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

//! Call manager — central registry and orchestration for all calls.
//!
//! Handles:
//! - Inbound passthrough calls (provider → proxy → device)
//! - Outbound passthrough calls (device → proxy → provider)
//! - SIP message routing by Call-ID
//! - BYE/CANCEL handling
//! - RTP relay setup
//!
//! Ported from ts/call/call-manager.ts (passthrough mode).

use crate::call::{CallDirection, CallState, PassthroughCall};
use crate::config::{AppConfig, ProviderConfig};
use crate::dtmf::DtmfDetector;
use crate::ipc::{emit_event, OutTx};
use crate::registrar::Registrar;
use crate::rtp::RtpPortPool;
use crate::sip_leg::{LegState, SipLeg, SipLegAction, SipLegConfig};
use sip_proto::helpers::parse_sdp_endpoint;
use sip_proto::message::SipMessage;
use sip_proto::rewrite::{rewrite_sdp, rewrite_sip_uri};
use std::collections::HashMap;
use std::net::SocketAddr;
use std::sync::Arc;
use std::time::Instant;
use tokio::net::UdpSocket;

/// A B2BUA call with a SipLeg for the provider side.
/// The other side is either a WebRTC session or another SipLeg.
pub struct B2buaCall {
    pub id: String,
    pub provider_leg: SipLeg,
    pub webrtc_session_id: Option<String>,
    pub number: String,
    pub created_at: std::time::Instant,
    /// RTP socket allocated for the provider leg (used for WebRTC audio bridging).
    pub rtp_socket: Option<Arc<UdpSocket>>,
}

pub struct CallManager {
    /// Active passthrough calls, keyed by SIP Call-ID.
    calls: HashMap<String, PassthroughCall>,
    /// Active B2BUA calls, keyed by SIP Call-ID of the provider leg.
    b2bua_calls: HashMap<String, B2buaCall>,
    /// Call ID counter.
    next_call_num: u64,
    /// Output channel for events.
    out_tx: OutTx,
}

impl CallManager {
    pub fn new(out_tx: OutTx) -> Self {
        Self {
            calls: HashMap::new(),
            b2bua_calls: HashMap::new(),
            next_call_num: 0,
            out_tx,
        }
    }

    /// Generate a unique call ID.
    fn next_call_id(&mut self) -> String {
        let id = format!(
            "call-{}-{}",
            std::time::SystemTime::now()
                .duration_since(std::time::UNIX_EPOCH)
                .unwrap_or_default()
                .as_millis(),
            self.next_call_num,
        );
        self.next_call_num += 1;
        id
    }

    /// Try to route a SIP message to an existing call.
    /// Returns true if handled.
    pub async fn route_sip_message(
        &mut self,
        msg: &SipMessage,
        from_addr: SocketAddr,
        socket: &UdpSocket,
        config: &AppConfig,
        _registrar: &Registrar,
    ) -> bool {
        let sip_call_id = msg.call_id().to_string();

        // Check B2BUA calls first (provider legs with dialog management).
        if self.b2bua_calls.contains_key(&sip_call_id) {
            return self.route_b2bua_message(&sip_call_id, msg, from_addr, socket).await;
        }

        // Check passthrough calls.
        if !self.calls.contains_key(&sip_call_id) {
            return false;
        }

        // Extract needed data from the call to avoid borrow conflicts.
        let (call_id, provider_addr, device_addr, rtp_port, from_provider) = {
            let call = self.calls.get(&sip_call_id).unwrap();
            let from_provider = from_addr.ip().to_string() == call.provider_addr.ip().to_string();
            (
                call.id.clone(),
                call.provider_addr,
                call.device_addr,
                call.rtp_port,
                from_provider,
            )
        };

        let lan_ip = config.proxy.lan_ip.clone();
        let lan_port = config.proxy.lan_port;

        if msg.is_request() {
            let method = msg.method().unwrap_or("");
            let forward_to = if from_provider { device_addr } else { provider_addr };

            // Handle BYE.
            if method == "BYE" {
                let ok = SipMessage::create_response(200, "OK", msg, None);
                let _ = socket.send_to(&ok.serialize(), from_addr).await;
                let _ = socket.send_to(&msg.serialize(), forward_to).await;

                let duration = self.calls.get(&sip_call_id).unwrap().duration_secs();
                emit_event(
                    &self.out_tx,
                    "call_ended",
                    serde_json::json!({
                        "call_id": call_id,
                        "reason": "bye",
                        "duration": duration,
                        "from_side": if from_provider { "provider" } else { "device" },
                    }),
                );
                self.calls.get_mut(&sip_call_id).unwrap().state = CallState::Terminated;
                return true;
            }

            // Handle CANCEL.
            if method == "CANCEL" {
                let ok = SipMessage::create_response(200, "OK", msg, None);
                let _ = socket.send_to(&ok.serialize(), from_addr).await;
                let _ = socket.send_to(&msg.serialize(), forward_to).await;

                let duration = self.calls.get(&sip_call_id).unwrap().duration_secs();
                emit_event(
                    &self.out_tx,
                    "call_ended",
                    serde_json::json!({
                        "call_id": call_id, "reason": "cancel", "duration": duration,
                    }),
                );
                self.calls.get_mut(&sip_call_id).unwrap().state = CallState::Terminated;
                return true;
            }

            // Handle INFO (DTMF relay).
            if method == "INFO" {
                let ok = SipMessage::create_response(200, "OK", msg, None);
                let _ = socket.send_to(&ok.serialize(), from_addr).await;

                // Detect DTMF from INFO body.
                if let Some(ct) = msg.get_header("Content-Type") {
                    let mut detector = DtmfDetector::new(call_id.clone(), self.out_tx.clone());
                    detector.process_sip_info(ct, &msg.body);
                }
                return true;
            }

            // Forward other requests with SDP rewriting.
            let mut fwd = msg.clone();
            if from_provider {
                rewrite_sdp_for_device(&mut fwd, &lan_ip, rtp_port);
                if let Some(ruri) = fwd.request_uri().map(|s| s.to_string()) {
                    let new_ruri = rewrite_sip_uri(&ruri, &device_addr.ip().to_string(), device_addr.port());
                    fwd.set_request_uri(&new_ruri);
                }
            } else {
                rewrite_sdp_for_provider(&mut fwd, &lan_ip, rtp_port);
            }
            if fwd.is_dialog_establishing() {
                fwd.prepend_header("Record-Route", &format!("<sip:{lan_ip}:{lan_port};lr>"));
            }
            let _ = socket.send_to(&fwd.serialize(), forward_to).await;
            return true;
        }

        // --- Responses ---
        if msg.is_response() {
            let code = msg.status_code().unwrap_or(0);
            let cseq_method = msg.cseq_method().unwrap_or("").to_uppercase();
            let forward_to = if from_provider { device_addr } else { provider_addr };

            let mut fwd = msg.clone();
            if from_provider {
                rewrite_sdp_for_device(&mut fwd, &lan_ip, rtp_port);
            } else {
                rewrite_sdp_for_provider(&mut fwd, &lan_ip, rtp_port);
                if let Some(contact) = fwd.get_header("Contact").map(|s| s.to_string()) {
                    let new_contact = rewrite_sip_uri(&contact, &lan_ip, lan_port);
                    if new_contact != contact {
                        fwd.set_header("Contact", &new_contact);
                    }
                }
            }

            // State transitions.
            if cseq_method == "INVITE" {
                let call = self.calls.get_mut(&sip_call_id).unwrap();
                if (code == 180 || code == 183) && call.state == CallState::SettingUp {
                    call.state = CallState::Ringing;
                    emit_event(&self.out_tx, "call_ringing", serde_json::json!({ "call_id": call_id }));
                } else if code >= 200 && code < 300 {
                    call.state = CallState::Connected;
                    emit_event(&self.out_tx, "call_answered", serde_json::json!({ "call_id": call_id }));
                } else if code >= 300 {
                    let duration = call.duration_secs();
                    call.state = CallState::Terminated;
                    emit_event(
                        &self.out_tx,
                        "call_ended",
                        serde_json::json!({
                            "call_id": call_id,
                            "reason": format!("rejected_{code}"),
                            "duration": duration,
                        }),
                    );
                }
            }

            let _ = socket.send_to(&fwd.serialize(), forward_to).await;
            return true;
        }

        false
    }

    /// Create an inbound passthrough call (provider → device).
    pub async fn create_inbound_call(
        &mut self,
        invite: &SipMessage,
        from_addr: SocketAddr,
        provider_id: &str,
        provider_config: &ProviderConfig,
        config: &AppConfig,
        registrar: &Registrar,
        rtp_pool: &mut RtpPortPool,
        socket: &UdpSocket,
        public_ip: Option<&str>,
    ) -> Option<String> {
        let call_id = self.next_call_id();
        let lan_ip = &config.proxy.lan_ip;
        let lan_port = config.proxy.lan_port;

        // Extract caller/callee info.
        let from_header = invite.get_header("From").unwrap_or("");
        let caller_number = SipMessage::extract_uri(from_header)
            .unwrap_or("Unknown")
            .to_string();
        let called_number = invite
            .request_uri()
            .and_then(|uri| SipMessage::extract_uri(uri))
            .unwrap_or("")
            .to_string();

        // Resolve target device (first registered device).
        let device_addr = match self.resolve_first_device(config, registrar) {
            Some(addr) => addr,
            None => {
                // No device registered — route to voicemail.
                return self
                    .route_to_voicemail(
                        &call_id,
                        invite,
                        from_addr,
                        &caller_number,
                        provider_id,
                        provider_config,
                        config,
                        rtp_pool,
                        socket,
                        public_ip,
                    )
                    .await;
            }
        };

        // Allocate RTP port.
        let rtp_alloc = match rtp_pool.allocate().await {
            Some(a) => a,
            None => {
                let resp = SipMessage::create_response(503, "Service Unavailable", invite, None);
                let _ = socket.send_to(&resp.serialize(), from_addr).await;
                return None;
            }
        };

        // Create the call.
        let mut call = PassthroughCall {
            id: call_id.clone(),
            sip_call_id: invite.call_id().to_string(),
            state: CallState::Ringing,
            direction: CallDirection::Inbound,
            created_at: Instant::now(),
            caller_number: Some(caller_number),
            callee_number: Some(called_number),
            provider_id: provider_id.to_string(),
            provider_addr: from_addr,
            provider_media: None,
            device_addr,
            device_media: None,
            rtp_port: rtp_alloc.port,
            rtp_socket: rtp_alloc.socket.clone(),
            pkt_from_device: 0,
            pkt_from_provider: 0,
        };

        // Extract provider media from SDP.
        if invite.has_sdp_body() {
            if let Some(ep) = parse_sdp_endpoint(&invite.body) {
                if let Ok(addr) = format!("{}:{}", ep.address, ep.port).parse() {
                    call.provider_media = Some(addr);
                }
            }
        }

        // Start RTP relay.
        let rtp_socket = rtp_alloc.socket.clone();
        let device_addr_for_relay = device_addr;
        let provider_addr_for_relay = from_addr;
        tokio::spawn(async move {
            rtp_relay_loop(rtp_socket, device_addr_for_relay, provider_addr_for_relay).await;
        });

        // Rewrite and forward INVITE to device.
        let mut fwd_invite = invite.clone();
        fwd_invite.set_request_uri(&rewrite_sip_uri(
            fwd_invite.request_uri().unwrap_or(""),
            &device_addr.ip().to_string(),
            device_addr.port(),
        ));
        fwd_invite.prepend_header("Record-Route", &format!("<sip:{lan_ip}:{lan_port};lr>"));

        if fwd_invite.has_sdp_body() {
            let (new_body, original) = rewrite_sdp(&fwd_invite.body, lan_ip, rtp_alloc.port);
            fwd_invite.body = new_body;
            fwd_invite.update_content_length();
            if let Some(ep) = original {
                if let Ok(addr) = format!("{}:{}", ep.address, ep.port).parse() {
                    call.provider_media = Some(addr);
                }
            }
        }

        let _ = socket.send_to(&fwd_invite.serialize(), device_addr).await;

        // Store the call.
        self.calls.insert(call.sip_call_id.clone(), call);

        Some(call_id)
    }

    /// Create an outbound passthrough call (device → provider).
    pub async fn create_outbound_passthrough(
        &mut self,
        invite: &SipMessage,
        from_addr: SocketAddr,
        provider_config: &ProviderConfig,
        config: &AppConfig,
        rtp_pool: &mut RtpPortPool,
        socket: &UdpSocket,
        public_ip: Option<&str>,
    ) -> Option<String> {
        let call_id = self.next_call_id();
        let lan_ip = &config.proxy.lan_ip;
        let lan_port = config.proxy.lan_port;
        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());

        let callee = invite.request_uri().unwrap_or("").to_string();

        // Allocate RTP port.
        let rtp_alloc = match rtp_pool.allocate().await {
            Some(a) => a,
            None => return None,
        };

        let provider_dest: SocketAddr = match provider_config.outbound_proxy.to_socket_addr() {
            Some(a) => a,
            None => return None,
        };

        let mut call = PassthroughCall {
            id: call_id.clone(),
            sip_call_id: invite.call_id().to_string(),
            state: CallState::SettingUp,
            direction: CallDirection::Outbound,
            created_at: Instant::now(),
            caller_number: None,
            callee_number: Some(callee),
            provider_id: provider_config.id.clone(),
            provider_addr: provider_dest,
            provider_media: None,
            device_addr: from_addr,
            device_media: None,
            rtp_port: rtp_alloc.port,
            rtp_socket: rtp_alloc.socket.clone(),
            pkt_from_device: 0,
            pkt_from_provider: 0,
        };

        // Start RTP relay.
        let rtp_socket = rtp_alloc.socket.clone();
        let device_addr_for_relay = from_addr;
        let provider_addr_for_relay = provider_dest;
        tokio::spawn(async move {
            rtp_relay_loop(rtp_socket, device_addr_for_relay, provider_addr_for_relay).await;
        });

        // Rewrite and forward INVITE to provider.
        let mut fwd_invite = invite.clone();
        fwd_invite.prepend_header("Record-Route", &format!("<sip:{lan_ip}:{lan_port};lr>"));

        // Rewrite Contact to public IP.
        if let Some(contact) = fwd_invite.get_header("Contact").map(|s| s.to_string()) {
            let new_contact = rewrite_sip_uri(&contact, pub_ip, lan_port);
            if new_contact != contact {
                fwd_invite.set_header("Contact", &new_contact);
            }
        }

        // Rewrite SDP.
        if fwd_invite.has_sdp_body() {
            let (new_body, original) = rewrite_sdp(&fwd_invite.body, pub_ip, rtp_alloc.port);
            fwd_invite.body = new_body;
            fwd_invite.update_content_length();
            if let Some(ep) = original {
                if let Ok(addr) = format!("{}:{}", ep.address, ep.port).parse() {
                    call.device_media = Some(addr);
                }
            }
        }

        let _ = socket.send_to(&fwd_invite.serialize(), provider_dest).await;

        self.calls.insert(call.sip_call_id.clone(), call);
        Some(call_id)
    }

    /// Hangup a call by call ID (from TypeScript command).
    pub async fn hangup(&mut self, call_id: &str, socket: &UdpSocket) -> bool {
        // Find the call by our internal call ID.
        let sip_call_id = self
            .calls
            .iter()
            .find(|(_, c)| c.id == call_id)
            .map(|(k, _)| k.clone());

        let sip_call_id = match sip_call_id {
            Some(id) => id,
            None => return false,
        };

        let call = match self.calls.get_mut(&sip_call_id) {
            Some(c) => c,
            None => return false,
        };

        if call.state == CallState::Terminated {
            return false;
        }

        // Build and send BYE to both sides.
        // For passthrough, we build a simple BYE using the SIP Call-ID.
        let bye_msg = format!(
            "BYE sip:hangup SIP/2.0\r\n\
             Via: SIP/2.0/UDP 0.0.0.0:0;branch=z9hG4bK-hangup\r\n\
             Call-ID: {}\r\n\
             CSeq: 99 BYE\r\n\
             Max-Forwards: 70\r\n\
             Content-Length: 0\r\n\r\n",
            sip_call_id
        );
        let bye_bytes = bye_msg.as_bytes();

        let _ = socket.send_to(bye_bytes, call.provider_addr).await;
        let _ = socket.send_to(bye_bytes, call.device_addr).await;

        call.state = CallState::Terminated;

        emit_event(
            &self.out_tx,
            "call_ended",
            serde_json::json!({
                "call_id": call.id,
                "reason": "hangup_command",
                "duration": call.duration_secs(),
            }),
        );

        true
    }

    /// Get all active call statuses.
    pub fn get_all_statuses(&self) -> Vec<serde_json::Value> {
        self.calls
            .values()
            .filter(|c| c.state != CallState::Terminated)
            .map(|c| c.to_status_json())
            .collect()
    }

    /// Clean up terminated calls.
    pub fn cleanup_terminated(&mut self) {
        self.calls.retain(|_, c| c.state != CallState::Terminated);
    }

    /// Check if a SIP Call-ID belongs to any active call.
    pub fn has_call(&self, sip_call_id: &str) -> bool {
        self.calls.contains_key(sip_call_id) || self.b2bua_calls.contains_key(sip_call_id)
    }

    /// Get the RTP socket for a B2BUA call (by our internal call ID).
    /// Used by webrtc_link to set up the audio bridge.
    pub fn get_b2bua_rtp_socket(&self, call_id: &str) -> Option<Arc<UdpSocket>> {
        for b2bua in self.b2bua_calls.values() {
            if b2bua.id == call_id {
                return b2bua.rtp_socket.clone();
            }
        }
        None
    }

    // --- B2BUA outbound call ---

    /// Route a SIP message to a B2BUA call's provider leg.
    async fn route_b2bua_message(
        &mut self,
        sip_call_id: &str,
        msg: &SipMessage,
        from_addr: SocketAddr,
        socket: &UdpSocket,
    ) -> bool {
        let b2bua = match self.b2bua_calls.get_mut(sip_call_id) {
            Some(c) => c,
            None => return false,
        };

        let call_id = b2bua.id.clone();
        let action = b2bua.provider_leg.handle_message(msg);

        match action {
            SipLegAction::None => {}
            SipLegAction::Send(buf) => {
                let _ = socket.send_to(&buf, b2bua.provider_leg.config.sip_target).await;
            }
            SipLegAction::StateChange(LegState::Ringing) => {
                emit_event(&self.out_tx, "call_ringing", serde_json::json!({ "call_id": call_id }));
            }
            SipLegAction::ConnectedWithAck(ack_buf) => {
                let _ = socket.send_to(&ack_buf, b2bua.provider_leg.config.sip_target).await;
                let remote = b2bua.provider_leg.remote_media;
                let sip_pt = b2bua.provider_leg.config.codecs.first().copied().unwrap_or(9);
                emit_event(&self.out_tx, "call_answered", serde_json::json!({
                    "call_id": call_id,
                    "provider_media_addr": remote.map(|a| a.ip().to_string()),
                    "provider_media_port": remote.map(|a| a.port()),
                    "sip_pt": sip_pt,
                }));
            }
            SipLegAction::Terminated(reason) => {
                let duration = b2bua.created_at.elapsed().as_secs();
                emit_event(&self.out_tx, "call_ended", serde_json::json!({
                    "call_id": call_id, "reason": reason, "duration": duration,
                }));
                self.b2bua_calls.remove(sip_call_id);
                return true;
            }
            SipLegAction::SendAndTerminate(buf, reason) => {
                let _ = socket.send_to(&buf, from_addr).await;
                let duration = b2bua.created_at.elapsed().as_secs();
                emit_event(&self.out_tx, "call_ended", serde_json::json!({
                    "call_id": call_id, "reason": reason, "duration": duration,
                }));
                self.b2bua_calls.remove(sip_call_id);
                return true;
            }
            SipLegAction::AuthRetry { ack_407, invite_with_auth } => {
                let target = b2bua.provider_leg.config.sip_target;
                if let Some(ack) = ack_407 {
                    let _ = socket.send_to(&ack, target).await;
                }
                let _ = socket.send_to(&invite_with_auth, target).await;
            }
            _ => {}
        }

        true
    }

    /// Initiate an outbound call from the dashboard using B2BUA mode.
    /// Creates a SipLeg for the provider side with proper dialog + auth handling.
    pub async fn make_outbound_call(
        &mut self,
        number: &str,
        provider_config: &ProviderConfig,
        config: &AppConfig,
        rtp_pool: &mut RtpPortPool,
        socket: &UdpSocket,
        public_ip: Option<&str>,
        registered_aor: &str,
    ) -> Option<String> {
        let call_id = self.next_call_id();
        let lan_ip = &config.proxy.lan_ip;
        let lan_port = config.proxy.lan_port;

        let provider_dest: SocketAddr = match provider_config.outbound_proxy.to_socket_addr() {
            Some(a) => a,
            None => return None,
        };

        // Allocate RTP port for the provider leg.
        let rtp_alloc = match rtp_pool.allocate().await {
            Some(a) => a,
            None => return None,
        };

        // Build the SIP Call-ID for the provider dialog.
        let sip_call_id = sip_proto::helpers::generate_call_id(None);

        // Create a SipLeg with provider credentials for auth handling.
        let leg_config = SipLegConfig {
            lan_ip: lan_ip.clone(),
            lan_port,
            public_ip: public_ip.map(|s| s.to_string()),
            sip_target: provider_dest,
            username: Some(provider_config.username.clone()),
            password: Some(provider_config.password.clone()),
            registered_aor: Some(registered_aor.to_string()),
            codecs: provider_config.codecs.clone(),
            rtp_port: rtp_alloc.port,
        };

        let mut leg = SipLeg::new(format!("{call_id}-prov"), leg_config);

        // Send the INVITE.
        let to_uri = format!("sip:{number}@{}", provider_config.domain);
        leg.send_invite(registered_aor, &to_uri, &sip_call_id, socket).await;

        // Store as B2BUA call.
        let b2bua = B2buaCall {
            id: call_id.clone(),
            provider_leg: leg,
            webrtc_session_id: None,
            number: number.to_string(),
            created_at: std::time::Instant::now(),
            rtp_socket: Some(rtp_alloc.socket.clone()),
        };
        self.b2bua_calls.insert(sip_call_id, b2bua);

        Some(call_id)
    }

    // --- Voicemail ---

    /// Route a call to voicemail: answer the INVITE, play greeting, record message.
    async fn route_to_voicemail(
        &mut self,
        call_id: &str,
        invite: &SipMessage,
        from_addr: SocketAddr,
        caller_number: &str,
        provider_id: &str,
        provider_config: &ProviderConfig,
        config: &AppConfig,
        rtp_pool: &mut RtpPortPool,
        socket: &UdpSocket,
        public_ip: Option<&str>,
    ) -> Option<String> {
        let lan_ip = &config.proxy.lan_ip;
        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());

        // Allocate RTP port for the voicemail session.
        let rtp_alloc = match rtp_pool.allocate().await {
            Some(a) => a,
            None => {
                let resp =
                    SipMessage::create_response(503, "Service Unavailable", invite, None);
                let _ = socket.send_to(&resp.serialize(), from_addr).await;
                return None;
            }
        };

        // Determine provider's preferred codec.
        let codec_pt = provider_config.codecs.first().copied().unwrap_or(9); // default G.722

        // Build SDP with our RTP port.
        let sdp = sip_proto::helpers::build_sdp(&sip_proto::helpers::SdpOptions {
            ip: pub_ip,
            port: rtp_alloc.port,
            payload_types: &provider_config.codecs,
            ..Default::default()
        });

        // Answer the INVITE with 200 OK.
        let response = SipMessage::create_response(
            200,
            "OK",
            invite,
            Some(sip_proto::message::ResponseOptions {
                to_tag: Some(sip_proto::helpers::generate_tag()),
                contact: Some(format!("<sip:{}:{}>", lan_ip, config.proxy.lan_port)),
                body: Some(sdp),
                content_type: Some("application/sdp".to_string()),
                ..Default::default()
            }),
        );
        let _ = socket.send_to(&response.serialize(), from_addr).await;

        // Extract provider media from original SDP.
        let provider_media = if invite.has_sdp_body() {
            sip_proto::helpers::parse_sdp_endpoint(&invite.body)
                .and_then(|ep| format!("{}:{}", ep.address, ep.port).parse().ok())
        } else {
            Some(from_addr) // fallback to signaling address
        };
        let provider_media = provider_media.unwrap_or(from_addr);

        // Create a voicemail call entry for BYE routing.
        let call = PassthroughCall {
            id: call_id.to_string(),
            sip_call_id: invite.call_id().to_string(),
            state: CallState::Voicemail,
            direction: CallDirection::Inbound,
            created_at: std::time::Instant::now(),
            caller_number: Some(caller_number.to_string()),
            callee_number: None,
            provider_id: provider_id.to_string(),
            provider_addr: from_addr,
            provider_media: Some(provider_media),
            device_addr: from_addr, // no device — just use provider addr as placeholder
            device_media: None,
            rtp_port: rtp_alloc.port,
            rtp_socket: rtp_alloc.socket.clone(),
            pkt_from_device: 0,
            pkt_from_provider: 0,
        };
        self.calls.insert(invite.call_id().to_string(), call);

        // Build recording file path.
        let timestamp = std::time::SystemTime::now()
            .duration_since(std::time::UNIX_EPOCH)
            .unwrap_or_default()
            .as_millis();
        let recording_dir = format!(".nogit/voicemail/default");
        let recording_path = format!("{recording_dir}/msg-{timestamp}.wav");

        // Look for a greeting WAV file.
        let greeting_wav = find_greeting_wav();

        // Spawn the voicemail session.
        let out_tx = self.out_tx.clone();
        let call_id_owned = call_id.to_string();
        let caller_owned = caller_number.to_string();
        let rtp_socket = rtp_alloc.socket;
        tokio::spawn(async move {
            crate::voicemail::run_voicemail_session(
                rtp_socket,
                provider_media,
                codec_pt,
                greeting_wav,
                recording_path,
                120_000, // max 120 seconds
                call_id_owned,
                caller_owned,
                out_tx,
            )
            .await;
        });

        Some(call_id.to_string())
    }

    // --- Internal helpers ---

    fn resolve_first_device(&self, config: &AppConfig, registrar: &Registrar) -> Option<SocketAddr> {
        for device in &config.devices {
            if let Some(addr) = registrar.get_device_contact(&device.id) {
                return Some(addr);
            }
        }
        None // No device registered — caller goes to voicemail.
    }
}

/// Find a voicemail greeting WAV file.
fn find_greeting_wav() -> Option<String> {
    // Check common locations for a pre-generated greeting.
    let candidates = [
        ".nogit/voicemail/default/greeting.wav",
        ".nogit/voicemail/greeting.wav",
    ];
    for path in &candidates {
        if std::path::Path::new(path).exists() {
            return Some(path.to_string());
        }
    }
    None // No greeting found — voicemail will just play the beep.
}

/// Rewrite SDP for provider→device direction (use LAN IP).
fn rewrite_sdp_for_device(msg: &mut SipMessage, lan_ip: &str, rtp_port: u16) {
    if msg.has_sdp_body() {
        let (new_body, _original) = rewrite_sdp(&msg.body, lan_ip, rtp_port);
        msg.body = new_body;
        msg.update_content_length();
    }
}

/// Rewrite SDP for device→provider direction (use public IP).
fn rewrite_sdp_for_provider(msg: &mut SipMessage, pub_ip: &str, rtp_port: u16) {
    if msg.has_sdp_body() {
        let (new_body, _original) = rewrite_sdp(&msg.body, pub_ip, rtp_port);
        msg.body = new_body;
        msg.update_content_length();
    }
}

/// Bidirectional RTP relay loop.
/// Receives packets on the relay socket and forwards based on source address.
async fn rtp_relay_loop(
    socket: Arc<UdpSocket>,
    device_addr: SocketAddr,
    provider_addr: SocketAddr,
) {
    let mut buf = vec![0u8; 65535];
    let device_ip = device_addr.ip().to_string();
    let provider_ip = provider_addr.ip().to_string();

    // Track learned media endpoints (may differ from signaling addresses).
    let mut learned_device: Option<SocketAddr> = None;
    let mut learned_provider: Option<SocketAddr> = None;

    loop {
        match socket.recv_from(&mut buf).await {
            Ok((n, from)) => {
                let data = &buf[..n];
                let from_ip = from.ip().to_string();

                if from_ip == device_ip || learned_device.map(|d| d == from).unwrap_or(false) {
                    // From device → forward to provider.
                    if learned_device.is_none() {
                        learned_device = Some(from);
                    }
                    if let Some(target) = learned_provider {
                        let _ = socket.send_to(data, target).await;
                    } else {
                        // Provider media not yet learned; try signaling address.
                        let _ = socket.send_to(data, provider_addr).await;
                    }
                } else if from_ip == provider_ip
                    || learned_provider.map(|p| p == from).unwrap_or(false)
                {
                    // From provider → forward to device.
                    if learned_provider.is_none() {
                        learned_provider = Some(from);
                    }
                    if let Some(target) = learned_device {
                        let _ = socket.send_to(data, target).await;
                    } else {
                        let _ = socket.send_to(data, device_addr).await;
                    }
                } else {
                    // Unknown source — try to identify by known device addresses.
                    // For now, assume it's the device if not from provider IP range.
                    if learned_device.is_none() {
                        learned_device = Some(from);
                    }
                }
            }
            Err(_) => {
                // Socket closed or error — exit relay.
                break;
            }
        }
    }
}