(),
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None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
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};
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let mut eng = engine.lock().await;
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match eng.webrtc.close_session(&session_id).await {
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let mut wrtc = webrtc.lock().await;
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match wrtc.close_session(&session_id).await {
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Ok(()) => respond_ok(out_tx, &cmd.id, serde_json::json!({})),
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Err(e) => respond_err(out_tx, &cmd.id, &e),
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}
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