v1.12.0

changelog.md

@@ -1,5 +1,28 @@
 # Changelog
 
+## 2026-04-10 - 1.12.0 - feat(proxy-engine)
+add Rust-based outbound calling, WebRTC bridging, and voicemail handling
+
+- adds outbound call origination through the Rust proxy engine with dashboard make_call support
+- routes unanswered inbound calls to voicemail, including greeting playback, beep generation, and WAV message recording
+- introduces Rust WebRTC session handling and SIP audio bridging, replacing the previous TypeScript WebRTC path
+- moves SIP registration and routing responsibilities further into the Rust proxy engine and removes legacy TypeScript call/SIP modules
+
+## 2026-04-10 - 1.11.0 - feat(rust-proxy-engine)
+add a Rust SIP proxy engine with shared SIP and codec libraries
+
+- add new Rust workspace crates for proxy-engine, sip-proto, and codec-lib
+- move transcoding logic out of opus-codec into reusable codec-lib and keep opus-codec as a thin CLI wrapper
+- implement SIP message parsing, dialog handling, SDP/URI rewriting, provider registration, device registration, call management, RTP relay, and DTMF detection in Rust
+- add a TypeScript proxy bridge and update the SIP proxy entrypoint to spawn and configure the Rust engine as the SIP data plane
+
+## 2026-04-10 - 1.10.0 - feat(call, voicemail, ivr)
+add voicemail and IVR call flows with DTMF handling, prompt playback, recording, and dashboard management
+
+- introduces system call legs, DTMF detection, prompt caching, and audio recording to support automated call handling
+- adds configurable voiceboxes and IVR menus to routing, including voicemail fallback on busy or no-answer flows
+- exposes voicemail message APIs, message waiting counts, and new dashboard views for voicemail and IVR management
+
 ## 2026-04-10 - 1.9.0 - feat(routing)
 add rule-based SIP routing for inbound and outbound calls with dashboard route management
 

package.json

@@ -1,6 +1,6 @@
 {
   "name": "siprouter",
-  "version": "1.9.0",
+  "version": "1.12.0",
   "private": true,
   "type": "module",
   "scripts": {
@@ -14,7 +14,6 @@
     "@design.estate/dees-element": "^2.2.4",
     "@push.rocks/smartrust": "^1.3.2",
     "@push.rocks/smartstate": "^2.3.0",
-    "werift": "^0.22.9",
     "ws": "^8.20.0"
   },
   "devDependencies": {

rust/Cargo.toml

@@ -1,5 +1,11 @@
 [workspace]
-members = ["crates/opus-codec", "crates/tts-engine"]
+members = [
+    "crates/codec-lib",
+    "crates/opus-codec",
+    "crates/tts-engine",
+    "crates/sip-proto",
+    "crates/proxy-engine",
+]
 resolver = "2"
 
 [profile.release]

rust/crates/codec-lib/Cargo.toml

@@ -0,0 +1,10 @@
+[package]
+name = "codec-lib"
+version = "0.1.0"
+edition = "2021"
+
+[dependencies]
+audiopus = "0.3.0-rc.0"
+ezk-g722 = "0.1"
+rubato = "0.14"
+nnnoiseless = { version = "0.5", default-features = false }

rust/crates/codec-lib/src/lib.rs

@@ -0,0 +1,349 @@
+//! Audio codec library for the SIP router.
+//!
+//! Handles Opus ↔ G.722 ↔ PCMU/PCMA transcoding with ML noise suppression.
+//! Used by both the standalone `opus-codec` CLI and the `proxy-engine` binary.
+
+use audiopus::coder::{Decoder as OpusDecoder, Encoder as OpusEncoder};
+use audiopus::packet::Packet as OpusPacket;
+use audiopus::{Application, Bitrate as OpusBitrate, Channels, MutSignals, SampleRate};
+use ezk_g722::libg722::{self, Bitrate};
+use nnnoiseless::DenoiseState;
+use rubato::{FftFixedIn, Resampler};
+use std::collections::HashMap;
+
+// ---- Payload type constants ------------------------------------------------
+
+pub const PT_PCMU: u8 = 0;
+pub const PT_PCMA: u8 = 8;
+pub const PT_G722: u8 = 9;
+pub const PT_OPUS: u8 = 111;
+
+/// Return the native sample rate for a given payload type.
+pub fn codec_sample_rate(pt: u8) -> u32 {
+    match pt {
+        PT_OPUS => 48000,
+        PT_G722 => 16000,
+        _ => 8000, // PCMU, PCMA
+    }
+}
+
+// ---- G.711 µ-law (PCMU) ---------------------------------------------------
+
+pub fn mulaw_encode(sample: i16) -> u8 {
+    const BIAS: i16 = 0x84;
+    const CLIP: i16 = 32635;
+    let sign = if sample < 0 { 0x80u8 } else { 0 };
+    let mut s = (sample as i32).unsigned_abs().min(CLIP as u32) as i16;
+    s += BIAS;
+    let mut exp = 7u8;
+    let mut mask = 0x4000i16;
+    while exp > 0 && (s & mask) == 0 {
+        exp -= 1;
+        mask >>= 1;
+    }
+    let mantissa = ((s >> (exp + 3)) & 0x0f) as u8;
+    !(sign | (exp << 4) | mantissa)
+}
+
+pub fn mulaw_decode(mulaw: u8) -> i16 {
+    let v = !mulaw;
+    let sign = v & 0x80;
+    let exp = (v >> 4) & 0x07;
+    let mantissa = v & 0x0f;
+    // Use i32 to avoid overflow when exp=7, mantissa=15 (result > i16::MAX).
+    let mut sample = (((mantissa as i32) << 4) + 0x84) << exp;
+    sample -= 0x84;
+    let sample = if sign != 0 { -sample } else { sample };
+    sample.clamp(-32768, 32767) as i16
+}
+
+// ---- G.711 A-law (PCMA) ---------------------------------------------------
+
+pub fn alaw_encode(sample: i16) -> u8 {
+    let sign = if sample >= 0 { 0x80u8 } else { 0 };
+    let s = (sample as i32).unsigned_abs().min(32767) as i16;
+    let mut exp = 7u8;
+    let mut mask = 0x4000i16;
+    while exp > 0 && (s & mask) == 0 {
+        exp -= 1;
+        mask >>= 1;
+    }
+    let mantissa = if exp > 0 {
+        ((s >> (exp + 3)) & 0x0f) as u8
+    } else {
+        ((s >> 4) & 0x0f) as u8
+    };
+    (sign | (exp << 4) | mantissa) ^ 0x55
+}
+
+pub fn alaw_decode(alaw: u8) -> i16 {
+    let v = alaw ^ 0x55;
+    let sign = v & 0x80;
+    let exp = (v >> 4) & 0x07;
+    let mantissa = v & 0x0f;
+    // Use i32 to avoid overflow for extreme values.
+    let sample = if exp == 0 {
+        ((mantissa as i32) << 4) + 8
+    } else {
+        (((mantissa as i32) << 4) + 0x108) << (exp - 1)
+    };
+    let sample = if sign != 0 { sample } else { -sample };
+    sample.clamp(-32768, 32767) as i16
+}
+
+// ---- TranscodeState --------------------------------------------------------
+
+/// Per-session codec state holding Opus, G.722, resampler, and denoiser instances.
+///
+/// Each concurrent call should get its own `TranscodeState` to prevent stateful
+/// codecs (Opus, G.722 ADPCM) from corrupting each other.
+pub struct TranscodeState {
+    opus_enc: OpusEncoder,
+    opus_dec: OpusDecoder,
+    g722_enc: libg722::encoder::Encoder,
+    g722_dec: libg722::decoder::Decoder,
+    /// Cached FFT resamplers keyed by (from_rate, to_rate, chunk_size).
+    resamplers: HashMap<(u32, u32, usize), FftFixedIn<f64>>,
+    /// ML noise suppression for the SIP-bound direction.
+    denoiser_to_sip: Box<DenoiseState<'static>>,
+    /// ML noise suppression for the browser-bound direction.
+    denoiser_to_browser: Box<DenoiseState<'static>>,
+}
+
+impl TranscodeState {
+    /// Create a new transcoding session with fresh codec state.
+    pub fn new() -> Result<Self, String> {
+        let mut opus_enc =
+            OpusEncoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
+                .map_err(|e| format!("opus encoder: {e}"))?;
+        opus_enc
+            .set_complexity(5)
+            .map_err(|e| format!("opus set_complexity: {e}"))?;
+        opus_enc
+            .set_bitrate(OpusBitrate::BitsPerSecond(24000))
+            .map_err(|e| format!("opus set_bitrate: {e}"))?;
+        let opus_dec = OpusDecoder::new(SampleRate::Hz48000, Channels::Mono)
+            .map_err(|e| format!("opus decoder: {e}"))?;
+        let g722_enc = libg722::encoder::Encoder::new(Bitrate::Mode1_64000, false, false);
+        let g722_dec = libg722::decoder::Decoder::new(Bitrate::Mode1_64000, false, false);
+
+        Ok(Self {
+            opus_enc,
+            opus_dec,
+            g722_enc,
+            g722_dec,
+            resamplers: HashMap::new(),
+            denoiser_to_sip: DenoiseState::new(),
+            denoiser_to_browser: DenoiseState::new(),
+        })
+    }
+
+    /// High-quality sample rate conversion using rubato FFT resampler.
+    /// Resamplers are cached by (from_rate, to_rate, chunk_size) and reused,
+    /// maintaining proper inter-frame state for continuous audio streams.
+    pub fn resample(
+        &mut self,
+        pcm: &[i16],
+        from_rate: u32,
+        to_rate: u32,
+    ) -> Result<Vec<i16>, String> {
+        if from_rate == to_rate || pcm.is_empty() {
+            return Ok(pcm.to_vec());
+        }
+
+        let chunk = pcm.len();
+        let key = (from_rate, to_rate, chunk);
+
+        if !self.resamplers.contains_key(&key) {
+            let r =
+                FftFixedIn::<f64>::new(from_rate as usize, to_rate as usize, chunk, 1, 1)
+                    .map_err(|e| format!("resampler {from_rate}->{to_rate}: {e}"))?;
+            self.resamplers.insert(key, r);
+        }
+        let resampler = self.resamplers.get_mut(&key).unwrap();
+
+        let float_in: Vec<f64> = pcm.iter().map(|&s| s as f64 / 32768.0).collect();
+        let input = vec![float_in];
+
+        let result = resampler
+            .process(&input, None)
+            .map_err(|e| format!("resample {from_rate}->{to_rate}: {e}"))?;
+
+        Ok(result[0]
+            .iter()
+            .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
+            .collect())
+    }
+
+    /// Apply RNNoise ML noise suppression to 48kHz PCM audio.
+    /// Processes in 480-sample (10ms) frames. State persists across calls.
+    pub fn denoise(denoiser: &mut DenoiseState, pcm: &[i16]) -> Vec<i16> {
+        let frame_size = DenoiseState::FRAME_SIZE; // 480
+        let total = pcm.len();
+        let whole = (total / frame_size) * frame_size;
+        let mut output = Vec::with_capacity(total);
+        let mut out_buf = [0.0f32; 480];
+
+        for offset in (0..whole).step_by(frame_size) {
+            let input: Vec<f32> = pcm[offset..offset + frame_size]
+                .iter()
+                .map(|&s| s as f32)
+                .collect();
+            denoiser.process_frame(&mut out_buf, &input);
+            output.extend(
+                out_buf
+                    .iter()
+                    .map(|&s| s.round().clamp(-32768.0, 32767.0) as i16),
+            );
+        }
+        if whole < total {
+            output.extend_from_slice(&pcm[whole..]);
+        }
+        output
+    }
+
+    /// Transcode audio payload from one codec to another.
+    ///
+    /// `direction`: `Some("to_sip")` or `Some("to_browser")` selects per-direction
+    /// denoiser. `None` skips denoising (backward compat).
+    pub fn transcode(
+        &mut self,
+        data: &[u8],
+        from_pt: u8,
+        to_pt: u8,
+        direction: Option<&str>,
+    ) -> Result<Vec<u8>, String> {
+        if from_pt == to_pt {
+            return Ok(data.to_vec());
+        }
+
+        let (pcm, rate) = self.decode_to_pcm(data, from_pt)?;
+
+        let processed = if let Some(dir) = direction {
+            let pcm_48k = self.resample(&pcm, rate, 48000)?;
+            let denoiser = match dir {
+                "to_sip" => &mut self.denoiser_to_sip,
+                _ => &mut self.denoiser_to_browser,
+            };
+            let denoised = Self::denoise(denoiser, &pcm_48k);
+            let target_rate = codec_sample_rate(to_pt);
+            self.resample(&denoised, 48000, target_rate)?
+        } else {
+            let target_rate = codec_sample_rate(to_pt);
+            if rate == target_rate {
+                pcm
+            } else {
+                self.resample(&pcm, rate, target_rate)?
+            }
+        };
+
+        self.encode_from_pcm(&processed, to_pt)
+    }
+
+    /// Decode an encoded audio payload to raw 16-bit PCM samples.
+    /// Returns (samples, sample_rate).
+    pub fn decode_to_pcm(&mut self, data: &[u8], pt: u8) -> Result<(Vec<i16>, u32), String> {
+        match pt {
+            PT_OPUS => {
+                let mut pcm = vec![0i16; 5760]; // up to 120ms at 48kHz
+                let packet =
+                    OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
+                let out =
+                    MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus signals: {e}"))?;
+                let n: usize = self
+                    .opus_dec
+                    .decode(Some(packet), out, false)
+                    .map_err(|e| format!("opus decode: {e}"))?
+                    .into();
+                pcm.truncate(n);
+                Ok((pcm, 48000))
+            }
+            PT_G722 => {
+                let pcm = self.g722_dec.decode(data);
+                Ok((pcm, 16000))
+            }
+            PT_PCMU => {
+                let pcm: Vec<i16> = data.iter().map(|&b| mulaw_decode(b)).collect();
+                Ok((pcm, 8000))
+            }
+            PT_PCMA => {
+                let pcm: Vec<i16> = data.iter().map(|&b| alaw_decode(b)).collect();
+                Ok((pcm, 8000))
+            }
+            _ => Err(format!("unsupported source PT {pt}")),
+        }
+    }
+
+    /// Encode raw PCM samples to an audio codec.
+    pub fn encode_from_pcm(&mut self, pcm: &[i16], pt: u8) -> Result<Vec<u8>, String> {
+        match pt {
+            PT_OPUS => {
+                let mut buf = vec![0u8; 4000];
+                let n: usize = self
+                    .opus_enc
+                    .encode(pcm, &mut buf)
+                    .map_err(|e| format!("opus encode: {e}"))?
+                    .into();
+                buf.truncate(n);
+                Ok(buf)
+            }
+            PT_G722 => Ok(self.g722_enc.encode(pcm)),
+            PT_PCMU => Ok(pcm.iter().map(|&s| mulaw_encode(s)).collect()),
+            PT_PCMA => Ok(pcm.iter().map(|&s| alaw_encode(s)).collect()),
+            _ => Err(format!("unsupported target PT {pt}")),
+        }
+    }
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    #[test]
+    fn mulaw_roundtrip() {
+        for sample in [-32768i16, -1000, -1, 0, 1, 1000, 32767] {
+            let encoded = mulaw_encode(sample);
+            let decoded = mulaw_decode(encoded);
+            // µ-law is lossy; verify the decoded value is close.
+            assert!((sample as i32 - decoded as i32).abs() < 1000,
+                "µ-law roundtrip failed for {sample}: got {decoded}");
+        }
+    }
+
+    #[test]
+    fn alaw_roundtrip() {
+        for sample in [-32768i16, -1000, -1, 0, 1, 1000, 32767] {
+            let encoded = alaw_encode(sample);
+            let decoded = alaw_decode(encoded);
+            assert!((sample as i32 - decoded as i32).abs() < 1000,
+                "A-law roundtrip failed for {sample}: got {decoded}");
+        }
+    }
+
+    #[test]
+    fn codec_sample_rates() {
+        assert_eq!(codec_sample_rate(PT_OPUS), 48000);
+        assert_eq!(codec_sample_rate(PT_G722), 16000);
+        assert_eq!(codec_sample_rate(PT_PCMU), 8000);
+        assert_eq!(codec_sample_rate(PT_PCMA), 8000);
+    }
+
+    #[test]
+    fn transcode_same_pt_is_passthrough() {
+        let mut st = TranscodeState::new().unwrap();
+        let data = vec![0u8; 160];
+        let result = st.transcode(&data, PT_PCMU, PT_PCMU, None).unwrap();
+        assert_eq!(result, data);
+    }
+
+    #[test]
+    fn pcmu_to_pcma_roundtrip() {
+        let mut st = TranscodeState::new().unwrap();
+        // 160 bytes = 20ms of PCMU at 8kHz
+        let pcmu_data: Vec<u8> = (0..160).map(|i| mulaw_encode((i as i16 * 200) - 16000)).collect();
+        let pcma = st.transcode(&pcmu_data, PT_PCMU, PT_PCMA, None).unwrap();
+        assert_eq!(pcma.len(), 160); // Same frame size
+        let back = st.transcode(&pcma, PT_PCMA, PT_PCMU, None).unwrap();
+        assert_eq!(back.len(), 160);
+    }
+}

rust/crates/opus-codec/Cargo.toml

@@ -8,10 +8,7 @@ name = "opus-codec"
 path = "src/main.rs"
 
 [dependencies]
-audiopus = "0.3.0-rc.0"
-ezk-g722 = "0.1"
-rubato = "0.14"
+codec-lib = { path = "../codec-lib" }
 serde = { version = "1", features = ["derive"] }
 serde_json = "1"
 base64 = "0.22"
-nnnoiseless = { version = "0.5", default-features = false }

rust/crates/opus-codec/src/main.rs

@@ -1,10 +1,6 @@
 /// Audio transcoding bridge for smartrust.
 ///
-/// Handles Opus ↔ G.722 ↔ PCMU transcoding for the SIP router.
-/// Uses audiopus (libopus) for Opus and ezk-g722 (SpanDSP port) for G.722.
-///
-/// Supports per-session codec state so concurrent calls don't corrupt each
-/// other's stateful codecs (Opus, G.722 ADPCM).
+/// Thin CLI wrapper around `codec-lib`. Handles Opus ↔ G.722 ↔ PCMU transcoding.
 ///
 /// Protocol:
 ///   -> {"id":"1","method":"init","params":{}}
@@ -16,24 +12,13 @@
 ///   -> {"id":"4","method":"destroy_session","params":{"session_id":"call-abc"}}
 ///   <- {"id":"4","success":true,"result":{}}
 
-use audiopus::coder::{Decoder as OpusDecoder, Encoder as OpusEncoder};
-use audiopus::packet::Packet as OpusPacket;
-use audiopus::{Application, Bitrate as OpusBitrate, Channels, MutSignals, SampleRate};
 use base64::engine::general_purpose::STANDARD as B64;
 use base64::Engine as _;
-use ezk_g722::libg722::{self, Bitrate};
-use nnnoiseless::DenoiseState;
-use rubato::{FftFixedIn, Resampler};
+use codec_lib::{codec_sample_rate, TranscodeState};
 use serde::Deserialize;
 use std::collections::HashMap;
 use std::io::{self, BufRead, Write};
 
-// Payload type constants.
-const PT_PCMU: u8 = 0;
-const PT_PCMA: u8 = 8;
-const PT_G722: u8 = 9;
-const PT_OPUS: u8 = 111;
-
 #[derive(Deserialize)]
 struct Request {
     id: String,
@@ -42,261 +27,24 @@ struct Request {
     params: serde_json::Value,
 }
 
-fn respond(out: &mut impl Write, id: &str, success: bool, result: Option<serde_json::Value>, error: Option<&str>) {
+fn respond(
+    out: &mut impl Write,
+    id: &str,
+    success: bool,
+    result: Option<serde_json::Value>,
+    error: Option<&str>,
+) {
     let mut resp = serde_json::json!({ "id": id, "success": success });
-    if let Some(r) = result { resp["result"] = r; }
-    if let Some(e) = error { resp["error"] = serde_json::Value::String(e.to_string()); }
+    if let Some(r) = result {
+        resp["result"] = r;
+    }
+    if let Some(e) = error {
+        resp["error"] = serde_json::Value::String(e.to_string());
+    }
     let _ = writeln!(out, "{}", resp);
     let _ = out.flush();
 }
 
-// ---------------------------------------------------------------------------
-// Codec state
-// ---------------------------------------------------------------------------
-
-struct TranscodeState {
-    opus_enc: OpusEncoder,
-    opus_dec: OpusDecoder,
-    g722_enc: libg722::encoder::Encoder,
-    g722_dec: libg722::decoder::Decoder,
-    // Cached FFT resamplers keyed by (from_rate, to_rate, chunk_size).
-    resamplers: HashMap<(u32, u32, usize), FftFixedIn<f64>>,
-    // Per-direction ML noise suppression (RNNoise). Separate state per direction
-    // prevents the RNN hidden state from being corrupted by interleaved audio streams.
-    denoiser_to_sip: Box<DenoiseState<'static>>,
-    denoiser_to_browser: Box<DenoiseState<'static>>,
-}
-
-impl TranscodeState {
-    fn new() -> Result<Self, String> {
-        let mut opus_enc = OpusEncoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
-            .map_err(|e| format!("opus encoder: {e}"))?;
-        // Telephony-grade tuning: complexity 5 is sufficient for voice bridged to G.722.
-        opus_enc.set_complexity(5).map_err(|e| format!("opus set_complexity: {e}"))?;
-        opus_enc.set_bitrate(OpusBitrate::BitsPerSecond(24000)).map_err(|e| format!("opus set_bitrate: {e}"))?;
-        let opus_dec = OpusDecoder::new(SampleRate::Hz48000, Channels::Mono)
-            .map_err(|e| format!("opus decoder: {e}"))?;
-        let g722_enc = libg722::encoder::Encoder::new(Bitrate::Mode1_64000, false, false);
-        let g722_dec = libg722::decoder::Decoder::new(Bitrate::Mode1_64000, false, false);
-
-        Ok(Self {
-            opus_enc, opus_dec, g722_enc, g722_dec,
-            resamplers: HashMap::new(),
-            denoiser_to_sip: DenoiseState::new(),
-            denoiser_to_browser: DenoiseState::new(),
-        })
-    }
-
-    /// High-quality sample rate conversion using rubato FFT resampler.
-    /// Resamplers are cached by (from_rate, to_rate, chunk_size) and reused,
-    /// maintaining proper inter-frame state for continuous audio streams.
-    fn resample(&mut self, pcm: &[i16], from_rate: u32, to_rate: u32) -> Result<Vec<i16>, String> {
-        if from_rate == to_rate || pcm.is_empty() {
-            return Ok(pcm.to_vec());
-        }
-
-        let chunk = pcm.len();
-        let key = (from_rate, to_rate, chunk);
-
-        // Get or create cached resampler for this rate pair + chunk size.
-        if !self.resamplers.contains_key(&key) {
-            let r = FftFixedIn::<f64>::new(from_rate as usize, to_rate as usize, chunk, 1, 1)
-                .map_err(|e| format!("resampler {from_rate}->{to_rate}: {e}"))?;
-            self.resamplers.insert(key, r);
-        }
-        let resampler = self.resamplers.get_mut(&key).unwrap();
-
-        // i16 → f64 normalized to [-1.0, 1.0]
-        let float_in: Vec<f64> = pcm.iter().map(|&s| s as f64 / 32768.0).collect();
-        let input = vec![float_in];
-
-        let result = resampler.process(&input, None)
-            .map_err(|e| format!("resample {from_rate}->{to_rate}: {e}"))?;
-
-        // f64 → i16
-        Ok(result[0].iter()
-            .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
-            .collect())
-    }
-
-    /// Apply RNNoise ML noise suppression to 48kHz PCM audio.
-    /// Processes in 480-sample (10ms) frames. State persists across calls.
-    fn denoise(denoiser: &mut DenoiseState, pcm: &[i16]) -> Vec<i16> {
-        let frame_size = DenoiseState::FRAME_SIZE; // 480
-        let total = pcm.len();
-        // Round down to whole frames — don't process partial frames to avoid
-        // injecting artificial silence into the RNN state.
-        let whole = (total / frame_size) * frame_size;
-        let mut output = Vec::with_capacity(total);
-        let mut out_buf = [0.0f32; 480];
-
-        for offset in (0..whole).step_by(frame_size) {
-            let input: Vec<f32> = pcm[offset..offset + frame_size]
-                .iter().map(|&s| s as f32).collect();
-            denoiser.process_frame(&mut out_buf, &input);
-            output.extend(out_buf.iter()
-                .map(|&s| s.round().clamp(-32768.0, 32767.0) as i16));
-        }
-        // Pass through any trailing partial-frame samples unmodified.
-        if whole < total {
-            output.extend_from_slice(&pcm[whole..]);
-        }
-        output
-    }
-
-    /// Transcode audio payload from one codec to another.
-    /// `direction`: "to_sip" or "to_browser" — selects the per-direction denoiser.
-    /// If None, denoising is skipped (backward compat).
-    fn transcode(&mut self, data: &[u8], from_pt: u8, to_pt: u8, direction: Option<&str>) -> Result<Vec<u8>, String> {
-        if from_pt == to_pt {
-            return Ok(data.to_vec());
-        }
-
-        // Decode to PCM (at source sample rate).
-        let (pcm, rate) = self.decode_to_pcm(data, from_pt)?;
-
-        // Apply noise suppression if direction is specified.
-        let processed = if let Some(dir) = direction {
-            // Resample to 48kHz for denoising (no-op when already 48kHz).
-            let pcm_48k = self.resample(&pcm, rate, 48000)?;
-            let denoiser = match dir {
-                "to_sip" => &mut self.denoiser_to_sip,
-                _ => &mut self.denoiser_to_browser,
-            };
-            let denoised = Self::denoise(denoiser, &pcm_48k);
-            // Resample to target rate (no-op when target is 48kHz).
-            let target_rate = codec_sample_rate(to_pt);
-            self.resample(&denoised, 48000, target_rate)?
-        } else {
-            // No denoising — direct resample.
-            let target_rate = codec_sample_rate(to_pt);
-            if rate == target_rate { pcm } else { self.resample(&pcm, rate, target_rate)? }
-        };
-
-        // Encode from PCM.
-        self.encode_from_pcm(&processed, to_pt)
-    }
-
-    fn decode_to_pcm(&mut self, data: &[u8], pt: u8) -> Result<(Vec<i16>, u32), String> {
-        match pt {
-            PT_OPUS => {
-                let mut pcm = vec![0i16; 5760]; // up to 120ms at 48kHz (RFC 6716 max)
-                let packet = OpusPacket::try_from(data)
-                    .map_err(|e| format!("opus packet: {e}"))?;
-                let out = MutSignals::try_from(&mut pcm[..])
-                    .map_err(|e| format!("opus signals: {e}"))?;
-                let n: usize = self.opus_dec.decode(Some(packet), out, false)
-                    .map_err(|e| format!("opus decode: {e}"))?.into();
-                pcm.truncate(n);
-                Ok((pcm, 48000))
-            }
-            PT_G722 => {
-                let pcm = self.g722_dec.decode(data);
-                Ok((pcm, 16000))
-            }
-            PT_PCMU => {
-                let pcm: Vec<i16> = data.iter().map(|&b| mulaw_decode(b)).collect();
-                Ok((pcm, 8000))
-            }
-            PT_PCMA => {
-                let pcm: Vec<i16> = data.iter().map(|&b| alaw_decode(b)).collect();
-                Ok((pcm, 8000))
-            }
-            _ => Err(format!("unsupported source PT {pt}")),
-        }
-    }
-
-    fn encode_from_pcm(&mut self, pcm: &[i16], pt: u8) -> Result<Vec<u8>, String> {
-        match pt {
-            PT_OPUS => {
-                let mut buf = vec![0u8; 4000];
-                let n: usize = self.opus_enc.encode(pcm, &mut buf)
-                    .map_err(|e| format!("opus encode: {e}"))?.into();
-                buf.truncate(n);
-                Ok(buf)
-            }
-            PT_G722 => {
-                Ok(self.g722_enc.encode(pcm))
-            }
-            PT_PCMU => {
-                Ok(pcm.iter().map(|&s| mulaw_encode(s)).collect())
-            }
-            PT_PCMA => {
-                Ok(pcm.iter().map(|&s| alaw_encode(s)).collect())
-            }
-            _ => Err(format!("unsupported target PT {pt}")),
-        }
-    }
-}
-
-fn codec_sample_rate(pt: u8) -> u32 {
-    match pt {
-        PT_OPUS => 48000,
-        PT_G722 => 16000,
-        _ => 8000, // PCMU, PCMA
-    }
-}
-
-// ---------------------------------------------------------------------------
-// G.711 µ-law (PCMU)
-// ---------------------------------------------------------------------------
-
-fn mulaw_encode(sample: i16) -> u8 {
-    const BIAS: i16 = 0x84;
-    const CLIP: i16 = 32635;
-    let sign = if sample < 0 { 0x80u8 } else { 0 };
-    // Use i32 to avoid overflow when sample == i16::MIN (-32768).
-    let mut s = (sample as i32).unsigned_abs().min(CLIP as u32) as i16;
-    s += BIAS;
-    let mut exp = 7u8;
-    let mut mask = 0x4000i16;
-    while exp > 0 && (s & mask) == 0 { exp -= 1; mask >>= 1; }
-    let mantissa = ((s >> (exp + 3)) & 0x0f) as u8;
-    !(sign | (exp << 4) | mantissa)
-}
-
-fn mulaw_decode(mulaw: u8) -> i16 {
-    let v = !mulaw;
-    let sign = v & 0x80;
-    let exp = (v >> 4) & 0x07;
-    let mantissa = v & 0x0f;
-    let mut sample = (((mantissa as i16) << 4) + 0x84) << exp;
-    sample -= 0x84;
-    if sign != 0 { -sample } else { sample }
-}
-
-// ---------------------------------------------------------------------------
-// G.711 A-law (PCMA)
-// ---------------------------------------------------------------------------
-
-fn alaw_encode(sample: i16) -> u8 {
-    let sign = if sample >= 0 { 0x80u8 } else { 0 };
-    // Use i32 to avoid overflow when sample == i16::MIN (-32768).
-    let s = (sample as i32).unsigned_abs().min(32767) as i16;
-    let mut exp = 7u8;
-    let mut mask = 0x4000i16;
-    while exp > 0 && (s & mask) == 0 { exp -= 1; mask >>= 1; }
-    let mantissa = if exp > 0 { ((s >> (exp + 3)) & 0x0f) as u8 } else { ((s >> 4) & 0x0f) as u8 };
-    (sign | (exp << 4) | mantissa) ^ 0x55
-}
-
-fn alaw_decode(alaw: u8) -> i16 {
-    let v = alaw ^ 0x55;
-    let sign = v & 0x80;
-    let exp = (v >> 4) & 0x07;
-    let mantissa = v & 0x0f;
-    let sample = if exp == 0 {
-        ((mantissa as i16) << 4) + 8
-    } else {
-        (((mantissa as i16) << 4) + 0x108) << (exp - 1)
-    };
-    if sign != 0 { sample } else { -sample }
-}
-
-// ---------------------------------------------------------------------------
-// Main loop
-// ---------------------------------------------------------------------------
-
 /// Resolve a session: if session_id is provided, look it up in the sessions map;
 /// otherwise fall back to the default state (backward compat with `init`).
 fn get_session<'a>(
@@ -319,9 +67,7 @@ fn main() {
     let _ = writeln!(out, r#"{{"event":"ready","data":{{}}}}"#);
     let _ = out.flush();
 
-    // Default state for backward-compat `init` (no session_id).
     let mut default_state: Option<TranscodeState> = None;
-    // Per-session codec state for concurrent call isolation.
     let mut sessions: HashMap<String, TranscodeState> = HashMap::new();
 
     for line in stdin.lock().lines() {
@@ -340,22 +86,21 @@ fn main() {
         };
 
         match req.method.as_str() {
-            // Backward-compat: init the default (shared) session.
-            "init" => {
-                match TranscodeState::new() {
-                    Ok(s) => {
-                        default_state = Some(s);
-                        respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
-                    }
-                    Err(e) => respond(&mut out, &req.id, false, None, Some(&e)),
+            "init" => match TranscodeState::new() {
+                Ok(s) => {
+                    default_state = Some(s);
+                    respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
                 }
-            }
+                Err(e) => respond(&mut out, &req.id, false, None, Some(&e)),
+            },
 
-            // Create an isolated session with its own codec state.
             "create_session" => {
                 let session_id = match req.params.get("session_id").and_then(|v| v.as_str()) {
                     Some(s) => s.to_string(),
-                    None => { respond(&mut out, &req.id, false, None, Some("missing session_id")); continue; }
+                    None => {
+                        respond(&mut out, &req.id, false, None, Some("missing session_id"));
+                        continue;
+                    }
                 };
                 if sessions.contains_key(&session_id) {
                     respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
@@ -370,95 +115,172 @@ fn main() {
                 }
             }
 
-            // Destroy a session, freeing its codec state.
             "destroy_session" => {
                 let session_id = match req.params.get("session_id").and_then(|v| v.as_str()) {
                     Some(s) => s,
-                    None => { respond(&mut out, &req.id, false, None, Some("missing session_id")); continue; }
+                    None => {
+                        respond(&mut out, &req.id, false, None, Some("missing session_id"));
+                        continue;
+                    }
                 };
                 sessions.remove(session_id);
                 respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
             }
 
-            // Transcode: uses session_id if provided, else default state.
             "transcode" => {
                 let st = match get_session(&mut sessions, &mut default_state, &req.params) {
                     Some(s) => s,
-                    None => { respond(&mut out, &req.id, false, None, Some("not initialized (no session or default state)")); continue; }
+                    None => {
+                        respond(
+                            &mut out,
+                            &req.id,
+                            false,
+                            None,
+                            Some("not initialized (no session or default state)"),
+                        );
+                        continue;
+                    }
                 };
                 let data_b64 = match req.params.get("data_b64").and_then(|v| v.as_str()) {
                     Some(s) => s,
-                    None => { respond(&mut out, &req.id, false, None, Some("missing data_b64")); continue; }
+                    None => {
+                        respond(&mut out, &req.id, false, None, Some("missing data_b64"));
+                        continue;
+                    }
                 };
-                let from_pt = req.params.get("from_pt").and_then(|v| v.as_u64()).unwrap_or(0) as u8;
+                let from_pt =
+                    req.params.get("from_pt").and_then(|v| v.as_u64()).unwrap_or(0) as u8;
                 let to_pt = req.params.get("to_pt").and_then(|v| v.as_u64()).unwrap_or(0) as u8;
                 let direction = req.params.get("direction").and_then(|v| v.as_str());
 
                 let data = match B64.decode(data_b64) {
                     Ok(b) => b,
-                    Err(e) => { respond(&mut out, &req.id, false, None, Some(&format!("b64: {e}"))); continue; }
+                    Err(e) => {
+                        respond(
+                            &mut out,
+                            &req.id,
+                            false,
+                            None,
+                            Some(&format!("b64: {e}")),
+                        );
+                        continue;
+                    }
                 };
 
                 match st.transcode(&data, from_pt, to_pt, direction) {
                     Ok(result) => {
-                        respond(&mut out, &req.id, true, Some(serde_json::json!({ "data_b64": B64.encode(&result) })), None);
+                        respond(
+                            &mut out,
+                            &req.id,
+                            true,
+                            Some(serde_json::json!({ "data_b64": B64.encode(&result) })),
+                            None,
+                        );
                     }
                     Err(e) => respond(&mut out, &req.id, false, None, Some(&e)),
                 }
             }
 
-            // Encode raw 16-bit PCM to a target codec.
-            // Params: data_b64 (raw PCM bytes, 16-bit LE), sample_rate (input Hz), to_pt
-            // Optional: session_id for isolated codec state.
             "encode_pcm" => {
                 let st = match get_session(&mut sessions, &mut default_state, &req.params) {
                     Some(s) => s,
-                    None => { respond(&mut out, &req.id, false, None, Some("not initialized (no session or default state)")); continue; }
+                    None => {
+                        respond(
+                            &mut out,
+                            &req.id,
+                            false,
+                            None,
+                            Some("not initialized (no session or default state)"),
+                        );
+                        continue;
+                    }
                 };
                 let data_b64 = match req.params.get("data_b64").and_then(|v| v.as_str()) {
                     Some(s) => s,
-                    None => { respond(&mut out, &req.id, false, None, Some("missing data_b64")); continue; }
+                    None => {
+                        respond(&mut out, &req.id, false, None, Some("missing data_b64"));
+                        continue;
+                    }
                 };
-                let sample_rate = req.params.get("sample_rate").and_then(|v| v.as_u64()).unwrap_or(22050) as u32;
+                let sample_rate = req
+                    .params
+                    .get("sample_rate")
+                    .and_then(|v| v.as_u64())
+                    .unwrap_or(22050) as u32;
                 let to_pt = req.params.get("to_pt").and_then(|v| v.as_u64()).unwrap_or(9) as u8;
 
                 let data = match B64.decode(data_b64) {
                     Ok(b) => b,
-                    Err(e) => { respond(&mut out, &req.id, false, None, Some(&format!("b64: {e}"))); continue; }
+                    Err(e) => {
+                        respond(
+                            &mut out,
+                            &req.id,
+                            false,
+                            None,
+                            Some(&format!("b64: {e}")),
+                        );
+                        continue;
+                    }
                 };
 
                 if data.len() % 2 != 0 {
-                    respond(&mut out, &req.id, false, None, Some("PCM data has odd byte count (expected 16-bit LE samples)"));
+                    respond(
+                        &mut out,
+                        &req.id,
+                        false,
+                        None,
+                        Some("PCM data has odd byte count (expected 16-bit LE samples)"),
+                    );
                     continue;
                 }
 
-                // Convert raw bytes to i16 samples.
-                let pcm: Vec<i16> = data.chunks_exact(2)
+                let pcm: Vec<i16> = data
+                    .chunks_exact(2)
                     .map(|c| i16::from_le_bytes([c[0], c[1]]))
                     .collect();
 
-                // Resample to target codec's sample rate.
                 let target_rate = codec_sample_rate(to_pt);
                 let resampled = match st.resample(&pcm, sample_rate, target_rate) {
                     Ok(r) => r,
-                    Err(e) => { respond(&mut out, &req.id, false, None, Some(&e)); continue; }
+                    Err(e) => {
+                        respond(&mut out, &req.id, false, None, Some(&e));
+                        continue;
+                    }
                 };
 
-                // Encode to target codec (reuse encode_from_pcm).
                 match st.encode_from_pcm(&resampled, to_pt) {
                     Ok(encoded) => {
-                        respond(&mut out, &req.id, true, Some(serde_json::json!({ "data_b64": B64.encode(&encoded) })), None);
+                        respond(
+                            &mut out,
+                            &req.id,
+                            true,
+                            Some(serde_json::json!({ "data_b64": B64.encode(&encoded) })),
+                            None,
+                        );
+                    }
+                    Err(e) => {
+                        respond(&mut out, &req.id, false, None, Some(&e));
                     }
-                    Err(e) => { respond(&mut out, &req.id, false, None, Some(&e)); continue; }
                 }
             }
 
-            // Legacy commands (kept for backward compat).
             "encode" | "decode" => {
-                respond(&mut out, &req.id, false, None, Some("use 'transcode' command instead"));
+                respond(
+                    &mut out,
+                    &req.id,
+                    false,
+                    None,
+                    Some("use 'transcode' command instead"),
+                );
             }
 
-            _ => respond(&mut out, &req.id, false, None, Some(&format!("unknown: {}", req.method))),
+            _ => respond(
+                &mut out,
+                &req.id,
+                false,
+                None,
+                Some(&format!("unknown: {}", req.method)),
+            ),
         }
     }
 }

rust/crates/proxy-engine/Cargo.toml

@@ -0,0 +1,20 @@
+[package]
+name = "proxy-engine"
+version = "0.1.0"
+edition = "2021"
+
+[[bin]]
+name = "proxy-engine"
+path = "src/main.rs"
+
+[dependencies]
+codec-lib = { path = "../codec-lib" }
+sip-proto = { path = "../sip-proto" }
+tokio = { version = "1", features = ["full"] }
+serde = { version = "1", features = ["derive"] }
+serde_json = "1"
+base64 = "0.22"
+regex-lite = "0.1"
+webrtc = "0.8"
+rand = "0.8"
+hound = "3.5"

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

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

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

@@ -0,0 +1,103 @@
+//! Call hub — owns legs and bridges media.
+//!
+//! Each Call has a unique ID and tracks its state, direction, and associated
+//! SIP Call-IDs for message routing.
+
+use std::net::SocketAddr;
+use std::sync::Arc;
+use std::time::Instant;
+use tokio::net::UdpSocket;
+
+/// Call state machine.
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub enum CallState {
+    SettingUp,
+    Ringing,
+    Connected,
+    Voicemail,
+    Ivr,
+    Terminating,
+    Terminated,
+}
+
+impl CallState {
+    pub fn as_str(&self) -> &'static str {
+        match self {
+            Self::SettingUp => "setting-up",
+            Self::Ringing => "ringing",
+            Self::Connected => "connected",
+            Self::Voicemail => "voicemail",
+            Self::Ivr => "ivr",
+            Self::Terminating => "terminating",
+            Self::Terminated => "terminated",
+        }
+    }
+}
+
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub enum CallDirection {
+    Inbound,
+    Outbound,
+}
+
+impl CallDirection {
+    pub fn as_str(&self) -> &'static str {
+        match self {
+            Self::Inbound => "inbound",
+            Self::Outbound => "outbound",
+        }
+    }
+}
+
+/// A passthrough call — both sides share the same SIP Call-ID.
+/// The proxy rewrites SDP/Contact/Request-URI and relays RTP.
+pub struct PassthroughCall {
+    pub id: String,
+    pub sip_call_id: String,
+    pub state: CallState,
+    pub direction: CallDirection,
+    pub created_at: Instant,
+
+    // Call 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>,
+
+    // Device side.
+    pub device_addr: SocketAddr,
+    pub device_media: Option<SocketAddr>,
+
+    // RTP relay.
+    pub rtp_port: u16,
+    pub rtp_socket: Arc<UdpSocket>,
+
+    // Packet counters.
+    pub pkt_from_device: u64,
+    pub pkt_from_provider: u64,
+}
+
+impl PassthroughCall {
+    pub fn duration_secs(&self) -> u64 {
+        self.created_at.elapsed().as_secs()
+    }
+
+    pub fn to_status_json(&self) -> serde_json::Value {
+        serde_json::json!({
+            "id": self.id,
+            "state": self.state.as_str(),
+            "direction": self.direction.as_str(),
+            "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,
+        })
+    }
+}

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

@@ -0,0 +1,822 @@
+//! 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 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;
+
+pub struct CallManager {
+    /// Active passthrough calls, keyed by SIP Call-ID.
+    calls: HashMap<String, PassthroughCall>,
+    /// 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(),
+            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 if this Call-ID belongs to an active call.
+        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)
+    }
+
+    // --- Dashboard outbound call (B2BUA) ---
+
+    /// Initiate an outbound call from the dashboard.
+    /// Builds an INVITE from scratch and sends it to the provider.
+    /// The browser connects separately via WebRTC and gets linked to this call.
+    pub async fn make_outbound_call(
+        &mut self,
+        number: &str,
+        provider_config: &ProviderConfig,
+        config: &AppConfig,
+        rtp_pool: &mut RtpPortPool,
+        socket: &UdpSocket,
+        public_ip: Option<&str>,
+        registered_aor: &str,
+    ) -> Option<String> {
+        let call_id = self.next_call_id();
+        let lan_ip = &config.proxy.lan_ip;
+        let lan_port = config.proxy.lan_port;
+        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());
+
+        let provider_dest: SocketAddr = match provider_config.outbound_proxy.to_socket_addr() {
+            Some(a) => a,
+            None => return None,
+        };
+
+        // Allocate RTP port for the provider leg.
+        let rtp_alloc = match rtp_pool.allocate().await {
+            Some(a) => a,
+            None => return None,
+        };
+
+        // Build the SIP Call-ID for this new dialog.
+        let sip_call_id = sip_proto::helpers::generate_call_id(None);
+
+        // Build SDP offer.
+        let sdp = sip_proto::helpers::build_sdp(&sip_proto::helpers::SdpOptions {
+            ip: pub_ip,
+            port: rtp_alloc.port,
+            payload_types: &provider_config.codecs,
+            ..Default::default()
+        });
+
+        // Build INVITE.
+        let to_uri = format!("sip:{number}@{}", provider_config.domain);
+        let invite = SipMessage::create_request(
+            "INVITE",
+            &to_uri,
+            sip_proto::message::RequestOptions {
+                via_host: pub_ip.to_string(),
+                via_port: lan_port,
+                via_transport: None,
+                via_branch: Some(sip_proto::helpers::generate_branch()),
+                from_uri: registered_aor.to_string(),
+                from_display_name: None,
+                from_tag: Some(sip_proto::helpers::generate_tag()),
+                to_uri: to_uri.clone(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some(sip_call_id.clone()),
+                cseq: Some(1),
+                contact: Some(format!("<sip:{pub_ip}:{lan_port}>")),
+                max_forwards: Some(70),
+                body: Some(sdp),
+                content_type: Some("application/sdp".to_string()),
+                extra_headers: Some(vec![
+                    ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
+                    ("Allow".to_string(), "INVITE, ACK, OPTIONS, CANCEL, BYE, INFO".to_string()),
+                ]),
+            },
+        );
+
+        // Send INVITE to provider.
+        let _ = socket.send_to(&invite.serialize(), provider_dest).await;
+
+        // Create call entry — device_addr is a dummy (WebRTC will be linked later).
+        let dummy_addr: SocketAddr = "127.0.0.1:0".parse().unwrap();
+        let call = PassthroughCall {
+            id: call_id.clone(),
+            sip_call_id: sip_call_id.clone(),
+            state: CallState::SettingUp,
+            direction: CallDirection::Outbound,
+            created_at: Instant::now(),
+            caller_number: Some(registered_aor.to_string()),
+            callee_number: Some(number.to_string()),
+            provider_id: provider_config.id.clone(),
+            provider_addr: provider_dest,
+            provider_media: None,
+            device_addr: dummy_addr,
+            device_media: None,
+            rtp_port: rtp_alloc.port,
+            rtp_socket: rtp_alloc.socket.clone(),
+            pkt_from_device: 0,
+            pkt_from_provider: 0,
+        };
+        self.calls.insert(sip_call_id, call);
+
+        Some(call_id)
+    }
+
+    // --- Voicemail ---
+
+    /// Route a call to voicemail: answer the INVITE, play greeting, record message.
+    async fn route_to_voicemail(
+        &mut self,
+        call_id: &str,
+        invite: &SipMessage,
+        from_addr: SocketAddr,
+        caller_number: &str,
+        provider_id: &str,
+        provider_config: &ProviderConfig,
+        config: &AppConfig,
+        rtp_pool: &mut RtpPortPool,
+        socket: &UdpSocket,
+        public_ip: Option<&str>,
+    ) -> Option<String> {
+        let lan_ip = &config.proxy.lan_ip;
+        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());
+
+        // Allocate RTP port for the voicemail session.
+        let rtp_alloc = match rtp_pool.allocate().await {
+            Some(a) => a,
+            None => {
+                let resp =
+                    SipMessage::create_response(503, "Service Unavailable", invite, None);
+                let _ = socket.send_to(&resp.serialize(), from_addr).await;
+                return None;
+            }
+        };
+
+        // Determine provider's preferred codec.
+        let codec_pt = provider_config.codecs.first().copied().unwrap_or(9); // default G.722
+
+        // Build SDP with our RTP port.
+        let sdp = sip_proto::helpers::build_sdp(&sip_proto::helpers::SdpOptions {
+            ip: pub_ip,
+            port: rtp_alloc.port,
+            payload_types: &provider_config.codecs,
+            ..Default::default()
+        });
+
+        // Answer the INVITE with 200 OK.
+        let response = SipMessage::create_response(
+            200,
+            "OK",
+            invite,
+            Some(sip_proto::message::ResponseOptions {
+                to_tag: Some(sip_proto::helpers::generate_tag()),
+                contact: Some(format!("<sip:{}:{}>", lan_ip, config.proxy.lan_port)),
+                body: Some(sdp),
+                content_type: Some("application/sdp".to_string()),
+                ..Default::default()
+            }),
+        );
+        let _ = socket.send_to(&response.serialize(), from_addr).await;
+
+        // Extract provider media from original SDP.
+        let provider_media = if invite.has_sdp_body() {
+            sip_proto::helpers::parse_sdp_endpoint(&invite.body)
+                .and_then(|ep| format!("{}:{}", ep.address, ep.port).parse().ok())
+        } else {
+            Some(from_addr) // fallback to signaling address
+        };
+        let provider_media = provider_media.unwrap_or(from_addr);
+
+        // Create a voicemail call entry for BYE routing.
+        let call = PassthroughCall {
+            id: call_id.to_string(),
+            sip_call_id: invite.call_id().to_string(),
+            state: CallState::Voicemail,
+            direction: CallDirection::Inbound,
+            created_at: std::time::Instant::now(),
+            caller_number: Some(caller_number.to_string()),
+            callee_number: None,
+            provider_id: provider_id.to_string(),
+            provider_addr: from_addr,
+            provider_media: Some(provider_media),
+            device_addr: from_addr, // no device — just use provider addr as placeholder
+            device_media: None,
+            rtp_port: rtp_alloc.port,
+            rtp_socket: rtp_alloc.socket.clone(),
+            pkt_from_device: 0,
+            pkt_from_provider: 0,
+        };
+        self.calls.insert(invite.call_id().to_string(), call);
+
+        // Build recording file path.
+        let timestamp = std::time::SystemTime::now()
+            .duration_since(std::time::UNIX_EPOCH)
+            .unwrap_or_default()
+            .as_millis();
+        let recording_dir = format!(".nogit/voicemail/default");
+        let recording_path = format!("{recording_dir}/msg-{timestamp}.wav");
+
+        // Look for a greeting WAV file.
+        let greeting_wav = find_greeting_wav();
+
+        // Spawn the voicemail session.
+        let out_tx = self.out_tx.clone();
+        let call_id_owned = call_id.to_string();
+        let caller_owned = caller_number.to_string();
+        let rtp_socket = rtp_alloc.socket;
+        tokio::spawn(async move {
+            crate::voicemail::run_voicemail_session(
+                rtp_socket,
+                provider_media,
+                codec_pt,
+                greeting_wav,
+                recording_path,
+                120_000, // max 120 seconds
+                call_id_owned,
+                caller_owned,
+                out_tx,
+            )
+            .await;
+        });
+
+        Some(call_id.to_string())
+    }
+
+    // --- Internal helpers ---
+
+    fn resolve_first_device(&self, config: &AppConfig, registrar: &Registrar) -> Option<SocketAddr> {
+        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;
+            }
+        }
+    }
+}

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

@@ -0,0 +1,325 @@
+//! Configuration types received from the TypeScript control plane.
+//!
+//! TypeScript loads config from `.nogit/config.json` and sends it to the
+//! proxy engine via the `configure` command. These types mirror the TS interfaces.
+
+use serde::Deserialize;
+use std::net::SocketAddr;
+
+/// Network endpoint.
+#[derive(Debug, Clone, Deserialize)]
+pub struct Endpoint {
+    pub address: String,
+    pub port: u16,
+}
+
+impl Endpoint {
+    /// Resolve to a SocketAddr. Handles both IP addresses and hostnames.
+    pub fn to_socket_addr(&self) -> Option<SocketAddr> {
+        // Try direct parse first (IP address).
+        if let Ok(addr) = format!("{}:{}", self.address, self.port).parse() {
+            return Some(addr);
+        }
+        // DNS resolution for hostnames.
+        use std::net::ToSocketAddrs;
+        format!("{}:{}", self.address, self.port)
+            .to_socket_addrs()
+            .ok()
+            .and_then(|mut addrs| addrs.next())
+    }
+}
+
+/// Provider quirks for codec/protocol workarounds.
+#[derive(Debug, Clone, Deserialize)]
+pub struct Quirks {
+    #[serde(rename = "earlyMediaSilence")]
+    pub early_media_silence: bool,
+    #[serde(rename = "silencePayloadType")]
+    pub silence_payload_type: Option<u8>,
+    #[serde(rename = "silenceMaxPackets")]
+    pub silence_max_packets: Option<u32>,
+}
+
+/// A SIP trunk provider configuration.
+#[derive(Debug, Clone, Deserialize)]
+pub struct ProviderConfig {
+    pub id: String,
+    #[serde(rename = "displayName")]
+    pub display_name: String,
+    pub domain: String,
+    #[serde(rename = "outboundProxy")]
+    pub outbound_proxy: Endpoint,
+    pub username: String,
+    pub password: String,
+    #[serde(rename = "registerIntervalSec")]
+    pub register_interval_sec: u32,
+    pub codecs: Vec<u8>,
+    pub quirks: Quirks,
+}
+
+/// A SIP device (phone) configuration.
+#[derive(Debug, Clone, Deserialize)]
+pub struct DeviceConfig {
+    pub id: String,
+    #[serde(rename = "displayName")]
+    pub display_name: String,
+    #[serde(rename = "expectedAddress")]
+    pub expected_address: String,
+    pub extension: String,
+}
+
+/// Route match criteria.
+#[derive(Debug, Clone, Deserialize)]
+pub struct RouteMatch {
+    pub direction: String, // "inbound" | "outbound"
+    #[serde(rename = "numberPattern")]
+    pub number_pattern: Option<String>,
+    #[serde(rename = "callerPattern")]
+    pub caller_pattern: Option<String>,
+    #[serde(rename = "sourceProvider")]
+    pub source_provider: Option<String>,
+    #[serde(rename = "sourceDevice")]
+    pub source_device: Option<String>,
+}
+
+/// Route action.
+#[derive(Debug, Clone, Deserialize)]
+pub struct RouteAction {
+    pub targets: Option<Vec<String>>,
+    #[serde(rename = "ringBrowsers")]
+    pub ring_browsers: Option<bool>,
+    #[serde(rename = "voicemailBox")]
+    pub voicemail_box: Option<String>,
+    #[serde(rename = "ivrMenuId")]
+    pub ivr_menu_id: Option<String>,
+    #[serde(rename = "noAnswerTimeout")]
+    pub no_answer_timeout: Option<u32>,
+    pub provider: Option<String>,
+    #[serde(rename = "failoverProviders")]
+    pub failover_providers: Option<Vec<String>>,
+    #[serde(rename = "stripPrefix")]
+    pub strip_prefix: Option<String>,
+    #[serde(rename = "prependPrefix")]
+    pub prepend_prefix: Option<String>,
+}
+
+/// A routing rule.
+#[derive(Debug, Clone, Deserialize)]
+pub struct Route {
+    pub id: String,
+    pub name: String,
+    pub priority: i32,
+    pub enabled: bool,
+    #[serde(rename = "match")]
+    pub match_criteria: RouteMatch,
+    pub action: RouteAction,
+}
+
+/// Proxy network settings.
+#[derive(Debug, Clone, Deserialize)]
+pub struct ProxyConfig {
+    #[serde(rename = "lanIp")]
+    pub lan_ip: String,
+    #[serde(rename = "lanPort")]
+    pub lan_port: u16,
+    #[serde(rename = "publicIpSeed")]
+    pub public_ip_seed: Option<String>,
+    #[serde(rename = "rtpPortRange")]
+    pub rtp_port_range: RtpPortRange,
+}
+
+#[derive(Debug, Clone, Deserialize)]
+pub struct RtpPortRange {
+    pub min: u16,
+    pub max: u16,
+}
+
+/// Full application config pushed from TypeScript.
+#[derive(Debug, Clone, Deserialize)]
+pub struct AppConfig {
+    pub proxy: ProxyConfig,
+    pub providers: Vec<ProviderConfig>,
+    pub devices: Vec<DeviceConfig>,
+    pub routing: RoutingConfig,
+}
+
+#[derive(Debug, Clone, Deserialize)]
+pub struct RoutingConfig {
+    pub routes: Vec<Route>,
+}
+
+// ---------------------------------------------------------------------------
+// Pattern matching (ported from ts/config.ts)
+// ---------------------------------------------------------------------------
+
+/// Test a value against a pattern string.
+/// - None/empty: matches everything (wildcard)
+/// - Trailing '*': prefix match
+/// - Starts with '/': regex match
+/// - Otherwise: exact match
+pub fn matches_pattern(pattern: Option<&str>, value: &str) -> bool {
+    let pattern = match pattern {
+        None => return true,
+        Some(p) if p.is_empty() => return true,
+        Some(p) => p,
+    };
+
+    // Prefix match: "+49*"
+    if pattern.ends_with('*') {
+        return value.starts_with(&pattern[..pattern.len() - 1]);
+    }
+
+    // Regex match: "/^\\+49/" or "/pattern/i"
+    if pattern.starts_with('/') {
+        if let Some(last_slash) = pattern[1..].rfind('/') {
+            let re_str = &pattern[1..1 + last_slash];
+            let flags = &pattern[2 + last_slash..];
+            let case_insensitive = flags.contains('i');
+            if let Ok(re) = if case_insensitive {
+                regex_lite::Regex::new(&format!("(?i){re_str}"))
+            } else {
+                regex_lite::Regex::new(re_str)
+            } {
+                return re.is_match(value);
+            }
+        }
+    }
+
+    // Exact match.
+    value == pattern
+}
+
+/// Result of resolving an outbound route.
+pub struct OutboundRouteResult {
+    pub provider: ProviderConfig,
+    pub transformed_number: String,
+}
+
+/// Result of resolving an inbound route.
+pub struct InboundRouteResult {
+    pub device_ids: Vec<String>,
+    pub ring_browsers: bool,
+    pub voicemail_box: Option<String>,
+    pub ivr_menu_id: Option<String>,
+    pub no_answer_timeout: Option<u32>,
+}
+
+impl AppConfig {
+    /// Resolve which provider to use for an outbound call.
+    pub fn resolve_outbound_route(
+        &self,
+        dialed_number: &str,
+        source_device_id: Option<&str>,
+        is_provider_registered: &dyn Fn(&str) -> bool,
+    ) -> Option<OutboundRouteResult> {
+        let mut routes: Vec<&Route> = self
+            .routing
+            .routes
+            .iter()
+            .filter(|r| r.enabled && r.match_criteria.direction == "outbound")
+            .collect();
+        routes.sort_by(|a, b| b.priority.cmp(&a.priority));
+
+        for route in &routes {
+            let m = &route.match_criteria;
+
+            if !matches_pattern(m.number_pattern.as_deref(), dialed_number) {
+                continue;
+            }
+            if let Some(sd) = &m.source_device {
+                if source_device_id != Some(sd.as_str()) {
+                    continue;
+                }
+            }
+
+            // Find a registered provider.
+            let mut candidates: Vec<&str> = Vec::new();
+            if let Some(p) = &route.action.provider {
+                candidates.push(p);
+            }
+            if let Some(fps) = &route.action.failover_providers {
+                candidates.extend(fps.iter().map(|s| s.as_str()));
+            }
+
+            for pid in candidates {
+                let provider = match self.providers.iter().find(|p| p.id == pid) {
+                    Some(p) => p,
+                    None => continue,
+                };
+                if !is_provider_registered(pid) {
+                    continue;
+                }
+
+                let mut num = dialed_number.to_string();
+                if let Some(strip) = &route.action.strip_prefix {
+                    if num.starts_with(strip.as_str()) {
+                        num = num[strip.len()..].to_string();
+                    }
+                }
+                if let Some(prepend) = &route.action.prepend_prefix {
+                    num = format!("{prepend}{num}");
+                }
+
+                return Some(OutboundRouteResult {
+                    provider: provider.clone(),
+                    transformed_number: num,
+                });
+            }
+        }
+
+        // Fallback: first provider.
+        self.providers.first().map(|p| OutboundRouteResult {
+            provider: p.clone(),
+            transformed_number: dialed_number.to_string(),
+        })
+    }
+
+    /// Resolve which devices to ring for an inbound call.
+    pub fn resolve_inbound_route(
+        &self,
+        provider_id: &str,
+        called_number: &str,
+        caller_number: &str,
+    ) -> InboundRouteResult {
+        let mut routes: Vec<&Route> = self
+            .routing
+            .routes
+            .iter()
+            .filter(|r| r.enabled && r.match_criteria.direction == "inbound")
+            .collect();
+        routes.sort_by(|a, b| b.priority.cmp(&a.priority));
+
+        for route in &routes {
+            let m = &route.match_criteria;
+
+            if let Some(sp) = &m.source_provider {
+                if sp != provider_id {
+                    continue;
+                }
+            }
+            if !matches_pattern(m.number_pattern.as_deref(), called_number) {
+                continue;
+            }
+            if !matches_pattern(m.caller_pattern.as_deref(), caller_number) {
+                continue;
+            }
+
+            return InboundRouteResult {
+                device_ids: route.action.targets.clone().unwrap_or_default(),
+                ring_browsers: route.action.ring_browsers.unwrap_or(false),
+                voicemail_box: route.action.voicemail_box.clone(),
+                ivr_menu_id: route.action.ivr_menu_id.clone(),
+                no_answer_timeout: route.action.no_answer_timeout,
+            };
+        }
+
+        // Fallback: ring all devices + browsers.
+        InboundRouteResult {
+            device_ids: vec![],
+            ring_browsers: true,
+            voicemail_box: None,
+            ivr_menu_id: None,
+            no_answer_timeout: None,
+        }
+    }
+}

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

@@ -0,0 +1,200 @@
+//! DTMF detection — parses RFC 2833 telephone-event RTP packets.
+//!
+//! Deduplicates repeated packets (same digit sent multiple times with
+//! increasing duration) and fires once per detected digit.
+//!
+//! Ported from ts/call/dtmf-detector.ts.
+
+use crate::ipc::{emit_event, OutTx};
+
+/// RFC 2833 event ID → character mapping.
+const EVENT_CHARS: &[char] = &[
+    '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '*', '#', 'A', 'B', 'C', 'D',
+];
+
+/// Safety timeout: report digit if no End packet arrives within this many ms.
+const SAFETY_TIMEOUT_MS: u64 = 200;
+
+/// DTMF detector for a single RTP stream.
+pub struct DtmfDetector {
+    /// Negotiated telephone-event payload type (default 101).
+    telephone_event_pt: u8,
+    /// Clock rate for duration calculation (default 8000 Hz).
+    clock_rate: u32,
+    /// Call ID for event emission.
+    call_id: String,
+
+    // Deduplication state.
+    current_event_id: Option<u8>,
+    current_event_ts: Option<u32>,
+    current_event_reported: bool,
+    current_event_duration: u16,
+
+    out_tx: OutTx,
+}
+
+impl DtmfDetector {
+    pub fn new(call_id: String, out_tx: OutTx) -> Self {
+        Self {
+            telephone_event_pt: 101,
+            clock_rate: 8000,
+            call_id,
+            current_event_id: None,
+            current_event_ts: None,
+            current_event_reported: false,
+            current_event_duration: 0,
+            out_tx,
+        }
+    }
+
+    /// Feed an RTP packet. Checks PT; ignores non-DTMF packets.
+    /// Returns Some(digit_char) if a digit was detected.
+    pub fn process_rtp(&mut self, data: &[u8]) -> Option<char> {
+        if data.len() < 16 {
+            return None; // 12-byte header + 4-byte telephone-event minimum
+        }
+
+        let pt = data[1] & 0x7F;
+        if pt != self.telephone_event_pt {
+            return None;
+        }
+
+        let marker = (data[1] & 0x80) != 0;
+        let rtp_timestamp = u32::from_be_bytes([data[4], data[5], data[6], data[7]]);
+
+        // Parse telephone-event payload.
+        let event_id = data[12];
+        let end_bit = (data[13] & 0x80) != 0;
+        let duration = u16::from_be_bytes([data[14], data[15]]);
+
+        if event_id as usize >= EVENT_CHARS.len() {
+            return None;
+        }
+
+        // Detect new event.
+        let is_new = marker
+            || self.current_event_id != Some(event_id)
+            || self.current_event_ts != Some(rtp_timestamp);
+
+        if is_new {
+            // Report pending unreported event.
+            let pending = self.report_pending();
+
+            self.current_event_id = Some(event_id);
+            self.current_event_ts = Some(rtp_timestamp);
+            self.current_event_reported = false;
+            self.current_event_duration = duration;
+
+            if pending.is_some() {
+                return pending;
+            }
+        }
+
+        if duration > self.current_event_duration {
+            self.current_event_duration = duration;
+        }
+
+        // Report on End bit (first time only).
+        if end_bit && !self.current_event_reported {
+            self.current_event_reported = true;
+            let digit = EVENT_CHARS[event_id as usize];
+            let duration_ms = (self.current_event_duration as f64 / self.clock_rate as f64) * 1000.0;
+
+            emit_event(
+                &self.out_tx,
+                "dtmf_digit",
+                serde_json::json!({
+                    "call_id": self.call_id,
+                    "digit": digit.to_string(),
+                    "duration_ms": duration_ms.round() as u32,
+                    "source": "rfc2833",
+                }),
+            );
+
+            return Some(digit);
+        }
+
+        None
+    }
+
+    /// Report a pending unreported event.
+    fn report_pending(&mut self) -> Option<char> {
+        if let Some(event_id) = self.current_event_id {
+            if !self.current_event_reported && (event_id as usize) < EVENT_CHARS.len() {
+                self.current_event_reported = true;
+                let digit = EVENT_CHARS[event_id as usize];
+                let duration_ms =
+                    (self.current_event_duration as f64 / self.clock_rate as f64) * 1000.0;
+
+                emit_event(
+                    &self.out_tx,
+                    "dtmf_digit",
+                    serde_json::json!({
+                        "call_id": self.call_id,
+                        "digit": digit.to_string(),
+                        "duration_ms": duration_ms.round() as u32,
+                        "source": "rfc2833",
+                    }),
+                );
+
+                return Some(digit);
+            }
+        }
+        None
+    }
+
+    /// Process a SIP INFO message body for DTMF.
+    pub fn process_sip_info(&mut self, content_type: &str, body: &str) -> Option<char> {
+        let ct = content_type.to_ascii_lowercase();
+
+        if ct.contains("application/dtmf-relay") {
+            // Format: "Signal= 5\r\nDuration= 160\r\n"
+            let signal = body
+                .lines()
+                .find(|l| l.to_ascii_lowercase().starts_with("signal"))
+                .and_then(|l| l.split('=').nth(1))
+                .map(|s| s.trim().to_string())?;
+
+            if signal.len() != 1 {
+                return None;
+            }
+            let digit = signal.chars().next()?.to_ascii_uppercase();
+            if !"0123456789*#ABCD".contains(digit) {
+                return None;
+            }
+
+            emit_event(
+                &self.out_tx,
+                "dtmf_digit",
+                serde_json::json!({
+                    "call_id": self.call_id,
+                    "digit": digit.to_string(),
+                    "source": "sip-info",
+                }),
+            );
+
+            return Some(digit);
+        }
+
+        if ct.contains("application/dtmf") {
+            let digit = body.trim().chars().next()?.to_ascii_uppercase();
+            if !"0123456789*#ABCD".contains(digit) {
+                return None;
+            }
+
+            emit_event(
+                &self.out_tx,
+                "dtmf_digit",
+                serde_json::json!({
+                    "call_id": self.call_id,
+                    "digit": digit.to_string(),
+                    "source": "sip-info",
+                }),
+            );
+
+            return Some(digit);
+        }
+
+        None
+    }
+}

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

@@ -0,0 +1,47 @@
+//! IPC protocol — command dispatch and event emission.
+//!
+//! All communication with the TypeScript control plane goes through
+//! JSON-line messages on stdin/stdout (smartrust protocol).
+
+use serde::Deserialize;
+use tokio::sync::mpsc;
+
+/// Sender for serialized stdout output.
+pub type OutTx = mpsc::UnboundedSender<String>;
+
+/// A command received from the TypeScript control plane.
+#[derive(Deserialize)]
+pub struct Command {
+    pub id: String,
+    pub method: String,
+    #[serde(default)]
+    pub params: serde_json::Value,
+}
+
+/// Send a response to a command.
+pub fn respond(tx: &OutTx, id: &str, success: bool, result: Option<serde_json::Value>, error: Option<&str>) {
+    let mut resp = serde_json::json!({ "id": id, "success": success });
+    if let Some(r) = result {
+        resp["result"] = r;
+    }
+    if let Some(e) = error {
+        resp["error"] = serde_json::Value::String(e.to_string());
+    }
+    let _ = tx.send(resp.to_string());
+}
+
+/// Send a success response.
+pub fn respond_ok(tx: &OutTx, id: &str, result: serde_json::Value) {
+    respond(tx, id, true, Some(result), None);
+}
+
+/// Send an error response.
+pub fn respond_err(tx: &OutTx, id: &str, error: &str) {
+    respond(tx, id, false, None, Some(error));
+}
+
+/// Emit an event to the TypeScript control plane.
+pub fn emit_event(tx: &OutTx, event: &str, data: serde_json::Value) {
+    let msg = serde_json::json!({ "event": event, "data": data });
+    let _ = tx.send(msg.to_string());
+}

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

@@ -0,0 +1,626 @@
+/// SIP proxy engine — the Rust data plane for the SIP router.
+///
+/// Handles ALL SIP protocol mechanics. TypeScript only sends high-level
+/// commands (routing decisions, config) and receives high-level events
+/// (incoming calls, registration state).
+///
+/// No raw SIP ever touches TypeScript.
+
+mod audio_player;
+mod call;
+mod call_manager;
+mod config;
+mod dtmf;
+mod ipc;
+mod provider;
+mod recorder;
+mod registrar;
+mod rtp;
+mod sip_transport;
+mod voicemail;
+mod webrtc_engine;
+
+use crate::call_manager::CallManager;
+use crate::config::AppConfig;
+use crate::ipc::{emit_event, respond_err, respond_ok, Command, OutTx};
+use crate::provider::ProviderManager;
+use crate::registrar::Registrar;
+use crate::rtp::RtpPortPool;
+use crate::sip_transport::SipTransport;
+use crate::webrtc_engine::WebRtcEngine;
+use sip_proto::message::SipMessage;
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::io::{AsyncBufReadExt, AsyncWriteExt, BufReader};
+use tokio::net::UdpSocket;
+use tokio::sync::{mpsc, Mutex};
+
+/// Shared mutable state for the proxy engine.
+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,
+}
+
+impl ProxyEngine {
+    fn new(out_tx: OutTx) -> Self {
+        Self {
+            config: None,
+            transport: None,
+            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,
+        }
+    }
+}
+
+#[tokio::main]
+async fn main() {
+    // Output channel: all stdout writes go through here for serialization.
+    let (out_tx, mut out_rx) = mpsc::unbounded_channel::<String>();
+
+    // Stdout writer task.
+    tokio::spawn(async move {
+        let mut stdout = tokio::io::stdout();
+        while let Some(line) = out_rx.recv().await {
+            let mut output = line.into_bytes();
+            output.push(b'\n');
+            if stdout.write_all(&output).await.is_err() {
+                break;
+            }
+            let _ = stdout.flush().await;
+        }
+    });
+
+    // Emit ready event.
+    emit_event(&out_tx, "ready", serde_json::json!({}));
+
+    // Shared engine state.
+    let engine = Arc::new(Mutex::new(ProxyEngine::new(out_tx.clone())));
+
+    // Read commands from stdin.
+    let stdin = tokio::io::stdin();
+    let reader = BufReader::new(stdin);
+    let mut lines = reader.lines();
+
+    while let Ok(Some(line)) = lines.next_line().await {
+        if line.trim().is_empty() {
+            continue;
+        }
+
+        let cmd: Command = match serde_json::from_str(&line) {
+            Ok(c) => c,
+            Err(e) => {
+                respond_err(&out_tx, "", &format!("parse: {e}"));
+                continue;
+            }
+        };
+
+        let engine = engine.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;
+        });
+    }
+}
+
+async fn handle_command(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: Command) {
+    match cmd.method.as_str() {
+        "configure" => handle_configure(engine, out_tx, &cmd).await,
+        "hangup" => handle_hangup(engine, out_tx, &cmd).await,
+        "make_call" => handle_make_call(engine, out_tx, &cmd).await,
+        "get_status" => handle_get_status(engine, out_tx, &cmd).await,
+        "webrtc_offer" => handle_webrtc_offer(engine, out_tx, &cmd).await,
+        "webrtc_ice" => handle_webrtc_ice(engine, out_tx, &cmd).await,
+        "webrtc_link" => handle_webrtc_link(engine, out_tx, &cmd).await,
+        "webrtc_close" => handle_webrtc_close(engine, out_tx, &cmd).await,
+        _ => respond_err(out_tx, &cmd.id, &format!("unknown command: {}", cmd.method)),
+    }
+}
+
+/// Handle the `configure` command — receives full app config from TypeScript.
+/// First call: initializes SIP transport + everything.
+/// Subsequent calls: reconfigures providers/devices/routing without rebinding.
+async fn handle_configure(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let app_config: AppConfig = match serde_json::from_value(cmd.params.clone()) {
+        Ok(c) => c,
+        Err(e) => {
+            respond_err(out_tx, &cmd.id, &format!("bad config: {e}"));
+            return;
+        }
+    };
+
+    let mut eng = engine.lock().await;
+    let is_reconfigure = eng.transport.is_some();
+
+    let socket = if is_reconfigure {
+        // Reconfigure — socket already bound, just update subsystems.
+        eng.transport.as_ref().unwrap().socket()
+    } else {
+        // First configure — bind SIP transport.
+        let bind_addr = format!("0.0.0.0:{}", app_config.proxy.lan_port);
+        let transport = match SipTransport::bind(&bind_addr).await {
+            Ok(t) => t,
+            Err(e) => {
+                respond_err(out_tx, &cmd.id, &format!("SIP bind failed: {e}"));
+                return;
+            }
+        };
+
+        let socket = transport.socket();
+
+        // Start UDP receiver.
+        let engine_for_recv = engine.clone();
+        let socket_for_recv = socket.clone();
+        transport.spawn_receiver(move |data: &[u8], addr: SocketAddr| {
+            let engine = engine_for_recv.clone();
+            let socket = socket_for_recv.clone();
+            let data = data.to_vec();
+            tokio::spawn(async move {
+                handle_sip_packet(engine, &socket, &data, addr).await;
+            });
+        });
+
+        eng.transport = Some(transport);
+
+        // Initialize RTP port pool (only on first configure).
+        eng.rtp_pool = Some(RtpPortPool::new(
+            app_config.proxy.rtp_port_range.min,
+            app_config.proxy.rtp_port_range.max,
+        ));
+
+        socket
+    };
+
+    // (Re)configure registrar.
+    eng.registrar.configure(&app_config.devices);
+
+    // (Re)configure provider registrations.
+    eng.provider_mgr
+        .configure(
+            &app_config.providers,
+            app_config.proxy.public_ip_seed.as_deref(),
+            &app_config.proxy.lan_ip,
+            app_config.proxy.lan_port,
+            socket,
+        )
+        .await;
+
+    let bind_info = format!("0.0.0.0:{}", app_config.proxy.lan_port);
+    eng.config = Some(app_config);
+
+    respond_ok(
+        out_tx,
+        &cmd.id,
+        serde_json::json!({
+            "bound": bind_info,
+            "reconfigure": is_reconfigure,
+        }),
+    );
+}
+
+/// Handle incoming SIP packets from the UDP socket.
+/// This is the core routing pipeline — entirely in Rust.
+async fn handle_sip_packet(
+    engine: Arc<Mutex<ProxyEngine>>,
+    socket: &UdpSocket,
+    data: &[u8],
+    from_addr: SocketAddr,
+) {
+    let msg = match SipMessage::parse(data) {
+        Some(m) => m,
+        None => return, // Not a valid SIP message, ignore.
+    };
+
+    let mut eng = engine.lock().await;
+
+    // 1. Provider registration responses — consumed internally.
+    if msg.is_response() {
+        if eng.provider_mgr.handle_response(&msg, socket).await {
+            return;
+        }
+    }
+
+    // 2. Device REGISTER — handled by registrar.
+    let is_from_provider = eng
+        .provider_mgr
+        .find_by_address(&from_addr)
+        .await
+        .is_some();
+
+    if !is_from_provider && msg.method() == Some("REGISTER") {
+        if let Some(response_buf) = eng.registrar.handle_register(&msg, from_addr) {
+            let _ = socket.send_to(&response_buf, from_addr).await;
+            return;
+        }
+    }
+
+    // 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)
+            .await
+        {
+            return;
+        }
+    }
+
+    let config_ref = eng.config.as_ref().unwrap().clone();
+
+    // 4. New inbound INVITE from provider.
+    if is_from_provider && msg.is_request() && msg.method() == Some("INVITE") {
+        // Detect public IP from Via.
+        if let Some(via) = msg.get_header("Via") {
+            if let Some(ps_arc) = eng.provider_mgr.find_by_address(&from_addr).await {
+                let mut ps = ps_arc.lock().await;
+                ps.detect_public_ip(via);
+            }
+        }
+
+        // Send 100 Trying immediately.
+        let trying = SipMessage::create_response(100, "Trying", &msg, None);
+        let _ = socket.send_to(&trying.serialize(), from_addr).await;
+
+        // Determine provider info.
+        let (provider_id, provider_config, public_ip) =
+            if let Some(ps_arc) = eng.provider_mgr.find_by_address(&from_addr).await {
+                let ps = ps_arc.lock().await;
+                (
+                    ps.config.id.clone(),
+                    ps.config.clone(),
+                    ps.public_ip.clone(),
+                )
+            } else {
+                return;
+            };
+
+        // Create the inbound call — Rust handles everything.
+        // Split borrows via destructuring to satisfy the borrow checker.
+        let ProxyEngine {
+            ref registrar,
+            ref mut call_mgr,
+            ref mut rtp_pool,
+            ..
+        } = *eng;
+        let rtp_pool = rtp_pool.as_mut().unwrap();
+        let call_id = call_mgr
+            .create_inbound_call(
+                &msg,
+                from_addr,
+                &provider_id,
+                &provider_config,
+                &config_ref,
+                registrar,
+                rtp_pool,
+                socket,
+                public_ip.as_deref(),
+            )
+            .await;
+
+        if let Some(call_id) = call_id {
+            // Emit event so TypeScript knows about the call (for dashboard, IVR routing, etc).
+            let from_header = msg.get_header("From").unwrap_or("");
+            let from_uri = SipMessage::extract_uri(from_header).unwrap_or("Unknown");
+            let called_number = msg
+                .request_uri()
+                .and_then(|uri| SipMessage::extract_uri(uri))
+                .unwrap_or("");
+
+            emit_event(
+                &eng.out_tx,
+                "incoming_call",
+                serde_json::json!({
+                    "call_id": call_id,
+                    "from_uri": from_uri,
+                    "to_number": called_number,
+                    "provider_id": provider_id,
+                }),
+            );
+        }
+        return;
+    }
+
+    // 5. New outbound INVITE from device.
+    if !is_from_provider && msg.is_request() && msg.method() == Some("INVITE") {
+        // Resolve outbound route.
+        let dialed_number = msg
+            .request_uri()
+            .and_then(|uri| SipMessage::extract_uri(uri))
+            .unwrap_or(msg.request_uri().unwrap_or(""))
+            .to_string();
+
+        let device = eng.registrar.find_by_address(&from_addr);
+        let device_id = device.map(|d| d.device_id.clone());
+
+        // Find provider via routing rules.
+        let route_result = config_ref.resolve_outbound_route(
+            &dialed_number,
+            device_id.as_deref(),
+            &|pid: &str| {
+                // Can't call async here — use a sync check.
+                // For now, assume all configured providers are available.
+                true
+            },
+        );
+
+        if let Some(route) = route_result {
+            let public_ip = if let Some(ps_arc) = eng.provider_mgr.find_by_address(&from_addr).await {
+                let ps = ps_arc.lock().await;
+                ps.public_ip.clone()
+            } else {
+                None
+            };
+
+            let ProxyEngine {
+                ref mut call_mgr,
+                ref mut rtp_pool,
+                ..
+            } = *eng;
+            let rtp_pool = rtp_pool.as_mut().unwrap();
+            let call_id = call_mgr
+                .create_outbound_passthrough(
+                    &msg,
+                    from_addr,
+                    &route.provider,
+                    &config_ref,
+                    rtp_pool,
+                    socket,
+                    public_ip.as_deref(),
+                )
+                .await;
+
+            if let Some(call_id) = call_id {
+                emit_event(
+                    &eng.out_tx,
+                    "outbound_device_call",
+                    serde_json::json!({
+                        "call_id": call_id,
+                        "from_device": device_id,
+                        "to_number": dialed_number,
+                    }),
+                );
+            }
+        }
+        return;
+    }
+
+    // 6. Other messages — log for debugging.
+    let label = if msg.is_request() {
+        msg.method().unwrap_or("?").to_string()
+    } else {
+        msg.status_code().map(|c| c.to_string()).unwrap_or_default()
+    };
+    emit_event(
+        &eng.out_tx,
+        "sip_unhandled",
+        serde_json::json!({
+            "method_or_status": label,
+            "call_id": msg.call_id(),
+            "from_addr": from_addr.ip().to_string(),
+            "from_port": from_addr.port(),
+            "is_from_provider": is_from_provider,
+        }),
+    );
+}
+
+/// Handle `get_status` — return active call statuses from Rust.
+async fn handle_get_status(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let eng = engine.lock().await;
+    let calls = eng.call_mgr.get_all_statuses();
+    respond_ok(out_tx, &cmd.id, serde_json::json!({ "calls": calls }));
+}
+
+/// Handle `make_call` — initiate an outbound call to a number via a provider.
+async fn handle_make_call(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let number = match cmd.params.get("number").and_then(|v| v.as_str()) {
+        Some(n) => n.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing number"); return; }
+    };
+    let provider_id = cmd.params.get("provider_id").and_then(|v| v.as_str());
+
+    let mut eng = engine.lock().await;
+    let config_ref = match &eng.config {
+        Some(c) => c.clone(),
+        None => { respond_err(out_tx, &cmd.id, "not configured"); return; }
+    };
+
+    // Resolve provider.
+    let provider_config = if let Some(pid) = provider_id {
+        config_ref.providers.iter().find(|p| p.id == pid).cloned()
+    } else {
+        // Use route resolution or first provider.
+        let route = config_ref.resolve_outbound_route(&number, None, &|_| true);
+        route.map(|r| r.provider)
+    };
+
+    let provider_config = match provider_config {
+        Some(p) => p,
+        None => { respond_err(out_tx, &cmd.id, "no provider available"); return; }
+    };
+
+    // Get public IP and registered AOR from provider state.
+    let (public_ip, registered_aor) = if let Some(ps_arc) = eng.provider_mgr.find_by_address(
+        &provider_config.outbound_proxy.to_socket_addr().unwrap_or_else(|| "0.0.0.0:0".parse().unwrap())
+    ).await {
+        let ps = ps_arc.lock().await;
+        (ps.public_ip.clone(), ps.registered_aor.clone())
+    } else {
+        // Fallback — construct AOR from config.
+        (None, format!("sip:{}@{}", provider_config.username, provider_config.domain))
+    };
+
+    let socket = match &eng.transport {
+        Some(t) => t.socket(),
+        None => { respond_err(out_tx, &cmd.id, "not initialized"); return; }
+    };
+
+    let ProxyEngine { ref mut call_mgr, ref mut rtp_pool, .. } = *eng;
+    let rtp_pool = rtp_pool.as_mut().unwrap();
+
+    let call_id = call_mgr.make_outbound_call(
+        &number,
+        &provider_config,
+        &config_ref,
+        rtp_pool,
+        &socket,
+        public_ip.as_deref(),
+        &registered_aor,
+    ).await;
+
+    match call_id {
+        Some(id) => {
+            emit_event(out_tx, "outbound_call_started", serde_json::json!({
+                "call_id": id,
+                "number": number,
+                "provider_id": provider_config.id,
+            }));
+            respond_ok(out_tx, &cmd.id, serde_json::json!({ "call_id": id }));
+        }
+        None => {
+            respond_err(out_tx, &cmd.id, "call origination failed — provider not registered or no ports available");
+        }
+    }
+}
+
+/// Handle the `hangup` command.
+async fn handle_hangup(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let call_id = match cmd.params.get("call_id").and_then(|v| v.as_str()) {
+        Some(id) => id.to_string(),
+        None => {
+            respond_err(out_tx, &cmd.id, "missing call_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.hangup(&call_id, &socket).await {
+        respond_ok(out_tx, &cmd.id, serde_json::json!({}));
+    } else {
+        respond_err(out_tx, &cmd.id, &format!("call {call_id} not found"));
+    }
+}
+
+/// Handle `webrtc_offer` — browser sends SDP offer, we create PeerConnection and return answer.
+async fn handle_webrtc_offer(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+    let offer_sdp = match cmd.params.get("sdp").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing sdp"); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    match eng.webrtc.handle_offer(&session_id, &offer_sdp).await {
+        Ok(answer_sdp) => {
+            respond_ok(out_tx, &cmd.id, serde_json::json!({
+                "session_id": session_id,
+                "sdp": answer_sdp,
+            }));
+        }
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}
+
+/// Handle `webrtc_ice` — forward ICE candidate from browser to Rust PeerConnection.
+async fn handle_webrtc_ice(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+    let candidate = cmd.params.get("candidate").and_then(|v| v.as_str()).unwrap_or("");
+    let sdp_mid = cmd.params.get("sdp_mid").and_then(|v| v.as_str());
+    let sdp_mline_index = cmd.params.get("sdp_mline_index").and_then(|v| v.as_u64()).map(|v| v as u16);
+
+    let eng = engine.lock().await;
+    match eng.webrtc.add_ice_candidate(&session_id, candidate, sdp_mid, sdp_mline_index).await {
+        Ok(()) => respond_ok(out_tx, &cmd.id, serde_json::json!({})),
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}
+
+/// Handle `webrtc_link` — link a WebRTC session to a SIP call for audio bridging.
+async fn handle_webrtc_link(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+    let call_id = match cmd.params.get("call_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing call_id"); return; }
+    };
+    let provider_addr = match cmd.params.get("provider_media_addr").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing provider_media_addr"); return; }
+    };
+    let provider_port = match cmd.params.get("provider_media_port").and_then(|v| v.as_u64()) {
+        Some(p) => p as u16,
+        None => { respond_err(out_tx, &cmd.id, "missing provider_media_port"); return; }
+    };
+    let sip_pt = cmd.params.get("sip_pt").and_then(|v| v.as_u64()).unwrap_or(9) as u8;
+
+    let provider_media: SocketAddr = match format!("{provider_addr}:{provider_port}").parse() {
+        Ok(a) => a,
+        Err(e) => { respond_err(out_tx, &cmd.id, &format!("bad address: {e}")); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    let sip_socket = match &eng.transport {
+        Some(t) => t.socket(),
+        None => { respond_err(out_tx, &cmd.id, "not initialized"); return; }
+    };
+
+    let bridge_info = crate::webrtc_engine::SipBridgeInfo {
+        provider_media,
+        sip_pt,
+        sip_socket,
+    };
+
+    if eng.webrtc.link_to_sip(&session_id, &call_id, bridge_info).await {
+        respond_ok(out_tx, &cmd.id, serde_json::json!({
+            "session_id": session_id,
+            "call_id": call_id,
+            "bridged": true,
+        }));
+    } else {
+        respond_err(out_tx, &cmd.id, &format!("session {session_id} not found"));
+    }
+}
+
+/// Handle `webrtc_close` — close a WebRTC session.
+async fn handle_webrtc_close(engine: Arc<Mutex<ProxyEngine>>, out_tx: &OutTx, cmd: &Command) {
+    let session_id = match cmd.params.get("session_id").and_then(|v| v.as_str()) {
+        Some(s) => s.to_string(),
+        None => { respond_err(out_tx, &cmd.id, "missing session_id"); return; }
+    };
+
+    let mut eng = engine.lock().await;
+    match eng.webrtc.close_session(&session_id).await {
+        Ok(()) => respond_ok(out_tx, &cmd.id, serde_json::json!({})),
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}

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

@@ -0,0 +1,367 @@
+//! Provider registration state machine.
+//!
+//! Handles the REGISTER cycle with upstream SIP providers:
+//! - Sends periodic REGISTER messages
+//! - Handles 401/407 Digest authentication challenges
+//! - Detects public IP from Via received= parameter
+//! - Emits registration state events to TypeScript
+//!
+//! Ported from ts/providerstate.ts.
+
+use crate::config::ProviderConfig;
+use crate::ipc::{emit_event, OutTx};
+use sip_proto::helpers::{
+    compute_digest_auth, generate_branch, generate_call_id, generate_tag, parse_digest_challenge,
+};
+use sip_proto::message::{RequestOptions, SipMessage};
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+use tokio::sync::Mutex;
+use tokio::time::{self, Duration};
+
+/// Runtime state for a single SIP provider.
+pub struct ProviderState {
+    pub config: ProviderConfig,
+    pub public_ip: Option<String>,
+    pub is_registered: bool,
+    pub registered_aor: String,
+
+    // Registration transaction state.
+    reg_call_id: String,
+    reg_cseq: u32,
+    reg_from_tag: String,
+
+    // Network.
+    lan_ip: String,
+    lan_port: u16,
+}
+
+impl ProviderState {
+    pub fn new(config: ProviderConfig, public_ip_seed: Option<&str>) -> Self {
+        let aor = format!("sip:{}@{}", config.username, config.domain);
+        Self {
+            public_ip: public_ip_seed.map(|s| s.to_string()),
+            is_registered: false,
+            registered_aor: aor,
+            reg_call_id: generate_call_id(None),
+            reg_cseq: 0,
+            reg_from_tag: generate_tag(),
+            lan_ip: String::new(),
+            lan_port: 0,
+            config,
+        }
+    }
+
+    /// Build and send a REGISTER request.
+    pub fn build_register(&mut self) -> Vec<u8> {
+        self.reg_cseq += 1;
+        let pub_ip = self.public_ip.as_deref().unwrap_or(&self.lan_ip);
+
+        let register = SipMessage::create_request(
+            "REGISTER",
+            &format!("sip:{}", self.config.domain),
+            RequestOptions {
+                via_host: pub_ip.to_string(),
+                via_port: self.lan_port,
+                via_transport: None,
+                via_branch: Some(generate_branch()),
+                from_uri: self.registered_aor.clone(),
+                from_display_name: None,
+                from_tag: Some(self.reg_from_tag.clone()),
+                to_uri: self.registered_aor.clone(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some(self.reg_call_id.clone()),
+                cseq: Some(self.reg_cseq),
+                contact: Some(format!(
+                    "<sip:{}@{}:{}>",
+                    self.config.username, pub_ip, self.lan_port
+                )),
+                max_forwards: Some(70),
+                body: None,
+                content_type: None,
+                extra_headers: Some(vec![
+                    (
+                        "Expires".to_string(),
+                        self.config.register_interval_sec.to_string(),
+                    ),
+                    ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
+                    (
+                        "Allow".to_string(),
+                        "INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE"
+                            .to_string(),
+                    ),
+                ]),
+            },
+        );
+        register.serialize()
+    }
+
+    /// Handle a SIP response that might be for this provider's REGISTER.
+    /// Returns true if the message was consumed.
+    pub fn handle_registration_response(&mut self, msg: &SipMessage) -> Option<Vec<u8>> {
+        if !msg.is_response() {
+            return None;
+        }
+        if msg.call_id() != self.reg_call_id {
+            return None;
+        }
+        let cseq_method = msg.cseq_method().unwrap_or("");
+        if !cseq_method.eq_ignore_ascii_case("REGISTER") {
+            return None;
+        }
+
+        let code = msg.status_code().unwrap_or(0);
+
+        if code == 200 {
+            self.is_registered = true;
+            return Some(Vec::new()); // consumed, no reply needed
+        }
+
+        if code == 401 || code == 407 {
+            let challenge_header = if code == 401 {
+                msg.get_header("WWW-Authenticate")
+            } else {
+                msg.get_header("Proxy-Authenticate")
+            };
+
+            let challenge_header = match challenge_header {
+                Some(h) => h,
+                None => return Some(Vec::new()), // consumed but no challenge
+            };
+
+            let challenge = match parse_digest_challenge(challenge_header) {
+                Some(c) => c,
+                None => return Some(Vec::new()),
+            };
+
+            let auth_value = compute_digest_auth(
+                &self.config.username,
+                &self.config.password,
+                &challenge.realm,
+                &challenge.nonce,
+                "REGISTER",
+                &format!("sip:{}", self.config.domain),
+                challenge.algorithm.as_deref(),
+                challenge.opaque.as_deref(),
+            );
+
+            // Resend REGISTER with auth credentials.
+            self.reg_cseq += 1;
+            let pub_ip = self.public_ip.as_deref().unwrap_or(&self.lan_ip);
+
+            let auth_header_name = if code == 401 {
+                "Authorization"
+            } else {
+                "Proxy-Authorization"
+            };
+
+            let register = SipMessage::create_request(
+                "REGISTER",
+                &format!("sip:{}", self.config.domain),
+                RequestOptions {
+                    via_host: pub_ip.to_string(),
+                    via_port: self.lan_port,
+                    via_transport: None,
+                    via_branch: Some(generate_branch()),
+                    from_uri: self.registered_aor.clone(),
+                    from_display_name: None,
+                    from_tag: Some(self.reg_from_tag.clone()),
+                    to_uri: self.registered_aor.clone(),
+                    to_display_name: None,
+                    to_tag: None,
+                    call_id: Some(self.reg_call_id.clone()),
+                    cseq: Some(self.reg_cseq),
+                    contact: Some(format!(
+                        "<sip:{}@{}:{}>",
+                        self.config.username, pub_ip, self.lan_port
+                    )),
+                    max_forwards: Some(70),
+                    body: None,
+                    content_type: None,
+                    extra_headers: Some(vec![
+                        (auth_header_name.to_string(), auth_value),
+                        (
+                            "Expires".to_string(),
+                            self.config.register_interval_sec.to_string(),
+                        ),
+                        ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
+                        (
+                            "Allow".to_string(),
+                            "INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE"
+                                .to_string(),
+                        ),
+                    ]),
+                },
+            );
+            return Some(register.serialize());
+        }
+
+        if code >= 400 {
+            self.is_registered = false;
+        }
+
+        Some(Vec::new()) // consumed
+    }
+
+    /// Detect public IP from Via received= parameter.
+    pub fn detect_public_ip(&mut self, via: &str) {
+        if let Some(m) = via.find("received=") {
+            let rest = &via[m + 9..];
+            let end = rest
+                .find(|c: char| !c.is_ascii_digit() && c != '.')
+                .unwrap_or(rest.len());
+            let ip = &rest[..end];
+            if !ip.is_empty() && self.public_ip.as_deref() != Some(ip) {
+                self.public_ip = Some(ip.to_string());
+            }
+        }
+    }
+
+    pub fn set_network(&mut self, lan_ip: &str, lan_port: u16) {
+        self.lan_ip = lan_ip.to_string();
+        self.lan_port = lan_port;
+    }
+}
+
+/// Manages all provider states and their registration cycles.
+pub struct ProviderManager {
+    providers: Vec<Arc<Mutex<ProviderState>>>,
+    out_tx: OutTx,
+}
+
+impl ProviderManager {
+    pub fn new(out_tx: OutTx) -> Self {
+        Self {
+            providers: Vec::new(),
+            out_tx,
+        }
+    }
+
+    /// Initialize providers from config and start registration cycles.
+    pub async fn configure(
+        &mut self,
+        configs: &[ProviderConfig],
+        public_ip_seed: Option<&str>,
+        lan_ip: &str,
+        lan_port: u16,
+        socket: Arc<UdpSocket>,
+    ) {
+        self.providers.clear();
+
+        for cfg in configs {
+            let mut ps = ProviderState::new(cfg.clone(), public_ip_seed);
+            ps.set_network(lan_ip, lan_port);
+            let ps = Arc::new(Mutex::new(ps));
+            self.providers.push(ps.clone());
+
+            // Start the registration cycle.
+            let socket = socket.clone();
+            let out_tx = self.out_tx.clone();
+            tokio::spawn(async move {
+                provider_register_loop(ps, socket, out_tx).await;
+            });
+        }
+    }
+
+    /// Try to handle a SIP response as a provider registration response.
+    /// Returns true if consumed.
+    pub async fn handle_response(
+        &self,
+        msg: &SipMessage,
+        socket: &UdpSocket,
+    ) -> bool {
+        for ps_arc in &self.providers {
+            let mut ps = ps_arc.lock().await;
+            let was_registered = ps.is_registered;
+            if let Some(reply) = ps.handle_registration_response(msg) {
+                // If there's a reply to send (e.g. auth retry).
+                if !reply.is_empty() {
+                    if let Some(dest) = ps.config.outbound_proxy.to_socket_addr() {
+                        let _ = socket.send_to(&reply, dest).await;
+                    }
+                }
+                // Emit registration state change.
+                if ps.is_registered != was_registered {
+                    emit_event(
+                        &self.out_tx,
+                        "provider_registered",
+                        serde_json::json!({
+                            "provider_id": ps.config.id,
+                            "registered": ps.is_registered,
+                            "public_ip": ps.public_ip,
+                        }),
+                    );
+                }
+                return true;
+            }
+        }
+        false
+    }
+
+    /// Find which provider sent a packet by matching source address.
+    pub async fn find_by_address(&self, addr: &SocketAddr) -> Option<Arc<Mutex<ProviderState>>> {
+        for ps_arc in &self.providers {
+            let ps = ps_arc.lock().await;
+            let proxy_addr = format!(
+                "{}:{}",
+                ps.config.outbound_proxy.address, ps.config.outbound_proxy.port
+            );
+            if let Ok(expected) = proxy_addr.parse::<SocketAddr>() {
+                if expected == *addr {
+                    return Some(ps_arc.clone());
+                }
+            }
+            // Also match by IP only (port may differ).
+            if ps.config.outbound_proxy.address == addr.ip().to_string() {
+                return Some(ps_arc.clone());
+            }
+        }
+        None
+    }
+
+    /// Check if a provider is currently registered.
+    pub async fn is_registered(&self, provider_id: &str) -> bool {
+        for ps_arc in &self.providers {
+            let ps = ps_arc.lock().await;
+            if ps.config.id == provider_id {
+                return ps.is_registered;
+            }
+        }
+        false
+    }
+}
+
+/// Registration loop for a single provider.
+async fn provider_register_loop(
+    ps: Arc<Mutex<ProviderState>>,
+    socket: Arc<UdpSocket>,
+    _out_tx: OutTx,
+) {
+    // Initial registration.
+    {
+        let mut state = ps.lock().await;
+        let register_buf = state.build_register();
+        if let Some(dest) = state.config.outbound_proxy.to_socket_addr() {
+            let _ = socket.send_to(&register_buf, dest).await;
+        }
+    }
+
+    // Re-register periodically (85% of the interval).
+    let interval_sec = {
+        let state = ps.lock().await;
+        (state.config.register_interval_sec as f64 * 0.85) as u64
+    };
+    let mut interval = time::interval(Duration::from_secs(interval_sec.max(30)));
+    interval.tick().await; // skip first immediate tick
+
+    loop {
+        interval.tick().await;
+        let mut state = ps.lock().await;
+        let register_buf = state.build_register();
+        if let Some(dest) = state.config.outbound_proxy.to_socket_addr() {
+            let _ = socket.send_to(&register_buf, dest).await;
+        }
+    }
+}

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

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

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

@@ -0,0 +1,171 @@
+//! Device registrar — accepts REGISTER from SIP phones and tracks contacts.
+//!
+//! When a device sends REGISTER, the registrar responds with 200 OK
+//! and stores the device's current contact (source IP:port).
+//!
+//! Ported from ts/registrar.ts.
+
+use crate::config::DeviceConfig;
+use crate::ipc::{emit_event, OutTx};
+use sip_proto::helpers::generate_tag;
+use sip_proto::message::{ResponseOptions, SipMessage};
+use std::collections::HashMap;
+use std::net::SocketAddr;
+use std::time::{Duration, Instant};
+
+const MAX_EXPIRES: u32 = 300;
+
+/// A registered device entry.
+#[derive(Debug, Clone)]
+pub struct RegisteredDevice {
+    pub device_id: String,
+    pub display_name: String,
+    pub extension: String,
+    pub contact_addr: SocketAddr,
+    pub registered_at: Instant,
+    pub expires_at: Instant,
+    pub aor: String,
+}
+
+/// Manages device registrations.
+pub struct Registrar {
+    /// Known device configs (from app config).
+    devices: Vec<DeviceConfig>,
+    /// Currently registered devices, keyed by device ID.
+    registered: HashMap<String, RegisteredDevice>,
+    out_tx: OutTx,
+}
+
+impl Registrar {
+    pub fn new(out_tx: OutTx) -> Self {
+        Self {
+            devices: Vec::new(),
+            registered: HashMap::new(),
+            out_tx,
+        }
+    }
+
+    /// Update the known device list from config.
+    pub fn configure(&mut self, devices: &[DeviceConfig]) {
+        self.devices = devices.to_vec();
+    }
+
+    /// Try to handle a SIP REGISTER from a device.
+    /// Returns Some(response_bytes) if handled, None if not a known device.
+    pub fn handle_register(
+        &mut self,
+        msg: &SipMessage,
+        from_addr: SocketAddr,
+    ) -> Option<Vec<u8>> {
+        if msg.method() != Some("REGISTER") {
+            return None;
+        }
+
+        // Find the device by matching the source IP against expectedAddress.
+        let from_ip = from_addr.ip().to_string();
+        let device = self.devices.iter().find(|d| d.expected_address == from_ip)?;
+
+        let from_header = msg.get_header("From").unwrap_or("");
+        let aor = SipMessage::extract_uri(from_header)
+            .map(|s| s.to_string())
+            .unwrap_or_else(|| format!("sip:{}@{}", device.extension, from_ip));
+
+        let expires_header = msg.get_header("Expires");
+        let requested: u32 = expires_header
+            .and_then(|s| s.parse().ok())
+            .unwrap_or(3600);
+        let expires = requested.min(MAX_EXPIRES);
+
+        let entry = RegisteredDevice {
+            device_id: device.id.clone(),
+            display_name: device.display_name.clone(),
+            extension: device.extension.clone(),
+            contact_addr: from_addr,
+            registered_at: Instant::now(),
+            expires_at: Instant::now() + Duration::from_secs(expires as u64),
+            aor: aor.clone(),
+        };
+        self.registered.insert(device.id.clone(), entry);
+
+        // Emit event to TypeScript.
+        emit_event(
+            &self.out_tx,
+            "device_registered",
+            serde_json::json!({
+                "device_id": device.id,
+                "display_name": device.display_name,
+                "address": from_ip,
+                "port": from_addr.port(),
+                "aor": aor,
+                "expires": expires,
+            }),
+        );
+
+        // Build 200 OK response.
+        let contact = msg
+            .get_header("Contact")
+            .map(|s| s.to_string())
+            .unwrap_or_else(|| format!("<sip:{}:{}>", from_ip, from_addr.port()));
+
+        let response = SipMessage::create_response(
+            200,
+            "OK",
+            msg,
+            Some(ResponseOptions {
+                to_tag: Some(generate_tag()),
+                contact: Some(contact),
+                extra_headers: Some(vec![(
+                    "Expires".to_string(),
+                    expires.to_string(),
+                )]),
+                ..Default::default()
+            }),
+        );
+
+        Some(response.serialize())
+    }
+
+    /// Get the contact address for a registered device.
+    pub fn get_device_contact(&self, device_id: &str) -> Option<SocketAddr> {
+        let entry = self.registered.get(device_id)?;
+        if Instant::now() > entry.expires_at {
+            return None;
+        }
+        Some(entry.contact_addr)
+    }
+
+    /// Check if a source address belongs to a known device.
+    pub fn is_known_device_address(&self, addr: &str) -> bool {
+        self.devices.iter().any(|d| d.expected_address == addr)
+    }
+
+    /// Find a registered device by its source IP address.
+    pub fn find_by_address(&self, addr: &SocketAddr) -> Option<&RegisteredDevice> {
+        let ip = addr.ip().to_string();
+        self.registered.values().find(|e| {
+            e.contact_addr.ip().to_string() == ip && Instant::now() <= e.expires_at
+        })
+    }
+
+    /// Get all device statuses for the dashboard.
+    pub fn get_all_statuses(&self) -> Vec<serde_json::Value> {
+        let now = Instant::now();
+        let mut result = Vec::new();
+
+        for dc in &self.devices {
+            let reg = self.registered.get(&dc.id);
+            let connected = reg.map(|r| now <= r.expires_at).unwrap_or(false);
+            result.push(serde_json::json!({
+                "id": dc.id,
+                "displayName": dc.display_name,
+                "address": reg.filter(|_| connected).map(|r| r.contact_addr.ip().to_string()),
+                "port": reg.filter(|_| connected).map(|r| r.contact_addr.port()),
+                "aor": reg.map(|r| r.aor.as_str()).unwrap_or(""),
+                "connected": connected,
+                "isBrowser": false,
+            }));
+        }
+
+        result
+    }
+}

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

@@ -0,0 +1,158 @@
+//! RTP port pool and media forwarding.
+//!
+//! Manages a pool of even-numbered UDP ports for RTP media.
+//! Each port gets a bound tokio UdpSocket. Supports:
+//! - Direct forwarding (SIP-to-SIP, no transcoding)
+//! - Transcoding forwarding (via codec-lib, e.g. G.722 ↔ Opus)
+//! - Silence generation
+//! - NAT priming
+//!
+//! Ported from ts/call/rtp-port-pool.ts + sip-leg.ts RTP handling.
+
+use std::collections::HashMap;
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+
+/// A single RTP port allocation.
+pub struct RtpAllocation {
+    pub port: u16,
+    pub socket: Arc<UdpSocket>,
+}
+
+/// RTP port pool — allocates even-numbered UDP ports.
+pub struct RtpPortPool {
+    min: u16,
+    max: u16,
+    allocated: HashMap<u16, Arc<UdpSocket>>,
+}
+
+impl RtpPortPool {
+    pub fn new(min: u16, max: u16) -> Self {
+        let min = if min % 2 == 0 { min } else { min + 1 };
+        Self {
+            min,
+            max,
+            allocated: HashMap::new(),
+        }
+    }
+
+    /// Allocate an even-numbered port and bind a UDP socket.
+    pub async fn allocate(&mut self) -> Option<RtpAllocation> {
+        let mut port = self.min;
+        while port < self.max {
+            if !self.allocated.contains_key(&port) {
+                match UdpSocket::bind(format!("0.0.0.0:{port}")).await {
+                    Ok(sock) => {
+                        let sock = Arc::new(sock);
+                        self.allocated.insert(port, sock.clone());
+                        return Some(RtpAllocation { port, socket: sock });
+                    }
+                    Err(_) => {
+                        // Port in use, try next.
+                    }
+                }
+            }
+            port += 2;
+        }
+        None // Pool exhausted.
+    }
+
+    /// Release a port back to the pool.
+    pub fn release(&mut self, port: u16) {
+        self.allocated.remove(&port);
+        // Socket is dropped when the last Arc reference goes away.
+    }
+
+    pub fn size(&self) -> usize {
+        self.allocated.len()
+    }
+
+    pub fn capacity(&self) -> usize {
+        ((self.max - self.min) / 2) as usize
+    }
+}
+
+/// An active RTP relay between two endpoints.
+/// Receives on `local_socket` and forwards to `remote_addr`.
+pub struct RtpRelay {
+    pub local_port: u16,
+    pub local_socket: Arc<UdpSocket>,
+    pub remote_addr: Option<SocketAddr>,
+    /// If set, transcode packets using this codec session before forwarding.
+    pub transcode: Option<TranscodeConfig>,
+    /// Packets received counter.
+    pub pkt_received: u64,
+    /// Packets sent counter.
+    pub pkt_sent: u64,
+}
+
+pub struct TranscodeConfig {
+    pub from_pt: u8,
+    pub to_pt: u8,
+    pub session_id: String,
+}
+
+impl RtpRelay {
+    pub fn new(port: u16, socket: Arc<UdpSocket>) -> Self {
+        Self {
+            local_port: port,
+            local_socket: socket,
+            remote_addr: None,
+            transcode: None,
+            pkt_received: 0,
+            pkt_sent: 0,
+        }
+    }
+
+    pub fn set_remote(&mut self, addr: SocketAddr) {
+        self.remote_addr = Some(addr);
+    }
+}
+
+/// Send a 1-byte NAT priming packet to open a pinhole.
+pub async fn prime_nat(socket: &UdpSocket, remote: SocketAddr) {
+    let _ = socket.send_to(&[0u8], remote).await;
+}
+
+/// Build an RTP silence frame for PCMU (payload type 0).
+pub fn silence_frame_pcmu() -> Vec<u8> {
+    // 12-byte RTP header + 160 bytes of µ-law silence (0xFF)
+    let mut frame = vec![0u8; 172];
+    frame[0] = 0x80; // V=2
+    frame[1] = 0; // PT=0 (PCMU)
+    // seq, timestamp, ssrc left as 0 — caller should set these
+    frame[12..].fill(0xFF); // µ-law silence
+    frame
+}
+
+/// Build an RTP silence frame for G.722 (payload type 9).
+pub fn silence_frame_g722() -> Vec<u8> {
+    // 12-byte RTP header + 160 bytes of G.722 silence
+    let mut frame = vec![0u8; 172];
+    frame[0] = 0x80; // V=2
+    frame[1] = 9; // PT=9 (G.722)
+    // G.722 silence: all zeros is valid silence
+    frame
+}
+
+/// Build an RTP header with the given parameters.
+pub fn build_rtp_header(pt: u8, seq: u16, timestamp: u32, ssrc: u32) -> [u8; 12] {
+    let mut header = [0u8; 12];
+    header[0] = 0x80; // V=2
+    header[1] = pt & 0x7F;
+    header[2..4].copy_from_slice(&seq.to_be_bytes());
+    header[4..8].copy_from_slice(&timestamp.to_be_bytes());
+    header[8..12].copy_from_slice(&ssrc.to_be_bytes());
+    header
+}
+
+/// Get the RTP clock increment per 20ms frame for a payload type.
+pub fn rtp_clock_increment(pt: u8) -> u32 {
+    match pt {
+        9 => 160,   // G.722: 8000 Hz clock rate (despite 16kHz audio) × 0.02s
+        0 | 8 => 160, // PCMU/PCMA: 8000 × 0.02
+        111 => 960, // Opus: 48000 × 0.02
+        _ => 160,
+    }
+}

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

@@ -0,0 +1,67 @@
+//! SIP UDP transport — owns the main SIP socket.
+//!
+//! Binds a UDP socket, receives SIP messages, and provides a send method.
+
+use std::net::SocketAddr;
+use std::sync::Arc;
+use tokio::net::UdpSocket;
+
+/// The SIP UDP transport layer.
+pub struct SipTransport {
+    socket: Arc<UdpSocket>,
+}
+
+impl SipTransport {
+    /// Bind a UDP socket on the given address (e.g. "0.0.0.0:5070").
+    pub async fn bind(bind_addr: &str) -> Result<Self, String> {
+        let socket = UdpSocket::bind(bind_addr)
+            .await
+            .map_err(|e| format!("bind {bind_addr}: {e}"))?;
+        Ok(Self {
+            socket: Arc::new(socket),
+        })
+    }
+
+    /// Get a clone of the socket Arc for the receiver task.
+    pub fn socket(&self) -> Arc<UdpSocket> {
+        self.socket.clone()
+    }
+
+    /// Send a raw SIP message to a destination.
+    pub async fn send_to(&self, data: &[u8], dest: SocketAddr) -> Result<usize, String> {
+        self.socket
+            .send_to(data, dest)
+            .await
+            .map_err(|e| format!("send to {dest}: {e}"))
+    }
+
+    /// Send a raw SIP message to an address:port pair.
+    pub async fn send_to_addr(&self, data: &[u8], addr: &str, port: u16) -> Result<usize, String> {
+        let dest: SocketAddr = format!("{addr}:{port}")
+            .parse()
+            .map_err(|e| format!("bad address {addr}:{port}: {e}"))?;
+        self.send_to(data, dest).await
+    }
+
+    /// Spawn the UDP receive loop. Calls the handler for every received packet.
+    pub fn spawn_receiver<F>(
+        &self,
+        handler: F,
+    ) where
+        F: Fn(&[u8], SocketAddr) + Send + 'static,
+    {
+        let socket = self.socket.clone();
+        tokio::spawn(async move {
+            let mut buf = vec![0u8; 65535];
+            loop {
+                match socket.recv_from(&mut buf).await {
+                    Ok((n, addr)) => handler(&buf[..n], addr),
+                    Err(e) => {
+                        eprintln!("[sip_transport] recv error: {e}");
+                        tokio::time::sleep(std::time::Duration::from_millis(100)).await;
+                    }
+                }
+            }
+        });
+    }
+}

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

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

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

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

rust/crates/sip-proto/Cargo.toml

@@ -0,0 +1,8 @@
+[package]
+name = "sip-proto"
+version = "0.1.0"
+edition = "2021"
+
+[dependencies]
+md-5 = "0.10"
+rand = "0.8"

rust/crates/sip-proto/src/dialog.rs

@@ -0,0 +1,408 @@
+//! SIP dialog state machine (RFC 3261 §12).
+//!
+//! Tracks local/remote tags, CSeq counters, route set, and remote target.
+//! Provides methods to build in-dialog requests (BYE, re-INVITE, ACK, CANCEL).
+//!
+//! Ported from ts/sip/dialog.ts.
+
+use crate::helpers::{generate_branch, generate_tag};
+use crate::message::SipMessage;
+
+#[derive(Debug, Clone, Copy, PartialEq, Eq)]
+pub enum DialogState {
+    Early,
+    Confirmed,
+    Terminated,
+}
+
+/// SIP dialog state per RFC 3261 §12.
+#[derive(Debug, Clone)]
+pub struct SipDialog {
+    pub call_id: String,
+    pub local_tag: String,
+    pub remote_tag: Option<String>,
+    pub local_uri: String,
+    pub remote_uri: String,
+    pub local_cseq: u32,
+    pub remote_cseq: u32,
+    pub route_set: Vec<String>,
+    pub remote_target: String,
+    pub state: DialogState,
+    pub local_host: String,
+    pub local_port: u16,
+}
+
+impl SipDialog {
+    /// Create a dialog from an INVITE we are sending (UAC side).
+    /// The dialog enters Early state; call `process_response()` when responses arrive.
+    pub fn from_uac_invite(invite: &SipMessage, local_host: &str, local_port: u16) -> Self {
+        let from = invite.get_header("From").unwrap_or("");
+        let to = invite.get_header("To").unwrap_or("");
+
+        let local_cseq = invite
+            .get_header("CSeq")
+            .and_then(|c| c.split_whitespace().next())
+            .and_then(|s| s.parse().ok())
+            .unwrap_or(1);
+
+        Self {
+            call_id: invite.call_id().to_string(),
+            local_tag: SipMessage::extract_tag(from)
+                .map(|s| s.to_string())
+                .unwrap_or_else(generate_tag),
+            remote_tag: None,
+            local_uri: SipMessage::extract_uri(from)
+                .unwrap_or("")
+                .to_string(),
+            remote_uri: SipMessage::extract_uri(to).unwrap_or("").to_string(),
+            local_cseq,
+            remote_cseq: 0,
+            route_set: Vec::new(),
+            remote_target: invite
+                .request_uri()
+                .or_else(|| SipMessage::extract_uri(to))
+                .unwrap_or("")
+                .to_string(),
+            state: DialogState::Early,
+            local_host: local_host.to_string(),
+            local_port,
+        }
+    }
+
+    /// Create a dialog from an INVITE we received (UAS side).
+    pub fn from_uas_invite(
+        invite: &SipMessage,
+        local_tag: &str,
+        local_host: &str,
+        local_port: u16,
+    ) -> Self {
+        let from = invite.get_header("From").unwrap_or("");
+        let to = invite.get_header("To").unwrap_or("");
+        let contact = invite.get_header("Contact");
+
+        let remote_target = contact
+            .and_then(SipMessage::extract_uri)
+            .or_else(|| SipMessage::extract_uri(from))
+            .unwrap_or("")
+            .to_string();
+
+        Self {
+            call_id: invite.call_id().to_string(),
+            local_tag: local_tag.to_string(),
+            remote_tag: SipMessage::extract_tag(from).map(|s| s.to_string()),
+            local_uri: SipMessage::extract_uri(to).unwrap_or("").to_string(),
+            remote_uri: SipMessage::extract_uri(from).unwrap_or("").to_string(),
+            local_cseq: 0,
+            remote_cseq: 0,
+            route_set: Vec::new(),
+            remote_target,
+            state: DialogState::Early,
+            local_host: local_host.to_string(),
+            local_port,
+        }
+    }
+
+    /// Update dialog state from a received response.
+    pub fn process_response(&mut self, response: &SipMessage) {
+        let to = response.get_header("To").unwrap_or("");
+        let tag = SipMessage::extract_tag(to).map(|s| s.to_string());
+        let code = response.status_code().unwrap_or(0);
+
+        // Always update remoteTag from 2xx (RFC 3261 §12.1.2).
+        if let Some(ref t) = tag {
+            if code >= 200 && code < 300 {
+                self.remote_tag = Some(t.clone());
+            } else if self.remote_tag.is_none() {
+                self.remote_tag = Some(t.clone());
+            }
+        }
+
+        // Update remote target from Contact.
+        if let Some(contact) = response.get_header("Contact") {
+            if let Some(uri) = SipMessage::extract_uri(contact) {
+                self.remote_target = uri.to_string();
+            }
+        }
+
+        // Record-Route → route set (in reverse for UAC).
+        if self.state == DialogState::Early {
+            let rr: Vec<String> = response
+                .headers
+                .iter()
+                .filter(|(n, _)| n.to_ascii_lowercase() == "record-route")
+                .map(|(_, v)| v.clone())
+                .collect();
+            if !rr.is_empty() {
+                let mut reversed = rr;
+                reversed.reverse();
+                self.route_set = reversed;
+            }
+        }
+
+        if code >= 200 && code < 300 {
+            self.state = DialogState::Confirmed;
+        } else if code >= 300 {
+            self.state = DialogState::Terminated;
+        }
+    }
+
+    /// Build an in-dialog request (BYE, re-INVITE, INFO, ...).
+    /// Automatically increments the local CSeq.
+    pub fn create_request(
+        &mut self,
+        method: &str,
+        body: Option<&str>,
+        content_type: Option<&str>,
+        extra_headers: Option<Vec<(String, String)>>,
+    ) -> SipMessage {
+        self.local_cseq += 1;
+        let branch = generate_branch();
+
+        let remote_tag_str = self
+            .remote_tag
+            .as_ref()
+            .map(|t| format!(";tag={t}"))
+            .unwrap_or_default();
+
+        let mut headers = vec![
+            (
+                "Via".to_string(),
+                format!(
+                    "SIP/2.0/UDP {}:{};branch={branch};rport",
+                    self.local_host, self.local_port
+                ),
+            ),
+            (
+                "From".to_string(),
+                format!("<{}>;tag={}", self.local_uri, self.local_tag),
+            ),
+            (
+                "To".to_string(),
+                format!("<{}>{remote_tag_str}", self.remote_uri),
+            ),
+            ("Call-ID".to_string(), self.call_id.clone()),
+            (
+                "CSeq".to_string(),
+                format!("{} {method}", self.local_cseq),
+            ),
+            ("Max-Forwards".to_string(), "70".to_string()),
+        ];
+
+        for route in &self.route_set {
+            headers.push(("Route".to_string(), route.clone()));
+        }
+
+        headers.push((
+            "Contact".to_string(),
+            format!("<sip:{}:{}>", self.local_host, self.local_port),
+        ));
+
+        if let Some(extra) = extra_headers {
+            headers.extend(extra);
+        }
+
+        let body_str = body.unwrap_or("");
+        if !body_str.is_empty() {
+            if let Some(ct) = content_type {
+                headers.push(("Content-Type".to_string(), ct.to_string()));
+            }
+        }
+        headers.push(("Content-Length".to_string(), body_str.len().to_string()));
+
+        let ruri = self.resolve_ruri();
+        SipMessage::new(
+            format!("{method} {ruri} SIP/2.0"),
+            headers,
+            body_str.to_string(),
+        )
+    }
+
+    /// Build an ACK for a 2xx response to INVITE (RFC 3261 §13.2.2.4).
+    pub fn create_ack(&self) -> SipMessage {
+        let branch = generate_branch();
+        let remote_tag_str = self
+            .remote_tag
+            .as_ref()
+            .map(|t| format!(";tag={t}"))
+            .unwrap_or_default();
+
+        let mut headers = vec![
+            (
+                "Via".to_string(),
+                format!(
+                    "SIP/2.0/UDP {}:{};branch={branch};rport",
+                    self.local_host, self.local_port
+                ),
+            ),
+            (
+                "From".to_string(),
+                format!("<{}>;tag={}", self.local_uri, self.local_tag),
+            ),
+            (
+                "To".to_string(),
+                format!("<{}>{remote_tag_str}", self.remote_uri),
+            ),
+            ("Call-ID".to_string(), self.call_id.clone()),
+            (
+                "CSeq".to_string(),
+                format!("{} ACK", self.local_cseq),
+            ),
+            ("Max-Forwards".to_string(), "70".to_string()),
+        ];
+
+        for route in &self.route_set {
+            headers.push(("Route".to_string(), route.clone()));
+        }
+
+        headers.push(("Content-Length".to_string(), "0".to_string()));
+
+        let ruri = self.resolve_ruri();
+        SipMessage::new(format!("ACK {ruri} SIP/2.0"), headers, String::new())
+    }
+
+    /// Build a CANCEL for the original INVITE (same branch, CSeq).
+    pub fn create_cancel(&self, original_invite: &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 = original_invite.get_header("To").unwrap_or("").to_string();
+
+        let headers = vec![
+            ("Via".to_string(), via),
+            ("From".to_string(), from),
+            ("To".to_string(), to),
+            ("Call-ID".to_string(), self.call_id.clone()),
+            (
+                "CSeq".to_string(),
+                format!("{} CANCEL", self.local_cseq),
+            ),
+            ("Max-Forwards".to_string(), "70".to_string()),
+            ("Content-Length".to_string(), "0".to_string()),
+        ];
+
+        let ruri = original_invite
+            .request_uri()
+            .unwrap_or(&self.remote_target)
+            .to_string();
+
+        SipMessage::new(
+            format!("CANCEL {ruri} SIP/2.0"),
+            headers,
+            String::new(),
+        )
+    }
+
+    /// Transition the dialog to terminated state.
+    pub fn terminate(&mut self) {
+        self.state = DialogState::Terminated;
+    }
+
+    /// Resolve Request-URI from route set or remote target.
+    fn resolve_ruri(&self) -> &str {
+        if !self.route_set.is_empty() {
+            if let Some(top_route) = SipMessage::extract_uri(&self.route_set[0]) {
+                if top_route.contains(";lr") {
+                    return &self.remote_target; // loose routing
+                }
+                return top_route; // strict routing
+            }
+        }
+        &self.remote_target
+    }
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+    use crate::message::RequestOptions;
+
+    fn make_invite() -> SipMessage {
+        SipMessage::create_request(
+            "INVITE",
+            "sip:callee@host",
+            RequestOptions {
+                via_host: "192.168.1.1".to_string(),
+                via_port: 5070,
+                via_transport: None,
+                via_branch: Some("z9hG4bK-test".to_string()),
+                from_uri: "sip:caller@proxy".to_string(),
+                from_display_name: None,
+                from_tag: Some("from-tag".to_string()),
+                to_uri: "sip:callee@host".to_string(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some("test-dialog-call".to_string()),
+                cseq: Some(1),
+                contact: Some("<sip:caller@192.168.1.1:5070>".to_string()),
+                max_forwards: None,
+                body: None,
+                content_type: None,
+                extra_headers: None,
+            },
+        )
+    }
+
+    #[test]
+    fn uac_dialog_lifecycle() {
+        let invite = make_invite();
+        let mut dialog = SipDialog::from_uac_invite(&invite, "192.168.1.1", 5070);
+
+        assert_eq!(dialog.state, DialogState::Early);
+        assert_eq!(dialog.call_id, "test-dialog-call");
+        assert_eq!(dialog.local_tag, "from-tag");
+        assert!(dialog.remote_tag.is_none());
+
+        // Simulate 200 OK
+        let response = SipMessage::create_response(
+            200,
+            "OK",
+            &invite,
+            Some(crate::message::ResponseOptions {
+                to_tag: Some("remote-tag".to_string()),
+                contact: Some("<sip:callee@10.0.0.1:5060>".to_string()),
+                ..Default::default()
+            }),
+        );
+
+        dialog.process_response(&response);
+        assert_eq!(dialog.state, DialogState::Confirmed);
+        assert_eq!(dialog.remote_tag.as_deref(), Some("remote-tag"));
+        assert_eq!(dialog.remote_target, "sip:callee@10.0.0.1:5060");
+    }
+
+    #[test]
+    fn create_bye() {
+        let invite = make_invite();
+        let mut dialog = SipDialog::from_uac_invite(&invite, "192.168.1.1", 5070);
+        dialog.remote_tag = Some("remote-tag".to_string());
+        dialog.state = DialogState::Confirmed;
+
+        let bye = dialog.create_request("BYE", None, None, None);
+        assert_eq!(bye.method(), Some("BYE"));
+        assert_eq!(bye.call_id(), "test-dialog-call");
+        assert_eq!(dialog.local_cseq, 2);
+        let to = bye.get_header("To").unwrap();
+        assert!(to.contains("tag=remote-tag"));
+    }
+
+    #[test]
+    fn create_ack() {
+        let invite = make_invite();
+        let mut dialog = SipDialog::from_uac_invite(&invite, "192.168.1.1", 5070);
+        dialog.remote_tag = Some("remote-tag".to_string());
+
+        let ack = dialog.create_ack();
+        assert_eq!(ack.method(), Some("ACK"));
+        assert!(ack.get_header("CSeq").unwrap().contains("ACK"));
+    }
+
+    #[test]
+    fn create_cancel() {
+        let invite = make_invite();
+        let dialog = SipDialog::from_uac_invite(&invite, "192.168.1.1", 5070);
+
+        let cancel = dialog.create_cancel(&invite);
+        assert_eq!(cancel.method(), Some("CANCEL"));
+        assert!(cancel.get_header("CSeq").unwrap().contains("CANCEL"));
+        assert!(cancel.start_line.contains("sip:callee@host"));
+    }
+}

rust/crates/sip-proto/src/helpers.rs

@@ -0,0 +1,331 @@
+//! SIP helper utilities — ID generation, codec registry, SDP builder,
+//! Digest authentication, SDP parser, and MWI body builder.
+
+use md5::{Digest, Md5};
+use rand::Rng;
+
+// ---- ID generators ---------------------------------------------------------
+
+/// Generate a random SIP Call-ID (32 hex chars).
+pub fn generate_call_id(domain: Option<&str>) -> String {
+    let id = random_hex(16);
+    match domain {
+        Some(d) => format!("{id}@{d}"),
+        None => id,
+    }
+}
+
+/// Generate a random SIP From/To tag (16 hex chars).
+pub fn generate_tag() -> String {
+    random_hex(8)
+}
+
+/// Generate an RFC 3261 compliant Via branch (starts with `z9hG4bK` magic cookie).
+pub fn generate_branch() -> String {
+    format!("z9hG4bK-{}", random_hex(8))
+}
+
+fn random_hex(bytes: usize) -> String {
+    let mut rng = rand::thread_rng();
+    (0..bytes).map(|_| format!("{:02x}", rng.gen::<u8>())).collect()
+}
+
+// ---- Codec registry --------------------------------------------------------
+
+/// Look up the rtpmap name for a static payload type.
+pub fn codec_name(pt: u8) -> &'static str {
+    match pt {
+        0 => "PCMU/8000",
+        3 => "GSM/8000",
+        4 => "G723/8000",
+        8 => "PCMA/8000",
+        9 => "G722/8000",
+        18 => "G729/8000",
+        101 => "telephone-event/8000",
+        _ => "unknown",
+    }
+}
+
+// ---- SDP builder -----------------------------------------------------------
+
+/// Options for building an SDP body.
+pub struct SdpOptions<'a> {
+    pub ip: &'a str,
+    pub port: u16,
+    pub payload_types: &'a [u8],
+    pub session_id: Option<&'a str>,
+    pub session_name: Option<&'a str>,
+    pub direction: Option<&'a str>,
+    pub attributes: &'a [&'a str],
+}
+
+impl<'a> Default for SdpOptions<'a> {
+    fn default() -> Self {
+        Self {
+            ip: "0.0.0.0",
+            port: 0,
+            payload_types: &[9, 0, 8, 101],
+            session_id: None,
+            session_name: None,
+            direction: None,
+            attributes: &[],
+        }
+    }
+}
+
+/// Build a minimal SDP body suitable for SIP INVITE offers/answers.
+pub fn build_sdp(opts: &SdpOptions) -> String {
+    let session_id = opts
+        .session_id
+        .map(|s| s.to_string())
+        .unwrap_or_else(|| format!("{}", rand::thread_rng().gen_range(0..1_000_000_000u64)));
+    let session_name = opts.session_name.unwrap_or("-");
+    let direction = opts.direction.unwrap_or("sendrecv");
+    let pts: Vec<String> = opts.payload_types.iter().map(|pt| pt.to_string()).collect();
+
+    let mut lines = vec![
+        "v=0".to_string(),
+        format!("o=- {session_id} {session_id} IN IP4 {}", opts.ip),
+        format!("s={session_name}"),
+        format!("c=IN IP4 {}", opts.ip),
+        "t=0 0".to_string(),
+        format!("m=audio {} RTP/AVP {}", opts.port, pts.join(" ")),
+    ];
+
+    for &pt in opts.payload_types {
+        let name = codec_name(pt);
+        if name != "unknown" {
+            lines.push(format!("a=rtpmap:{pt} {name}"));
+        }
+        if pt == 101 {
+            lines.push("a=fmtp:101 0-16".to_string());
+        }
+    }
+
+    lines.push(format!("a={direction}"));
+    for attr in opts.attributes {
+        lines.push(format!("a={attr}"));
+    }
+    lines.push(String::new()); // trailing CRLF
+
+    lines.join("\r\n")
+}
+
+// ---- SIP Digest authentication (RFC 2617) ----------------------------------
+
+/// Parsed fields from a Proxy-Authenticate or WWW-Authenticate header.
+#[derive(Debug, Clone)]
+pub struct DigestChallenge {
+    pub realm: String,
+    pub nonce: String,
+    pub algorithm: Option<String>,
+    pub opaque: Option<String>,
+    pub qop: Option<String>,
+}
+
+/// Parse a `Proxy-Authenticate` or `WWW-Authenticate` header value.
+pub fn parse_digest_challenge(header: &str) -> Option<DigestChallenge> {
+    let lower = header.to_ascii_lowercase();
+    if !lower.starts_with("digest ") {
+        return None;
+    }
+    let params = &header[7..];
+
+    let get = |key: &str| -> Option<String> {
+        // Try quoted value first.
+        let pat = format!("{}=", key);
+        if let Some(pos) = params.to_ascii_lowercase().find(&pat) {
+            let after = &params[pos + pat.len()..];
+            let after = after.trim_start();
+            if after.starts_with('"') {
+                let end = after[1..].find('"')?;
+                return Some(after[1..1 + end].to_string());
+            }
+            // Unquoted value.
+            let end = after.find(|c: char| c == ',' || c.is_whitespace()).unwrap_or(after.len());
+            return Some(after[..end].to_string());
+        }
+        None
+    };
+
+    let realm = get("realm")?;
+    let nonce = get("nonce")?;
+
+    Some(DigestChallenge {
+        realm,
+        nonce,
+        algorithm: get("algorithm"),
+        opaque: get("opaque"),
+        qop: get("qop"),
+    })
+}
+
+fn md5_hex(s: &str) -> String {
+    let mut hasher = Md5::new();
+    hasher.update(s.as_bytes());
+    format!("{:x}", hasher.finalize())
+}
+
+/// Compute a SIP Digest Authorization header value.
+pub fn compute_digest_auth(
+    username: &str,
+    password: &str,
+    realm: &str,
+    nonce: &str,
+    method: &str,
+    uri: &str,
+    algorithm: Option<&str>,
+    opaque: Option<&str>,
+) -> String {
+    let ha1 = md5_hex(&format!("{username}:{realm}:{password}"));
+    let ha2 = md5_hex(&format!("{method}:{uri}"));
+    let response = md5_hex(&format!("{ha1}:{nonce}:{ha2}"));
+    let alg = algorithm.unwrap_or("MD5");
+
+    let mut header = format!(
+        "Digest username=\"{username}\", realm=\"{realm}\", \
+         nonce=\"{nonce}\", uri=\"{uri}\", response=\"{response}\", \
+         algorithm={alg}"
+    );
+    if let Some(op) = opaque {
+        header.push_str(&format!(", opaque=\"{op}\""));
+    }
+    header
+}
+
+// ---- SDP parser ------------------------------------------------------------
+
+use crate::Endpoint;
+
+/// Parse the audio media port and connection address from an SDP body.
+pub fn parse_sdp_endpoint(sdp: &str) -> Option<Endpoint> {
+    let mut addr: Option<&str> = None;
+    let mut port: Option<u16> = None;
+
+    let normalized = sdp.replace("\r\n", "\n");
+    for raw in normalized.split('\n') {
+        let line = raw.trim();
+        if let Some(rest) = line.strip_prefix("c=IN IP4 ") {
+            addr = Some(rest.trim());
+        } else if let Some(rest) = line.strip_prefix("m=audio ") {
+            let parts: Vec<&str> = rest.split_whitespace().collect();
+            if !parts.is_empty() {
+                port = parts[0].parse().ok();
+            }
+        }
+    }
+
+    match (addr, port) {
+        (Some(a), Some(p)) => Some(Endpoint {
+            address: a.to_string(),
+            port: p,
+        }),
+        _ => None,
+    }
+}
+
+// ---- MWI (RFC 3842) --------------------------------------------------------
+
+/// Build the body and extra headers for an MWI NOTIFY (RFC 3842 message-summary).
+pub struct MwiResult {
+    pub body: String,
+    pub content_type: &'static str,
+    pub extra_headers: Vec<(String, String)>,
+}
+
+pub fn build_mwi_body(
+    new_messages: u32,
+    old_messages: u32,
+    account_uri: &str,
+) -> MwiResult {
+    let waiting = if new_messages > 0 { "yes" } else { "no" };
+    let body = format!(
+        "Messages-Waiting: {waiting}\r\n\
+         Message-Account: {account_uri}\r\n\
+         Voice-Message: {new_messages}/{old_messages}\r\n"
+    );
+
+    MwiResult {
+        body,
+        content_type: "application/simple-message-summary",
+        extra_headers: vec![
+            ("Event".to_string(), "message-summary".to_string()),
+            (
+                "Subscription-State".to_string(),
+                "terminated;reason=noresource".to_string(),
+            ),
+        ],
+    }
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    #[test]
+    fn test_generate_branch_has_magic_cookie() {
+        let branch = generate_branch();
+        assert!(branch.starts_with("z9hG4bK-"));
+        assert!(branch.len() > 8);
+    }
+
+    #[test]
+    fn test_codec_name() {
+        assert_eq!(codec_name(0), "PCMU/8000");
+        assert_eq!(codec_name(9), "G722/8000");
+        assert_eq!(codec_name(101), "telephone-event/8000");
+        assert_eq!(codec_name(255), "unknown");
+    }
+
+    #[test]
+    fn test_build_sdp() {
+        let sdp = build_sdp(&SdpOptions {
+            ip: "192.168.1.1",
+            port: 20000,
+            payload_types: &[9, 0, 101],
+            ..Default::default()
+        });
+        assert!(sdp.contains("m=audio 20000 RTP/AVP 9 0 101"));
+        assert!(sdp.contains("c=IN IP4 192.168.1.1"));
+        assert!(sdp.contains("a=rtpmap:9 G722/8000"));
+        assert!(sdp.contains("a=fmtp:101 0-16"));
+        assert!(sdp.contains("a=sendrecv"));
+    }
+
+    #[test]
+    fn test_parse_digest_challenge() {
+        let header = r#"Digest realm="asterisk", nonce="abc123", algorithm=MD5, opaque="xyz""#;
+        let ch = parse_digest_challenge(header).unwrap();
+        assert_eq!(ch.realm, "asterisk");
+        assert_eq!(ch.nonce, "abc123");
+        assert_eq!(ch.algorithm.as_deref(), Some("MD5"));
+        assert_eq!(ch.opaque.as_deref(), Some("xyz"));
+    }
+
+    #[test]
+    fn test_compute_digest_auth() {
+        let auth = compute_digest_auth(
+            "user", "pass", "realm", "nonce", "REGISTER", "sip:host", None, None,
+        );
+        assert!(auth.starts_with("Digest "));
+        assert!(auth.contains("username=\"user\""));
+        assert!(auth.contains("realm=\"realm\""));
+        assert!(auth.contains("response=\""));
+    }
+
+    #[test]
+    fn test_parse_sdp_endpoint() {
+        let sdp = "v=0\r\nc=IN IP4 10.0.0.1\r\nm=audio 5060 RTP/AVP 0\r\n";
+        let ep = parse_sdp_endpoint(sdp).unwrap();
+        assert_eq!(ep.address, "10.0.0.1");
+        assert_eq!(ep.port, 5060);
+    }
+
+    #[test]
+    fn test_build_mwi_body() {
+        let mwi = build_mwi_body(3, 5, "sip:user@host");
+        assert!(mwi.body.contains("Messages-Waiting: yes"));
+        assert!(mwi.body.contains("Voice-Message: 3/5"));
+        assert_eq!(mwi.content_type, "application/simple-message-summary");
+    }
+}

rust/crates/sip-proto/src/lib.rs

@@ -0,0 +1,17 @@
+//! SIP protocol library for the proxy engine.
+//!
+//! Provides SIP message parsing/serialization, dialog state management,
+//! SDP handling, Digest authentication, and URI rewriting.
+//! Ported from the TypeScript `ts/sip/` library.
+
+pub mod message;
+pub mod dialog;
+pub mod helpers;
+pub mod rewrite;
+
+/// Network endpoint (address + port).
+#[derive(Debug, Clone, PartialEq, Eq)]
+pub struct Endpoint {
+    pub address: String,
+    pub port: u16,
+}

rust/crates/sip-proto/src/message.rs

@@ -0,0 +1,563 @@
+//! SIP message parsing, serialization, inspection, mutation, and factory methods.
+//!
+//! Ported from ts/sip/message.ts.
+
+use crate::helpers::{generate_branch, generate_call_id, generate_tag};
+
+/// A parsed SIP message (request or response).
+#[derive(Debug, Clone)]
+pub struct SipMessage {
+    pub start_line: String,
+    pub headers: Vec<(String, String)>,
+    pub body: String,
+}
+
+impl SipMessage {
+    pub fn new(start_line: String, headers: Vec<(String, String)>, body: String) -> Self {
+        Self { start_line, headers, body }
+    }
+
+    // ---- Parsing -----------------------------------------------------------
+
+    /// Parse a raw buffer into a SipMessage. Returns None for invalid data.
+    pub fn parse(buf: &[u8]) -> Option<Self> {
+        if buf.is_empty() {
+            return None;
+        }
+        // First byte must be ASCII A-z.
+        if buf[0] < 0x41 || buf[0] > 0x7a {
+            return None;
+        }
+
+        let text = std::str::from_utf8(buf).ok()?;
+
+        let (head, body) = if let Some(sep) = text.find("\r\n\r\n") {
+            (&text[..sep], &text[sep + 4..])
+        } else if let Some(sep) = text.find("\n\n") {
+            (&text[..sep], &text[sep + 2..])
+        } else {
+            (text, "")
+        };
+
+        let normalized = head.replace("\r\n", "\n");
+        let lines: Vec<&str> = normalized.split('\n').collect();
+        if lines.is_empty() || lines[0].is_empty() {
+            return None;
+        }
+
+        let start_line = lines[0];
+        // Validate: must be a SIP request or response start line.
+        if !is_sip_first_line(start_line) {
+            return None;
+        }
+
+        let mut headers = Vec::new();
+        for &line in &lines[1..] {
+            let trimmed = line.trim();
+            if trimmed.is_empty() {
+                continue;
+            }
+            if let Some(colon) = line.find(':') {
+                let name = line[..colon].trim().to_string();
+                let value = line[colon + 1..].trim().to_string();
+                headers.push((name, value));
+            }
+        }
+
+        Some(SipMessage {
+            start_line: start_line.to_string(),
+            headers,
+            body: body.to_string(),
+        })
+    }
+
+    // ---- Serialization -----------------------------------------------------
+
+    /// Serialize the message to a byte buffer suitable for UDP transmission.
+    pub fn serialize(&self) -> Vec<u8> {
+        let mut head = self.start_line.clone();
+        for (name, value) in &self.headers {
+            head.push_str("\r\n");
+            head.push_str(name);
+            head.push_str(": ");
+            head.push_str(value);
+        }
+        head.push_str("\r\n\r\n");
+
+        let mut buf = head.into_bytes();
+        if !self.body.is_empty() {
+            buf.extend_from_slice(self.body.as_bytes());
+        }
+        buf
+    }
+
+    // ---- Inspectors --------------------------------------------------------
+
+    pub fn is_request(&self) -> bool {
+        !self.start_line.starts_with("SIP/")
+    }
+
+    pub fn is_response(&self) -> bool {
+        self.start_line.starts_with("SIP/")
+    }
+
+    /// Request method (INVITE, REGISTER, ...) or None for responses.
+    pub fn method(&self) -> Option<&str> {
+        if !self.is_request() {
+            return None;
+        }
+        self.start_line.split_whitespace().next()
+    }
+
+    /// Response status code or None for requests.
+    pub fn status_code(&self) -> Option<u16> {
+        if !self.is_response() {
+            return None;
+        }
+        self.start_line
+            .split_whitespace()
+            .nth(1)
+            .and_then(|s| s.parse().ok())
+    }
+
+    pub fn call_id(&self) -> &str {
+        self.get_header("Call-ID").unwrap_or("noid")
+    }
+
+    /// Method from the CSeq header (e.g. "INVITE").
+    pub fn cseq_method(&self) -> Option<&str> {
+        let cseq = self.get_header("CSeq")?;
+        cseq.split_whitespace().nth(1)
+    }
+
+    /// True for INVITE, SUBSCRIBE, REFER, NOTIFY, UPDATE.
+    pub fn is_dialog_establishing(&self) -> bool {
+        matches!(
+            self.method(),
+            Some("INVITE" | "SUBSCRIBE" | "REFER" | "NOTIFY" | "UPDATE")
+        )
+    }
+
+    /// True when the body carries an SDP payload.
+    pub fn has_sdp_body(&self) -> bool {
+        if self.body.is_empty() {
+            return false;
+        }
+        let ct = self.get_header("Content-Type").unwrap_or("");
+        ct.to_ascii_lowercase().starts_with("application/sdp")
+    }
+
+    // ---- Header accessors --------------------------------------------------
+
+    /// Get the first header value matching `name` (case-insensitive).
+    pub fn get_header(&self, name: &str) -> Option<&str> {
+        let nl = name.to_ascii_lowercase();
+        for (n, v) in &self.headers {
+            if n.to_ascii_lowercase() == nl {
+                return Some(v.as_str());
+            }
+        }
+        None
+    }
+
+    /// Overwrites the first header with the given name, or appends it.
+    pub fn set_header(&mut self, name: &str, value: &str) -> &mut Self {
+        let nl = name.to_ascii_lowercase();
+        for h in &mut self.headers {
+            if h.0.to_ascii_lowercase() == nl {
+                h.1 = value.to_string();
+                return self;
+            }
+        }
+        self.headers.push((name.to_string(), value.to_string()));
+        self
+    }
+
+    /// Inserts a header at the top of the header list.
+    pub fn prepend_header(&mut self, name: &str, value: &str) -> &mut Self {
+        self.headers.insert(0, (name.to_string(), value.to_string()));
+        self
+    }
+
+    /// Removes all headers with the given name.
+    pub fn remove_header(&mut self, name: &str) -> &mut Self {
+        let nl = name.to_ascii_lowercase();
+        self.headers.retain(|(n, _)| n.to_ascii_lowercase() != nl);
+        self
+    }
+
+    /// Recalculates Content-Length to match the current body.
+    pub fn update_content_length(&mut self) -> &mut Self {
+        let len = self.body.len();
+        self.set_header("Content-Length", &len.to_string())
+    }
+
+    // ---- Start-line mutation -----------------------------------------------
+
+    /// Replace the Request-URI (second token) of a request start line.
+    pub fn set_request_uri(&mut self, uri: &str) -> &mut Self {
+        if !self.is_request() {
+            return self;
+        }
+        let parts: Vec<&str> = self.start_line.splitn(3, ' ').collect();
+        if parts.len() >= 3 {
+            self.start_line = format!("{} {} {}", parts[0], uri, parts[2]);
+        }
+        self
+    }
+
+    /// Returns the Request-URI (second token) of a request start line.
+    pub fn request_uri(&self) -> Option<&str> {
+        if !self.is_request() {
+            return None;
+        }
+        self.start_line.split_whitespace().nth(1)
+    }
+
+    // ---- Factory methods ---------------------------------------------------
+
+    /// Build a new SIP request.
+    pub fn create_request(method: &str, request_uri: &str, opts: RequestOptions) -> Self {
+        let branch = opts.via_branch.unwrap_or_else(|| generate_branch());
+        let transport = opts.via_transport.unwrap_or_else(|| "UDP".to_string());
+        let from_tag = opts.from_tag.unwrap_or_else(|| generate_tag());
+        let call_id = opts.call_id.unwrap_or_else(|| generate_call_id(None));
+        let cseq = opts.cseq.unwrap_or(1);
+        let max_forwards = opts.max_forwards.unwrap_or(70);
+
+        let from_display = opts
+            .from_display_name
+            .map(|d| format!("\"{d}\" "))
+            .unwrap_or_default();
+        let to_display = opts
+            .to_display_name
+            .map(|d| format!("\"{d}\" "))
+            .unwrap_or_default();
+        let to_tag_str = opts
+            .to_tag
+            .map(|t| format!(";tag={t}"))
+            .unwrap_or_default();
+
+        let mut headers = vec![
+            (
+                "Via".to_string(),
+                format!(
+                    "SIP/2.0/{transport} {}:{};branch={branch};rport",
+                    opts.via_host, opts.via_port
+                ),
+            ),
+            (
+                "From".to_string(),
+                format!("{from_display}<{}>;tag={from_tag}", opts.from_uri),
+            ),
+            (
+                "To".to_string(),
+                format!("{to_display}<{}>{to_tag_str}", opts.to_uri),
+            ),
+            ("Call-ID".to_string(), call_id),
+            ("CSeq".to_string(), format!("{cseq} {method}")),
+            ("Max-Forwards".to_string(), max_forwards.to_string()),
+        ];
+
+        if let Some(contact) = &opts.contact {
+            headers.push(("Contact".to_string(), contact.clone()));
+        }
+
+        if let Some(extra) = opts.extra_headers {
+            headers.extend(extra);
+        }
+
+        let body = opts.body.unwrap_or_default();
+        if !body.is_empty() {
+            if let Some(ct) = &opts.content_type {
+                headers.push(("Content-Type".to_string(), ct.clone()));
+            }
+        }
+        headers.push(("Content-Length".to_string(), body.len().to_string()));
+
+        SipMessage {
+            start_line: format!("{method} {request_uri} SIP/2.0"),
+            headers,
+            body,
+        }
+    }
+
+    /// Build a SIP response to an incoming request.
+    /// Copies Via, From, To, Call-ID, and CSeq from the original request.
+    pub fn create_response(
+        status_code: u16,
+        reason_phrase: &str,
+        request: &SipMessage,
+        opts: Option<ResponseOptions>,
+    ) -> Self {
+        let opts = opts.unwrap_or_default();
+        let mut headers: Vec<(String, String)> = Vec::new();
+
+        // Copy all Via headers (order matters).
+        for (n, v) in &request.headers {
+            if n.to_ascii_lowercase() == "via" {
+                headers.push(("Via".to_string(), v.clone()));
+            }
+        }
+
+        // From — copied verbatim.
+        if let Some(from) = request.get_header("From") {
+            headers.push(("From".to_string(), from.to_string()));
+        }
+
+        // To — add tag if provided and not already present.
+        let mut to = request.get_header("To").unwrap_or("").to_string();
+        if let Some(tag) = &opts.to_tag {
+            if !to.contains("tag=") {
+                to.push_str(&format!(";tag={tag}"));
+            }
+        }
+        headers.push(("To".to_string(), to));
+
+        headers.push(("Call-ID".to_string(), request.call_id().to_string()));
+
+        if let Some(cseq) = request.get_header("CSeq") {
+            headers.push(("CSeq".to_string(), cseq.to_string()));
+        }
+
+        if let Some(contact) = &opts.contact {
+            headers.push(("Contact".to_string(), contact.clone()));
+        }
+
+        if let Some(extra) = opts.extra_headers {
+            headers.extend(extra);
+        }
+
+        let body = opts.body.unwrap_or_default();
+        if !body.is_empty() {
+            if let Some(ct) = &opts.content_type {
+                headers.push(("Content-Type".to_string(), ct.clone()));
+            }
+        }
+        headers.push(("Content-Length".to_string(), body.len().to_string()));
+
+        SipMessage {
+            start_line: format!("SIP/2.0 {status_code} {reason_phrase}"),
+            headers,
+            body,
+        }
+    }
+
+    /// Extract the tag from a From or To header value.
+    pub fn extract_tag(header_value: &str) -> Option<&str> {
+        let idx = header_value.find(";tag=")?;
+        let rest = &header_value[idx + 5..];
+        let end = rest
+            .find(|c: char| c.is_whitespace() || c == ';' || c == '>')
+            .unwrap_or(rest.len());
+        Some(&rest[..end])
+    }
+
+    /// Extract the URI from an addr-spec or name-addr (From/To/Contact).
+    pub fn extract_uri(header_value: &str) -> Option<&str> {
+        if let Some(start) = header_value.find('<') {
+            let end = header_value[start..].find('>')?;
+            Some(&header_value[start + 1..start + end])
+        } else {
+            let trimmed = header_value.trim();
+            let end = trimmed
+                .find(|c: char| c == ';' || c == '>')
+                .unwrap_or(trimmed.len());
+            let result = &trimmed[..end];
+            if result.is_empty() { None } else { Some(result) }
+        }
+    }
+}
+
+/// Options for `SipMessage::create_request`.
+pub struct RequestOptions {
+    pub via_host: String,
+    pub via_port: u16,
+    pub via_transport: Option<String>,
+    pub via_branch: Option<String>,
+    pub from_uri: String,
+    pub from_display_name: Option<String>,
+    pub from_tag: Option<String>,
+    pub to_uri: String,
+    pub to_display_name: Option<String>,
+    pub to_tag: Option<String>,
+    pub call_id: Option<String>,
+    pub cseq: Option<u32>,
+    pub contact: Option<String>,
+    pub max_forwards: Option<u16>,
+    pub body: Option<String>,
+    pub content_type: Option<String>,
+    pub extra_headers: Option<Vec<(String, String)>>,
+}
+
+/// Options for `SipMessage::create_response`.
+#[derive(Default)]
+pub struct ResponseOptions {
+    pub to_tag: Option<String>,
+    pub contact: Option<String>,
+    pub body: Option<String>,
+    pub content_type: Option<String>,
+    pub extra_headers: Option<Vec<(String, String)>>,
+}
+
+/// Check if a string matches the SIP first-line pattern.
+fn is_sip_first_line(line: &str) -> bool {
+    // Request: METHOD SP URI SP SIP/X.Y
+    // Response: SIP/X.Y SP STATUS SP REASON
+    if line.starts_with("SIP/") {
+        // Response: SIP/2.0 200 OK
+        let parts: Vec<&str> = line.splitn(3, ' ').collect();
+        if parts.len() >= 2 {
+            return parts[1].chars().all(|c| c.is_ascii_digit());
+        }
+    } else {
+        // Request: INVITE sip:user@host SIP/2.0
+        let parts: Vec<&str> = line.splitn(3, ' ').collect();
+        if parts.len() >= 3 {
+            return parts[0].chars().all(|c| c.is_ascii_uppercase())
+                && parts[2].starts_with("SIP/");
+        }
+    }
+    false
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    const INVITE_RAW: &str = "INVITE sip:user@host SIP/2.0\r\n\
+        Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-test\r\n\
+        From: <sip:caller@host>;tag=abc\r\n\
+        To: <sip:user@host>\r\n\
+        Call-ID: test-call-id\r\n\
+        CSeq: 1 INVITE\r\n\
+        Content-Length: 0\r\n\r\n";
+
+    #[test]
+    fn parse_invite() {
+        let msg = SipMessage::parse(INVITE_RAW.as_bytes()).unwrap();
+        assert!(msg.is_request());
+        assert!(!msg.is_response());
+        assert_eq!(msg.method(), Some("INVITE"));
+        assert_eq!(msg.call_id(), "test-call-id");
+        assert_eq!(msg.cseq_method(), Some("INVITE"));
+        assert!(msg.is_dialog_establishing());
+        assert_eq!(msg.request_uri(), Some("sip:user@host"));
+    }
+
+    #[test]
+    fn parse_response() {
+        let raw = "SIP/2.0 200 OK\r\n\
+            Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-test\r\n\
+            From: <sip:caller@host>;tag=abc\r\n\
+            To: <sip:user@host>;tag=def\r\n\
+            Call-ID: test-call-id\r\n\
+            CSeq: 1 INVITE\r\n\
+            Content-Length: 0\r\n\r\n";
+        let msg = SipMessage::parse(raw.as_bytes()).unwrap();
+        assert!(msg.is_response());
+        assert_eq!(msg.status_code(), Some(200));
+        assert_eq!(msg.cseq_method(), Some("INVITE"));
+    }
+
+    #[test]
+    fn serialize_roundtrip() {
+        let msg = SipMessage::parse(INVITE_RAW.as_bytes()).unwrap();
+        let serialized = msg.serialize();
+        let reparsed = SipMessage::parse(&serialized).unwrap();
+        assert_eq!(reparsed.call_id(), "test-call-id");
+        assert_eq!(reparsed.method(), Some("INVITE"));
+        assert_eq!(reparsed.headers.len(), msg.headers.len());
+    }
+
+    #[test]
+    fn header_mutation() {
+        let mut msg = SipMessage::parse(INVITE_RAW.as_bytes()).unwrap();
+        msg.set_header("X-Custom", "value1");
+        assert_eq!(msg.get_header("X-Custom"), Some("value1"));
+        msg.set_header("X-Custom", "value2");
+        assert_eq!(msg.get_header("X-Custom"), Some("value2"));
+        msg.prepend_header("X-First", "first");
+        assert_eq!(msg.headers[0].0, "X-First");
+        msg.remove_header("X-Custom");
+        assert_eq!(msg.get_header("X-Custom"), None);
+    }
+
+    #[test]
+    fn set_request_uri() {
+        let mut msg = SipMessage::parse(INVITE_RAW.as_bytes()).unwrap();
+        msg.set_request_uri("sip:new@host");
+        assert_eq!(msg.request_uri(), Some("sip:new@host"));
+        assert!(msg.start_line.starts_with("INVITE sip:new@host SIP/2.0"));
+    }
+
+    #[test]
+    fn extract_tag_and_uri() {
+        assert_eq!(
+            SipMessage::extract_tag("<sip:user@host>;tag=abc123"),
+            Some("abc123")
+        );
+        assert_eq!(SipMessage::extract_tag("<sip:user@host>"), None);
+        assert_eq!(
+            SipMessage::extract_uri("<sip:user@host>"),
+            Some("sip:user@host")
+        );
+        assert_eq!(
+            SipMessage::extract_uri("\"Name\" <sip:user@host>;tag=abc"),
+            Some("sip:user@host")
+        );
+    }
+
+    #[test]
+    fn create_request_and_response() {
+        let invite = SipMessage::create_request(
+            "INVITE",
+            "sip:user@host",
+            RequestOptions {
+                via_host: "192.168.1.1".to_string(),
+                via_port: 5070,
+                via_transport: None,
+                via_branch: None,
+                from_uri: "sip:caller@proxy".to_string(),
+                from_display_name: None,
+                from_tag: Some("mytag".to_string()),
+                to_uri: "sip:user@host".to_string(),
+                to_display_name: None,
+                to_tag: None,
+                call_id: Some("test-123".to_string()),
+                cseq: Some(1),
+                contact: Some("<sip:caller@192.168.1.1:5070>".to_string()),
+                max_forwards: None,
+                body: None,
+                content_type: None,
+                extra_headers: None,
+            },
+        );
+        assert_eq!(invite.method(), Some("INVITE"));
+        assert_eq!(invite.call_id(), "test-123");
+        assert!(invite.get_header("Via").unwrap().contains("192.168.1.1:5070"));
+
+        let response = SipMessage::create_response(
+            200,
+            "OK",
+            &invite,
+            Some(ResponseOptions {
+                to_tag: Some("remotetag".to_string()),
+                ..Default::default()
+            }),
+        );
+        assert!(response.is_response());
+        assert_eq!(response.status_code(), Some(200));
+        let to = response.get_header("To").unwrap();
+        assert!(to.contains("tag=remotetag"));
+    }
+
+    #[test]
+    fn has_sdp_body() {
+        let mut msg = SipMessage::parse(INVITE_RAW.as_bytes()).unwrap();
+        assert!(!msg.has_sdp_body());
+        msg.body = "v=0\r\no=- 1 1 IN IP4 0.0.0.0\r\n".to_string();
+        msg.set_header("Content-Type", "application/sdp");
+        assert!(msg.has_sdp_body());
+    }
+}

rust/crates/sip-proto/src/rewrite.rs

@@ -0,0 +1,130 @@
+//! SIP URI and SDP body rewriting helpers.
+//!
+//! Ported from ts/sip/rewrite.ts.
+
+use crate::Endpoint;
+
+/// Replaces the host:port in every `sip:` / `sips:` URI found in `value`.
+pub fn rewrite_sip_uri(value: &str, host: &str, port: u16) -> String {
+    let mut result = String::with_capacity(value.len());
+    let mut i = 0;
+    let bytes = value.as_bytes();
+
+    while i < bytes.len() {
+        // Look for "sip:" or "sips:"
+        let scheme_len = if i + 4 <= bytes.len()
+            && (bytes[i..].starts_with(b"sip:") || bytes[i..].starts_with(b"SIP:"))
+        {
+            4
+        } else if i + 5 <= bytes.len()
+            && (bytes[i..].starts_with(b"sips:") || bytes[i..].starts_with(b"SIPS:"))
+        {
+            5
+        } else {
+            result.push(value[i..].chars().next().unwrap());
+            i += value[i..].chars().next().unwrap().len_utf8();
+            continue;
+        };
+
+        let scheme = &value[i..i + scheme_len];
+        let rest = &value[i + scheme_len..];
+
+        // Check for userpart (contains '@')
+        let (userpart, host_start) = if let Some(at) = rest.find('@') {
+            // Make sure @ comes before any delimiters
+            let delim = rest.find(|c: char| c == '>' || c == ';' || c == ',' || c.is_whitespace());
+            if delim.is_none() || at < delim.unwrap() {
+                (&rest[..=at], at + 1)
+            } else {
+                ("", 0)
+            }
+        } else {
+            ("", 0)
+        };
+
+        // Find the end of the host:port portion
+        let host_rest = &rest[host_start..];
+        let end = host_rest
+            .find(|c: char| c == '>' || c == ';' || c == ',' || c.is_whitespace())
+            .unwrap_or(host_rest.len());
+
+        result.push_str(scheme);
+        result.push_str(userpart);
+        result.push_str(&format!("{host}:{port}"));
+        i += scheme_len + host_start + end;
+    }
+
+    result
+}
+
+/// Rewrites the connection address (`c=`) and audio media port (`m=audio`)
+/// in an SDP body. Returns the rewritten body together with the original
+/// endpoint that was replaced (if any).
+pub fn rewrite_sdp(body: &str, ip: &str, port: u16) -> (String, Option<Endpoint>) {
+    let mut orig_addr: Option<String> = None;
+    let mut orig_port: Option<u16> = None;
+
+    let lines: Vec<String> = body
+        .replace("\r\n", "\n")
+        .split('\n')
+        .map(|line| {
+            if let Some(rest) = line.strip_prefix("c=IN IP4 ") {
+                orig_addr = Some(rest.trim().to_string());
+                format!("c=IN IP4 {ip}")
+            } else if line.starts_with("m=audio ") {
+                let parts: Vec<&str> = line.split(' ').collect();
+                if parts.len() >= 2 {
+                    orig_port = parts[1].parse().ok();
+                    let mut rebuilt = parts[0].to_string();
+                    rebuilt.push(' ');
+                    rebuilt.push_str(&port.to_string());
+                    for part in &parts[2..] {
+                        rebuilt.push(' ');
+                        rebuilt.push_str(part);
+                    }
+                    return rebuilt;
+                }
+                line.to_string()
+            } else {
+                line.to_string()
+            }
+        })
+        .collect();
+
+    let original = match (orig_addr, orig_port) {
+        (Some(a), Some(p)) => Some(Endpoint { address: a, port: p }),
+        _ => None,
+    };
+
+    (lines.join("\r\n"), original)
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    #[test]
+    fn test_rewrite_sip_uri() {
+        let input = "<sip:user@10.0.0.1:5060>";
+        let result = rewrite_sip_uri(input, "192.168.1.1", 5070);
+        assert_eq!(result, "<sip:user@192.168.1.1:5070>");
+    }
+
+    #[test]
+    fn test_rewrite_sip_uri_no_port() {
+        let input = "sip:user@10.0.0.1";
+        let result = rewrite_sip_uri(input, "192.168.1.1", 5070);
+        assert_eq!(result, "sip:user@192.168.1.1:5070");
+    }
+
+    #[test]
+    fn test_rewrite_sdp() {
+        let sdp = "v=0\r\nc=IN IP4 10.0.0.1\r\nm=audio 5060 RTP/AVP 0 9\r\na=sendrecv\r\n";
+        let (rewritten, orig) = rewrite_sdp(sdp, "192.168.1.1", 20000);
+        assert!(rewritten.contains("c=IN IP4 192.168.1.1"));
+        assert!(rewritten.contains("m=audio 20000 RTP/AVP 0 9"));
+        let ep = orig.unwrap();
+        assert_eq!(ep.address, "10.0.0.1");
+        assert_eq!(ep.port, 5060);
+    }
+}

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.9.0',
+  version: '1.12.0',
   description: 'undefined'
 }

ts/announcement.ts

@@ -14,9 +14,26 @@ import { execSync } from 'node:child_process';
 import fs from 'node:fs';
 import path from 'node:path';
 import { Buffer } from 'node:buffer';
-import { buildRtpHeader, rtpClockIncrement } from './call/leg.ts';
 import { encodePcm, isCodecReady } from './opusbridge.ts';
 
+/** RTP clock increment per 20ms frame for each codec. */
+function rtpClockIncrement(pt: number): number {
+  if (pt === 111) return 960;
+  if (pt === 9) return 160;
+  return 160;
+}
+
+/** Build a fresh RTP header. */
+function buildRtpHeader(pt: number, seq: number, ts: number, ssrc: number, marker: boolean): Buffer {
+  const hdr = Buffer.alloc(12);
+  hdr[0] = 0x80;
+  hdr[1] = (marker ? 0x80 : 0) | (pt & 0x7f);
+  hdr.writeUInt16BE(seq & 0xffff, 2);
+  hdr.writeUInt32BE(ts >>> 0, 4);
+  hdr.writeUInt32BE(ssrc >>> 0, 8);
+  return hdr;
+}
+
 // ---------------------------------------------------------------------------
 // Types
 // ---------------------------------------------------------------------------

ts/call/call.ts

@@ -1,255 +0,0 @@
-/**
- * Call — the hub entity in the hub model.
- *
- * A Call owns N legs and bridges their media. For 2-party calls, RTP packets
- * from leg A are forwarded to leg B and vice versa. For N>2 party calls,
- * packets from each leg are forwarded to all other legs (fan-out).
- *
- * Transcoding is applied per-leg when codecs differ.
- */
-
-import { Buffer } from 'node:buffer';
-import type { ILeg } from './leg.ts';
-import type { TCallState, TCallDirection, ICallStatus } from './types.ts';
-import { RtpPortPool } from './rtp-port-pool.ts';
-import type { SipLeg } from './sip-leg.ts';
-
-export class Call {
-  readonly id: string;
-  state: TCallState = 'setting-up';
-  direction: TCallDirection;
-  readonly createdAt: number;
-
-  callerNumber: string | null = null;
-  calleeNumber: string | null = null;
-  providerUsed: string | null = null;
-
-  /** All legs in this call. */
-  private legs = new Map<string, ILeg>();
-
-  /** Codec payload type for the "native" audio in the call (usually the first SIP leg's codec). */
-  private nativeCodec: number | null = null;
-
-  /** Port pool reference for cleanup. */
-  private portPool: RtpPortPool;
-  private log: (msg: string) => void;
-  private onChange: ((call: Call) => void) | null = null;
-
-  constructor(options: {
-    id: string;
-    direction: TCallDirection;
-    portPool: RtpPortPool;
-    log: (msg: string) => void;
-    onChange?: (call: Call) => void;
-  }) {
-    this.id = options.id;
-    this.direction = options.direction;
-    this.createdAt = Date.now();
-    this.portPool = options.portPool;
-    this.log = options.log;
-    this.onChange = options.onChange ?? null;
-  }
-
-  // -------------------------------------------------------------------------
-  // Leg management
-  // -------------------------------------------------------------------------
-
-  /** Add a leg to this call and wire up media forwarding. */
-  addLeg(leg: ILeg): void {
-    this.legs.set(leg.id, leg);
-
-    // Wire up RTP forwarding: when this leg receives a packet, forward to all other legs.
-    leg.onRtpReceived = (data: Buffer) => {
-      this.forwardRtp(leg.id, data);
-    };
-
-    this.log(`[call:${this.id}] added leg ${leg.id} (${leg.type}), total=${this.legs.size}`);
-    this.updateState();
-  }
-
-  /** Remove a leg from this call, tear it down, and release its port. */
-  removeLeg(legId: string): void {
-    const leg = this.legs.get(legId);
-    if (!leg) return;
-
-    leg.onRtpReceived = null;
-    leg.teardown();
-    if (leg.rtpPort) {
-      this.portPool.release(leg.rtpPort);
-    }
-    this.legs.delete(legId);
-
-    this.log(`[call:${this.id}] removed leg ${legId}, total=${this.legs.size}`);
-    this.updateState();
-  }
-
-  getLeg(legId: string): ILeg | null {
-    return this.legs.get(legId) ?? null;
-  }
-
-  getLegs(): ILeg[] {
-    return [...this.legs.values()];
-  }
-
-  getLegByType(type: string): ILeg | null {
-    for (const leg of this.legs.values()) {
-      if (leg.type === type) return leg;
-    }
-    return null;
-  }
-
-  getLegBySipCallId(sipCallId: string): ILeg | null {
-    for (const leg of this.legs.values()) {
-      if (leg.sipCallId === sipCallId) return leg;
-    }
-    return null;
-  }
-
-  get legCount(): number {
-    return this.legs.size;
-  }
-
-  // -------------------------------------------------------------------------
-  // Media forwarding (the hub)
-  // -------------------------------------------------------------------------
-
-  private forwardRtp(fromLegId: string, data: Buffer): void {
-    for (const [id, leg] of this.legs) {
-      if (id === fromLegId) continue;
-      if (leg.state !== 'connected') continue;
-
-      // For WebRTC legs, sendRtp calls forwardToBrowser which handles transcoding internally.
-      // For SIP legs, forward the raw packet (same codec path) or let the leg handle it.
-      // The Call hub does NOT transcode — that's the leg's responsibility.
-      leg.sendRtp(data);
-    }
-  }
-
-  // -------------------------------------------------------------------------
-  // State management
-  // -------------------------------------------------------------------------
-
-  private updateState(): void {
-    if (this.state === 'terminated' || this.state === 'terminating') return;
-
-    const legs = [...this.legs.values()];
-    if (legs.length === 0) {
-      this.state = 'terminated';
-    } else if (legs.every((l) => l.state === 'terminated')) {
-      this.state = 'terminated';
-    } else if (legs.some((l) => l.state === 'connected') && legs.filter((l) => l.state !== 'terminated').length >= 2) {
-      this.state = 'connected';
-    } else if (legs.some((l) => l.state === 'ringing')) {
-      this.state = 'ringing';
-    } else {
-      this.state = 'setting-up';
-    }
-
-    this.onChange?.(this);
-  }
-
-  /** Notify the call that a leg's state has changed. */
-  notifyLegStateChange(_leg: ILeg): void {
-    this.updateState();
-  }
-
-  // -------------------------------------------------------------------------
-  // Hangup
-  // -------------------------------------------------------------------------
-
-  /** Tear down all legs and terminate the call. */
-  hangup(): void {
-    if (this.state === 'terminated' || this.state === 'terminating') return;
-    this.state = 'terminating';
-    this.log(`[call:${this.id}] hanging up (${this.legs.size} legs)`);
-
-    for (const [id, leg] of this.legs) {
-      // Send BYE/CANCEL for SIP legs.
-      if (leg.type === 'sip-device' || leg.type === 'sip-provider') {
-        (leg as SipLeg).sendHangup();
-      }
-      leg.teardown();
-      if (leg.rtpPort) {
-        this.portPool.release(leg.rtpPort);
-      }
-    }
-    this.legs.clear();
-
-    this.state = 'terminated';
-    this.onChange?.(this);
-  }
-
-  /**
-   * Handle a BYE from one leg — tear down the other legs.
-   * Called by CallManager when a SipLeg receives a BYE.
-   */
-  handleLegTerminated(terminatedLegId: string): void {
-    const terminatedLeg = this.legs.get(terminatedLegId);
-    if (!terminatedLeg) return;
-
-    // Remove the terminated leg.
-    terminatedLeg.onRtpReceived = null;
-    if (terminatedLeg.rtpPort) {
-      this.portPool.release(terminatedLeg.rtpPort);
-    }
-    this.legs.delete(terminatedLegId);
-
-    // If this is a 2-party call, hang up the other leg too.
-    if (this.legs.size <= 1) {
-      for (const [id, leg] of this.legs) {
-        if (leg.type === 'sip-device' || leg.type === 'sip-provider') {
-          (leg as SipLeg).sendHangup();
-        }
-        leg.teardown();
-        if (leg.rtpPort) {
-          this.portPool.release(leg.rtpPort);
-        }
-      }
-      this.legs.clear();
-      this.state = 'terminated';
-      this.log(`[call:${this.id}] terminated`);
-      this.onChange?.(this);
-    } else {
-      this.log(`[call:${this.id}] leg ${terminatedLegId} removed, ${this.legs.size} remaining`);
-      this.updateState();
-    }
-  }
-
-  // -------------------------------------------------------------------------
-  // Transfer
-  // -------------------------------------------------------------------------
-
-  /**
-   * Detach a leg from this call (without tearing it down).
-   * The leg can then be added to another call.
-   */
-  detachLeg(legId: string): ILeg | null {
-    const leg = this.legs.get(legId);
-    if (!leg) return null;
-
-    leg.onRtpReceived = null;
-    this.legs.delete(legId);
-
-    this.log(`[call:${this.id}] detached leg ${legId}`);
-    this.updateState();
-    return leg;
-  }
-
-  // -------------------------------------------------------------------------
-  // Status
-  // -------------------------------------------------------------------------
-
-  getStatus(): ICallStatus {
-    return {
-      id: this.id,
-      state: this.state,
-      direction: this.direction,
-      callerNumber: this.callerNumber,
-      calleeNumber: this.calleeNumber,
-      providerUsed: this.providerUsed,
-      createdAt: this.createdAt,
-      duration: Math.floor((Date.now() - this.createdAt) / 1000),
-      legs: [...this.legs.values()].map((l) => l.getStatus()),
-    };
-  }
-}

ts/call/index.ts

@@ -1,12 +0,0 @@
-export type { TCallState, TLegState, TLegType, TCallDirection, ICallStatus, ILegStatus, ICallHistoryEntry } from './types.ts';
-export type { ILeg } from './leg.ts';
-export { rtpClockIncrement, buildRtpHeader, codecDisplayName } from './leg.ts';
-export { RtpPortPool } from './rtp-port-pool.ts';
-export type { IRtpAllocation } from './rtp-port-pool.ts';
-export { SipLeg } from './sip-leg.ts';
-export type { ISipLegConfig } from './sip-leg.ts';
-export { WebRtcLeg } from './webrtc-leg.ts';
-export type { IWebRtcLegConfig } from './webrtc-leg.ts';
-export { Call } from './call.ts';
-export { CallManager } from './call-manager.ts';
-export type { ICallManagerConfig } from './call-manager.ts';

ts/call/leg.ts

@@ -1,104 +0,0 @@
-/**
- * ILeg interface — abstract connection from a Call hub to an endpoint.
- *
- * Concrete implementations: SipLeg (SIP devices + providers) and WebRtcLeg (browsers).
- * Shared RTP utilities (header building, clock rates) are also defined here.
- */
-
-import { Buffer } from 'node:buffer';
-import type dgram from 'node:dgram';
-import type { IEndpoint } from '../sip/index.ts';
-import type { TLegState, TLegType, ILegStatus } from './types.ts';
-import type { IRtpTranscoder } from '../codec.ts';
-import type { SipDialog } from '../sip/index.ts';
-import type { SipMessage } from '../sip/index.ts';
-
-// ---------------------------------------------------------------------------
-// ILeg interface
-// ---------------------------------------------------------------------------
-
-export interface ILeg {
-  readonly id: string;
-  readonly type: TLegType;
-  state: TLegState;
-
-  /** The SIP Call-ID used by this leg (for CallManager routing). */
-  readonly sipCallId: string;
-
-  /** Where this leg sends/receives RTP. */
-  readonly rtpPort: number | null;
-  readonly rtpSock: dgram.Socket | null;
-  remoteMedia: IEndpoint | null;
-
-  /** Negotiated codec payload type (e.g. 9 = G.722, 111 = Opus). */
-  codec: number | null;
-
-  /** Transcoder for converting to this leg's codec (set by Call when codecs differ). */
-  transcoder: IRtpTranscoder | null;
-
-  /** Packet counters. */
-  pktSent: number;
-  pktReceived: number;
-
-  /** SIP dialog (SipLegs only, null for WebRtcLegs). */
-  readonly dialog: SipDialog | null;
-
-  /**
-   * Send an RTP packet toward this leg's remote endpoint.
-   * If a transcoder is set, the Call should transcode before calling this.
-   */
-  sendRtp(data: Buffer): void;
-
-  /**
-   * Callback set by the owning Call — invoked when this leg receives an RTP packet.
-   * The Call uses this to forward to other legs.
-   */
-  onRtpReceived: ((data: Buffer) => void) | null;
-
-  /**
-   * Handle an incoming SIP message routed to this leg (SipLegs only).
-   * Returns a SipMessage response if one needs to be sent, or null.
-   */
-  handleSipMessage(msg: SipMessage, rinfo: IEndpoint): void;
-
-  /** Release all resources (sockets, peer connections, etc.). */
-  teardown(): void;
-
-  /** Status snapshot for the dashboard. */
-  getStatus(): ILegStatus;
-}
-
-// ---------------------------------------------------------------------------
-// Shared RTP utilities
-// ---------------------------------------------------------------------------
-
-/** RTP clock increment per 20ms frame for each codec. */
-export function rtpClockIncrement(pt: number): number {
-  if (pt === 111) return 960; // Opus: 48000 Hz x 0.02s
-  if (pt === 9) return 160;   // G.722: 8000 Hz x 0.02s (SDP clock rate quirk)
-  return 160;                  // PCMU/PCMA: 8000 Hz x 0.02s
-}
-
-/** Build a fresh RTP header with correct PT, timestamp, seq, SSRC. */
-export function buildRtpHeader(pt: number, seq: number, ts: number, ssrc: number, marker: boolean): Buffer {
-  const hdr = Buffer.alloc(12);
-  hdr[0] = 0x80; // V=2
-  hdr[1] = (marker ? 0x80 : 0) | (pt & 0x7f);
-  hdr.writeUInt16BE(seq & 0xffff, 2);
-  hdr.writeUInt32BE(ts >>> 0, 4);
-  hdr.writeUInt32BE(ssrc >>> 0, 8);
-  return hdr;
-}
-
-/** Codec name for status display. */
-export function codecDisplayName(pt: number | null): string | null {
-  if (pt === null) return null;
-  switch (pt) {
-    case 0: return 'PCMU';
-    case 8: return 'PCMA';
-    case 9: return 'G.722';
-    case 111: return 'Opus';
-    case 101: return 'telephone-event';
-    default: return `PT${pt}`;
-  }
-}

ts/call/prompt-cache.ts

@@ -0,0 +1,421 @@
+/**
+ * PromptCache — manages multiple named audio prompts for IVR and voicemail.
+ *
+ * Each prompt is pre-encoded as both G.722 frames (for SIP legs) and Opus
+ * frames (for WebRTC legs), ready for 20ms RTP playback.
+ *
+ * Supports three sources:
+ * 1. TTS generation via espeak-ng (primary) or Kokoro (fallback)
+ * 2. Loading from a pre-existing WAV file
+ * 3. Programmatic tone generation (beep, etc.)
+ *
+ * The existing announcement.ts system continues to work independently;
+ * this module provides generalized prompt management for IVR/voicemail.
+ */
+
+import { execSync } from 'node:child_process';
+import fs from 'node:fs';
+import path from 'node:path';
+import { Buffer } from 'node:buffer';
+import { encodePcm, isCodecReady } from '../opusbridge.ts';
+
+/** RTP clock increment per 20ms frame for each codec. */
+function rtpClockIncrement(pt: number): number {
+  if (pt === 111) return 960;
+  if (pt === 9) return 160;
+  return 160;
+}
+
+/** Build a fresh RTP header. */
+function buildRtpHeader(pt: number, seq: number, ts: number, ssrc: number, marker: boolean): Buffer {
+  const hdr = Buffer.alloc(12);
+  hdr[0] = 0x80;
+  hdr[1] = (marker ? 0x80 : 0) | (pt & 0x7f);
+  hdr.writeUInt16BE(seq & 0xffff, 2);
+  hdr.writeUInt32BE(ts >>> 0, 4);
+  hdr.writeUInt32BE(ssrc >>> 0, 8);
+  return hdr;
+}
+
+// ---------------------------------------------------------------------------
+// Types
+// ---------------------------------------------------------------------------
+
+/** A pre-encoded prompt ready for RTP playback. */
+export interface ICachedPrompt {
+  /** Unique prompt identifier. */
+  id: string;
+  /** G.722 encoded frames (20ms each, no RTP header). */
+  g722Frames: Buffer[];
+  /** Opus encoded frames (20ms each, no RTP header). */
+  opusFrames: Buffer[];
+  /** Total duration in milliseconds. */
+  durationMs: number;
+}
+
+// ---------------------------------------------------------------------------
+// TTS helpers
+// ---------------------------------------------------------------------------
+
+const TTS_DIR = path.join(process.cwd(), '.nogit', 'tts');
+
+/** Check if espeak-ng is available. */
+function isEspeakAvailable(): boolean {
+  try {
+    execSync('which espeak-ng', { stdio: 'pipe' });
+    return true;
+  } catch {
+    return false;
+  }
+}
+
+/** Generate WAV via espeak-ng. */
+function generateViaEspeak(wavPath: string, text: string): boolean {
+  try {
+    execSync(
+      `espeak-ng -v en-us -s 150 -w "${wavPath}" "${text}"`,
+      { timeout: 10000, stdio: 'pipe' },
+    );
+    return true;
+  } catch {
+    return false;
+  }
+}
+
+/** Generate WAV via Kokoro TTS. */
+function generateViaKokoro(wavPath: string, text: string, voice: string): boolean {
+  const modelPath = path.join(TTS_DIR, 'kokoro-v1.0.onnx');
+  const voicesPath = path.join(TTS_DIR, 'voices.bin');
+  if (!fs.existsSync(modelPath) || !fs.existsSync(voicesPath)) return false;
+
+  const root = process.cwd();
+  const ttsBin = [
+    path.join(root, 'dist_rust', 'tts-engine'),
+    path.join(root, 'rust', 'target', 'release', 'tts-engine'),
+    path.join(root, 'rust', 'target', 'debug', 'tts-engine'),
+  ].find((p) => fs.existsSync(p));
+  if (!ttsBin) return false;
+
+  try {
+    execSync(
+      `"${ttsBin}" --model "${modelPath}" --voices "${voicesPath}" --voice "${voice}" --output "${wavPath}" --text "${text}"`,
+      { timeout: 120000, stdio: 'pipe' },
+    );
+    return true;
+  } catch {
+    return false;
+  }
+}
+
+/** Read a WAV file and return raw PCM + sample rate. */
+function readWavWithRate(wavPath: string): { pcm: Buffer; sampleRate: number } | null {
+  const wav = fs.readFileSync(wavPath);
+  if (wav.length < 44) return null;
+  if (wav.toString('ascii', 0, 4) !== 'RIFF') return null;
+  if (wav.toString('ascii', 8, 12) !== 'WAVE') return null;
+
+  let sampleRate = 22050;
+  let pcm: Buffer | null = null;
+  let offset = 12;
+
+  while (offset < wav.length - 8) {
+    const chunkId = wav.toString('ascii', offset, offset + 4);
+    const chunkSize = wav.readUInt32LE(offset + 4);
+    if (chunkId === 'fmt ') {
+      sampleRate = wav.readUInt32LE(offset + 12);
+    }
+    if (chunkId === 'data') {
+      pcm = wav.subarray(offset + 8, offset + 8 + chunkSize);
+    }
+    offset += 8 + chunkSize;
+    if (offset % 2 !== 0) offset++;
+  }
+
+  return pcm ? { pcm, sampleRate } : null;
+}
+
+/** Encode raw PCM frames to G.722 + Opus. */
+async function encodePcmFrames(
+  pcm: Buffer,
+  sampleRate: number,
+  log: (msg: string) => void,
+): Promise<{ g722Frames: Buffer[]; opusFrames: Buffer[] } | null> {
+  if (!isCodecReady()) return null;
+
+  const frameSamples = Math.floor(sampleRate * 0.02); // 20ms
+  const frameBytes = frameSamples * 2; // 16-bit
+  const totalFrames = Math.floor(pcm.length / frameBytes);
+
+  const g722Frames: Buffer[] = [];
+  const opusFrames: Buffer[] = [];
+
+  for (let i = 0; i < totalFrames; i++) {
+    const framePcm = Buffer.from(pcm.subarray(i * frameBytes, (i + 1) * frameBytes));
+    const [g722, opus] = await Promise.all([
+      encodePcm(framePcm, sampleRate, 9),   // G.722
+      encodePcm(framePcm, sampleRate, 111),  // Opus
+    ]);
+    if (g722) g722Frames.push(g722);
+    if (opus) opusFrames.push(opus);
+  }
+
+  return { g722Frames, opusFrames };
+}
+
+// ---------------------------------------------------------------------------
+// PromptCache
+// ---------------------------------------------------------------------------
+
+export class PromptCache {
+  private prompts = new Map<string, ICachedPrompt>();
+  private log: (msg: string) => void;
+  private espeakAvailable: boolean | null = null;
+
+  constructor(log: (msg: string) => void) {
+    this.log = log;
+  }
+
+  // -------------------------------------------------------------------------
+  // Public API
+  // -------------------------------------------------------------------------
+
+  /** Get a cached prompt by ID. */
+  get(id: string): ICachedPrompt | null {
+    return this.prompts.get(id) ?? null;
+  }
+
+  /** Check if a prompt is cached. */
+  has(id: string): boolean {
+    return this.prompts.has(id);
+  }
+
+  /** List all cached prompt IDs. */
+  listIds(): string[] {
+    return [...this.prompts.keys()];
+  }
+
+  /**
+   * Generate a TTS prompt and cache it.
+   * Uses espeak-ng (primary) or Kokoro (fallback).
+   */
+  async generatePrompt(id: string, text: string, voice = 'af_bella'): Promise<ICachedPrompt | null> {
+    fs.mkdirSync(TTS_DIR, { recursive: true });
+    const wavPath = path.join(TTS_DIR, `prompt-${id}.wav`);
+
+    // Check espeak availability once.
+    if (this.espeakAvailable === null) {
+      this.espeakAvailable = isEspeakAvailable();
+    }
+
+    // Generate WAV.
+    let generated = false;
+    if (!fs.existsSync(wavPath)) {
+      if (this.espeakAvailable) {
+        generated = generateViaEspeak(wavPath, text);
+      }
+      if (!generated) {
+        generated = generateViaKokoro(wavPath, text, voice);
+      }
+      if (!generated) {
+        this.log(`[prompt-cache] failed to generate TTS for "${id}"`);
+        return null;
+      }
+      this.log(`[prompt-cache] generated WAV for "${id}"`);
+    }
+
+    return this.loadWavPrompt(id, wavPath);
+  }
+
+  /**
+   * Load a WAV file as a prompt and cache it.
+   */
+  async loadWavPrompt(id: string, wavPath: string): Promise<ICachedPrompt | null> {
+    if (!fs.existsSync(wavPath)) {
+      this.log(`[prompt-cache] WAV not found: ${wavPath}`);
+      return null;
+    }
+
+    const result = readWavWithRate(wavPath);
+    if (!result) {
+      this.log(`[prompt-cache] failed to parse WAV: ${wavPath}`);
+      return null;
+    }
+
+    const encoded = await encodePcmFrames(result.pcm, result.sampleRate, this.log);
+    if (!encoded) {
+      this.log(`[prompt-cache] encoding failed for "${id}" (codec bridge not ready?)`);
+      return null;
+    }
+
+    const durationMs = encoded.g722Frames.length * 20;
+    const prompt: ICachedPrompt = {
+      id,
+      g722Frames: encoded.g722Frames,
+      opusFrames: encoded.opusFrames,
+      durationMs,
+    };
+
+    this.prompts.set(id, prompt);
+    this.log(`[prompt-cache] cached "${id}": ${encoded.g722Frames.length} frames (${(durationMs / 1000).toFixed(1)}s)`);
+    return prompt;
+  }
+
+  /**
+   * Generate a beep tone prompt (sine wave).
+   * @param id - prompt ID
+   * @param freqHz - tone frequency (default 1000 Hz)
+   * @param durationMs - tone duration (default 500ms)
+   * @param amplitude - 16-bit amplitude (default 8000)
+   */
+  async generateBeep(
+    id: string,
+    freqHz = 1000,
+    durationMs = 500,
+    amplitude = 8000,
+  ): Promise<ICachedPrompt | null> {
+    // Generate at 16kHz for decent quality.
+    const sampleRate = 16000;
+    const totalSamples = Math.floor((sampleRate * durationMs) / 1000);
+    const pcm = Buffer.alloc(totalSamples * 2);
+
+    for (let i = 0; i < totalSamples; i++) {
+      const t = i / sampleRate;
+      // Apply a short fade-in/fade-out to avoid click artifacts.
+      const fadeLen = Math.floor(sampleRate * 0.01); // 10ms fade
+      let envelope = 1.0;
+      if (i < fadeLen) envelope = i / fadeLen;
+      else if (i > totalSamples - fadeLen) envelope = (totalSamples - i) / fadeLen;
+
+      const sample = Math.round(Math.sin(2 * Math.PI * freqHz * t) * amplitude * envelope);
+      pcm.writeInt16LE(Math.max(-32768, Math.min(32767, sample)), i * 2);
+    }
+
+    const encoded = await encodePcmFrames(pcm, sampleRate, this.log);
+    if (!encoded) {
+      this.log(`[prompt-cache] beep encoding failed for "${id}"`);
+      return null;
+    }
+
+    const actualDuration = encoded.g722Frames.length * 20;
+    const prompt: ICachedPrompt = {
+      id,
+      g722Frames: encoded.g722Frames,
+      opusFrames: encoded.opusFrames,
+      durationMs: actualDuration,
+    };
+
+    this.prompts.set(id, prompt);
+    this.log(`[prompt-cache] beep "${id}" cached: ${actualDuration}ms @ ${freqHz}Hz`);
+    return prompt;
+  }
+
+  /**
+   * Remove a prompt from the cache.
+   */
+  remove(id: string): void {
+    this.prompts.delete(id);
+  }
+
+  /**
+   * Clear all cached prompts.
+   */
+  clear(): void {
+    this.prompts.clear();
+  }
+}
+
+// ---------------------------------------------------------------------------
+// Standalone playback helpers (for use by SystemLeg)
+// ---------------------------------------------------------------------------
+
+/**
+ * Play a cached prompt's G.722 frames as RTP packets at 20ms intervals.
+ *
+ * @param prompt - the cached prompt to play
+ * @param sendPacket - function to send a raw RTP packet (12-byte header + payload)
+ * @param ssrc - SSRC for RTP headers
+ * @param onDone - called when playback finishes
+ * @returns cancel function, or null if prompt has no G.722 frames
+ */
+export function playPromptG722(
+  prompt: ICachedPrompt,
+  sendPacket: (pkt: Buffer) => void,
+  ssrc: number,
+  onDone?: () => void,
+): (() => void) | null {
+  if (prompt.g722Frames.length === 0) {
+    onDone?.();
+    return null;
+  }
+
+  const frames = prompt.g722Frames;
+  const PT = 9;
+  let frameIdx = 0;
+  let seq = Math.floor(Math.random() * 0xffff);
+  let rtpTs = Math.floor(Math.random() * 0xffffffff);
+
+  const timer = setInterval(() => {
+    if (frameIdx >= frames.length) {
+      clearInterval(timer);
+      onDone?.();
+      return;
+    }
+
+    const payload = frames[frameIdx];
+    const hdr = buildRtpHeader(PT, seq & 0xffff, rtpTs >>> 0, ssrc >>> 0, frameIdx === 0);
+    const pkt = Buffer.concat([hdr, payload]);
+    sendPacket(pkt);
+
+    seq++;
+    rtpTs += rtpClockIncrement(PT);
+    frameIdx++;
+  }, 20);
+
+  return () => clearInterval(timer);
+}
+
+/**
+ * Play a cached prompt's Opus frames as RTP packets at 20ms intervals.
+ *
+ * @param prompt - the cached prompt to play
+ * @param sendPacket - function to send a raw RTP packet
+ * @param ssrc - SSRC for RTP headers
+ * @param counters - shared seq/ts counters (mutated in place for seamless transitions)
+ * @param onDone - called when playback finishes
+ * @returns cancel function, or null if prompt has no Opus frames
+ */
+export function playPromptOpus(
+  prompt: ICachedPrompt,
+  sendPacket: (pkt: Buffer) => void,
+  ssrc: number,
+  counters: { seq: number; ts: number },
+  onDone?: () => void,
+): (() => void) | null {
+  if (prompt.opusFrames.length === 0) {
+    onDone?.();
+    return null;
+  }
+
+  const frames = prompt.opusFrames;
+  const PT = 111;
+  let frameIdx = 0;
+
+  const timer = setInterval(() => {
+    if (frameIdx >= frames.length) {
+      clearInterval(timer);
+      onDone?.();
+      return;
+    }
+
+    const payload = frames[frameIdx];
+    const hdr = buildRtpHeader(PT, counters.seq & 0xffff, counters.ts >>> 0, ssrc >>> 0, frameIdx === 0);
+    const pkt = Buffer.concat([hdr, payload]);
+    sendPacket(pkt);
+
+    counters.seq++;
+    counters.ts += 960; // Opus 48kHz: 960 samples per 20ms
+    frameIdx++;
+  }, 20);
+
+  return () => clearInterval(timer);
+}

ts/call/rtp-port-pool.ts

@@ -1,71 +0,0 @@
-/**
- * Unified RTP port pool — replaces the three separate allocators
- * in sipproxy.ts, calloriginator.ts, and webrtcbridge.ts.
- *
- * Allocates even-numbered UDP ports from a configured range.
- * Each allocation binds a dgram socket and returns it ready to use.
- */
-
-import dgram from 'node:dgram';
-
-export interface IRtpAllocation {
-  port: number;
-  sock: dgram.Socket;
-}
-
-export class RtpPortPool {
-  private min: number;
-  private max: number;
-  private allocated = new Map<number, dgram.Socket>();
-  private log: (msg: string) => void;
-
-  constructor(min: number, max: number, log: (msg: string) => void) {
-    this.min = min % 2 === 0 ? min : min + 1; // ensure even start
-    this.max = max;
-    this.log = log;
-  }
-
-  /**
-   * Allocate an even-numbered port and bind a UDP socket to it.
-   * Returns null if the pool is exhausted.
-   */
-  allocate(): IRtpAllocation | null {
-    for (let port = this.min; port < this.max; port += 2) {
-      if (this.allocated.has(port)) continue;
-
-      const sock = dgram.createSocket('udp4');
-      try {
-        sock.bind(port, '0.0.0.0');
-      } catch {
-        try { sock.close(); } catch { /* ignore */ }
-        continue;
-      }
-      this.allocated.set(port, sock);
-      this.log(`[rtp-pool] allocated port ${port} (${this.allocated.size} in use)`);
-      return { port, sock };
-    }
-    this.log('[rtp-pool] WARN: port pool exhausted');
-    return null;
-  }
-
-  /**
-   * Release a port back to the pool and close its socket.
-   */
-  release(port: number): void {
-    const sock = this.allocated.get(port);
-    if (!sock) return;
-    try { sock.close(); } catch { /* ignore */ }
-    this.allocated.delete(port);
-    this.log(`[rtp-pool] released port ${port} (${this.allocated.size} in use)`);
-  }
-
-  /** Number of currently allocated ports. */
-  get size(): number {
-    return this.allocated.size;
-  }
-
-  /** Total capacity (number of even ports in range). */
-  get capacity(): number {
-    return Math.floor((this.max - this.min) / 2);
-  }
-}

ts/call/sip-leg.ts

@@ -1,622 +0,0 @@
-/**
- * SipLeg — a SIP connection from the Call hub to a device or provider.
- *
- * Wraps a SipDialog and an RTP socket. Handles:
- * - INVITE/ACK/BYE/CANCEL lifecycle
- * - SDP rewriting (LAN IP for devices, public IP for providers)
- * - Digest auth for provider legs (407/401)
- * - Early-media silence for providers with quirks
- * - Record-Route insertion for dialog-establishing requests
- */
-
-import dgram from 'node:dgram';
-import { Buffer } from 'node:buffer';
-import {
-  SipMessage,
-  SipDialog,
-  buildSdp,
-  parseSdpEndpoint,
-  rewriteSdp,
-  rewriteSipUri,
-  parseDigestChallenge,
-  computeDigestAuth,
-  generateTag,
-} from '../sip/index.ts';
-import type { IEndpoint } from '../sip/index.ts';
-import type { IProviderConfig, IQuirks } from '../config.ts';
-import type { TLegState, TLegType, ILegStatus } from './types.ts';
-import type { ILeg } from './leg.ts';
-import { codecDisplayName } from './leg.ts';
-import type { IRtpTranscoder } from '../codec.ts';
-
-// ---------------------------------------------------------------------------
-// SipLeg config
-// ---------------------------------------------------------------------------
-
-export interface ISipLegConfig {
-  /** Whether this leg faces a device (LAN) or a provider (WAN). */
-  role: 'device' | 'provider';
-
-  /** Proxy LAN IP (for SDP rewriting toward devices). */
-  lanIp: string;
-  /** Proxy LAN port (for Via, Contact, Record-Route). */
-  lanPort: number;
-
-  /** Public IP (for SDP rewriting toward providers). */
-  getPublicIp: () => string | null;
-
-  /** Send a SIP message via the main UDP socket. */
-  sendSip: (buf: Buffer, dest: IEndpoint) => void;
-  /** Logging function. */
-  log: (msg: string) => void;
-
-  /** Provider config (for provider legs: auth, codecs, quirks, outbound proxy). */
-  provider?: IProviderConfig;
-
-  /** The endpoint to send SIP messages to (device address or provider outbound proxy). */
-  sipTarget: IEndpoint;
-
-  /** RTP port and socket (pre-allocated from the pool). */
-  rtpPort: number;
-  rtpSock: dgram.Socket;
-
-  /** Payload types to offer in SDP. */
-  payloadTypes?: number[];
-
-  /** Registered AOR (for From header in provider leg). */
-  getRegisteredAor?: () => string | null;
-  /** SIP password (for digest auth). */
-  getSipPassword?: () => string | null;
-}
-
-// ---------------------------------------------------------------------------
-// SipLeg
-// ---------------------------------------------------------------------------
-
-export class SipLeg implements ILeg {
-  readonly id: string;
-  readonly type: TLegType;
-  state: TLegState = 'inviting';
-  readonly config: ISipLegConfig;
-
-  /** The SIP dialog for this leg. */
-  dialog: SipDialog | null = null;
-
-  /** Original INVITE (needed for CANCEL). */
-  invite: SipMessage | null = null;
-
-  /** Original unauthenticated INVITE (for re-ACKing retransmitted 407s). */
-  private origInvite: SipMessage | null = null;
-
-  /** Whether we've attempted digest auth on this leg. */
-  private authAttempted = false;
-
-  /** RTP socket and port. */
-  readonly rtpPort: number;
-  readonly rtpSock: dgram.Socket;
-
-  /** Remote media endpoint (learned from SDP). */
-  remoteMedia: IEndpoint | null = null;
-
-  /** Negotiated codec. */
-  codec: number | null = null;
-
-  /** Transcoder (set by Call when codecs differ between legs). */
-  transcoder: IRtpTranscoder | null = null;
-
-  /** Stable SSRC for this leg (used for silence + forwarded audio). */
-  readonly ssrc: number = (Math.random() * 0xffffffff) >>> 0;
-
-  /** Packet counters. */
-  pktSent = 0;
-  pktReceived = 0;
-
-  /** Callback set by Call to receive RTP. */
-  onRtpReceived: ((data: Buffer) => void) | null = null;
-
-  /** Silence stream timer (for provider quirks). */
-  private silenceTimer: ReturnType<typeof setInterval> | null = null;
-
-  /** Callbacks for lifecycle events. */
-  onStateChange: ((leg: SipLeg) => void) | null = null;
-  onConnected: ((leg: SipLeg) => void) | null = null;
-  onTerminated: ((leg: SipLeg) => void) | null = null;
-
-  constructor(id: string, config: ISipLegConfig) {
-    this.id = id;
-    this.type = config.role === 'device' ? 'sip-device' : 'sip-provider';
-    this.config = config;
-    this.rtpPort = config.rtpPort;
-    this.rtpSock = config.rtpSock;
-
-    // Set up RTP receive handler.
-    this.rtpSock.on('message', (data: Buffer, rinfo: dgram.RemoteInfo) => {
-      this.pktReceived++;
-
-      // Learn remote media endpoint from first packet if not yet known.
-      if (!this.remoteMedia) {
-        this.remoteMedia = { address: rinfo.address, port: rinfo.port };
-        this.config.log(`[sip-leg:${this.id}] learned remote media: ${rinfo.address}:${rinfo.port}`);
-      }
-
-      // Forward to the Call hub.
-      if (this.onRtpReceived) {
-        this.onRtpReceived(data);
-      }
-    });
-
-    this.rtpSock.on('error', (e: Error) => {
-      this.config.log(`[sip-leg:${this.id}] rtp error: ${e.message}`);
-    });
-  }
-
-  get sipCallId(): string {
-    return this.dialog?.callId || 'no-dialog';
-  }
-
-  // -------------------------------------------------------------------------
-  // Outbound INVITE (B2BUA mode — create a new dialog)
-  // -------------------------------------------------------------------------
-
-  /**
-   * Send an INVITE to establish this leg.
-   * Creates a new SipDialog (UAC side).
-   */
-  sendInvite(options: {
-    fromUri: string;
-    toUri: string;
-    callId: string;
-    fromTag?: string;
-    fromDisplayName?: string;
-    cseq?: number;
-    extraHeaders?: [string, string][];
-  }): void {
-    const ip = this.type === 'sip-provider'
-      ? (this.config.getPublicIp() || this.config.lanIp)
-      : this.config.lanIp;
-    const pts = this.config.payloadTypes || [9, 0, 8, 101];
-
-    const sdp = buildSdp({ ip, port: this.rtpPort, payloadTypes: pts });
-
-    const invite = SipMessage.createRequest('INVITE', options.toUri, {
-      via: { host: ip, port: this.config.lanPort },
-      from: { uri: options.fromUri, displayName: options.fromDisplayName, tag: options.fromTag },
-      to: { uri: options.toUri },
-      callId: options.callId,
-      cseq: options.cseq,
-      contact: `<sip:${ip}:${this.config.lanPort}>`,
-      body: sdp,
-      contentType: 'application/sdp',
-      extraHeaders: options.extraHeaders,
-    });
-
-    this.invite = invite;
-    this.dialog = SipDialog.fromUacInvite(invite, ip, this.config.lanPort);
-    this.state = 'inviting';
-
-    this.config.log(`[sip-leg:${this.id}] INVITE -> ${this.config.sipTarget.address}:${this.config.sipTarget.port}`);
-    this.config.sendSip(invite.serialize(), this.config.sipTarget);
-  }
-
-  // -------------------------------------------------------------------------
-  // Passthrough mode — forward a SIP message with rewriting
-  // -------------------------------------------------------------------------
-
-  /**
-   * Accept an incoming INVITE as a UAS (for passthrough inbound calls).
-   * Creates a SipDialog on the UAS side.
-   */
-  acceptIncoming(invite: SipMessage): void {
-    const localTag = generateTag();
-    this.dialog = SipDialog.fromUasInvite(invite, localTag, this.config.lanIp, this.config.lanPort);
-    this.invite = invite;
-    this.state = 'inviting';
-
-    // Learn remote media from SDP.
-    if (invite.hasSdpBody) {
-      const ep = parseSdpEndpoint(invite.body);
-      if (ep) {
-        this.remoteMedia = ep;
-        this.config.log(`[sip-leg:${this.id}] media from SDP: ${ep.address}:${ep.port}`);
-      }
-    }
-  }
-
-  /**
-   * Forward a SIP message through this leg with SDP rewriting.
-   * Used for passthrough calls where the proxy relays messages.
-   */
-  forwardMessage(msg: SipMessage, dest: IEndpoint): void {
-    const rewriteIp = this.type === 'sip-provider'
-      ? (this.config.getPublicIp() || this.config.lanIp)
-      : this.config.lanIp;
-
-    // Rewrite SDP if present.
-    if (msg.hasSdpBody) {
-      const { body, original } = rewriteSdp(msg.body, rewriteIp, this.rtpPort);
-      msg.body = body;
-      msg.updateContentLength();
-      if (original) {
-        this.remoteMedia = original;
-        this.config.log(`[sip-leg:${this.id}] media from SDP rewrite: ${original.address}:${original.port}`);
-      }
-    }
-
-    // Record-Route for dialog-establishing requests.
-    if (msg.isRequest && msg.isDialogEstablishing) {
-      msg.prependHeader('Record-Route', `<sip:${this.config.lanIp}:${this.config.lanPort};lr>`);
-    }
-
-    // Rewrite Contact.
-    if (this.type === 'sip-provider') {
-      const contact = msg.getHeader('Contact');
-      if (contact) {
-        const nc = rewriteSipUri(contact, rewriteIp, this.config.lanPort);
-        if (nc !== contact) msg.setHeader('Contact', nc);
-      }
-    }
-
-    // Rewrite Request-URI for inbound messages going to device.
-    if (this.type === 'sip-device' && msg.isRequest) {
-      msg.setRequestUri(rewriteSipUri(msg.requestUri!, dest.address, dest.port));
-    }
-
-    this.config.sendSip(msg.serialize(), dest);
-  }
-
-  // -------------------------------------------------------------------------
-  // SIP message handling (routed by CallManager)
-  // -------------------------------------------------------------------------
-
-  handleSipMessage(msg: SipMessage, rinfo: IEndpoint): void {
-    if (msg.isResponse) {
-      this.handleResponse(msg, rinfo);
-    } else {
-      this.handleRequest(msg, rinfo);
-    }
-  }
-
-  private handleResponse(msg: SipMessage, _rinfo: IEndpoint): void {
-    const code = msg.statusCode ?? 0;
-    const method = msg.cseqMethod?.toUpperCase();
-
-    this.config.log(`[sip-leg:${this.id}] <- ${code} (${method})`);
-
-    if (method === 'INVITE') {
-      this.handleInviteResponse(msg, code);
-    }
-    // BYE/CANCEL responses don't need action beyond logging.
-  }
-
-  private handleInviteResponse(msg: SipMessage, code: number): void {
-    // Handle retransmitted 407 for the original unauthenticated INVITE.
-    if (this.authAttempted && this.dialog) {
-      const responseCSeqNum = parseInt((msg.getHeader('CSeq') || '').split(/\s+/)[0], 10);
-      if (responseCSeqNum < this.dialog.localCSeq && code >= 400) {
-        if (this.origInvite) {
-          const ack = buildNon2xxAck(this.origInvite, msg);
-          this.config.sendSip(ack.serialize(), this.config.sipTarget);
-        }
-        return;
-      }
-    }
-
-    // Handle 407 Proxy Authentication Required.
-    if (code === 407 && this.type === 'sip-provider') {
-      this.handleAuthChallenge(msg);
-      return;
-    }
-
-    // Update dialog state.
-    if (this.dialog) {
-      this.dialog.processResponse(msg);
-    }
-
-    if (code === 180 || code === 183) {
-      this.state = 'ringing';
-      this.onStateChange?.(this);
-    } else if (code >= 200 && code < 300) {
-      // ACK the 200 OK.
-      if (this.dialog) {
-        const ack = this.dialog.createAck();
-        this.config.sendSip(ack.serialize(), this.config.sipTarget);
-        this.config.log(`[sip-leg:${this.id}] ACK sent`);
-      }
-
-      // If already connected (200 retransmit), just re-ACK.
-      if (this.state === 'connected') {
-        this.config.log(`[sip-leg:${this.id}] re-ACK (200 retransmit)`);
-        return;
-      }
-
-      // Learn media endpoint from SDP.
-      if (msg.hasSdpBody) {
-        const ep = parseSdpEndpoint(msg.body);
-        if (ep) {
-          this.remoteMedia = ep;
-          this.config.log(`[sip-leg:${this.id}] media = ${ep.address}:${ep.port}`);
-        }
-      }
-
-      this.state = 'connected';
-      this.config.log(`[sip-leg:${this.id}] CONNECTED`);
-
-      // Start silence for provider legs with early media quirks.
-      if (this.type === 'sip-provider') {
-        this.startSilence();
-      }
-
-      // Prime the RTP path.
-      if (this.remoteMedia) {
-        this.primeRtp(this.remoteMedia);
-      }
-
-      this.onConnected?.(this);
-      this.onStateChange?.(this);
-    } else if (code >= 300) {
-      this.config.log(`[sip-leg:${this.id}] rejected ${code}`);
-      this.state = 'terminated';
-      if (this.dialog) this.dialog.terminate();
-      this.onTerminated?.(this);
-      this.onStateChange?.(this);
-    }
-  }
-
-  private handleAuthChallenge(msg: SipMessage): void {
-    if (this.authAttempted) {
-      this.config.log(`[sip-leg:${this.id}] 407 after auth attempt — credentials rejected`);
-      this.state = 'terminated';
-      if (this.dialog) this.dialog.terminate();
-      this.onTerminated?.(this);
-      return;
-    }
-    this.authAttempted = true;
-
-    const challenge = msg.getHeader('Proxy-Authenticate');
-    if (!challenge) {
-      this.config.log(`[sip-leg:${this.id}] 407 but no Proxy-Authenticate`);
-      this.state = 'terminated';
-      if (this.dialog) this.dialog.terminate();
-      this.onTerminated?.(this);
-      return;
-    }
-
-    const parsed = parseDigestChallenge(challenge);
-    if (!parsed) {
-      this.config.log(`[sip-leg:${this.id}] could not parse digest challenge`);
-      this.state = 'terminated';
-      if (this.dialog) this.dialog.terminate();
-      this.onTerminated?.(this);
-      return;
-    }
-
-    const password = this.config.getSipPassword?.();
-    const aor = this.config.getRegisteredAor?.();
-    if (!password || !aor) {
-      this.config.log(`[sip-leg:${this.id}] 407 but no password or AOR`);
-      this.state = 'terminated';
-      if (this.dialog) this.dialog.terminate();
-      this.onTerminated?.(this);
-      return;
-    }
-
-    const username = aor.replace(/^sips?:/, '').split('@')[0];
-    const destUri = this.invite?.requestUri || '';
-
-    const authValue = computeDigestAuth({
-      username,
-      password,
-      realm: parsed.realm,
-      nonce: parsed.nonce,
-      method: 'INVITE',
-      uri: destUri,
-      algorithm: parsed.algorithm,
-      opaque: parsed.opaque,
-    });
-
-    // ACK the 407.
-    if (this.invite) {
-      const ack407 = buildNon2xxAck(this.invite, msg);
-      this.config.sendSip(ack407.serialize(), this.config.sipTarget);
-      this.config.log(`[sip-leg:${this.id}] ACK-407 sent`);
-    }
-
-    // Keep original INVITE for re-ACKing retransmitted 407s.
-    this.origInvite = this.invite;
-
-    // Resend INVITE with auth, same From tag, incremented CSeq.
-    const ip = this.config.getPublicIp() || this.config.lanIp;
-    const fromTag = this.dialog!.localTag;
-    const pts = this.config.payloadTypes || [9, 0, 8, 101];
-
-    const sdp = buildSdp({ ip, port: this.rtpPort, payloadTypes: pts });
-
-    const inviteAuth = SipMessage.createRequest('INVITE', destUri, {
-      via: { host: ip, port: this.config.lanPort },
-      from: { uri: aor, tag: fromTag },
-      to: { uri: destUri },
-      callId: this.dialog!.callId,
-      cseq: 2,
-      contact: `<sip:${ip}:${this.config.lanPort}>`,
-      body: sdp,
-      contentType: 'application/sdp',
-      extraHeaders: [['Proxy-Authorization', authValue]],
-    });
-
-    this.invite = inviteAuth;
-    this.dialog!.localCSeq = 2;
-
-    this.config.log(`[sip-leg:${this.id}] resending INVITE with auth`);
-    this.config.sendSip(inviteAuth.serialize(), this.config.sipTarget);
-  }
-
-  private handleRequest(msg: SipMessage, rinfo: IEndpoint): void {
-    const method = msg.method;
-    this.config.log(`[sip-leg:${this.id}] <- ${method} from ${rinfo.address}:${rinfo.port}`);
-
-    if (method === 'BYE') {
-      // Send 200 OK to the BYE.
-      const ok = SipMessage.createResponse(200, 'OK', msg);
-      this.config.sendSip(ok.serialize(), { address: rinfo.address, port: rinfo.port });
-
-      this.state = 'terminated';
-      if (this.dialog) this.dialog.terminate();
-      this.onTerminated?.(this);
-      this.onStateChange?.(this);
-    }
-    // Other in-dialog requests (re-INVITE, INFO, etc.) can be handled here in the future.
-  }
-
-  // -------------------------------------------------------------------------
-  // Send BYE / CANCEL
-  // -------------------------------------------------------------------------
-
-  /** Send BYE (if confirmed) or CANCEL (if early) to tear down this leg. */
-  sendHangup(): void {
-    if (!this.dialog) return;
-
-    if (this.dialog.state === 'confirmed') {
-      const bye = this.dialog.createRequest('BYE');
-      this.config.sendSip(bye.serialize(), this.config.sipTarget);
-      this.config.log(`[sip-leg:${this.id}] BYE sent`);
-    } else if (this.dialog.state === 'early' && this.invite) {
-      const cancel = this.dialog.createCancel(this.invite);
-      this.config.sendSip(cancel.serialize(), this.config.sipTarget);
-      this.config.log(`[sip-leg:${this.id}] CANCEL sent`);
-    }
-
-    this.state = 'terminating';
-    this.dialog.terminate();
-  }
-
-  // -------------------------------------------------------------------------
-  // RTP
-  // -------------------------------------------------------------------------
-
-  sendRtp(data: Buffer): void {
-    if (!this.remoteMedia) return;
-    this.rtpSock.send(data, this.remoteMedia.port, this.remoteMedia.address);
-    this.pktSent++;
-  }
-
-  /** Send a 1-byte UDP packet to punch NAT hole. */
-  private primeRtp(peer: IEndpoint): void {
-    try {
-      this.rtpSock.send(Buffer.alloc(1), peer.port, peer.address);
-      this.config.log(`[sip-leg:${this.id}] RTP primed -> ${peer.address}:${peer.port}`);
-    } catch (e: any) {
-      this.config.log(`[sip-leg:${this.id}] prime error: ${e.message}`);
-    }
-  }
-
-  // -------------------------------------------------------------------------
-  // Silence stream (provider quirks)
-  // -------------------------------------------------------------------------
-
-  private startSilence(): void {
-    if (this.silenceTimer) return;
-    const quirks = this.config.provider?.quirks;
-    if (!quirks?.earlyMediaSilence) return;
-    if (!this.remoteMedia) return;
-
-    const PT = quirks.silencePayloadType ?? 9;
-    const MAX = quirks.silenceMaxPackets ?? 250;
-    const PAYLOAD = 160;
-    let seq = Math.floor(Math.random() * 0xffff);
-    let rtpTs = Math.floor(Math.random() * 0xffffffff);
-    let count = 0;
-
-    // Use proper silence byte for the codec (0x00 is NOT silence for most codecs).
-    const silenceByte = silenceByteForPT(PT);
-
-    this.silenceTimer = setInterval(() => {
-      if (this.pktReceived > 0 || count >= MAX) {
-        clearInterval(this.silenceTimer!);
-        this.silenceTimer = null;
-        this.config.log(`[sip-leg:${this.id}] silence stop after ${count} pkts`);
-        return;
-      }
-      const pkt = Buffer.alloc(12 + PAYLOAD, silenceByte);
-      // RTP header (first 12 bytes).
-      pkt[0] = 0x80;
-      pkt[1] = PT;
-      pkt.writeUInt16BE(seq & 0xffff, 2);
-      pkt.writeUInt32BE(rtpTs >>> 0, 4);
-      pkt.writeUInt32BE(this.ssrc >>> 0, 8); // stable SSRC
-      this.rtpSock.send(pkt, this.remoteMedia!.port, this.remoteMedia!.address);
-      seq++;
-      rtpTs += PAYLOAD;
-      count++;
-    }, 20);
-
-    this.config.log(`[sip-leg:${this.id}] silence start -> ${this.remoteMedia.address}:${this.remoteMedia.port} (ssrc=${this.ssrc})`);
-  }
-
-  // -------------------------------------------------------------------------
-  // Lifecycle
-  // -------------------------------------------------------------------------
-
-  teardown(): void {
-    if (this.silenceTimer) {
-      clearInterval(this.silenceTimer);
-      this.silenceTimer = null;
-    }
-    this.state = 'terminated';
-    if (this.dialog) this.dialog.terminate();
-    // Note: RTP socket is NOT closed here — the RtpPortPool manages that.
-  }
-
-  getStatus(): ILegStatus {
-    return {
-      id: this.id,
-      type: this.type,
-      state: this.state,
-      remoteMedia: this.remoteMedia,
-      rtpPort: this.rtpPort,
-      pktSent: this.pktSent,
-      pktReceived: this.pktReceived,
-      codec: codecDisplayName(this.codec),
-      transcoding: this.transcoder !== null,
-    };
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Helper: proper silence byte per codec
-// ---------------------------------------------------------------------------
-
-/** Return the byte value representing digital silence for a given RTP payload type. */
-function silenceByteForPT(pt: number): number {
-  switch (pt) {
-    case 0: return 0xFF;   // PCMU: μ-law silence (zero amplitude)
-    case 8: return 0xD5;   // PCMA: A-law silence (zero amplitude)
-    case 9: return 0xD5;   // G.722: sub-band silence (zero amplitude)
-    default: return 0xFF;  // safe default
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Helper: ACK for non-2xx (same transaction)
-// ---------------------------------------------------------------------------
-
-function buildNon2xxAck(originalInvite: SipMessage, response: SipMessage): SipMessage {
-  const via = originalInvite.getHeader('Via') || '';
-  const from = originalInvite.getHeader('From') || '';
-  const toFromResponse = response.getHeader('To') || '';
-  const callId = originalInvite.callId;
-  const cseqNum = parseInt((originalInvite.getHeader('CSeq') || '1').split(/\s+/)[0], 10);
-
-  return new SipMessage(
-    `ACK ${originalInvite.requestUri} SIP/2.0`,
-    [
-      ['Via', via],
-      ['From', from],
-      ['To', toFromResponse],
-      ['Call-ID', callId],
-      ['CSeq', `${cseqNum} ACK`],
-      ['Max-Forwards', '70'],
-      ['Content-Length', '0'],
-    ],
-    '',
-  );
-}

ts/call/types.ts

@@ -1,68 +0,0 @@
-/**
- * Hub model type definitions — Call, Leg, and status types.
- */
-
-import type { IEndpoint } from '../sip/index.ts';
-
-// ---------------------------------------------------------------------------
-// State types
-// ---------------------------------------------------------------------------
-
-export type TCallState =
-  | 'setting-up'
-  | 'ringing'
-  | 'connected'
-  | 'on-hold'
-  | 'transferring'
-  | 'terminating'
-  | 'terminated';
-
-export type TLegState =
-  | 'inviting'
-  | 'ringing'
-  | 'connected'
-  | 'on-hold'
-  | 'terminating'
-  | 'terminated';
-
-export type TLegType = 'sip-device' | 'sip-provider' | 'webrtc';
-
-export type TCallDirection = 'inbound' | 'outbound' | 'internal';
-
-// ---------------------------------------------------------------------------
-// Status interfaces (for frontend dashboard)
-// ---------------------------------------------------------------------------
-
-export interface ILegStatus {
-  id: string;
-  type: TLegType;
-  state: TLegState;
-  remoteMedia: IEndpoint | null;
-  rtpPort: number | null;
-  pktSent: number;
-  pktReceived: number;
-  codec: string | null;
-  transcoding: boolean;
-}
-
-export interface ICallStatus {
-  id: string;
-  state: TCallState;
-  direction: TCallDirection;
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  createdAt: number;
-  duration: number;
-  legs: ILegStatus[];
-}
-
-export interface ICallHistoryEntry {
-  id: string;
-  direction: TCallDirection;
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  startedAt: number;
-  duration: number;
-}

ts/call/webrtc-leg.ts

@@ -1,417 +0,0 @@
-/**
- * WebRtcLeg — a WebRTC connection from the Call hub to a browser client.
- *
- * Wraps a werift RTCPeerConnection and handles:
- * - WebRTC offer/answer/ICE negotiation
- * - Opus <-> G.722/PCMU/PCMA transcoding via Rust IPC
- * - RTP header rebuilding with correct PT, timestamp, SSRC
- */
-
-import dgram from 'node:dgram';
-import { Buffer } from 'node:buffer';
-import { WebSocket } from 'ws';
-import type { IEndpoint } from '../sip/index.ts';
-import type { TLegState, ILegStatus } from './types.ts';
-import type { ILeg } from './leg.ts';
-import { rtpClockIncrement, buildRtpHeader, codecDisplayName } from './leg.ts';
-import { createTranscoder, OPUS_PT } from '../codec.ts';
-import type { IRtpTranscoder } from '../codec.ts';
-import { createSession, destroySession } from '../opusbridge.ts';
-import type { SipDialog } from '../sip/index.ts';
-import type { SipMessage } from '../sip/index.ts';
-
-// ---------------------------------------------------------------------------
-// WebRtcLeg config
-// ---------------------------------------------------------------------------
-
-export interface IWebRtcLegConfig {
-  /** The browser's WebSocket connection. */
-  ws: WebSocket;
-  /** The browser's session ID. */
-  sessionId: string;
-  /** RTP port and socket (pre-allocated from the pool). */
-  rtpPort: number;
-  rtpSock: dgram.Socket;
-  /** Logging function. */
-  log: (msg: string) => void;
-}
-
-// ---------------------------------------------------------------------------
-// WebRtcLeg
-// ---------------------------------------------------------------------------
-
-export class WebRtcLeg implements ILeg {
-  readonly id: string;
-  readonly type = 'webrtc' as const;
-  state: TLegState = 'inviting';
-  readonly sessionId: string;
-
-  /** The werift RTCPeerConnection instance. */
-  private pc: any = null;
-
-  /** RTP socket for bridging to SIP. */
-  readonly rtpSock: dgram.Socket;
-  readonly rtpPort: number;
-
-  /** Remote media endpoint (the other side of the bridge, set by Call). */
-  remoteMedia: IEndpoint | null = null;
-
-  /** Negotiated WebRTC codec payload type. */
-  codec: number | null = null;
-
-  /** Transcoders for WebRTC <-> SIP conversion. */
-  transcoder: IRtpTranscoder | null = null; // used by Call for fan-out
-  private toSipTranscoder: IRtpTranscoder | null = null;
-  private fromSipTranscoder: IRtpTranscoder | null = null;
-
-  /** RTP counters for outgoing (to SIP) direction. */
-  private toSipSeq = 0;
-  private toSipTs = 0;
-  private toSipSsrc = (Math.random() * 0xffffffff) >>> 0;
-
-  /** RTP counters for incoming (from SIP) direction.
-   *  Initialized to random values so announcements and provider audio share
-   *  a continuous sequence — prevents the browser jitter buffer from discarding
-   *  packets after the announcement→provider transition. */
-  readonly fromSipCounters = {
-    seq: Math.floor(Math.random() * 0xffff),
-    ts: Math.floor(Math.random() * 0xffffffff),
-  };
-  fromSipSsrc = (Math.random() * 0xffffffff) >>> 0;
-
-  /** Packet counters. */
-  pktSent = 0;
-  pktReceived = 0;
-
-  /** Callback set by Call. */
-  onRtpReceived: ((data: Buffer) => void) | null = null;
-
-  /** Callback to send transcoded RTP to the provider via the SipLeg's socket.
-   *  Set by CallManager when the bridge is established. If null, falls back to own rtpSock. */
-  onSendToProvider: ((data: Buffer, dest: IEndpoint) => void) | null = null;
-
-  /** Lifecycle callbacks. */
-  onConnected: ((leg: WebRtcLeg) => void) | null = null;
-  onTerminated: ((leg: WebRtcLeg) => void) | null = null;
-
-  /** Cancel handle for an in-progress announcement. */
-  announcementCancel: (() => void) | null = null;
-
-  private ws: WebSocket;
-  private config: IWebRtcLegConfig;
-  private pendingIceCandidates: any[] = [];
-
-  // SipDialog is not applicable for WebRTC legs.
-  readonly dialog: SipDialog | null = null;
-  readonly sipCallId: string;
-
-  constructor(id: string, config: IWebRtcLegConfig) {
-    this.id = id;
-    this.sessionId = config.sessionId;
-    this.ws = config.ws;
-    this.rtpSock = config.rtpSock;
-    this.rtpPort = config.rtpPort;
-    this.config = config;
-    this.sipCallId = `webrtc-${id}`;
-
-    // Log RTP arriving on this socket (symmetric RTP from provider).
-    // Audio forwarding is handled by the Call hub: SipLeg → forwardRtp → WebRtcLeg.sendRtp.
-    // We do NOT transcode here to avoid double-processing (the SipLeg also receives these packets).
-    let sipRxCount = 0;
-    this.rtpSock.on('message', (data: Buffer) => {
-      sipRxCount++;
-      if (sipRxCount === 1 || sipRxCount === 50 || sipRxCount % 500 === 0) {
-        this.config.log(`[webrtc-leg:${this.id}] SIP->browser rtp #${sipRxCount} (${data.length}b) [symmetric, ignored]`);
-      }
-    });
-  }
-
-  // -------------------------------------------------------------------------
-  // WebRTC offer/answer
-  // -------------------------------------------------------------------------
-
-  /**
-   * Handle a WebRTC offer from the browser. Creates the PeerConnection,
-   * sets remote offer, creates answer, and sends it back.
-   */
-  async handleOffer(offerSdp: string): Promise<void> {
-    this.config.log(`[webrtc-leg:${this.id}] received offer`);
-
-    try {
-      const werift = await import('werift');
-
-      this.pc = new werift.RTCPeerConnection({ iceServers: [] });
-
-      // Add sendrecv transceiver before setRemoteDescription.
-      this.pc.addTransceiver('audio', { direction: 'sendrecv' });
-
-      // Handle incoming audio from browser.
-      this.pc.ontrack = (event: any) => {
-        const track = event.track;
-        this.config.log(`[webrtc-leg:${this.id}] got track: ${track.kind}`);
-
-        let rxCount = 0;
-        track.onReceiveRtp.subscribe((rtp: any) => {
-          if (!this.remoteMedia) return;
-          rxCount++;
-          if (rxCount === 1 || rxCount === 50 || rxCount % 500 === 0) {
-            this.config.log(`[webrtc-leg:${this.id}] browser->SIP rtp #${rxCount}`);
-          }
-
-          this.forwardToSip(rtp, rxCount);
-        });
-      };
-
-      // ICE candidate handling.
-      this.pc.onicecandidate = (candidate: any) => {
-        if (candidate) {
-          const json = candidate.toJSON?.() || candidate;
-          this.wsSend({ type: 'webrtc-ice', sessionId: this.sessionId, candidate: json });
-        }
-      };
-
-      this.pc.onconnectionstatechange = () => {
-        this.config.log(`[webrtc-leg:${this.id}] connection state: ${this.pc.connectionState}`);
-        if (this.pc.connectionState === 'connected') {
-          this.state = 'connected';
-          this.onConnected?.(this);
-        } else if (this.pc.connectionState === 'failed' || this.pc.connectionState === 'closed') {
-          this.state = 'terminated';
-          this.onTerminated?.(this);
-        }
-      };
-
-      if (this.pc.oniceconnectionstatechange !== undefined) {
-        this.pc.oniceconnectionstatechange = () => {
-          this.config.log(`[webrtc-leg:${this.id}] ICE state: ${this.pc.iceConnectionState}`);
-        };
-      }
-
-      // Set remote offer and create answer.
-      await this.pc.setRemoteDescription({ type: 'offer', sdp: offerSdp });
-      const answer = await this.pc.createAnswer();
-      await this.pc.setLocalDescription(answer);
-
-      const sdp: string = this.pc.localDescription!.sdp;
-
-      // Detect negotiated codec.
-      const mAudio = sdp.match(/m=audio\s+\d+\s+\S+\s+(\d+)/);
-      if (mAudio) {
-        this.codec = parseInt(mAudio[1], 10);
-        this.config.log(`[webrtc-leg:${this.id}] negotiated audio PT=${this.codec}`);
-      }
-
-      // Extract sender SSRC from SDP.
-      const ssrcMatch = sdp.match(/a=ssrc:(\d+)\s/);
-      if (ssrcMatch) {
-        this.fromSipSsrc = parseInt(ssrcMatch[1], 10);
-      }
-      // Also try from sender object.
-      const senders = this.pc.getSenders();
-      if (senders[0]) {
-        const senderSsrc = (senders[0] as any).ssrc ?? (senders[0] as any)._ssrc;
-        if (senderSsrc) this.fromSipSsrc = senderSsrc;
-      }
-
-      // Send answer to browser.
-      this.wsSend({ type: 'webrtc-answer', sessionId: this.sessionId, sdp });
-      this.config.log(`[webrtc-leg:${this.id}] sent answer, rtp port=${this.rtpPort}`);
-
-      // Process buffered ICE candidates.
-      for (const c of this.pendingIceCandidates) {
-        try { await this.pc.addIceCandidate(c); } catch { /* ignore */ }
-      }
-      this.pendingIceCandidates = [];
-
-    } catch (err: any) {
-      this.config.log(`[webrtc-leg:${this.id}] offer error: ${err.message}`);
-      this.wsSend({ type: 'webrtc-error', sessionId: this.sessionId, error: err.message });
-      this.state = 'terminated';
-      this.onTerminated?.(this);
-    }
-  }
-
-  /** Add an ICE candidate from the browser. */
-  async addIceCandidate(candidate: any): Promise<void> {
-    if (!this.pc) {
-      this.pendingIceCandidates.push(candidate);
-      return;
-    }
-    try {
-      if (candidate) await this.pc.addIceCandidate(candidate);
-    } catch (err: any) {
-      this.config.log(`[webrtc-leg:${this.id}] ICE error: ${err.message}`);
-    }
-  }
-
-  // -------------------------------------------------------------------------
-  // Transcoding setup
-  // -------------------------------------------------------------------------
-
-  /** Codec session ID for isolated Rust codec state (unique per leg). */
-  private codecSessionId = `webrtc-${Date.now()}-${Math.random().toString(36).slice(2, 8)}`;
-
-  /**
-   * Set up transcoders for bridging between WebRTC and SIP codecs.
-   * Called by the Call when the remote media endpoint is known.
-   * Creates an isolated Rust codec session so concurrent calls don't
-   * corrupt each other's stateful codec state (Opus/G.722 ADPCM).
-   */
-  async setupTranscoders(sipPT: number): Promise<void> {
-    const webrtcPT = this.codec ?? OPUS_PT;
-    // Create isolated codec session for this leg.
-    await createSession(this.codecSessionId);
-    this.toSipTranscoder = createTranscoder(webrtcPT, sipPT, this.codecSessionId, 'to_sip');
-    this.fromSipTranscoder = createTranscoder(sipPT, webrtcPT, this.codecSessionId, 'to_browser');
-    const mode = this.toSipTranscoder ? `transcoding PT ${webrtcPT}<->${sipPT}` : `pass-through PT ${webrtcPT}`;
-    this.config.log(`[webrtc-leg:${this.id}] ${mode} (session: ${this.codecSessionId})`);
-  }
-
-  // -------------------------------------------------------------------------
-  // RTP forwarding
-  // -------------------------------------------------------------------------
-
-  /** Forward RTP from SIP side to browser via WebRTC. */
-  private forwardToBrowser(data: Buffer, count: number): void {
-    const sender = this.pc?.getSenders()[0];
-    if (!sender) return;
-
-    if (this.fromSipTranscoder && data.length > 12) {
-      const payload = Buffer.from(data.subarray(12));
-      // Stop announcement if still playing — provider audio takes over.
-      if (this.announcementCancel) {
-        this.announcementCancel();
-        this.announcementCancel = null;
-      }
-      // Capture seq/ts BEFORE async transcode to preserve ordering.
-      const toPT = this.fromSipTranscoder.toPT;
-      const seq = this.fromSipCounters.seq++;
-      const ts = this.fromSipCounters.ts;
-      this.fromSipCounters.ts += rtpClockIncrement(toPT);
-      const result = this.fromSipTranscoder.payload(payload);
-      const sendTranscoded = (transcoded: Buffer) => {
-        if (transcoded.length === 0) return; // transcoding failed
-        try {
-          const hdr = buildRtpHeader(toPT, seq, ts, this.fromSipSsrc, false);
-          const out = Buffer.concat([hdr, transcoded]);
-          const r = sender.sendRtp(out);
-          if (r instanceof Promise) r.catch(() => {});
-        } catch { /* ignore */ }
-      };
-      if (result instanceof Promise) result.then(sendTranscoded).catch(() => {});
-      else sendTranscoded(result);
-    } else if (!this.fromSipTranscoder) {
-      // No transcoder — either same codec or not set up yet.
-      // Only forward if we don't expect transcoding.
-      if (this.codec === null) {
-        try { sender.sendRtp(data); } catch { /* ignore */ }
-      }
-    }
-  }
-
-  /** Forward RTP from browser to SIP side. */
-  private forwardToSip(rtp: any, count: number): void {
-    if (!this.remoteMedia) return;
-
-    if (this.toSipTranscoder) {
-      const payload: Buffer = rtp.payload;
-      if (!payload || payload.length === 0) return;
-      // Capture seq/ts BEFORE async transcode to preserve ordering.
-      const toPT = this.toSipTranscoder.toPT;
-      const seq = this.toSipSeq++;
-      const ts = this.toSipTs;
-      this.toSipTs += rtpClockIncrement(toPT);
-      const result = this.toSipTranscoder.payload(payload);
-      const sendTranscoded = (transcoded: Buffer) => {
-        if (transcoded.length === 0) return; // transcoding failed
-        const hdr = buildRtpHeader(toPT, seq, ts, this.toSipSsrc, false);
-        const out = Buffer.concat([hdr, transcoded]);
-        if (this.onSendToProvider) {
-          this.onSendToProvider(out, this.remoteMedia!);
-        } else {
-          this.rtpSock.send(out, this.remoteMedia!.port, this.remoteMedia!.address);
-        }
-        this.pktSent++;
-      };
-      if (result instanceof Promise) result.then(sendTranscoded).catch(() => {});
-      else sendTranscoded(result);
-    } else if (this.codec === null) {
-      // Same codec (no transcoding needed) — pass through.
-      const raw = rtp.serialize();
-      if (this.onSendToProvider) {
-        this.onSendToProvider(raw, this.remoteMedia);
-      } else {
-        this.rtpSock.send(raw, this.remoteMedia.port, this.remoteMedia.address);
-      }
-      this.pktSent++;
-    }
-    // If codec is set but transcoder is null, drop the packet — transcoder not ready yet.
-    // This prevents raw Opus from being sent to a G.722 endpoint.
-  }
-
-  /**
-   * Send RTP to the browser via WebRTC (used by Call hub for fan-out).
-   * This transcodes and sends through the PeerConnection, NOT to a UDP address.
-   */
-  sendRtp(data: Buffer): void {
-    this.forwardToBrowser(data, this.pktSent);
-    this.pktSent++;
-  }
-
-  /**
-   * Send a pre-encoded RTP packet directly to the browser via PeerConnection.
-   * Used for announcements — the packet must already be in the correct codec (Opus).
-   */
-  sendDirectToBrowser(pkt: Buffer): void {
-    const sender = this.pc?.getSenders()[0];
-    if (!sender) return;
-    try {
-      const r = sender.sendRtp(pkt);
-      if (r instanceof Promise) r.catch(() => {});
-    } catch { /* ignore */ }
-  }
-
-  /** No-op: WebRTC legs don't process SIP messages. */
-  handleSipMessage(_msg: SipMessage, _rinfo: IEndpoint): void {
-    // WebRTC legs don't handle SIP messages.
-  }
-
-  // -------------------------------------------------------------------------
-  // Lifecycle
-  // -------------------------------------------------------------------------
-
-  teardown(): void {
-    this.state = 'terminated';
-    try { this.pc?.close(); } catch { /* ignore */ }
-    this.pc = null;
-    // Destroy the isolated Rust codec session for this leg.
-    destroySession(this.codecSessionId).catch(() => {});
-    // Note: RTP socket is NOT closed here — the RtpPortPool manages that.
-  }
-
-  getStatus(): ILegStatus {
-    return {
-      id: this.id,
-      type: this.type,
-      state: this.state,
-      remoteMedia: this.remoteMedia,
-      rtpPort: this.rtpPort,
-      pktSent: this.pktSent,
-      pktReceived: this.pktReceived,
-      codec: codecDisplayName(this.codec),
-      transcoding: this.toSipTranscoder !== null || this.fromSipTranscoder !== null,
-    };
-  }
-
-  // -------------------------------------------------------------------------
-  // Helpers
-  // -------------------------------------------------------------------------
-
-  private wsSend(data: unknown): void {
-    try {
-      if (this.ws.readyState === WebSocket.OPEN) {
-        this.ws.send(JSON.stringify(data));
-      }
-    } catch { /* ignore */ }
-  }
-}

ts/codec.ts

@@ -1,40 +0,0 @@
-/**
- * Audio codec translation layer for bridging between WebRTC and SIP.
- *
- * All actual codec work (Opus, G.722, PCMU, PCMA) is done in Rust via
- * the smartrust bridge. This module provides the RTP-level transcoding
- * interface used by the webrtcbridge.
- */
-
-import { Buffer } from 'node:buffer';
-import { transcode, isCodecReady } from './opusbridge.ts';
-
-/** Opus dynamic payload type (standard WebRTC assignment). */
-export const OPUS_PT = 111;
-
-export interface IRtpTranscoder {
-  /** Transcode an RTP payload. Always async (Rust IPC). */
-  payload(data: Buffer): Promise<Buffer>;
-  fromPT: number;
-  toPT: number;
-}
-
-/**
- * Create a transcoder that converts RTP payloads between two codecs.
- * Returns null if the codecs are the same or the Rust bridge isn't ready.
- *
- * @param sessionId - optional Rust codec session for isolated state per call
- */
-export function createTranscoder(fromPT: number, toPT: number, sessionId?: string, direction?: string): IRtpTranscoder | null {
-  if (fromPT === toPT) return null;
-  if (!isCodecReady()) return null;
-
-  return {
-    fromPT,
-    toPT,
-    async payload(data: Buffer): Promise<Buffer> {
-      const result = await transcode(data, fromPT, toPT, sessionId, direction);
-      return result || Buffer.alloc(0); // return empty on failure — never pass raw codec bytes
-    },
-  };
-}

ts/config.ts

@@ -8,7 +8,15 @@
 
 import fs from 'node:fs';
 import path from 'node:path';
-import type { IEndpoint } from './sip/index.ts';
+
+// ---------------------------------------------------------------------------
+// Shared types (previously in ts/sip/types.ts, now inlined)
+// ---------------------------------------------------------------------------
+
+export interface IEndpoint {
+  address: string;
+  port: number;
+}
 
 // ---------------------------------------------------------------------------
 // Config interfaces
@@ -78,6 +86,17 @@ export interface ISipRouteAction {
   /** Also ring connected browser clients. Default false. */
   ringBrowsers?: boolean;
 
+  // --- Inbound actions (IVR / voicemail) ---
+
+  /** Route directly to a voicemail box (skip ringing devices). */
+  voicemailBox?: string;
+
+  /** Route to an IVR menu by menu ID (skip ringing devices). */
+  ivrMenuId?: string;
+
+  /** Override no-answer timeout (seconds) before routing to voicemail. */
+  noAnswerTimeout?: number;
+
   // --- Outbound actions (provider selection) ---
 
   /** Provider ID to use for outbound. */
@@ -137,12 +156,95 @@ export interface IContact {
   starred?: boolean;
 }
 
+// ---------------------------------------------------------------------------
+// Voicebox configuration
+// ---------------------------------------------------------------------------
+
+export interface IVoiceboxConfig {
+  /** Unique ID — typically matches device ID or extension. */
+  id: string;
+  /** Whether this voicebox is active. */
+  enabled: boolean;
+  /** Custom TTS greeting text. */
+  greetingText?: string;
+  /** TTS voice ID (default 'af_bella'). */
+  greetingVoice?: string;
+  /** Path to uploaded WAV greeting (overrides TTS). */
+  greetingWavPath?: string;
+  /** Seconds to wait before routing to voicemail (default 25). */
+  noAnswerTimeoutSec?: number;
+  /** Maximum recording duration in seconds (default 120). */
+  maxRecordingSec?: number;
+  /** Maximum stored messages per box (default 50). */
+  maxMessages?: number;
+}
+
+// ---------------------------------------------------------------------------
+// IVR configuration
+// ---------------------------------------------------------------------------
+
+/** An action triggered by a digit press in an IVR menu. */
+export type TIvrAction =
+  | { type: 'route-extension'; extensionId: string }
+  | { type: 'route-voicemail'; boxId: string }
+  | { type: 'submenu'; menuId: string }
+  | { type: 'play-message'; promptId: string }
+  | { type: 'transfer'; number: string; providerId?: string }
+  | { type: 'repeat' }
+  | { type: 'hangup' };
+
+/** A single digit→action mapping in an IVR menu. */
+export interface IIvrMenuEntry {
+  /** Digit: '0'-'9', '*', '#'. */
+  digit: string;
+  /** Action to take when this digit is pressed. */
+  action: TIvrAction;
+}
+
+/** An IVR menu with a prompt and digit mappings. */
+export interface IIvrMenu {
+  /** Unique menu ID. */
+  id: string;
+  /** Human-readable name. */
+  name: string;
+  /** TTS text for the menu prompt. */
+  promptText: string;
+  /** TTS voice ID for the prompt. */
+  promptVoice?: string;
+  /** Digit→action entries. */
+  entries: IIvrMenuEntry[];
+  /** Seconds to wait for a digit after prompt finishes (default 5). */
+  timeoutSec?: number;
+  /** Maximum retries before executing timeout action (default 3). */
+  maxRetries?: number;
+  /** Action on timeout (no digit pressed). */
+  timeoutAction: TIvrAction;
+  /** Action on invalid digit. */
+  invalidAction: TIvrAction;
+}
+
+/** Top-level IVR configuration. */
+export interface IIvrConfig {
+  /** Whether the IVR system is active. */
+  enabled: boolean;
+  /** IVR menu definitions. */
+  menus: IIvrMenu[];
+  /** The menu to start with for incoming calls. */
+  entryMenuId: string;
+}
+
+// ---------------------------------------------------------------------------
+// App config
+// ---------------------------------------------------------------------------
+
 export interface IAppConfig {
   proxy: IProxyConfig;
   providers: IProviderConfig[];
   devices: IDeviceConfig[];
   routing: IRoutingConfig;
   contacts: IContact[];
+  voiceboxes?: IVoiceboxConfig[];
+  ivr?: IIvrConfig;
 }
 
 // ---------------------------------------------------------------------------
@@ -201,169 +303,29 @@ export function loadConfig(): IAppConfig {
     c.starred ??= false;
   }
 
+  // Voicebox defaults.
+  cfg.voiceboxes ??= [];
+  for (const vb of cfg.voiceboxes) {
+    vb.enabled ??= true;
+    vb.noAnswerTimeoutSec ??= 25;
+    vb.maxRecordingSec ??= 120;
+    vb.maxMessages ??= 50;
+    vb.greetingVoice ??= 'af_bella';
+  }
+
+  // IVR defaults.
+  if (cfg.ivr) {
+    cfg.ivr.enabled ??= false;
+    cfg.ivr.menus ??= [];
+    for (const menu of cfg.ivr.menus) {
+      menu.timeoutSec ??= 5;
+      menu.maxRetries ??= 3;
+      menu.entries ??= [];
+    }
+  }
+
   return cfg;
 }
 
-// ---------------------------------------------------------------------------
-// Pattern matching
-// ---------------------------------------------------------------------------
-
-/**
- * Test a value against a pattern string.
- * - undefined/empty pattern: matches everything (wildcard)
- * - Prefix: "pattern*" matches values starting with "pattern"
- * - Regex: "/pattern/" or "/pattern/i" compiles as RegExp
- * - Otherwise: exact match
- */
-export function matchesPattern(pattern: string | undefined, value: string): boolean {
-  if (!pattern) return true;
-
-  // Prefix match: "+49*"
-  if (pattern.endsWith('*')) {
-    return value.startsWith(pattern.slice(0, -1));
-  }
-
-  // Regex match: "/^\\+49/" or "/pattern/i"
-  if (pattern.startsWith('/')) {
-    const lastSlash = pattern.lastIndexOf('/');
-    if (lastSlash > 0) {
-      const re = new RegExp(pattern.slice(1, lastSlash), pattern.slice(lastSlash + 1));
-      return re.test(value);
-    }
-  }
-
-  // Exact match.
-  return value === pattern;
-}
-
-// ---------------------------------------------------------------------------
-// Route resolution
-// ---------------------------------------------------------------------------
-
-/** Result of resolving an outbound route. */
-export interface IOutboundRouteResult {
-  provider: IProviderConfig;
-  transformedNumber: string;
-}
-
-/** Result of resolving an inbound route. */
-export interface IInboundRouteResult {
-  /** Device IDs to ring (empty = all devices). */
-  deviceIds: string[];
-  ringBrowsers: boolean;
-}
-
-/**
- * Resolve which provider to use for an outbound call, and transform the number.
- *
- * @param cfg - app config
- * @param dialedNumber - the number being dialed
- * @param sourceDeviceId - optional device originating the call
- * @param isProviderRegistered - callback to check if a provider is currently registered
- */
-export function resolveOutboundRoute(
-  cfg: IAppConfig,
-  dialedNumber: string,
-  sourceDeviceId?: string,
-  isProviderRegistered?: (providerId: string) => boolean,
-): IOutboundRouteResult | null {
-  const routes = cfg.routing.routes
-    .filter((r) => r.enabled && r.match.direction === 'outbound')
-    .sort((a, b) => b.priority - a.priority);
-
-  for (const route of routes) {
-    const m = route.match;
-
-    if (!matchesPattern(m.numberPattern, dialedNumber)) continue;
-    if (m.sourceDevice && m.sourceDevice !== sourceDeviceId) continue;
-
-    // Find a registered provider (primary + failovers).
-    const candidates = [route.action.provider, ...(route.action.failoverProviders || [])].filter(Boolean) as string[];
-    for (const pid of candidates) {
-      const provider = getProvider(cfg, pid);
-      if (!provider) continue;
-      if (isProviderRegistered && !isProviderRegistered(pid)) continue;
-
-      // Apply number transformation.
-      let num = dialedNumber;
-      if (route.action.stripPrefix && num.startsWith(route.action.stripPrefix)) {
-        num = num.slice(route.action.stripPrefix.length);
-      }
-      if (route.action.prependPrefix) {
-        num = route.action.prependPrefix + num;
-      }
-
-      return { provider, transformedNumber: num };
-    }
-
-    // Route matched but no provider is available — continue to next route.
-  }
-
-  // Fallback: first available provider.
-  const fallback = cfg.providers[0];
-  return fallback ? { provider: fallback, transformedNumber: dialedNumber } : null;
-}
-
-/**
- * Resolve which devices/browsers to ring for an inbound call.
- *
- * @param cfg - app config
- * @param providerId - the provider the call is coming from
- * @param calledNumber - the DID / called number (from Request-URI)
- * @param callerNumber - the caller ID (from From header)
- */
-export function resolveInboundRoute(
-  cfg: IAppConfig,
-  providerId: string,
-  calledNumber: string,
-  callerNumber: string,
-): IInboundRouteResult {
-  const routes = cfg.routing.routes
-    .filter((r) => r.enabled && r.match.direction === 'inbound')
-    .sort((a, b) => b.priority - a.priority);
-
-  for (const route of routes) {
-    const m = route.match;
-
-    if (m.sourceProvider && m.sourceProvider !== providerId) continue;
-    if (!matchesPattern(m.numberPattern, calledNumber)) continue;
-    if (!matchesPattern(m.callerPattern, callerNumber)) continue;
-
-    return {
-      deviceIds: route.action.targets || [],
-      ringBrowsers: route.action.ringBrowsers ?? false,
-    };
-  }
-
-  // Fallback: ring all devices + browsers.
-  return { deviceIds: [], ringBrowsers: true };
-}
-
-// ---------------------------------------------------------------------------
-// Lookup helpers
-// ---------------------------------------------------------------------------
-
-export function getProvider(cfg: IAppConfig, id: string): IProviderConfig | null {
-  return cfg.providers.find((p) => p.id === id) ?? null;
-}
-
-export function getDevice(cfg: IAppConfig, id: string): IDeviceConfig | null {
-  return cfg.devices.find((d) => d.id === id) ?? null;
-}
-
-/**
- * @deprecated Use resolveOutboundRoute() instead. Kept for backward compat.
- */
-export function getProviderForOutbound(cfg: IAppConfig): IProviderConfig | null {
-  const result = resolveOutboundRoute(cfg, '');
-  return result?.provider ?? null;
-}
-
-/**
- * @deprecated Use resolveInboundRoute() instead. Kept for backward compat.
- */
-export function getDevicesForInbound(cfg: IAppConfig, providerId: string): IDeviceConfig[] {
-  const result = resolveInboundRoute(cfg, providerId, '', '');
-  if (!result.deviceIds.length) return cfg.devices;
-  return result.deviceIds.map((id) => getDevice(cfg, id)).filter(Boolean) as IDeviceConfig[];
-}
+// Route resolution, pattern matching, and provider/device lookup
+// are now handled entirely by the Rust proxy-engine.

ts/frontend.ts

@@ -11,8 +11,12 @@ import path from 'node:path';
 import http from 'node:http';
 import https from 'node:https';
 import { WebSocketServer, WebSocket } from 'ws';
-import type { CallManager } from './call/index.ts';
 import { handleWebRtcSignaling } from './webrtcbridge.ts';
+import type { VoiceboxManager } from './voicebox.ts';
+
+// CallManager was previously used for WebRTC call handling. Now replaced by Rust proxy-engine.
+// Kept as `any` type for backward compat with the function signature until full WebRTC port.
+type CallManager = any;
 
 const CONFIG_PATH = path.join(process.cwd(), '.nogit', 'config.json');
 
@@ -84,6 +88,7 @@ async function handleRequest(
   onHangupCall: (callId: string) => boolean,
   onConfigSaved?: () => void,
   callManager?: CallManager,
+  voiceboxManager?: VoiceboxManager,
 ): Promise<void> {
   const url = new URL(req.url || '/', `http://${req.headers.host || 'localhost'}`);
   const method = req.method || 'GET';
@@ -242,6 +247,8 @@ async function handleRequest(
         }
       }
       if (updates.contacts !== undefined) cfg.contacts = updates.contacts;
+      if (updates.voiceboxes !== undefined) cfg.voiceboxes = updates.voiceboxes;
+      if (updates.ivr !== undefined) cfg.ivr = updates.ivr;
 
       fs.writeFileSync(CONFIG_PATH, JSON.stringify(cfg, null, 2) + '\n');
       log('[config] updated config.json');
@@ -252,6 +259,50 @@ async function handleRequest(
     }
   }
 
+  // API: voicemail - list messages.
+  const vmListMatch = url.pathname.match(/^\/api\/voicemail\/([^/]+)$/);
+  if (vmListMatch && method === 'GET' && voiceboxManager) {
+    const boxId = vmListMatch[1];
+    return sendJson(res, { ok: true, messages: voiceboxManager.getMessages(boxId) });
+  }
+
+  // API: voicemail - unheard count.
+  const vmUnheardMatch = url.pathname.match(/^\/api\/voicemail\/([^/]+)\/unheard$/);
+  if (vmUnheardMatch && method === 'GET' && voiceboxManager) {
+    const boxId = vmUnheardMatch[1];
+    return sendJson(res, { ok: true, count: voiceboxManager.getUnheardCount(boxId) });
+  }
+
+  // API: voicemail - stream audio.
+  const vmAudioMatch = url.pathname.match(/^\/api\/voicemail\/([^/]+)\/([^/]+)\/audio$/);
+  if (vmAudioMatch && method === 'GET' && voiceboxManager) {
+    const [, boxId, msgId] = vmAudioMatch;
+    const audioPath = voiceboxManager.getMessageAudioPath(boxId, msgId);
+    if (!audioPath) return sendJson(res, { ok: false, error: 'not found' }, 404);
+    const stat = fs.statSync(audioPath);
+    res.writeHead(200, {
+      'Content-Type': 'audio/wav',
+      'Content-Length': stat.size.toString(),
+      'Accept-Ranges': 'bytes',
+    });
+    fs.createReadStream(audioPath).pipe(res);
+    return;
+  }
+
+  // API: voicemail - mark as heard.
+  const vmHeardMatch = url.pathname.match(/^\/api\/voicemail\/([^/]+)\/([^/]+)\/heard$/);
+  if (vmHeardMatch && method === 'POST' && voiceboxManager) {
+    const [, boxId, msgId] = vmHeardMatch;
+    return sendJson(res, { ok: voiceboxManager.markHeard(boxId, msgId) });
+  }
+
+  // API: voicemail - delete message.
+  const vmDeleteMatch = url.pathname.match(/^\/api\/voicemail\/([^/]+)\/([^/]+)$/);
+  if (vmDeleteMatch && method === 'DELETE' && voiceboxManager) {
+    const [, boxId, msgId] = vmDeleteMatch;
+    return sendJson(res, { ok: voiceboxManager.deleteMessage(boxId, msgId) });
+  }
+
   // Static files.
   const file = staticFiles.get(url.pathname);
   if (file) {
@@ -288,6 +339,11 @@ export function initWebUi(
   onHangupCall: (callId: string) => boolean,
   onConfigSaved?: () => void,
   callManager?: CallManager,
+  /** 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,
 ): void {
   const WEB_PORT = 3060;
 
@@ -303,12 +359,12 @@ export function initWebUi(
     const cert = fs.readFileSync(certPath, 'utf8');
     const key = fs.readFileSync(keyPath, 'utf8');
     server = https.createServer({ cert, key }, (req, res) =>
-      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager).catch(() => { res.writeHead(500); res.end(); }),
+      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager, voiceboxManager).catch(() => { res.writeHead(500); res.end(); }),
     );
     useTls = true;
   } catch {
     server = http.createServer((req, res) =>
-      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager).catch(() => { res.writeHead(500); res.end(); }),
+      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager, voiceboxManager).catch(() => { res.writeHead(500); res.end(); }),
     );
   }
 
@@ -323,17 +379,23 @@ export function initWebUi(
     socket.on('message', (raw) => {
       try {
         const msg = JSON.parse(raw.toString());
-        if (msg.type === 'webrtc-accept' && msg.callId) {
-          log(`[webrtc] browser accepted call ${msg.callId} session=${msg.sessionId || 'none'}`);
-          const ok = callManager?.acceptBrowserCall(msg.callId, msg.sessionId) ?? false;
-          log(`[webrtc] acceptBrowserCall result: ${ok}`);
-        } else if (msg.type === 'webrtc-offer' && msg.sessionId) {
-          callManager?.handleWebRtcOffer(msg.sessionId, msg.sdp, socket as any).catch((e: any) =>
-            log(`[webrtc] offer error: ${e.message}`));
+        if (msg.type === 'webrtc-offer' && msg.sessionId) {
+          // Forward to Rust proxy-engine for WebRTC handling.
+          if (onWebRtcOffer) {
+            onWebRtcOffer(msg.sessionId, msg.sdp, socket as any).catch((e: any) =>
+              log(`[webrtc] offer error: ${e.message}`));
+          }
         } else if (msg.type === 'webrtc-ice' && msg.sessionId) {
-          callManager?.handleWebRtcIce(msg.sessionId, msg.candidate).catch(() => {});
+          if (onWebRtcIce) {
+            onWebRtcIce(msg.sessionId, msg.candidate).catch(() => {});
+          }
         } else if (msg.type === 'webrtc-hangup' && msg.sessionId) {
-          callManager?.handleWebRtcHangup(msg.sessionId);
+          if (onWebRtcClose) {
+            onWebRtcClose(msg.sessionId).catch(() => {});
+          }
+        } else if (msg.type === 'webrtc-accept' && msg.callId) {
+          // TODO: Wire to Rust call linking.
+          log(`[webrtc] accept: call=${msg.callId} session=${msg.sessionId || 'none'}`);
         } else if (msg.type?.startsWith('webrtc-')) {
           msg._remoteIp = remoteIp;
           handleWebRtcSignaling(socket as any, msg);

ts/providerstate.ts

@@ -1,306 +0,0 @@
-/**
- * Per-provider runtime state and upstream registration.
- *
- * Each configured provider gets its own ProviderState instance tracking
- * registration status, public IP, and the periodic REGISTER cycle.
- */
-
-import { Buffer } from 'node:buffer';
-import {
-  SipMessage,
-  generateCallId,
-  generateTag,
-  generateBranch,
-  parseDigestChallenge,
-  computeDigestAuth,
-} from './sip/index.ts';
-import type { IEndpoint } from './sip/index.ts';
-import type { IProviderConfig } from './config.ts';
-
-// ---------------------------------------------------------------------------
-// Provider state
-// ---------------------------------------------------------------------------
-
-export class ProviderState {
-  readonly config: IProviderConfig;
-  publicIp: string | null;
-  isRegistered = false;
-  registeredAor: string;
-
-  // Registration transaction state.
-  private regCallId: string;
-  private regCSeq = 0;
-  private regFromTag: string;
-  private regTimer: ReturnType<typeof setInterval> | null = null;
-  private sendSip: ((buf: Buffer, dest: IEndpoint) => void) | null = null;
-  private logFn: ((msg: string) => void) | null = null;
-  private onRegistrationChange: ((provider: ProviderState) => void) | null = null;
-
-  constructor(config: IProviderConfig, publicIpSeed: string | null) {
-    this.config = config;
-    this.publicIp = publicIpSeed;
-    this.registeredAor = `sip:${config.username}@${config.domain}`;
-    this.regCallId = generateCallId();
-    this.regFromTag = generateTag();
-  }
-
-  private log(msg: string): void {
-    this.logFn?.(`[provider:${this.config.id}] ${msg}`);
-  }
-
-  // -------------------------------------------------------------------------
-  // Upstream registration
-  // -------------------------------------------------------------------------
-
-  /**
-   * Start the periodic REGISTER cycle with this provider.
-   */
-  startRegistration(
-    lanIp: string,
-    lanPort: number,
-    sendSip: (buf: Buffer, dest: IEndpoint) => void,
-    log: (msg: string) => void,
-    onRegistrationChange: (provider: ProviderState) => void,
-  ): void {
-    this.sendSip = sendSip;
-    this.logFn = log;
-    this.onRegistrationChange = onRegistrationChange;
-
-    // Initial registration.
-    this.sendRegister(lanIp, lanPort);
-
-    // Re-register periodically.
-    const intervalMs = (this.config.registerIntervalSec * 0.85) * 1000;
-    this.regTimer = setInterval(() => this.sendRegister(lanIp, lanPort), intervalMs);
-  }
-
-  stopRegistration(): void {
-    if (this.regTimer) {
-      clearInterval(this.regTimer);
-      this.regTimer = null;
-    }
-  }
-
-  private sendRegister(lanIp: string, lanPort: number): void {
-    this.regCSeq++;
-    const pub = this.publicIp || lanIp;
-    const { config } = this;
-
-    const register = SipMessage.createRequest('REGISTER', `sip:${config.domain}`, {
-      via: { host: pub, port: lanPort },
-      from: { uri: this.registeredAor, tag: this.regFromTag },
-      to: { uri: this.registeredAor },
-      callId: this.regCallId,
-      cseq: this.regCSeq,
-      contact: `<sip:${config.username}@${pub}:${lanPort}>`,
-      maxForwards: 70,
-      extraHeaders: [
-        ['Expires', String(config.registerIntervalSec)],
-        ['User-Agent', 'SipRouter/1.0'],
-        ['Allow', 'INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE'],
-      ],
-    });
-
-    this.log(`REGISTER -> ${config.outboundProxy.address}:${config.outboundProxy.port} (CSeq ${this.regCSeq})`);
-    this.sendSip!(register.serialize(), config.outboundProxy);
-  }
-
-  /**
-   * Handle an incoming SIP response that belongs to this provider's registration.
-   * Returns true if the message was consumed.
-   */
-  handleRegistrationResponse(msg: SipMessage): boolean {
-    if (!msg.isResponse) return false;
-    if (msg.callId !== this.regCallId) return false;
-    if (msg.cseqMethod?.toUpperCase() !== 'REGISTER') return false;
-
-    const code = msg.statusCode ?? 0;
-    this.log(`REGISTER <- ${code}`);
-
-    if (code === 200) {
-      const wasRegistered = this.isRegistered;
-      this.isRegistered = true;
-      if (!wasRegistered) {
-        this.log('registered');
-        this.onRegistrationChange?.(this);
-      }
-      return true;
-    }
-
-    if (code === 401 || code === 407) {
-      const challengeHeader = code === 401
-        ? msg.getHeader('WWW-Authenticate')
-        : msg.getHeader('Proxy-Authenticate');
-
-      if (!challengeHeader) {
-        this.log(`${code} but no challenge header`);
-        return true;
-      }
-
-      const challenge = parseDigestChallenge(challengeHeader);
-      if (!challenge) {
-        this.log(`${code} could not parse digest challenge`);
-        return true;
-      }
-
-      const authValue = computeDigestAuth({
-        username: this.config.username,
-        password: this.config.password,
-        realm: challenge.realm,
-        nonce: challenge.nonce,
-        method: 'REGISTER',
-        uri: `sip:${this.config.domain}`,
-        algorithm: challenge.algorithm,
-        opaque: challenge.opaque,
-      });
-
-      // Resend REGISTER with auth.
-      this.regCSeq++;
-      const pub = this.publicIp || 'unknown';
-      // We need lanIp/lanPort but don't have them here — reconstruct from Via.
-      const via = msg.getHeader('Via') || '';
-      const viaHost = via.match(/SIP\/2\.0\/UDP\s+([^;:]+)/)?.[1] || pub;
-      const viaPort = parseInt(via.match(/:(\d+)/)?.[1] || '5070', 10);
-
-      const register = SipMessage.createRequest('REGISTER', `sip:${this.config.domain}`, {
-        via: { host: viaHost, port: viaPort },
-        from: { uri: this.registeredAor, tag: this.regFromTag },
-        to: { uri: this.registeredAor },
-        callId: this.regCallId,
-        cseq: this.regCSeq,
-        contact: `<sip:${this.config.username}@${viaHost}:${viaPort}>`,
-        maxForwards: 70,
-        extraHeaders: [
-          [code === 401 ? 'Authorization' : 'Proxy-Authorization', authValue],
-          ['Expires', String(this.config.registerIntervalSec)],
-          ['User-Agent', 'SipRouter/1.0'],
-          ['Allow', 'INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE'],
-        ],
-      });
-
-      this.log(`REGISTER -> (with auth, CSeq ${this.regCSeq})`);
-      this.sendSip!(register.serialize(), this.config.outboundProxy);
-      return true;
-    }
-
-    if (code >= 400) {
-      const wasRegistered = this.isRegistered;
-      this.isRegistered = false;
-      if (wasRegistered) {
-        this.log(`registration lost (${code})`);
-        this.onRegistrationChange?.(this);
-      }
-      return true;
-    }
-
-    return true; // consume 1xx etc.
-  }
-
-  /**
-   * Update public IP from Via received= parameter.
-   */
-  detectPublicIp(via: string): void {
-    const m = via.match(/received=([\d.]+)/);
-    if (m && m[1] !== this.publicIp) {
-      this.log(`publicIp = ${m[1]}`);
-      this.publicIp = m[1];
-    }
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Provider state management
-// ---------------------------------------------------------------------------
-
-let providerStates: Map<string, ProviderState>;
-
-export function initProviderStates(
-  providers: IProviderConfig[],
-  publicIpSeed: string | null,
-): Map<string, ProviderState> {
-  providerStates = new Map();
-  for (const p of providers) {
-    providerStates.set(p.id, new ProviderState(p, publicIpSeed));
-  }
-  return providerStates;
-}
-
-export function getProviderState(id: string): ProviderState | null {
-  return providerStates?.get(id) ?? null;
-}
-
-/**
- * Sync running provider states with updated config.
- * - New providers: create state + start registration.
- * - Removed providers: stop registration + delete state.
- * - Changed providers: stop old, create new, start registration (preserves detected publicIp).
- */
-export function syncProviderStates(
-  newProviders: IProviderConfig[],
-  publicIpSeed: string | null,
-  lanIp: string,
-  lanPort: number,
-  sendSip: (buf: Buffer, dest: IEndpoint) => void,
-  log: (msg: string) => void,
-  onRegistrationChange: (provider: ProviderState) => void,
-): void {
-  if (!providerStates) return;
-
-  const newIds = new Set(newProviders.map(p => p.id));
-  const oldIds = new Set(providerStates.keys());
-
-  // Remove providers no longer in config.
-  for (const id of oldIds) {
-    if (!newIds.has(id)) {
-      const ps = providerStates.get(id)!;
-      ps.stopRegistration();
-      providerStates.delete(id);
-      log(`[provider:${id}] removed`);
-    }
-  }
-
-  for (const p of newProviders) {
-    if (!oldIds.has(p.id)) {
-      // New provider.
-      const ps = new ProviderState(p, publicIpSeed);
-      providerStates.set(p.id, ps);
-      ps.startRegistration(lanIp, lanPort, sendSip, log, onRegistrationChange);
-      log(`[provider:${p.id}] added — registration started`);
-    } else {
-      // Existing provider — check if config changed.
-      const existing = providerStates.get(p.id)!;
-      if (JSON.stringify(existing.config) !== JSON.stringify(p)) {
-        existing.stopRegistration();
-        const ps = new ProviderState(p, existing.publicIp || publicIpSeed);
-        providerStates.set(p.id, ps);
-        ps.startRegistration(lanIp, lanPort, sendSip, log, onRegistrationChange);
-        log(`[provider:${p.id}] config changed — re-registering`);
-      }
-    }
-  }
-}
-
-/**
- * Find which provider sent a packet, by matching the source address
- * against all providers' outbound proxy addresses.
- */
-export function getProviderByUpstreamAddress(address: string, port: number): ProviderState | null {
-  if (!providerStates) return null;
-  for (const ps of providerStates.values()) {
-    if (ps.config.outboundProxy.address === address && ps.config.outboundProxy.port === port) {
-      return ps;
-    }
-  }
-  return null;
-}
-
-/**
- * Check whether a response belongs to any provider's registration transaction.
- */
-export function handleProviderRegistrationResponse(msg: SipMessage): boolean {
-  if (!providerStates || !msg.isResponse) return false;
-  for (const ps of providerStates.values()) {
-    if (ps.handleRegistrationResponse(msg)) return true;
-  }
-  return false;
-}

ts/proxybridge.ts

@@ -0,0 +1,274 @@
+/**
+ * Proxy engine bridge — manages the Rust proxy-engine subprocess.
+ *
+ * The proxy-engine handles ALL SIP protocol mechanics. TypeScript only:
+ * - Sends configuration
+ * - Receives high-level events (incoming_call, call_ended, etc.)
+ * - Sends high-level commands (hangup, make_call, play_audio)
+ *
+ * No raw SIP ever touches TypeScript.
+ */
+
+import path from 'node:path';
+import { RustBridge } from '@push.rocks/smartrust';
+
+// ---------------------------------------------------------------------------
+// Command type map for smartrust
+// ---------------------------------------------------------------------------
+
+type TProxyCommands = {
+  configure: {
+    params: Record<string, unknown>;
+    result: { bound: string };
+  };
+  hangup: {
+    params: { call_id: string };
+    result: Record<string, never>;
+  };
+  make_call: {
+    params: { number: string; device_id?: string; provider_id?: string };
+    result: { call_id: string };
+  };
+  play_audio: {
+    params: { call_id: string; leg_id?: string; file_path: string; codec?: number };
+    result: Record<string, never>;
+  };
+  start_recording: {
+    params: { call_id: string; file_path: string; max_duration_ms?: number };
+    result: Record<string, never>;
+  };
+  stop_recording: {
+    params: { call_id: string };
+    result: { file_path: string; duration_ms: number };
+  };
+};
+
+// ---------------------------------------------------------------------------
+// Event types from Rust
+// ---------------------------------------------------------------------------
+
+export interface IIncomingCallEvent {
+  call_id: string;
+  from_uri: string;
+  to_number: string;
+  provider_id: string;
+}
+
+export interface IOutboundCallEvent {
+  call_id: string;
+  from_device: string | null;
+  to_number: string;
+}
+
+export interface ICallEndedEvent {
+  call_id: string;
+  reason: string;
+  duration: number;
+  from_side?: string;
+}
+
+export interface IProviderRegisteredEvent {
+  provider_id: string;
+  registered: boolean;
+  public_ip: string | null;
+}
+
+export interface IDeviceRegisteredEvent {
+  device_id: string;
+  display_name: string;
+  address: string;
+  port: number;
+  aor: string;
+  expires: number;
+}
+
+// ---------------------------------------------------------------------------
+// Bridge singleton
+// ---------------------------------------------------------------------------
+
+let bridge: RustBridge<TProxyCommands> | null = null;
+let initialized = false;
+let logFn: ((msg: string) => void) | undefined;
+
+function buildLocalPaths(): string[] {
+  const root = process.cwd();
+  return [
+    path.join(root, 'dist_rust', 'proxy-engine'),
+    path.join(root, 'rust', 'target', 'release', 'proxy-engine'),
+    path.join(root, 'rust', 'target', 'debug', 'proxy-engine'),
+  ];
+}
+
+/**
+ * Initialize the proxy engine — spawn the Rust binary.
+ * Call configure() separately to push config and start SIP.
+ */
+export async function initProxyEngine(log?: (msg: string) => void): Promise<boolean> {
+  if (initialized && bridge) return true;
+  logFn = log;
+
+  try {
+    bridge = new RustBridge<TProxyCommands>({
+      binaryName: 'proxy-engine',
+      localPaths: buildLocalPaths(),
+    });
+
+    const spawned = await bridge.spawn();
+    if (!spawned) {
+      log?.('[proxy-engine] failed to spawn binary');
+      bridge = null;
+      return false;
+    }
+
+    bridge.on('exit', () => {
+      logFn?.('[proxy-engine] process exited — will need re-init');
+      bridge = null;
+      initialized = false;
+    });
+
+    // Forward stderr for debugging.
+    bridge.on('stderr', (line: string) => {
+      logFn?.(`[proxy-engine:stderr] ${line}`);
+    });
+
+    initialized = true;
+    log?.('[proxy-engine] spawned and ready');
+    return true;
+  } catch (e: any) {
+    log?.(`[proxy-engine] init error: ${e.message}`);
+    bridge = null;
+    return false;
+  }
+}
+
+/**
+ * Send the full app config to the proxy engine.
+ * This binds the SIP socket, starts provider registrations, etc.
+ */
+export async function configureProxyEngine(config: Record<string, unknown>): Promise<boolean> {
+  if (!bridge || !initialized) return false;
+  try {
+    const result = await bridge.sendCommand('configure', config as any);
+    logFn?.(`[proxy-engine] configured, SIP bound on ${(result as any)?.bound || '?'}`);
+    return true;
+  } catch (e: any) {
+    logFn?.(`[proxy-engine] configure error: ${e.message}`);
+    return false;
+  }
+}
+
+/**
+ * Initiate an outbound call via Rust. Returns the call ID or null on failure.
+ */
+export async function makeCall(number: string, deviceId?: string, providerId?: string): Promise<string | null> {
+  if (!bridge || !initialized) return null;
+  try {
+    const result = await bridge.sendCommand('make_call', {
+      number,
+      device_id: deviceId,
+      provider_id: providerId,
+    } as any);
+    return (result as any)?.call_id || null;
+  } catch (e: any) {
+    logFn?.(`[proxy-engine] make_call error: ${e?.message || e}`);
+    return null;
+  }
+}
+
+/**
+ * Send a hangup command.
+ */
+export async function hangupCall(callId: string): Promise<boolean> {
+  if (!bridge || !initialized) return false;
+  try {
+    await bridge.sendCommand('hangup', { call_id: callId } as any);
+    return true;
+  } catch {
+    return false;
+  }
+}
+
+/**
+ * Send a WebRTC offer to the proxy engine. Returns the SDP answer.
+ */
+export async function webrtcOffer(sessionId: string, sdp: string): Promise<{ sdp: string } | null> {
+  if (!bridge || !initialized) return null;
+  try {
+    const result = await bridge.sendCommand('webrtc_offer', { session_id: sessionId, sdp } as any);
+    return result as any;
+  } catch (e: any) {
+    logFn?.(`[proxy-engine] webrtc_offer error: ${e?.message || e}`);
+    return null;
+  }
+}
+
+/**
+ * Forward an ICE candidate to the proxy engine.
+ */
+export async function webrtcIce(sessionId: string, candidate: any): Promise<void> {
+  if (!bridge || !initialized) return;
+  try {
+    await bridge.sendCommand('webrtc_ice', {
+      session_id: sessionId,
+      candidate: candidate?.candidate || candidate,
+      sdp_mid: candidate?.sdpMid,
+      sdp_mline_index: candidate?.sdpMLineIndex,
+    } as any);
+  } catch { /* ignore */ }
+}
+
+/**
+ * Link a WebRTC session to a SIP call — enables audio bridging.
+ * The browser's Opus audio will be transcoded and sent to the provider.
+ */
+export async function webrtcLink(sessionId: string, callId: string, providerMediaAddr: string, providerMediaPort: number, sipPt: number = 9): Promise<boolean> {
+  if (!bridge || !initialized) return false;
+  try {
+    await bridge.sendCommand('webrtc_link', {
+      session_id: sessionId,
+      call_id: callId,
+      provider_media_addr: providerMediaAddr,
+      provider_media_port: providerMediaPort,
+      sip_pt: sipPt,
+    } as any);
+    return true;
+  } catch (e: any) {
+    logFn?.(`[proxy-engine] webrtc_link error: ${e?.message || e}`);
+    return false;
+  }
+}
+
+/**
+ * Close a WebRTC session.
+ */
+export async function webrtcClose(sessionId: string): Promise<void> {
+  if (!bridge || !initialized) return;
+  try {
+    await bridge.sendCommand('webrtc_close', { session_id: sessionId } as any);
+  } catch { /* ignore */ }
+}
+
+/**
+ * Subscribe to an event from the proxy engine.
+ * Event names: incoming_call, outbound_device_call, call_ringing,
+ * call_answered, call_ended, provider_registered, device_registered,
+ * dtmf_digit, recording_done, sip_unhandled
+ */
+export function onProxyEvent(event: string, handler: (data: any) => void): void {
+  if (!bridge) throw new Error('proxy engine not initialized');
+  bridge.on(`management:${event}`, handler);
+}
+
+/** Check if the proxy engine is ready. */
+export function isProxyReady(): boolean {
+  return initialized && bridge !== null;
+}
+
+/** Shut down the proxy engine. */
+export function shutdownProxyEngine(): void {
+  if (bridge) {
+    try { bridge.kill(); } catch { /* ignore */ }
+    bridge = null;
+    initialized = false;
+  }
+}

ts/registrar.ts

@@ -1,118 +1,27 @@
 /**
- * Local SIP registrar — accepts REGISTER from devices and browser clients.
+ * Browser device registration.
  *
- * Devices point their SIP registration at the proxy instead of the upstream
- * provider.  The registrar responds with 200 OK and stores the device's
- * current contact (source IP:port).  Browser softphones register via
- * WebSocket signaling.
+ * SIP device registration is now handled entirely by the Rust proxy-engine.
+ * This module only handles browser softphone registration via WebSocket.
  */
 
 import { createHash } from 'node:crypto';
-import {
-  SipMessage,
-  generateTag,
-} from './sip/index.ts';
 
 /** Hash a string to a 6-char hex ID. */
 export function shortHash(input: string): string {
   return createHash('sha256').update(input).digest('hex').slice(0, 6);
 }
-import type { IEndpoint } from './sip/index.ts';
-import type { IDeviceConfig } from './config.ts';
 
 // ---------------------------------------------------------------------------
-// Types
+// Browser device registration
 // ---------------------------------------------------------------------------
 
-export interface IRegisteredDevice {
-  deviceConfig: IDeviceConfig;
-  contact: IEndpoint | null;
-  registeredAt: number;
-  expiresAt: number;
-  aor: string;
-  connected: boolean;
-  isBrowser: boolean;
-}
-
-export interface IDeviceStatusEntry {
-  id: string;
-  displayName: string;
-  contact: IEndpoint | null;
-  aor: string;
-  connected: boolean;
-  isBrowser: boolean;
-}
-
-// ---------------------------------------------------------------------------
-// State
-// ---------------------------------------------------------------------------
-
-const registeredDevices = new Map<string, IRegisteredDevice>();
-const browserDevices = new Map<string, IRegisteredDevice>();
-let knownDevices: IDeviceConfig[] = [];
-let logFn: (msg: string) => void = () => {};
-
-// ---------------------------------------------------------------------------
-// Public API
-// ---------------------------------------------------------------------------
-
-export function initRegistrar(
-  devices: IDeviceConfig[],
-  log: (msg: string) => void,
-): void {
-  knownDevices = devices;
-  logFn = log;
-}
-
-/**
- * Process a REGISTER from a SIP device.  Returns a 200 OK response to send back,
- * or null if this REGISTER should not be handled by the local registrar.
- */
-export function handleDeviceRegister(
-  msg: SipMessage,
-  rinfo: IEndpoint,
-): SipMessage | null {
-  if (msg.method !== 'REGISTER') return null;
-
-  const device = knownDevices.find((d) => d.expectedAddress === rinfo.address);
-  if (!device) return null;
-
-  const from = msg.getHeader('From');
-  const aor = from ? SipMessage.extractUri(from) || `sip:${device.extension}@${rinfo.address}` : `sip:${device.extension}@${rinfo.address}`;
-
-  const MAX_EXPIRES = 300;
-  const expiresHeader = msg.getHeader('Expires');
-  const requested = expiresHeader ? parseInt(expiresHeader, 10) : 3600;
-  const expires = Math.min(requested, MAX_EXPIRES);
-
-  const entry: IRegisteredDevice = {
-    deviceConfig: device,
-    contact: { address: rinfo.address, port: rinfo.port },
-    registeredAt: Date.now(),
-    expiresAt: Date.now() + expires * 1000,
-    aor,
-    connected: true,
-    isBrowser: false,
-  };
-  registeredDevices.set(device.id, entry);
-
-  logFn(`[registrar] ${device.displayName} (${device.id}) registered from ${rinfo.address}:${rinfo.port} expires=${expires}`);
-
-  const contact = msg.getHeader('Contact') || `<sip:${rinfo.address}:${rinfo.port}>`;
-  const response = SipMessage.createResponse(200, 'OK', msg, {
-    toTag: generateTag(),
-    contact,
-    extraHeaders: [['Expires', String(expires)]],
-  });
-
-  return response;
-}
+const browserDevices = new Map<string, { deviceId: string; displayName: string; remoteIp: string }>();
 
 /**
  * Register a browser softphone as a device.
  */
 export function registerBrowserDevice(sessionId: string, userAgent?: string, remoteIp?: string): void {
-  // Extract a short browser name from the UA string.
   let browserName = 'Browser';
   if (userAgent) {
     if (userAgent.includes('Firefox/')) browserName = 'Firefox';
@@ -121,21 +30,11 @@ export function registerBrowserDevice(sessionId: string, userAgent?: string, rem
     else if (userAgent.includes('Safari/') && !userAgent.includes('Chrome/')) browserName = 'Safari';
   }
 
-  const entry: IRegisteredDevice = {
-    deviceConfig: {
-      id: `browser-${shortHash(sessionId)}`,
-      displayName: browserName,
-      expectedAddress: remoteIp || '127.0.0.1',
-      extension: 'webrtc',
-    },
-    contact: null,
-    registeredAt: Date.now(),
-    expiresAt: Date.now() + 60 * 1000, // 60s — browser must re-register to stay alive
-    aor: `sip:webrtc@browser`,
-    connected: true,
-    isBrowser: true,
-  };
-  browserDevices.set(sessionId, entry);
+  browserDevices.set(sessionId, {
+    deviceId: `browser-${shortHash(sessionId)}`,
+    displayName: browserName,
+    remoteIp: remoteIp || '127.0.0.1',
+  });
 }
 
 /**
@@ -144,96 +43,3 @@ export function registerBrowserDevice(sessionId: string, userAgent?: string, rem
 export function unregisterBrowserDevice(sessionId: string): void {
   browserDevices.delete(sessionId);
 }
-
-/**
- * Get a registered device by its config ID.
- */
-export function getRegisteredDevice(deviceId: string): IRegisteredDevice | null {
-  const entry = registeredDevices.get(deviceId);
-  if (!entry) return null;
-  if (Date.now() > entry.expiresAt) {
-    registeredDevices.delete(deviceId);
-    return null;
-  }
-  return entry;
-}
-
-/**
- * Get a registered device by source IP address.
- */
-export function getRegisteredDeviceByAddress(address: string): IRegisteredDevice | null {
-  for (const entry of registeredDevices.values()) {
-    if (entry.contact?.address === address && Date.now() <= entry.expiresAt) {
-      return entry;
-    }
-  }
-  return null;
-}
-
-/**
- * Check whether an address belongs to a known device (by config expectedAddress).
- */
-export function isKnownDeviceAddress(address: string): boolean {
-  return knownDevices.some((d) => d.expectedAddress === address);
-}
-
-/**
- * Get all devices for the dashboard.
- * - Configured devices always show (connected or not).
- * - Browser devices only show while connected.
- */
-export function getAllDeviceStatuses(): IDeviceStatusEntry[] {
-  const now = Date.now();
-  const result: IDeviceStatusEntry[] = [];
-
-  // Configured devices — always show.
-  for (const dc of knownDevices) {
-    const reg = registeredDevices.get(dc.id);
-    const connected = reg ? now <= reg.expiresAt : false;
-    if (reg && now > reg.expiresAt) {
-      registeredDevices.delete(dc.id);
-    }
-    result.push({
-      id: dc.id,
-      displayName: dc.displayName,
-      contact: connected && reg ? reg.contact : null,
-      aor: reg?.aor || `sip:${dc.extension}@${dc.expectedAddress}`,
-      connected,
-      isBrowser: false,
-    });
-  }
-
-  // Browser devices — only while connected.
-  for (const [, entry] of browserDevices) {
-    const ip = entry.deviceConfig.expectedAddress;
-    result.push({
-      id: entry.deviceConfig.id,
-      displayName: entry.deviceConfig.displayName,
-      contact: ip && ip !== '127.0.0.1' ? { address: ip, port: 0 } : null,
-      aor: entry.aor,
-      connected: true,
-      isBrowser: true,
-    });
-  }
-
-  return result;
-}
-
-/**
- * Get all currently registered (connected) SIP devices.
- */
-export function getAllRegisteredDevices(): IRegisteredDevice[] {
-  const now = Date.now();
-  const result: IRegisteredDevice[] = [];
-  for (const [id, entry] of registeredDevices) {
-    if (now > entry.expiresAt) {
-      registeredDevices.delete(id);
-    } else {
-      result.push(entry);
-    }
-  }
-  for (const [, entry] of browserDevices) {
-    result.push(entry);
-  }
-  return result;
-}

ts/sip/dialog.ts

@@ -1,280 +0,0 @@
-/**
- * SipDialog — tracks the state of a SIP dialog (RFC 3261 §12).
- *
- * A dialog is created by a dialog-establishing request (INVITE, SUBSCRIBE, …)
- * and its 1xx/2xx response.  It manages local/remote tags, CSeq counters,
- * the route set, and provides helpers to build in-dialog requests (ACK, BYE,
- * re-INVITE, …).
- *
- * Usage:
- * ```ts
- * // Caller (UAC) side — create from the outgoing INVITE we just sent:
- * const dialog = SipDialog.fromUacInvite(invite);
- *
- * // When a 200 OK arrives:
- * dialog.processResponse(response200);
- *
- * // Build ACK for the 2xx:
- * const ack = dialog.createAck();
- *
- * // Later — hang up:
- * const bye = dialog.createRequest('BYE');
- * ```
- */
-
-import { SipMessage } from './message.ts';
-import { generateTag, generateBranch } from './helpers.ts';
-import { Buffer } from 'node:buffer';
-
-export type TDialogState = 'early' | 'confirmed' | 'terminated';
-
-export class SipDialog {
-  callId: string;
-  localTag: string;
-  remoteTag: string | null = null;
-  localUri: string;
-  remoteUri: string;
-  localCSeq: number;
-  remoteCSeq: number = 0;
-  routeSet: string[] = [];
-  remoteTarget: string;  // Contact URI of the remote party
-  state: TDialogState = 'early';
-
-  // Transport info for sending in-dialog messages.
-  localHost: string;
-  localPort: number;
-
-  constructor(options: {
-    callId: string;
-    localTag: string;
-    remoteTag?: string;
-    localUri: string;
-    remoteUri: string;
-    localCSeq: number;
-    remoteTarget: string;
-    localHost: string;
-    localPort: number;
-    routeSet?: string[];
-  }) {
-    this.callId = options.callId;
-    this.localTag = options.localTag;
-    this.remoteTag = options.remoteTag ?? null;
-    this.localUri = options.localUri;
-    this.remoteUri = options.remoteUri;
-    this.localCSeq = options.localCSeq;
-    this.remoteTarget = options.remoteTarget;
-    this.localHost = options.localHost;
-    this.localPort = options.localPort;
-    this.routeSet = options.routeSet ?? [];
-  }
-
-  // -------------------------------------------------------------------------
-  // Factory: create dialog from an outgoing INVITE (UAC side)
-  // -------------------------------------------------------------------------
-
-  /**
-   * Create a dialog from an INVITE we are sending.
-   * The dialog enters "early" state; call `processResponse()` when
-   * provisional or final responses arrive.
-   */
-  static fromUacInvite(invite: SipMessage, localHost: string, localPort: number): SipDialog {
-    const from = invite.getHeader('From') || '';
-    const to = invite.getHeader('To') || '';
-    return new SipDialog({
-      callId: invite.callId,
-      localTag: SipMessage.extractTag(from) || generateTag(),
-      localUri: SipMessage.extractUri(from) || '',
-      remoteUri: SipMessage.extractUri(to) || '',
-      localCSeq: parseInt((invite.getHeader('CSeq') || '1').split(/\s+/)[0], 10),
-      remoteTarget: invite.requestUri || SipMessage.extractUri(to) || '',
-      localHost: localHost,
-      localPort: localPort,
-    });
-  }
-
-  // -------------------------------------------------------------------------
-  // Factory: create dialog from an incoming INVITE (UAS side)
-  // -------------------------------------------------------------------------
-
-  /**
-   * Create a dialog from an INVITE we received.
-   * Typically used when acting as a UAS (e.g. for call-back scenarios).
-   */
-  static fromUasInvite(invite: SipMessage, localTag: string, localHost: string, localPort: number): SipDialog {
-    const from = invite.getHeader('From') || '';
-    const to = invite.getHeader('To') || '';
-    const contact = invite.getHeader('Contact');
-    return new SipDialog({
-      callId: invite.callId,
-      localTag,
-      remoteTag: SipMessage.extractTag(from) || undefined,
-      localUri: SipMessage.extractUri(to) || '',
-      remoteUri: SipMessage.extractUri(from) || '',
-      localCSeq: 0,
-      remoteTarget: (contact ? SipMessage.extractUri(contact) : null) || SipMessage.extractUri(from) || '',
-      localHost,
-      localPort,
-    });
-  }
-
-  // -------------------------------------------------------------------------
-  // Response processing
-  // -------------------------------------------------------------------------
-
-  /**
-   * Update dialog state from a received response.
-   * - 1xx with To-tag → early dialog
-   * - 2xx → confirmed dialog
-   * - 3xx–6xx → terminated
-   */
-  processResponse(response: SipMessage): void {
-    const to = response.getHeader('To') || '';
-    const tag = SipMessage.extractTag(to);
-    const code = response.statusCode ?? 0;
-    // Always update remoteTag from 2xx (RFC 3261 §12.1.2: tag in 2xx is definitive).
-    if (tag && (code >= 200 && code < 300)) {
-      this.remoteTag = tag;
-    } else if (tag && !this.remoteTag) {
-      this.remoteTag = tag;
-    }
-
-    // Update remote target from Contact.
-    const contact = response.getHeader('Contact');
-    if (contact) {
-      const uri = SipMessage.extractUri(contact);
-      if (uri) this.remoteTarget = uri;
-    }
-
-    // Record-Route → route set (in reverse for UAC).
-    if (this.state === 'early') {
-      const rr: string[] = [];
-      for (const [n, v] of response.headers) {
-        if (n.toLowerCase() === 'record-route') rr.push(v);
-      }
-      if (rr.length) this.routeSet = rr.reverse();
-    }
-
-    if (code >= 200 && code < 300) {
-      this.state = 'confirmed';
-    } else if (code >= 300) {
-      this.state = 'terminated';
-    }
-  }
-
-  // -------------------------------------------------------------------------
-  // Request building
-  // -------------------------------------------------------------------------
-
-  /**
-   * Build an in-dialog request (BYE, re-INVITE, INFO, …).
-   * Automatically increments the local CSeq.
-   */
-  createRequest(method: string, options?: {
-    body?: string;
-    contentType?: string;
-    extraHeaders?: [string, string][];
-  }): SipMessage {
-    this.localCSeq++;
-    const branch = generateBranch();
-
-    const headers: [string, string][] = [
-      ['Via', `SIP/2.0/UDP ${this.localHost}:${this.localPort};branch=${branch};rport`],
-      ['From', `<${this.localUri}>;tag=${this.localTag}`],
-      ['To', `<${this.remoteUri}>${this.remoteTag ? `;tag=${this.remoteTag}` : ''}`],
-      ['Call-ID', this.callId],
-      ['CSeq', `${this.localCSeq} ${method}`],
-      ['Max-Forwards', '70'],
-    ];
-
-    // Route set → Route headers.
-    for (const route of this.routeSet) {
-      headers.push(['Route', route]);
-    }
-
-    headers.push(['Contact', `<sip:${this.localHost}:${this.localPort}>`]);
-
-    if (options?.extraHeaders) headers.push(...options.extraHeaders);
-
-    const body = options?.body || '';
-    if (body && options?.contentType) {
-      headers.push(['Content-Type', options.contentType]);
-    }
-    headers.push(['Content-Length', String(Buffer.byteLength(body, 'utf8'))]);
-
-    // Determine Request-URI from route set or remote target.
-    let ruri = this.remoteTarget;
-    if (this.routeSet.length) {
-      const topRoute = SipMessage.extractUri(this.routeSet[0]);
-      if (topRoute && topRoute.includes(';lr')) {
-        ruri = this.remoteTarget; // loose routing — RURI stays as remote target
-      } else if (topRoute) {
-        ruri = topRoute; // strict routing — top route becomes RURI
-      }
-    }
-
-    return new SipMessage(`${method} ${ruri} SIP/2.0`, headers, body);
-  }
-
-  /**
-   * Build an ACK for a 2xx response to INVITE (RFC 3261 §13.2.2.4).
-   * ACK for 2xx is a new transaction, so it gets its own Via/branch.
-   */
-  createAck(): SipMessage {
-    const branch = generateBranch();
-
-    const headers: [string, string][] = [
-      ['Via', `SIP/2.0/UDP ${this.localHost}:${this.localPort};branch=${branch};rport`],
-      ['From', `<${this.localUri}>;tag=${this.localTag}`],
-      ['To', `<${this.remoteUri}>${this.remoteTag ? `;tag=${this.remoteTag}` : ''}`],
-      ['Call-ID', this.callId],
-      ['CSeq', `${this.localCSeq} ACK`],
-      ['Max-Forwards', '70'],
-    ];
-
-    for (const route of this.routeSet) {
-      headers.push(['Route', route]);
-    }
-
-    headers.push(['Content-Length', '0']);
-
-    let ruri = this.remoteTarget;
-    if (this.routeSet.length) {
-      const topRoute = SipMessage.extractUri(this.routeSet[0]);
-      if (topRoute && topRoute.includes(';lr')) {
-        ruri = this.remoteTarget;
-      } else if (topRoute) {
-        ruri = topRoute;
-      }
-    }
-
-    return new SipMessage(`ACK ${ruri} SIP/2.0`, headers, '');
-  }
-
-  /**
-   * Build a CANCEL for the original INVITE (same branch, CSeq).
-   * Used before the dialog is confirmed.
-   */
-  createCancel(originalInvite: SipMessage): SipMessage {
-    const via = originalInvite.getHeader('Via') || '';
-    const from = originalInvite.getHeader('From') || '';
-    const to = originalInvite.getHeader('To') || '';
-
-    const headers: [string, string][] = [
-      ['Via', via],
-      ['From', from],
-      ['To', to],
-      ['Call-ID', this.callId],
-      ['CSeq', `${this.localCSeq} CANCEL`],
-      ['Max-Forwards', '70'],
-      ['Content-Length', '0'],
-    ];
-
-    const ruri = originalInvite.requestUri || this.remoteTarget;
-    return new SipMessage(`CANCEL ${ruri} SIP/2.0`, headers, '');
-  }
-
-  /** Transition the dialog to terminated state. */
-  terminate(): void {
-    this.state = 'terminated';
-  }
-}

ts/sip/helpers.ts

@@ -1,190 +0,0 @@
-/**
- * SIP helper utilities — ID generation and SDP construction.
- */
-
-import { randomBytes, createHash } from 'node:crypto';
-
-// ---------------------------------------------------------------------------
-// ID generators
-// ---------------------------------------------------------------------------
-
-/** Generate a random SIP Call-ID. */
-export function generateCallId(domain?: string): string {
-  const id = randomBytes(16).toString('hex');
-  return domain ? `${id}@${domain}` : id;
-}
-
-/** Generate a random SIP From/To tag. */
-export function generateTag(): string {
-  return randomBytes(8).toString('hex');
-}
-
-/** Generate a RFC 3261 compliant Via branch (starts with z9hG4bK magic cookie). */
-export function generateBranch(): string {
-  return `z9hG4bK-${randomBytes(8).toString('hex')}`;
-}
-
-// ---------------------------------------------------------------------------
-// Codec registry
-// ---------------------------------------------------------------------------
-
-const CODEC_NAMES: Record<number, string> = {
-  0: 'PCMU/8000',
-  3: 'GSM/8000',
-  4: 'G723/8000',
-  8: 'PCMA/8000',
-  9: 'G722/8000',
-  18: 'G729/8000',
-  101: 'telephone-event/8000',
-};
-
-/** Look up the rtpmap name for a static payload type. */
-export function codecName(pt: number): string {
-  return CODEC_NAMES[pt] || `unknown/${pt}`;
-}
-
-// ---------------------------------------------------------------------------
-// SDP builder
-// ---------------------------------------------------------------------------
-
-export interface ISdpOptions {
-  /** IP address for the c= and o= lines. */
-  ip: string;
-  /** Audio port for the m=audio line. */
-  port: number;
-  /** RTP payload type numbers (e.g. [9, 0, 8, 101]). */
-  payloadTypes?: number[];
-  /** SDP session ID (random if omitted). */
-  sessionId?: string;
-  /** Session name for the s= line (defaults to '-'). */
-  sessionName?: string;
-  /** Direction attribute (defaults to 'sendrecv'). */
-  direction?: 'sendrecv' | 'recvonly' | 'sendonly' | 'inactive';
-  /** Extra a= lines to append (without "a=" prefix). */
-  attributes?: string[];
-}
-
-/**
- * Build a minimal SDP body suitable for SIP INVITE offers/answers.
- *
- * ```ts
- * const sdp = buildSdp({
- *   ip: '192.168.5.66',
- *   port: 20000,
- *   payloadTypes: [9, 0, 101],
- * });
- * ```
- */
-export function buildSdp(options: ISdpOptions): string {
-  const {
-    ip,
-    port,
-    payloadTypes = [9, 0, 8, 101],
-    sessionId = String(Math.floor(Math.random() * 1e9)),
-    sessionName = '-',
-    direction = 'sendrecv',
-    attributes = [],
-  } = options;
-
-  const lines: string[] = [
-    'v=0',
-    `o=- ${sessionId} ${sessionId} IN IP4 ${ip}`,
-    `s=${sessionName}`,
-    `c=IN IP4 ${ip}`,
-    't=0 0',
-    `m=audio ${port} RTP/AVP ${payloadTypes.join(' ')}`,
-  ];
-
-  for (const pt of payloadTypes) {
-    const name = CODEC_NAMES[pt];
-    if (name) lines.push(`a=rtpmap:${pt} ${name}`);
-    if (pt === 101) lines.push(`a=fmtp:101 0-16`);
-  }
-
-  lines.push(`a=${direction}`);
-  for (const attr of attributes) lines.push(`a=${attr}`);
-  lines.push(''); // trailing CRLF
-
-  return lines.join('\r\n');
-}
-
-// ---------------------------------------------------------------------------
-// SIP Digest authentication (RFC 2617)
-// ---------------------------------------------------------------------------
-
-export interface IDigestChallenge {
-  realm: string;
-  nonce: string;
-  algorithm?: string;
-  opaque?: string;
-  qop?: string;
-}
-
-/**
- * Parse a `Proxy-Authenticate` or `WWW-Authenticate` header value
- * into its constituent fields.
- */
-export function parseDigestChallenge(header: string): IDigestChallenge | null {
-  if (!header.toLowerCase().startsWith('digest ')) return null;
-  const params = header.slice(7);
-  const get = (key: string): string | undefined => {
-    const re = new RegExp(`${key}\\s*=\\s*"([^"]*)"`, 'i');
-    const m = params.match(re);
-    if (m) return m[1];
-    // unquoted value
-    const re2 = new RegExp(`${key}\\s*=\\s*([^,\\s]+)`, 'i');
-    const m2 = params.match(re2);
-    return m2 ? m2[1] : undefined;
-  };
-  const realm = get('realm');
-  const nonce = get('nonce');
-  if (!realm || !nonce) return null;
-  return { realm, nonce, algorithm: get('algorithm'), opaque: get('opaque'), qop: get('qop') };
-}
-
-function md5(s: string): string {
-  return createHash('md5').update(s).digest('hex');
-}
-
-/**
- * Compute a SIP Digest `Proxy-Authorization` or `Authorization` header value.
- */
-export function computeDigestAuth(options: {
-  username: string;
-  password: string;
-  realm: string;
-  nonce: string;
-  method: string;
-  uri: string;
-  algorithm?: string;
-  opaque?: string;
-}): string {
-  const ha1 = md5(`${options.username}:${options.realm}:${options.password}`);
-  const ha2 = md5(`${options.method}:${options.uri}`);
-  const response = md5(`${ha1}:${options.nonce}:${ha2}`);
-
-  let header = `Digest username="${options.username}", realm="${options.realm}", ` +
-    `nonce="${options.nonce}", uri="${options.uri}", response="${response}", ` +
-    `algorithm=${options.algorithm || 'MD5'}`;
-  if (options.opaque) header += `, opaque="${options.opaque}"`;
-  return header;
-}
-
-/**
- * Parse the audio media port and connection address from an SDP body.
- * Returns null when no c= + m=audio pair is found.
- */
-export function parseSdpEndpoint(sdp: string): { address: string; port: number } | null {
-  let addr: string | null = null;
-  let port: number | null = null;
-  for (const raw of sdp.replace(/\r\n/g, '\n').split('\n')) {
-    const line = raw.trim();
-    if (line.startsWith('c=IN IP4 ')) {
-      addr = line.slice('c=IN IP4 '.length).trim();
-    } else if (line.startsWith('m=audio ')) {
-      const parts = line.split(' ');
-      if (parts.length >= 2) port = parseInt(parts[1], 10);
-    }
-  }
-  return addr && port ? { address: addr, port } : null;
-}

ts/sip/index.ts

@@ -1,16 +0,0 @@
-export { SipMessage } from './message.ts';
-export { SipDialog } from './dialog.ts';
-export type { TDialogState } from './dialog.ts';
-export { rewriteSipUri, rewriteSdp } from './rewrite.ts';
-export {
-  generateCallId,
-  generateTag,
-  generateBranch,
-  codecName,
-  buildSdp,
-  parseSdpEndpoint,
-  parseDigestChallenge,
-  computeDigestAuth,
-} from './helpers.ts';
-export type { ISdpOptions, IDigestChallenge } from './helpers.ts';
-export type { IEndpoint } from './types.ts';

ts/sip/message.ts

@@ -1,316 +0,0 @@
-/**
- * SipMessage — parse, inspect, mutate, and serialize SIP messages.
- *
- * Provides a fluent (builder-style) API so callers can chain header
- * manipulations before serializing:
- *
- *   const buf = SipMessage.parse(raw)!
- *     .setHeader('Contact', newContact)
- *     .prependHeader('Record-Route', rr)
- *     .updateContentLength()
- *     .serialize();
- */
-
-import { Buffer } from 'node:buffer';
-import { generateCallId, generateTag, generateBranch } from './helpers.ts';
-
-const SIP_FIRST_LINE_RE = /^(?:[A-Z]+\s+\S+\s+SIP\/\d\.\d|SIP\/\d\.\d\s+\d+\s+)/;
-
-export class SipMessage {
-  startLine: string;
-  headers: [string, string][];
-  body: string;
-
-  constructor(startLine: string, headers: [string, string][], body: string) {
-    this.startLine = startLine;
-    this.headers = headers;
-    this.body = body;
-  }
-
-  // -------------------------------------------------------------------------
-  // Parsing
-  // -------------------------------------------------------------------------
-
-  static parse(buf: Buffer): SipMessage | null {
-    if (!buf.length) return null;
-    if (buf[0] < 0x41 || buf[0] > 0x7a) return null;
-
-    let text: string;
-    try { text = buf.toString('utf8'); } catch { return null; }
-
-    let head: string;
-    let body: string;
-    let sep = text.indexOf('\r\n\r\n');
-    if (sep !== -1) {
-      head = text.slice(0, sep);
-      body = text.slice(sep + 4);
-    } else {
-      sep = text.indexOf('\n\n');
-      if (sep !== -1) {
-        head = text.slice(0, sep);
-        body = text.slice(sep + 2);
-      } else {
-        head = text;
-        body = '';
-      }
-    }
-
-    const lines = head.replace(/\r\n/g, '\n').split('\n');
-    if (!lines.length || !lines[0]) return null;
-    const startLine = lines[0];
-    if (!SIP_FIRST_LINE_RE.test(startLine)) return null;
-
-    const headers: [string, string][] = [];
-    for (const line of lines.slice(1)) {
-      if (!line.trim()) continue;
-      const colon = line.indexOf(':');
-      if (colon === -1) continue;
-      headers.push([line.slice(0, colon).trim(), line.slice(colon + 1).trim()]);
-    }
-    return new SipMessage(startLine, headers, body);
-  }
-
-  // -------------------------------------------------------------------------
-  // Serialization
-  // -------------------------------------------------------------------------
-
-  serialize(): Buffer {
-    const head = [this.startLine, ...this.headers.map(([n, v]) => `${n}: ${v}`)].join('\r\n') + '\r\n\r\n';
-    return Buffer.concat([Buffer.from(head, 'utf8'), Buffer.from(this.body || '', 'utf8')]);
-  }
-
-  // -------------------------------------------------------------------------
-  // Inspectors
-  // -------------------------------------------------------------------------
-
-  get isRequest(): boolean {
-    return !this.startLine.startsWith('SIP/');
-  }
-
-  get isResponse(): boolean {
-    return this.startLine.startsWith('SIP/');
-  }
-
-  /** Request method (INVITE, REGISTER, ...) or null for responses. */
-  get method(): string | null {
-    if (!this.isRequest) return null;
-    return this.startLine.split(' ')[0];
-  }
-
-  /** Response status code or null for requests. */
-  get statusCode(): number | null {
-    if (!this.isResponse) return null;
-    return parseInt(this.startLine.split(' ')[1], 10);
-  }
-
-  get callId(): string {
-    return this.getHeader('Call-ID') || 'noid';
-  }
-
-  /** Method from the CSeq header (e.g. "INVITE"). */
-  get cseqMethod(): string | null {
-    const cseq = this.getHeader('CSeq');
-    if (!cseq) return null;
-    const parts = cseq.trim().split(/\s+/);
-    return parts.length >= 2 ? parts[1] : null;
-  }
-
-  /** True for INVITE, SUBSCRIBE, REFER, NOTIFY, UPDATE. */
-  get isDialogEstablishing(): boolean {
-    return /^(INVITE|SUBSCRIBE|REFER|NOTIFY|UPDATE)\s/.test(this.startLine);
-  }
-
-  /** True when the body carries an SDP payload. */
-  get hasSdpBody(): boolean {
-    const ct = (this.getHeader('Content-Type') || '').toLowerCase();
-    return !!this.body && ct.startsWith('application/sdp');
-  }
-
-  // -------------------------------------------------------------------------
-  // Header accessors (fluent)
-  // -------------------------------------------------------------------------
-
-  getHeader(name: string): string | null {
-    const nl = name.toLowerCase();
-    for (const [n, v] of this.headers) if (n.toLowerCase() === nl) return v;
-    return null;
-  }
-
-  /** Overwrites the first header with the given name, or appends it. */
-  setHeader(name: string, value: string): this {
-    const nl = name.toLowerCase();
-    for (const h of this.headers) {
-      if (h[0].toLowerCase() === nl) { h[1] = value; return this; }
-    }
-    this.headers.push([name, value]);
-    return this;
-  }
-
-  /** Inserts a header at the top of the header list. */
-  prependHeader(name: string, value: string): this {
-    this.headers.unshift([name, value]);
-    return this;
-  }
-
-  /** Removes all headers with the given name. */
-  removeHeader(name: string): this {
-    const nl = name.toLowerCase();
-    this.headers = this.headers.filter(([n]) => n.toLowerCase() !== nl);
-    return this;
-  }
-
-  /** Recalculates Content-Length to match the current body. */
-  updateContentLength(): this {
-    const len = Buffer.byteLength(this.body || '', 'utf8');
-    return this.setHeader('Content-Length', String(len));
-  }
-
-  // -------------------------------------------------------------------------
-  // Start-line mutation
-  // -------------------------------------------------------------------------
-
-  /** Replaces the Request-URI (second token) of a request start line. */
-  setRequestUri(uri: string): this {
-    if (!this.isRequest) return this;
-    const parts = this.startLine.split(' ');
-    if (parts.length >= 2) {
-      parts[1] = uri;
-      this.startLine = parts.join(' ');
-    }
-    return this;
-  }
-
-  /** Returns the Request-URI (second token) of a request start line. */
-  get requestUri(): string | null {
-    if (!this.isRequest) return null;
-    return this.startLine.split(' ')[1] || null;
-  }
-
-  // -------------------------------------------------------------------------
-  // Factory methods — build new SIP messages from scratch
-  // -------------------------------------------------------------------------
-
-  /**
-   * Build a new SIP request.
-   *
-   * ```ts
-   * const invite = SipMessage.createRequest('INVITE', 'sip:user@host', {
-   *   from: { uri: 'sip:me@proxy', tag: 'abc' },
-   *   to:   { uri: 'sip:user@host' },
-   *   via:  { host: '192.168.5.66', port: 5070 },
-   *   contact: '<sip:me@192.168.5.66:5070>',
-   * });
-   * ```
-   */
-  static createRequest(method: string, requestUri: string, options: {
-    via: { host: string; port: number; transport?: string; branch?: string };
-    from: { uri: string; displayName?: string; tag?: string };
-    to: { uri: string; displayName?: string; tag?: string };
-    callId?: string;
-    cseq?: number;
-    contact?: string;
-    maxForwards?: number;
-    body?: string;
-    contentType?: string;
-    extraHeaders?: [string, string][];
-  }): SipMessage {
-    const branch = options.via.branch || generateBranch();
-    const transport = options.via.transport || 'UDP';
-    const fromTag = options.from.tag || generateTag();
-    const callId = options.callId || generateCallId();
-    const cseq = options.cseq ?? 1;
-
-    const fromDisplay = options.from.displayName ? `"${options.from.displayName}" ` : '';
-    const toDisplay = options.to.displayName ? `"${options.to.displayName}" ` : '';
-    const toTag = options.to.tag ? `;tag=${options.to.tag}` : '';
-
-    const headers: [string, string][] = [
-      ['Via', `SIP/2.0/${transport} ${options.via.host}:${options.via.port};branch=${branch};rport`],
-      ['From', `${fromDisplay}<${options.from.uri}>;tag=${fromTag}`],
-      ['To', `${toDisplay}<${options.to.uri}>${toTag}`],
-      ['Call-ID', callId],
-      ['CSeq', `${cseq} ${method}`],
-      ['Max-Forwards', String(options.maxForwards ?? 70)],
-    ];
-
-    if (options.contact) {
-      headers.push(['Contact', options.contact]);
-    }
-
-    if (options.extraHeaders) {
-      headers.push(...options.extraHeaders);
-    }
-
-    const body = options.body || '';
-    if (body && options.contentType) {
-      headers.push(['Content-Type', options.contentType]);
-    }
-    headers.push(['Content-Length', String(Buffer.byteLength(body, 'utf8'))]);
-
-    return new SipMessage(`${method} ${requestUri} SIP/2.0`, headers, body);
-  }
-
-  /**
-   * Build a SIP response to an incoming request.
-   *
-   * Copies Via, From, To, Call-ID, and CSeq from the original request.
-   */
-  static createResponse(
-    statusCode: number,
-    reasonPhrase: string,
-    request: SipMessage,
-    options?: {
-      toTag?: string;
-      contact?: string;
-      body?: string;
-      contentType?: string;
-      extraHeaders?: [string, string][];
-    },
-  ): SipMessage {
-    const headers: [string, string][] = [];
-
-    // Copy all Via headers (order matters).
-    for (const [n, v] of request.headers) {
-      if (n.toLowerCase() === 'via') headers.push(['Via', v]);
-    }
-
-    // From — copied verbatim.
-    const from = request.getHeader('From');
-    if (from) headers.push(['From', from]);
-
-    // To — add tag if provided and not already present.
-    let to = request.getHeader('To') || '';
-    if (options?.toTag && !to.includes('tag=')) {
-      to += `;tag=${options.toTag}`;
-    }
-    headers.push(['To', to]);
-
-    headers.push(['Call-ID', request.callId]);
-
-    const cseq = request.getHeader('CSeq');
-    if (cseq) headers.push(['CSeq', cseq]);
-
-    if (options?.contact) headers.push(['Contact', options.contact]);
-    if (options?.extraHeaders) headers.push(...options.extraHeaders);
-
-    const body = options?.body || '';
-    if (body && options?.contentType) {
-      headers.push(['Content-Type', options.contentType]);
-    }
-    headers.push(['Content-Length', String(Buffer.byteLength(body, 'utf8'))]);
-
-    return new SipMessage(`SIP/2.0 ${statusCode} ${reasonPhrase}`, headers, body);
-  }
-
-  /** Extract the tag from a From or To header value. */
-  static extractTag(headerValue: string): string | null {
-    const m = headerValue.match(/;tag=([^\s;>]+)/);
-    return m ? m[1] : null;
-  }
-
-  /** Extract the URI from an addr-spec or name-addr (From/To/Contact). */
-  static extractUri(headerValue: string): string | null {
-    const m = headerValue.match(/<([^>]+)>/);
-    return m ? m[1] : headerValue.trim().split(/[;>]/)[0] || null;
-  }
-}

ts/sip/readme.md

@@ -1,228 +0,0 @@
-# ts/sip — SIP Protocol Library
-
-A zero-dependency SIP (Session Initiation Protocol) library for Deno / Node.
-Provides parsing, construction, mutation, and dialog management for SIP
-messages, plus helpers for SDP bodies and URI rewriting.
-
-## Modules
-
-| File | Purpose |
-|------|---------|
-| `message.ts` | `SipMessage` class — parse, inspect, mutate, serialize |
-| `dialog.ts` | `SipDialog` class — track dialog state, build in-dialog requests |
-| `helpers.ts` | ID generators, codec registry, SDP builder/parser |
-| `rewrite.ts` | SIP URI and SDP body rewriting |
-| `types.ts` | Shared types (`IEndpoint`) |
-| `index.ts` | Barrel re-export |
-
-## Quick Start
-
-```ts
-import {
-  SipMessage,
-  SipDialog,
-  buildSdp,
-  parseSdpEndpoint,
-  rewriteSipUri,
-  rewriteSdp,
-  generateCallId,
-  generateTag,
-  generateBranch,
-} from './sip/index.ts';
-```
-
-## SipMessage
-
-### Parsing
-
-```ts
-import { Buffer } from 'node:buffer';
-
-const raw = Buffer.from(
-  'INVITE sip:user@example.com SIP/2.0\r\n' +
-  'Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK776\r\n' +
-  'From: <sip:alice@example.com>;tag=abc\r\n' +
-  'To: <sip:bob@example.com>\r\n' +
-  'Call-ID: a84b4c76e66710@10.0.0.1\r\n' +
-  'CSeq: 1 INVITE\r\n' +
-  'Content-Length: 0\r\n\r\n'
-);
-
-const msg = SipMessage.parse(raw);
-// msg.method        → "INVITE"
-// msg.isRequest     → true
-// msg.callId        → "a84b4c76e66710@10.0.0.1"
-// msg.cseqMethod    → "INVITE"
-// msg.isDialogEstablishing → true
-```
-
-### Fluent mutation
-
-All setter methods return `this` for chaining:
-
-```ts
-const buf = SipMessage.parse(raw)!
-  .setHeader('Contact', '<sip:proxy@192.168.1.1:5070>')
-  .prependHeader('Record-Route', '<sip:192.168.1.1:5070;lr>')
-  .updateContentLength()
-  .serialize();
-```
-
-### Building requests from scratch
-
-```ts
-const invite = SipMessage.createRequest('INVITE', 'sip:+4930123@voip.example.com', {
-  via: { host: '192.168.5.66', port: 5070 },
-  from: { uri: 'sip:alice@example.com', displayName: 'Alice' },
-  to: { uri: 'sip:+4930123@voip.example.com' },
-  contact: '<sip:192.168.5.66:5070>',
-  body: sdpBody,
-  contentType: 'application/sdp',
-});
-// Call-ID, From tag, Via branch are auto-generated if not provided.
-```
-
-### Building responses
-
-```ts
-const ok = SipMessage.createResponse(200, 'OK', incomingInvite, {
-  toTag: generateTag(),
-  contact: '<sip:192.168.5.66:5070>',
-  body: answerSdp,
-  contentType: 'application/sdp',
-});
-```
-
-### Inspectors
-
-| Property | Type | Description |
-|----------|------|-------------|
-| `isRequest` | `boolean` | True for requests (INVITE, BYE, ...) |
-| `isResponse` | `boolean` | True for responses (SIP/2.0 200 OK, ...) |
-| `method` | `string \| null` | Request method or null |
-| `statusCode` | `number \| null` | Response status code or null |
-| `callId` | `string` | Call-ID header value |
-| `cseqMethod` | `string \| null` | Method from CSeq header |
-| `requestUri` | `string \| null` | Request-URI (second token of start line) |
-| `isDialogEstablishing` | `boolean` | INVITE, SUBSCRIBE, REFER, NOTIFY, UPDATE |
-| `hasSdpBody` | `boolean` | Body present with Content-Type: application/sdp |
-
-### Static helpers
-
-```ts
-SipMessage.extractTag('<sip:alice@x.com>;tag=abc')   // → "abc"
-SipMessage.extractUri('"Alice" <sip:alice@x.com>')   // → "sip:alice@x.com"
-```
-
-## SipDialog
-
-Tracks dialog state per RFC 3261 §12. A dialog is created from a
-dialog-establishing request and updated as responses arrive.
-
-### UAC (caller) side
-
-```ts
-// 1. Build and send INVITE
-const invite = SipMessage.createRequest('INVITE', destUri, { ... });
-const dialog = SipDialog.fromUacInvite(invite, '192.168.5.66', 5070);
-
-// 2. Process responses as they arrive
-dialog.processResponse(trying100);   // state stays 'early'
-dialog.processResponse(ringing180);  // state stays 'early', remoteTag learned
-dialog.processResponse(ok200);       // state → 'confirmed'
-
-// 3. ACK the 200
-const ack = dialog.createAck();
-
-// 4. In-dialog requests
-const bye = dialog.createRequest('BYE');
-dialog.terminate();
-```
-
-### UAS (callee) side
-
-```ts
-const dialog = SipDialog.fromUasInvite(incomingInvite, generateTag(), localHost, localPort);
-```
-
-### CANCEL (before answer)
-
-```ts
-const cancel = dialog.createCancel(originalInvite);
-```
-
-### Dialog states
-
-`'early'` → `'confirmed'` → `'terminated'`
-
-## Helpers
-
-### ID generation
-
-```ts
-generateCallId()               // → "a3f8b2c1d4e5f6a7b8c9d0e1f2a3b4c5"
-generateCallId('example.com')  // → "a3f8b2c1...@example.com"
-generateTag()                  // → "1a2b3c4d5e6f7a8b"
-generateBranch()               // → "z9hG4bK-1a2b3c4d5e6f7a8b"
-```
-
-### SDP builder
-
-```ts
-const sdp = buildSdp({
-  ip: '192.168.5.66',
-  port: 20000,
-  payloadTypes: [9, 0, 8, 101],  // G.722, PCMU, PCMA, telephone-event
-  direction: 'sendrecv',
-});
-```
-
-### SDP parser
-
-```ts
-const ep = parseSdpEndpoint(sdpBody);
-// → { address: '10.0.0.1', port: 20000 } or null
-```
-
-### Codec names
-
-```ts
-codecName(9)   // → "G722/8000"
-codecName(0)   // → "PCMU/8000"
-codecName(101) // → "telephone-event/8000"
-```
-
-## Rewriting
-
-### SIP URI
-
-Replaces the host:port in all `sip:` / `sips:` URIs found in a header value:
-
-```ts
-rewriteSipUri('<sip:user@10.0.0.1:5060>', '203.0.113.1', 5070)
-// → '<sip:user@203.0.113.1:5070>'
-```
-
-### SDP body
-
-Rewrites the connection address and audio media port, returning the original
-endpoint that was replaced:
-
-```ts
-const { body, original } = rewriteSdp(sdpBody, '203.0.113.1', 20000);
-// original → { address: '10.0.0.1', port: 8000 }
-```
-
-## Architecture Notes
-
-This library is intentionally low-level — it operates on individual messages
-and dialogs rather than providing a full SIP stack with transport and
-transaction layers. This makes it suitable for building:
-
-- **SIP proxies** — parse, rewrite headers/SDP, serialize, forward
-- **B2BUA (back-to-back user agents)** — manage two dialogs, bridge media
-- **SIP testing tools** — craft and send arbitrary messages
-- **Protocol analyzers** — parse and inspect SIP traffic
-
-The library does not manage sockets, timers, or retransmissions — those
-concerns belong to the application layer.

ts/sip/rewrite.ts

@@ -1,54 +0,0 @@
-/**
- * SIP URI and SDP body rewriting helpers.
- */
-
-import type { IEndpoint } from './types.ts';
-
-const SIP_URI_RE = /(sips?:)([^@>;,\s]+@)?([^>;,\s:]+)(:\d+)?/g;
-
-/**
- * Replaces the host:port in every `sip:` / `sips:` URI found in `value`.
- */
-export function rewriteSipUri(value: string, host: string, port: number): string {
-  return value.replace(SIP_URI_RE, (_m, scheme: string, userpart?: string) =>
-    `${scheme}${userpart || ''}${host}:${port}`);
-}
-
-/**
- * Rewrites the connection address (`c=`) and audio media port (`m=audio`)
- * in an SDP body.  Returns the rewritten body together with the original
- * endpoint that was replaced (if any).
- */
-export function rewriteSdp(
-  body: string,
-  ip: string,
-  port: number,
-): { body: string; original: IEndpoint | null } {
-  let origAddr: string | null = null;
-  let origPort: number | null = null;
-
-  const out = body
-    .replace(/\r\n/g, '\n')
-    .split('\n')
-    .map((line) => {
-      if (line.startsWith('c=IN IP4 ')) {
-        origAddr = line.slice('c=IN IP4 '.length).trim();
-        return `c=IN IP4 ${ip}`;
-      }
-      if (line.startsWith('m=audio ')) {
-        const parts = line.split(' ');
-        if (parts.length >= 2) {
-          origPort = parseInt(parts[1], 10);
-          parts[1] = String(port);
-        }
-        return parts.join(' ');
-      }
-      return line;
-    })
-    .join('\r\n');
-
-  return {
-    body: out,
-    original: origAddr && origPort ? { address: origAddr, port: origPort } : null,
-  };
-}

ts/sip/types.ts

@@ -1,8 +0,0 @@
-/**
- * Shared SIP types.
- */
-
-export interface IEndpoint {
-  address: string;
-  port: number;
-}

ts/sipproxy.ts

@@ -1,39 +1,22 @@
 /**
- * SIP proxy — hub model entry point.
+ * SIP proxy — entry point.
  *
- * Thin bootstrap that wires together:
- * - UDP socket for all SIP signaling
- * - CallManager (the hub model core)
- * - Provider registration
- * - Local device registrar
- * - WebRTC signaling
- * - Web dashboard
- * - Rust codec bridge
+ * Spawns the Rust proxy-engine which handles ALL SIP protocol mechanics.
+ * TypeScript is the control plane:
+ * - Loads config and pushes it to Rust
+ * - Receives high-level events (incoming calls, registration, etc.)
+ * - Drives the web dashboard
+ * - Manages IVR, voicemail, announcements
+ * - Handles WebRTC browser signaling (forwarded to Rust in Phase 2)
  *
- * All call/media logic lives in ts/call/.
+ * No raw SIP ever touches TypeScript.
  */
 
-import dgram from 'node:dgram';
 import fs from 'node:fs';
 import path from 'node:path';
-import { Buffer } from 'node:buffer';
 
-import { SipMessage } from './sip/index.ts';
-import type { IEndpoint } from './sip/index.ts';
-import { loadConfig, resolveOutboundRoute } from './config.ts';
-import type { IAppConfig, IProviderConfig } from './config.ts';
-import {
-  initProviderStates,
-  syncProviderStates,
-  getProviderByUpstreamAddress,
-  handleProviderRegistrationResponse,
-} from './providerstate.ts';
-import {
-  initRegistrar,
-  handleDeviceRegister,
-  isKnownDeviceAddress,
-  getAllDeviceStatuses,
-} from './registrar.ts';
+import { loadConfig } from './config.ts';
+import type { IAppConfig } from './config.ts';
 import { broadcastWs, initWebUi } from './frontend.ts';
 import {
   initWebRtcSignaling,
@@ -43,17 +26,32 @@ import {
 } from './webrtcbridge.ts';
 import { initCodecBridge } from './opusbridge.ts';
 import { initAnnouncement } from './announcement.ts';
-import { CallManager } from './call/index.ts';
+import { PromptCache } from './call/prompt-cache.ts';
+import { VoiceboxManager } from './voicebox.ts';
+import {
+  initProxyEngine,
+  configureProxyEngine,
+  onProxyEvent,
+  hangupCall,
+  makeCall,
+  shutdownProxyEngine,
+  webrtcOffer,
+  webrtcIce,
+  webrtcClose,
+} from './proxybridge.ts';
+import type {
+  IIncomingCallEvent,
+  IOutboundCallEvent,
+  ICallEndedEvent,
+  IProviderRegisteredEvent,
+  IDeviceRegisteredEvent,
+} from './proxybridge.ts';
 
 // ---------------------------------------------------------------------------
 // Config
 // ---------------------------------------------------------------------------
 
 let appConfig: IAppConfig = loadConfig();
-const { proxy } = appConfig;
-
-const LAN_IP   = proxy.lanIp;
-const LAN_PORT = proxy.lanPort;
 
 const LOG_PATH = path.join(process.cwd(), 'sip_trace.log');
 
@@ -75,35 +73,82 @@ function log(msg: string): void {
   broadcastWs('log', { message: msg });
 }
 
-function logPacket(label: string, data: Buffer): void {
-  const head = `\n========== ${now()} ${label} (${data.length}b) ==========\n`;
-  const looksText = data.length > 0 && data[0] >= 0x41 && data[0] <= 0x7a;
-  const body = looksText
-    ? data.toString('utf8')
-    : `[${data.length} bytes binary] ${data.toString('hex').slice(0, 80)}`;
-  fs.appendFileSync(LOG_PATH, head + body + '\n');
+// ---------------------------------------------------------------------------
+// Shadow state — maintained from Rust events for the dashboard
+// ---------------------------------------------------------------------------
+
+interface IProviderStatus {
+  id: string;
+  displayName: string;
+  registered: boolean;
+  publicIp: string | null;
+}
+
+interface IDeviceStatus {
+  id: string;
+  displayName: string;
+  address: string | null;
+  port: number;
+  connected: boolean;
+  isBrowser: boolean;
+}
+
+interface IActiveCall {
+  id: string;
+  direction: string;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  state: string;
+  startedAt: number;
+}
+
+interface ICallHistoryEntry {
+  id: string;
+  direction: string;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  startedAt: number;
+  duration: number;
+}
+
+const providerStatuses = new Map<string, IProviderStatus>();
+const deviceStatuses = new Map<string, IDeviceStatus>();
+const activeCalls = new Map<string, IActiveCall>();
+const callHistory: ICallHistoryEntry[] = [];
+const MAX_HISTORY = 100;
+
+// Initialize provider statuses from config (all start as unregistered).
+for (const p of appConfig.providers) {
+  providerStatuses.set(p.id, {
+    id: p.id,
+    displayName: p.displayName,
+    registered: false,
+    publicIp: null,
+  });
+}
+
+// Initialize device statuses from config.
+for (const d of appConfig.devices) {
+  deviceStatuses.set(d.id, {
+    id: d.id,
+    displayName: d.displayName,
+    address: null,
+    port: 0,
+    connected: false,
+    isBrowser: false,
+  });
 }
 
 // ---------------------------------------------------------------------------
 // Initialize subsystems
 // ---------------------------------------------------------------------------
 
-const providerStates = initProviderStates(appConfig.providers, proxy.publicIpSeed);
+const promptCache = new PromptCache(log);
+const voiceboxManager = new VoiceboxManager(log);
+voiceboxManager.init(appConfig.voiceboxes ?? []);
 
-initRegistrar(appConfig.devices, log);
-
-const callManager = new CallManager({
-  appConfig,
-  sendSip: (buf, dest) => sock.send(buf, dest.port, dest.address),
-  log,
-  broadcastWs,
-  getProviderState: (id) => providerStates.get(id),
-  getAllBrowserDeviceIds,
-  sendToBrowserDevice,
-  getBrowserDeviceWs,
-});
-
-// Initialize WebRTC signaling (browser device registration only).
+// WebRTC signaling (browser device registration).
 initWebRtcSignaling({ log });
 
 // ---------------------------------------------------------------------------
@@ -111,149 +156,245 @@ initWebRtcSignaling({ log });
 // ---------------------------------------------------------------------------
 
 function getStatus() {
-  const providers: unknown[] = [];
-  for (const ps of providerStates.values()) {
-    providers.push({
-      id: ps.config.id,
-      displayName: ps.config.displayName,
-      registered: ps.isRegistered,
-      publicIp: ps.publicIp,
+  // Merge SIP devices (from Rust) + browser devices (from TS WebSocket).
+  const devices = [...deviceStatuses.values()];
+  for (const bid of getAllBrowserDeviceIds()) {
+    devices.push({
+      id: bid,
+      displayName: 'Browser',
+      address: null,
+      port: 0,
+      connected: true,
+      isBrowser: true,
     });
   }
 
   return {
     instanceId,
     uptime: Math.floor((Date.now() - startTime) / 1000),
-    lanIp: LAN_IP,
-    providers,
-    devices: getAllDeviceStatuses(),
-    calls: callManager.getStatus(),
-    callHistory: callManager.getHistory(),
+    lanIp: appConfig.proxy.lanIp,
+    providers: [...providerStatuses.values()],
+    devices,
+    calls: [...activeCalls.values()].map((c) => ({
+      ...c,
+      duration: Math.floor((Date.now() - c.startedAt) / 1000),
+      legs: [],
+    })),
+    callHistory,
     contacts: appConfig.contacts || [],
+    voicemailCounts: voiceboxManager.getAllUnheardCounts(),
   };
 }
 
 // ---------------------------------------------------------------------------
-// Main UDP socket
+// Start Rust proxy engine
 // ---------------------------------------------------------------------------
 
-const sock = dgram.createSocket('udp4');
-
-sock.on('message', (data: Buffer, rinfo: dgram.RemoteInfo) => {
-  try {
-    const ps = getProviderByUpstreamAddress(rinfo.address, rinfo.port);
-
-    const msg = SipMessage.parse(data);
-    if (!msg) {
-      // Non-SIP data — forward raw based on direction.
-      if (ps) {
-        // From provider, forward to... nowhere useful without a call context.
-        logPacket(`UP->DN RAW (unparsed) from ${rinfo.address}:${rinfo.port}`, data);
-      } else {
-        // From device, forward to default provider.
-        const dp = resolveOutboundRoute(appConfig, '');
-        if (dp) sock.send(data, dp.provider.outboundProxy.port, dp.provider.outboundProxy.address);
-      }
-      return;
-    }
-
-    // 1. Provider registration responses — consumed by providerstate.
-    if (handleProviderRegistrationResponse(msg)) return;
-
-    // 2. Device REGISTER — handled by local registrar.
-    if (!ps && msg.method === 'REGISTER') {
-      const response = handleDeviceRegister(msg, { address: rinfo.address, port: rinfo.port });
-      if (response) {
-        sock.send(response.serialize(), rinfo.port, rinfo.address);
-        return;
-      }
-    }
-
-    // 3. Route to existing call by SIP Call-ID.
-    if (callManager.routeSipMessage(msg, rinfo)) {
-      return;
-    }
-
-    // 4. New inbound call from provider.
-    if (ps && msg.isRequest && msg.method === 'INVITE') {
-      logPacket(`[new inbound] INVITE from ${rinfo.address}:${rinfo.port}`, data);
-
-      // Detect public IP from Via.
-      const via = msg.getHeader('Via');
-      if (via) ps.detectPublicIp(via);
-
-      callManager.createInboundCall(ps, msg, { address: rinfo.address, port: rinfo.port });
-      return;
-    }
-
-    // 5. New outbound call from device (passthrough).
-    if (!ps && msg.isRequest && msg.method === 'INVITE') {
-      logPacket(`[new outbound] INVITE from ${rinfo.address}:${rinfo.port}`, data);
-      const dialedNumber = SipMessage.extractUri(msg.requestUri || '') || '';
-      const routeResult = resolveOutboundRoute(
-        appConfig,
-        dialedNumber,
-        undefined,
-        (pid) => !!providerStates.get(pid)?.registeredAor,
-      );
-      if (routeResult) {
-        const provState = providerStates.get(routeResult.provider.id);
-        if (provState) {
-          // Apply number transformation to the INVITE if needed.
-          if (routeResult.transformedNumber !== dialedNumber) {
-            const newUri = msg.requestUri?.replace(dialedNumber, routeResult.transformedNumber);
-            if (newUri) msg.setRequestUri(newUri);
-          }
-          callManager.handlePassthroughOutbound(msg, { address: rinfo.address, port: rinfo.port }, routeResult.provider, provState);
-        }
-      }
-      return;
-    }
-
-    // 6. Fallback: forward based on direction (for mid-dialog messages
-    //    that don't match any tracked call, e.g. OPTIONS, NOTIFY).
-    if (ps) {
-      // From provider -> forward to device.
-      logPacket(`[fallback inbound] from ${rinfo.address}:${rinfo.port}`, data);
-      const via = msg.getHeader('Via');
-      if (via) ps.detectPublicIp(via);
-      // Try to figure out where to send it...
-      // For now, just log. These should become rare once all calls are tracked.
-      log(`[fallback] unrouted inbound ${msg.isRequest ? msg.method : msg.statusCode} Call-ID=${msg.callId.slice(0, 30)}`);
-    } else {
-      // From device -> forward to provider.
-      logPacket(`[fallback outbound] from ${rinfo.address}:${rinfo.port}`, data);
-      const fallback = resolveOutboundRoute(appConfig, '');
-      if (fallback) sock.send(msg.serialize(), fallback.provider.outboundProxy.port, fallback.provider.outboundProxy.address);
-    }
-  } catch (e: any) {
-    log(`[err] ${e?.stack || e}`);
+async function startProxyEngine(): Promise<void> {
+  const ok = await initProxyEngine(log);
+  if (!ok) {
+    log('[FATAL] failed to start proxy engine');
+    process.exit(1);
   }
-});
 
-sock.on('error', (err: Error) => log(`[main] sock err: ${err.message}`));
+  // Subscribe to events from Rust BEFORE sending configure.
+  onProxyEvent('provider_registered', (data: IProviderRegisteredEvent) => {
+    const ps = providerStatuses.get(data.provider_id);
+    if (ps) {
+      const wasRegistered = ps.registered;
+      ps.registered = data.registered;
+      ps.publicIp = data.public_ip;
+      if (data.registered && !wasRegistered) {
+        log(`[provider:${data.provider_id}] registered (publicIp=${data.public_ip})`);
+      } else if (!data.registered && wasRegistered) {
+        log(`[provider:${data.provider_id}] registration lost`);
+      }
+      broadcastWs('registration', { providerId: data.provider_id, registered: data.registered });
+    }
+  });
+
+  onProxyEvent('device_registered', (data: IDeviceRegisteredEvent) => {
+    const ds = deviceStatuses.get(data.device_id);
+    if (ds) {
+      ds.address = data.address;
+      ds.port = data.port;
+      ds.connected = true;
+      log(`[registrar] ${data.display_name} registered from ${data.address}:${data.port}`);
+    }
+  });
+
+  onProxyEvent('incoming_call', (data: IIncomingCallEvent) => {
+    log(`[call] incoming: ${data.from_uri} → ${data.to_number} via ${data.provider_id} (${data.call_id})`);
+    activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'inbound',
+      callerNumber: data.from_uri,
+      calleeNumber: data.to_number,
+      providerUsed: data.provider_id,
+      state: 'ringing',
+      startedAt: Date.now(),
+    });
+
+    // Notify browsers of incoming call.
+    const browserIds = getAllBrowserDeviceIds();
+    for (const bid of browserIds) {
+      sendToBrowserDevice(bid, {
+        type: 'webrtc-incoming',
+        callId: data.call_id,
+        from: data.from_uri,
+        deviceId: bid,
+      });
+    }
+  });
+
+  onProxyEvent('outbound_device_call', (data: IOutboundCallEvent) => {
+    log(`[call] outbound: device ${data.from_device} → ${data.to_number} (${data.call_id})`);
+    activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'outbound',
+      callerNumber: data.from_device,
+      calleeNumber: data.to_number,
+      providerUsed: null,
+      state: 'setting-up',
+      startedAt: Date.now(),
+    });
+  });
+
+  onProxyEvent('outbound_call_started', (data: any) => {
+    log(`[call] outbound started: ${data.call_id} → ${data.number} via ${data.provider_id}`);
+    activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'outbound',
+      callerNumber: null,
+      calleeNumber: data.number,
+      providerUsed: data.provider_id,
+      state: 'setting-up',
+      startedAt: Date.now(),
+    });
+  });
+
+  onProxyEvent('call_ringing', (data: { call_id: string }) => {
+    const call = activeCalls.get(data.call_id);
+    if (call) call.state = 'ringing';
+  });
+
+  onProxyEvent('call_answered', (data: { call_id: string }) => {
+    const call = activeCalls.get(data.call_id);
+    if (call) {
+      call.state = 'connected';
+      log(`[call] ${data.call_id} connected`);
+    }
+  });
+
+  onProxyEvent('call_ended', (data: ICallEndedEvent) => {
+    const call = activeCalls.get(data.call_id);
+    if (call) {
+      log(`[call] ${data.call_id} ended: ${data.reason} (${data.duration}s)`);
+      // Move to history.
+      callHistory.unshift({
+        id: call.id,
+        direction: call.direction,
+        callerNumber: call.callerNumber,
+        calleeNumber: call.calleeNumber,
+        startedAt: call.startedAt,
+        duration: data.duration,
+      });
+      if (callHistory.length > MAX_HISTORY) callHistory.pop();
+      activeCalls.delete(data.call_id);
+    }
+  });
+
+  onProxyEvent('sip_unhandled', (data: any) => {
+    log(`[sip] unhandled ${data.method_or_status} Call-ID=${data.call_id?.slice(0, 20)} from=${data.from_addr}:${data.from_port}`);
+  });
+
+  // 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.
+    broadcastWs('webrtc-ice', {
+      sessionId: data.session_id,
+      candidate: { candidate: data.candidate, sdpMid: data.sdp_mid, sdpMLineIndex: data.sdp_mline_index },
+    });
+  });
+
+  onProxyEvent('webrtc_state', (data: any) => {
+    log(`[webrtc] session=${data.session_id?.slice(0, 8)} state=${data.state}`);
+  });
+
+  onProxyEvent('webrtc_track', (data: any) => {
+    log(`[webrtc] session=${data.session_id?.slice(0, 8)} track=${data.kind} codec=${data.codec}`);
+  });
+
+  onProxyEvent('webrtc_audio_rx', (data: any) => {
+    if (data.packet_count === 1 || data.packet_count === 50) {
+      log(`[webrtc] session=${data.session_id?.slice(0, 8)} browser audio rx #${data.packet_count}`);
+    }
+  });
+
+  // Voicemail events.
+  onProxyEvent('voicemail_started', (data: any) => {
+    log(`[voicemail] started for call ${data.call_id} caller=${data.caller_number}`);
+  });
+
+  onProxyEvent('recording_done', (data: any) => {
+    log(`[voicemail] recording done: ${data.file_path} (${data.duration_ms}ms) caller=${data.caller_number}`);
+    // Save voicemail metadata via VoiceboxManager.
+    voiceboxManager.addMessage?.('default', {
+      callerNumber: data.caller_number || 'Unknown',
+      callerName: null,
+      fileName: data.file_path,
+      durationMs: data.duration_ms,
+    });
+  });
+
+  onProxyEvent('voicemail_error', (data: any) => {
+    log(`[voicemail] error: ${data.error} call=${data.call_id}`);
+  });
+
+  // Send full config to Rust — this binds the SIP socket and starts registrations.
+  const configured = await configureProxyEngine({
+    proxy: appConfig.proxy,
+    providers: appConfig.providers,
+    devices: appConfig.devices,
+    routing: appConfig.routing,
+  });
+
+  if (!configured) {
+    log('[FATAL] failed to configure proxy engine');
+    process.exit(1);
+  }
 
-sock.bind(LAN_PORT, '0.0.0.0', () => {
   const providerList = appConfig.providers.map((p) => p.displayName).join(', ');
   const deviceList = appConfig.devices.map((d) => d.displayName).join(', ');
-  log(`sip proxy bound 0.0.0.0:${LAN_PORT} | providers: ${providerList} | devices: ${deviceList}`);
+  log(`proxy engine started | LAN ${appConfig.proxy.lanIp}:${appConfig.proxy.lanPort} | providers: ${providerList} | devices: ${deviceList}`);
 
-  // Start upstream provider registrations.
-  for (const ps of providerStates.values()) {
-    ps.startRegistration(
-      LAN_IP,
-      LAN_PORT,
-      (buf, dest) => sock.send(buf, dest.port, dest.address),
-      log,
-      (provider) => broadcastWs('registration', { providerId: provider.config.id, registered: provider.isRegistered }),
-    );
+  // Initialize audio codec bridge (still needed for WebRTC transcoding).
+  try {
+    await initCodecBridge(log);
+    await initAnnouncement(log);
+
+    // Pre-generate prompts.
+    await promptCache.generateBeep('voicemail-beep', 1000, 500, 8000);
+    for (const vb of appConfig.voiceboxes ?? []) {
+      if (!vb.enabled) continue;
+      const promptId = `voicemail-greeting-${vb.id}`;
+      if (vb.greetingWavPath) {
+        await promptCache.loadWavPrompt(promptId, vb.greetingWavPath);
+      } else {
+        const text = vb.greetingText || 'The person you are trying to reach is not available. Please leave a message after the tone.';
+        await promptCache.generatePrompt(promptId, text, vb.greetingVoice || 'af_bella');
+      }
+    }
+    if (appConfig.ivr?.enabled) {
+      for (const menu of appConfig.ivr.menus) {
+        await promptCache.generatePrompt(`ivr-menu-${menu.id}`, menu.promptText, menu.promptVoice || 'af_bella');
+      }
+    }
+    log(`[startup] prompts cached: ${promptCache.listIds().join(', ') || 'none'}`);
+  } catch (e) {
+    log(`[codec] init failed: ${e}`);
   }
-
-  // Initialize audio codec bridge (Rust binary via smartrust).
-  initCodecBridge(log)
-    .then(() => initAnnouncement(log))
-    .catch((e) => log(`[codec] init failed: ${e}`));
-});
+}
 
 // ---------------------------------------------------------------------------
 // Web UI
@@ -263,32 +404,92 @@ initWebUi(
   getStatus,
   log,
   (number, deviceId, providerId) => {
-    const call = callManager.createOutboundCall(number, deviceId, providerId);
-    return call ? { id: call.id } : null;
+    // Outbound calls from dashboard — send make_call command to Rust.
+    log(`[dashboard] start call: ${number} device=${deviceId || 'any'} provider=${providerId || 'auto'}`);
+    // Fire-and-forget — the async result comes via events.
+    makeCall(number, deviceId, providerId).then((callId) => {
+      if (callId) {
+        log(`[dashboard] call started: ${callId}`);
+        activeCalls.set(callId, {
+          id: callId,
+          direction: 'outbound',
+          callerNumber: null,
+          calleeNumber: number,
+          providerUsed: providerId || null,
+          state: 'setting-up',
+          startedAt: Date.now(),
+        });
+      } else {
+        log(`[dashboard] call failed for ${number}`);
+      }
+    });
+    // Return a temporary ID so the frontend doesn't show "failed" immediately.
+    return { id: `pending-${Date.now()}` };
+  },
+  (callId) => {
+    hangupCall(callId);
+    return true;
   },
-  (callId) => callManager.hangup(callId),
   () => {
-    // Reload config after UI save.
+    // Config saved — reconfigure Rust engine.
     try {
       const fresh = loadConfig();
       Object.assign(appConfig, fresh);
-      // Sync provider registrations: add new, remove deleted, re-register changed.
-      syncProviderStates(
-        fresh.providers,
-        proxy.publicIpSeed,
-        LAN_IP,
-        LAN_PORT,
-        (buf, dest) => sock.send(buf, dest.port, dest.address),
-        log,
-        (provider) => broadcastWs('registration', { providerId: provider.config.id, registered: provider.isRegistered }),
-      );
-      log('[config] reloaded config after save');
+
+      // Update shadow state.
+      for (const p of fresh.providers) {
+        if (!providerStatuses.has(p.id)) {
+          providerStatuses.set(p.id, {
+            id: p.id, displayName: p.displayName, registered: false, publicIp: null,
+          });
+        }
+      }
+      for (const d of fresh.devices) {
+        if (!deviceStatuses.has(d.id)) {
+          deviceStatuses.set(d.id, {
+            id: d.id, displayName: d.displayName, address: null, port: 0, connected: false, isBrowser: false,
+          });
+        }
+      }
+
+      // Re-send config to Rust.
+      configureProxyEngine({
+        proxy: fresh.proxy,
+        providers: fresh.providers,
+        devices: fresh.devices,
+        routing: fresh.routing,
+      }).then((ok) => {
+        if (ok) log('[config] reloaded — proxy engine reconfigured');
+        else log('[config] reload failed — proxy engine rejected config');
+      });
     } catch (e: any) {
       log(`[config] reload failed: ${e.message}`);
     }
   },
-  callManager,
+  undefined, // callManager — legacy, replaced by Rust proxy-engine
+  voiceboxManager,
+  // WebRTC signaling → forwarded to Rust proxy-engine.
+  async (sessionId, sdp, ws) => {
+    log(`[webrtc] offer from browser session=${sessionId.slice(0, 8)}`);
+    const result = await webrtcOffer(sessionId, sdp);
+    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)}`);
+    }
+  },
+  async (sessionId, candidate) => {
+    await webrtcIce(sessionId, candidate);
+  },
+  async (sessionId) => {
+    await webrtcClose(sessionId);
+  },
 );
 
-process.on('SIGINT', () => { log('SIGINT, exiting'); process.exit(0); });
-process.on('SIGTERM', () => { log('SIGTERM, exiting'); process.exit(0); });
+// ---------------------------------------------------------------------------
+// Start
+// ---------------------------------------------------------------------------
+
+startProxyEngine();
+
+process.on('SIGINT', () => { log('SIGINT, exiting'); shutdownProxyEngine(); process.exit(0); });
+process.on('SIGTERM', () => { log('SIGTERM, exiting'); shutdownProxyEngine(); process.exit(0); });

ts/voicebox.ts

@@ -0,0 +1,314 @@
+/**
+ * VoiceboxManager — manages voicemail boxes, message storage, and MWI.
+ *
+ * Each voicebox corresponds to a device/extension. Messages are stored
+ * as WAV files with JSON metadata in .nogit/voicemail/{boxId}/.
+ *
+ * Supports:
+ * - Per-box configurable TTS greetings (text + voice) or uploaded WAV
+ * - Message CRUD: save, list, mark heard, delete
+ * - Unheard count for MWI (Message Waiting Indicator)
+ * - Storage limit (max messages per box)
+ */
+
+import fs from 'node:fs';
+import path from 'node:path';
+
+// ---------------------------------------------------------------------------
+// Types
+// ---------------------------------------------------------------------------
+
+export interface IVoiceboxConfig {
+  /** Unique ID — typically matches device ID or extension. */
+  id: string;
+  /** Whether this voicebox is active. */
+  enabled: boolean;
+  /** Custom TTS greeting text (overrides default). */
+  greetingText?: string;
+  /** Kokoro TTS voice ID for the greeting (default 'af_bella'). */
+  greetingVoice?: string;
+  /** Path to uploaded WAV greeting (overrides TTS). */
+  greetingWavPath?: string;
+  /** Seconds to wait before routing to voicemail (default 25). */
+  noAnswerTimeoutSec: number;
+  /** Maximum recording duration in seconds (default 120). */
+  maxRecordingSec: number;
+  /** Maximum stored messages per box (default 50). */
+  maxMessages: number;
+}
+
+export interface IVoicemailMessage {
+  /** Unique message ID. */
+  id: string;
+  /** Which voicebox this message belongs to. */
+  boxId: string;
+  /** Caller's phone number. */
+  callerNumber: string;
+  /** Caller's display name (if available from SIP From header). */
+  callerName?: string;
+  /** Unix timestamp (ms) when the message was recorded. */
+  timestamp: number;
+  /** Duration in milliseconds. */
+  durationMs: number;
+  /** Relative path to the WAV file (within the box directory). */
+  fileName: string;
+  /** Whether the message has been listened to. */
+  heard: boolean;
+}
+
+// Default greeting text when no custom text is configured.
+const DEFAULT_GREETING = 'The person you are trying to reach is not available. Please leave a message after the tone.';
+
+// ---------------------------------------------------------------------------
+// VoiceboxManager
+// ---------------------------------------------------------------------------
+
+export class VoiceboxManager {
+  private boxes = new Map<string, IVoiceboxConfig>();
+  private basePath: string;
+  private log: (msg: string) => void;
+
+  constructor(log: (msg: string) => void) {
+    this.basePath = path.join(process.cwd(), '.nogit', 'voicemail');
+    this.log = log;
+  }
+
+  // -------------------------------------------------------------------------
+  // Initialization
+  // -------------------------------------------------------------------------
+
+  /**
+   * Load voicebox configurations from the app config.
+   */
+  init(voiceboxConfigs: IVoiceboxConfig[]): void {
+    this.boxes.clear();
+
+    for (const cfg of voiceboxConfigs) {
+      // Apply defaults.
+      cfg.noAnswerTimeoutSec ??= 25;
+      cfg.maxRecordingSec ??= 120;
+      cfg.maxMessages ??= 50;
+      cfg.greetingVoice ??= 'af_bella';
+
+      this.boxes.set(cfg.id, cfg);
+    }
+
+    // Ensure base directory exists.
+    fs.mkdirSync(this.basePath, { recursive: true });
+
+    this.log(`[voicebox] initialized ${this.boxes.size} voicebox(es)`);
+  }
+
+  // -------------------------------------------------------------------------
+  // Box management
+  // -------------------------------------------------------------------------
+
+  /** Get config for a specific voicebox. */
+  getBox(boxId: string): IVoiceboxConfig | null {
+    return this.boxes.get(boxId) ?? null;
+  }
+
+  /** Get all configured voicebox IDs. */
+  getBoxIds(): string[] {
+    return [...this.boxes.keys()];
+  }
+
+  /** Get the greeting text for a voicebox. */
+  getGreetingText(boxId: string): string {
+    const box = this.boxes.get(boxId);
+    return box?.greetingText || DEFAULT_GREETING;
+  }
+
+  /** Get the greeting voice for a voicebox. */
+  getGreetingVoice(boxId: string): string {
+    const box = this.boxes.get(boxId);
+    return box?.greetingVoice || 'af_bella';
+  }
+
+  /** Check if a voicebox has a custom WAV greeting. */
+  hasCustomGreetingWav(boxId: string): boolean {
+    const box = this.boxes.get(boxId);
+    if (!box?.greetingWavPath) return false;
+    return fs.existsSync(box.greetingWavPath);
+  }
+
+  /** Get the greeting WAV path (custom or null). */
+  getCustomGreetingWavPath(boxId: string): string | null {
+    const box = this.boxes.get(boxId);
+    if (!box?.greetingWavPath) return null;
+    return fs.existsSync(box.greetingWavPath) ? box.greetingWavPath : null;
+  }
+
+  /** Get the directory path for a voicebox. */
+  getBoxDir(boxId: string): string {
+    return path.join(this.basePath, boxId);
+  }
+
+  // -------------------------------------------------------------------------
+  // Message CRUD
+  // -------------------------------------------------------------------------
+
+  /**
+   * Save a new voicemail message.
+   * The WAV file should already exist at the expected path.
+   */
+  saveMessage(msg: IVoicemailMessage): void {
+    const boxDir = this.getBoxDir(msg.boxId);
+    fs.mkdirSync(boxDir, { recursive: true });
+
+    const messages = this.loadMessages(msg.boxId);
+    messages.unshift(msg); // newest first
+
+    // Enforce max messages — delete oldest.
+    const box = this.boxes.get(msg.boxId);
+    const maxMessages = box?.maxMessages ?? 50;
+    while (messages.length > maxMessages) {
+      const old = messages.pop()!;
+      const oldPath = path.join(boxDir, old.fileName);
+      try {
+        if (fs.existsSync(oldPath)) fs.unlinkSync(oldPath);
+      } catch { /* best effort */ }
+    }
+
+    this.writeMessages(msg.boxId, messages);
+    this.log(`[voicebox] saved message ${msg.id} in box "${msg.boxId}" (${msg.durationMs}ms from ${msg.callerNumber})`);
+  }
+
+  /**
+   * List messages for a voicebox (newest first).
+   */
+  getMessages(boxId: string): IVoicemailMessage[] {
+    return this.loadMessages(boxId);
+  }
+
+  /**
+   * Get a single message by ID.
+   */
+  getMessage(boxId: string, messageId: string): IVoicemailMessage | null {
+    const messages = this.loadMessages(boxId);
+    return messages.find((m) => m.id === messageId) ?? null;
+  }
+
+  /**
+   * Mark a message as heard.
+   */
+  markHeard(boxId: string, messageId: string): boolean {
+    const messages = this.loadMessages(boxId);
+    const msg = messages.find((m) => m.id === messageId);
+    if (!msg) return false;
+
+    msg.heard = true;
+    this.writeMessages(boxId, messages);
+    return true;
+  }
+
+  /**
+   * Delete a message (both metadata and WAV file).
+   */
+  deleteMessage(boxId: string, messageId: string): boolean {
+    const messages = this.loadMessages(boxId);
+    const idx = messages.findIndex((m) => m.id === messageId);
+    if (idx === -1) return false;
+
+    const msg = messages[idx];
+    const boxDir = this.getBoxDir(boxId);
+    const wavPath = path.join(boxDir, msg.fileName);
+
+    // Delete WAV file.
+    try {
+      if (fs.existsSync(wavPath)) fs.unlinkSync(wavPath);
+    } catch { /* best effort */ }
+
+    // Remove from list and save.
+    messages.splice(idx, 1);
+    this.writeMessages(boxId, messages);
+    this.log(`[voicebox] deleted message ${messageId} from box "${boxId}"`);
+    return true;
+  }
+
+  /**
+   * Get the full file path for a message's WAV file.
+   */
+  getMessageAudioPath(boxId: string, messageId: string): string | null {
+    const msg = this.getMessage(boxId, messageId);
+    if (!msg) return null;
+    const filePath = path.join(this.getBoxDir(boxId), msg.fileName);
+    return fs.existsSync(filePath) ? filePath : null;
+  }
+
+  // -------------------------------------------------------------------------
+  // Counts
+  // -------------------------------------------------------------------------
+
+  /** Get count of unheard messages for a voicebox. */
+  getUnheardCount(boxId: string): number {
+    const messages = this.loadMessages(boxId);
+    return messages.filter((m) => !m.heard).length;
+  }
+
+  /** Get total message count for a voicebox. */
+  getTotalCount(boxId: string): number {
+    return this.loadMessages(boxId).length;
+  }
+
+  /** Get unheard counts for all voiceboxes. */
+  getAllUnheardCounts(): Record<string, number> {
+    const counts: Record<string, number> = {};
+    for (const boxId of this.boxes.keys()) {
+      counts[boxId] = this.getUnheardCount(boxId);
+    }
+    return counts;
+  }
+
+  // -------------------------------------------------------------------------
+  // Greeting management
+  // -------------------------------------------------------------------------
+
+  /**
+   * Save a custom greeting WAV file for a voicebox.
+   */
+  saveCustomGreeting(boxId: string, wavData: Buffer): string {
+    const boxDir = this.getBoxDir(boxId);
+    fs.mkdirSync(boxDir, { recursive: true });
+    const greetingPath = path.join(boxDir, 'greeting.wav');
+    fs.writeFileSync(greetingPath, wavData);
+    this.log(`[voicebox] saved custom greeting for box "${boxId}"`);
+    return greetingPath;
+  }
+
+  /**
+   * Delete the custom greeting for a voicebox (falls back to TTS).
+   */
+  deleteCustomGreeting(boxId: string): void {
+    const boxDir = this.getBoxDir(boxId);
+    const greetingPath = path.join(boxDir, 'greeting.wav');
+    try {
+      if (fs.existsSync(greetingPath)) fs.unlinkSync(greetingPath);
+    } catch { /* best effort */ }
+  }
+
+  // -------------------------------------------------------------------------
+  // Internal: JSON persistence
+  // -------------------------------------------------------------------------
+
+  private messagesPath(boxId: string): string {
+    return path.join(this.getBoxDir(boxId), 'messages.json');
+  }
+
+  private loadMessages(boxId: string): IVoicemailMessage[] {
+    const filePath = this.messagesPath(boxId);
+    try {
+      if (!fs.existsSync(filePath)) return [];
+      const raw = fs.readFileSync(filePath, 'utf8');
+      return JSON.parse(raw) as IVoicemailMessage[];
+    } catch {
+      return [];
+    }
+  }
+
+  private writeMessages(boxId: string, messages: IVoicemailMessage[]): void {
+    const boxDir = this.getBoxDir(boxId);
+    fs.mkdirSync(boxDir, { recursive: true });
+    fs.writeFileSync(this.messagesPath(boxId), JSON.stringify(messages, null, 2), 'utf8');
+  }
+}

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.9.0',
+  version: '1.12.0',
   description: 'undefined'
 }

ts_web/elements/index.ts

@@ -5,8 +5,10 @@ export * from './sipproxy-view-calls.js';
 export * from './sipproxy-view-phone.js';
 export * from './sipproxy-view-contacts.js';
 export * from './sipproxy-view-providers.js';
+export * from './sipproxy-view-voicemail.js';
 export * from './sipproxy-view-log.js';
 export * from './sipproxy-view-routes.js';
+export * from './sipproxy-view-ivr.js';
 
 // Sub-components (used within views)
 export * from './sipproxy-devices.js';

ts_web/elements/sipproxy-app.ts

@@ -9,12 +9,16 @@ import { SipproxyViewContacts } from './sipproxy-view-contacts.js';
 import { SipproxyViewProviders } from './sipproxy-view-providers.js';
 import { SipproxyViewLog } from './sipproxy-view-log.js';
 import { SipproxyViewRoutes } from './sipproxy-view-routes.js';
+import { SipproxyViewVoicemail } from './sipproxy-view-voicemail.js';
+import { SipproxyViewIvr } from './sipproxy-view-ivr.js';
 
 const VIEW_TABS = [
   { name: 'Overview', iconName: 'lucide:layoutDashboard', element: SipproxyViewOverview },
   { name: 'Calls', iconName: 'lucide:phone', element: SipproxyViewCalls },
   { name: 'Phone', iconName: 'lucide:headset', element: SipproxyViewPhone },
   { name: 'Routes', iconName: 'lucide:route', element: SipproxyViewRoutes },
+  { name: 'Voicemail', iconName: 'lucide:voicemail', element: SipproxyViewVoicemail },
+  { name: 'IVR', iconName: 'lucide:list-tree', element: SipproxyViewIvr },
   { name: 'Contacts', iconName: 'lucide:contactRound', element: SipproxyViewContacts },
   { name: 'Providers', iconName: 'lucide:server', element: SipproxyViewProviders },
   { name: 'Log', iconName: 'lucide:scrollText', element: SipproxyViewLog },

ts_web/elements/sipproxy-view-ivr.ts

@@ -0,0 +1,657 @@
+import { DeesElement, customElement, html, css, cssManager, state, type TemplateResult } from '../plugins.js';
+import { deesCatalog } from '../plugins.js';
+import { appState, type IAppState } from '../state/appstate.js';
+import { viewHostCss } from './shared/index.js';
+import type { IStatsTile } from '@design.estate/dees-catalog';
+
+const { DeesModal, DeesToast } = deesCatalog;
+
+// ---------------------------------------------------------------------------
+// IVR types (mirrors ts/config.ts)
+// ---------------------------------------------------------------------------
+
+type TIvrAction =
+  | { type: 'route-extension'; extensionId: string }
+  | { type: 'route-voicemail'; boxId: string }
+  | { type: 'submenu'; menuId: string }
+  | { type: 'play-message'; promptId: string }
+  | { type: 'transfer'; number: string; providerId?: string }
+  | { type: 'repeat' }
+  | { type: 'hangup' };
+
+interface IIvrMenuEntry {
+  digit: string;
+  action: TIvrAction;
+}
+
+interface IIvrMenu {
+  id: string;
+  name: string;
+  promptText: string;
+  promptVoice?: string;
+  entries: IIvrMenuEntry[];
+  timeoutSec?: number;
+  maxRetries?: number;
+  timeoutAction: TIvrAction;
+  invalidAction: TIvrAction;
+}
+
+interface IIvrConfig {
+  enabled: boolean;
+  menus: IIvrMenu[];
+  entryMenuId: string;
+}
+
+// ---------------------------------------------------------------------------
+// Helpers
+// ---------------------------------------------------------------------------
+
+function slugify(name: string): string {
+  return name
+    .toLowerCase()
+    .replace(/[^a-z0-9]+/g, '-')
+    .replace(/^-|-$/g, '') || `menu-${Date.now()}`;
+}
+
+const VOICE_OPTIONS = [
+  { option: 'af_bella (Female)', key: 'af_bella' },
+  { option: 'af_sarah (Female)', key: 'af_sarah' },
+  { option: 'am_adam (Male)', key: 'am_adam' },
+  { option: 'bf_alice (Female)', key: 'bf_alice' },
+];
+
+const ACTION_TYPE_OPTIONS = [
+  { option: 'Route to Extension', key: 'route-extension' },
+  { option: 'Route to Voicemail', key: 'route-voicemail' },
+  { option: 'Submenu', key: 'submenu' },
+  { option: 'Play Message', key: 'play-message' },
+  { option: 'Transfer', key: 'transfer' },
+  { option: 'Repeat', key: 'repeat' },
+  { option: 'Hangup', key: 'hangup' },
+];
+
+const DIGIT_OPTIONS = ['1', '2', '3', '4', '5', '6', '7', '8', '9', '0', '*', '#'];
+
+function describeAction(action: TIvrAction): string {
+  switch (action.type) {
+    case 'route-extension': return `Extension: ${action.extensionId}`;
+    case 'route-voicemail': return `Voicemail: ${action.boxId}`;
+    case 'submenu': return `Submenu: ${action.menuId}`;
+    case 'play-message': return `Play: ${action.promptId}`;
+    case 'transfer': return `Transfer: ${action.number}${action.providerId ? ` (${action.providerId})` : ''}`;
+    case 'repeat': return 'Repeat';
+    case 'hangup': return 'Hangup';
+    default: return 'Unknown';
+  }
+}
+
+function makeDefaultAction(): TIvrAction {
+  return { type: 'hangup' };
+}
+
+// ---------------------------------------------------------------------------
+// View element
+// ---------------------------------------------------------------------------
+
+@customElement('sipproxy-view-ivr')
+export class SipproxyViewIvr extends DeesElement {
+  @state() accessor appData: IAppState = appState.getState();
+  @state() accessor config: any = null;
+
+  public static styles = [
+    cssManager.defaultStyles,
+    viewHostCss,
+    css`
+      .view-section { margin-bottom: 24px; }
+    `,
+  ];
+
+  // ---- lifecycle -----------------------------------------------------------
+
+  connectedCallback() {
+    super.connectedCallback();
+    this.rxSubscriptions.push({
+      unsubscribe: appState.subscribe((s) => { this.appData = s; }),
+    } as any);
+    this.loadConfig();
+  }
+
+  private async loadConfig() {
+    try {
+      this.config = await appState.apiGetConfig();
+    } catch {
+      // Will show empty state.
+    }
+  }
+
+  private getIvrConfig(): IIvrConfig {
+    return this.config?.ivr || { enabled: false, menus: [], entryMenuId: '' };
+  }
+
+  // ---- stats tiles ---------------------------------------------------------
+
+  private getStatsTiles(): IStatsTile[] {
+    const ivr = this.getIvrConfig();
+    const entryMenu = ivr.menus.find((m) => m.id === ivr.entryMenuId);
+
+    return [
+      {
+        id: 'total-menus',
+        title: 'Total Menus',
+        value: ivr.menus.length,
+        type: 'number',
+        icon: 'lucide:list-tree',
+        description: 'IVR menu definitions',
+      },
+      {
+        id: 'entry-menu',
+        title: 'Entry Menu',
+        value: entryMenu?.name || '(none)',
+        type: 'text' as any,
+        icon: 'lucide:door-open',
+        description: entryMenu ? `ID: ${entryMenu.id}` : 'No entry menu set',
+      },
+      {
+        id: 'status',
+        title: 'Status',
+        value: ivr.enabled ? 'Enabled' : 'Disabled',
+        type: 'text' as any,
+        icon: ivr.enabled ? 'lucide:check-circle' : 'lucide:x-circle',
+        color: ivr.enabled ? 'hsl(142.1 76.2% 36.3%)' : 'hsl(0 84.2% 60.2%)',
+        description: ivr.enabled ? 'IVR is active' : 'IVR is inactive',
+      },
+    ];
+  }
+
+  // ---- table columns -------------------------------------------------------
+
+  private getColumns() {
+    const ivr = this.getIvrConfig();
+    return [
+      {
+        key: 'name',
+        header: 'Name',
+        sortable: true,
+        renderer: (val: string, row: IIvrMenu) => {
+          const isEntry = row.id === ivr.entryMenuId;
+          return html`
+            <span>${val}</span>
+            ${isEntry ? html`<span style="display:inline-block;margin-left:8px;padding:1px 6px;border-radius:4px;font-size:.65rem;font-weight:600;text-transform:uppercase;background:#1e3a5f;color:#60a5fa">entry</span>` : ''}
+          `;
+        },
+      },
+      {
+        key: 'promptText',
+        header: 'Prompt',
+        renderer: (val: string) => {
+          const truncated = val && val.length > 60 ? val.slice(0, 60) + '...' : val || '--';
+          return html`<span style="font-size:.82rem;color:#94a3b8">${truncated}</span>`;
+        },
+      },
+      {
+        key: 'entries',
+        header: 'Digits',
+        renderer: (_val: any, row: IIvrMenu) => {
+          const digits = (row.entries || []).map((e) => e.digit).join(', ');
+          return html`<span style="font-family:'JetBrains Mono',monospace;font-size:.75rem">${digits || '(none)'}</span>`;
+        },
+      },
+      {
+        key: 'timeoutAction',
+        header: 'Timeout Action',
+        renderer: (_val: any, row: IIvrMenu) => {
+          return html`<span style="font-size:.82rem;color:#94a3b8">${describeAction(row.timeoutAction)}</span>`;
+        },
+      },
+    ];
+  }
+
+  // ---- table actions -------------------------------------------------------
+
+  private getDataActions() {
+    return [
+      {
+        name: 'Add Menu',
+        iconName: 'lucide:plus' as any,
+        type: ['header'] as any,
+        actionFunc: async () => {
+          await this.openMenuEditor(null);
+        },
+      },
+      {
+        name: 'Edit',
+        iconName: 'lucide:pencil' as any,
+        type: ['inRow'] as any,
+        actionFunc: async ({ item }: { item: IIvrMenu }) => {
+          await this.openMenuEditor(item);
+        },
+      },
+      {
+        name: 'Set as Entry',
+        iconName: 'lucide:door-open' as any,
+        type: ['inRow'] as any,
+        actionFunc: async ({ item }: { item: IIvrMenu }) => {
+          await this.setEntryMenu(item.id);
+        },
+      },
+      {
+        name: 'Delete',
+        iconName: 'lucide:trash-2' as any,
+        type: ['inRow'] as any,
+        actionFunc: async ({ item }: { item: IIvrMenu }) => {
+          await this.confirmDeleteMenu(item);
+        },
+      },
+    ];
+  }
+
+  // ---- toggle enabled ------------------------------------------------------
+
+  private async toggleEnabled() {
+    const ivr = this.getIvrConfig();
+    const updated: IIvrConfig = { ...ivr, enabled: !ivr.enabled };
+    const result = await appState.apiSaveConfig({ ivr: updated });
+    if (result.ok) {
+      DeesToast.success(updated.enabled ? 'IVR enabled' : 'IVR disabled');
+      await this.loadConfig();
+    } else {
+      DeesToast.error('Failed to update IVR status');
+    }
+  }
+
+  // ---- set entry menu ------------------------------------------------------
+
+  private async setEntryMenu(menuId: string) {
+    const ivr = this.getIvrConfig();
+    const updated: IIvrConfig = { ...ivr, entryMenuId: menuId };
+    const result = await appState.apiSaveConfig({ ivr: updated });
+    if (result.ok) {
+      DeesToast.success('Entry menu updated');
+      await this.loadConfig();
+    } else {
+      DeesToast.error('Failed to set entry menu');
+    }
+  }
+
+  // ---- delete menu ---------------------------------------------------------
+
+  private async confirmDeleteMenu(menu: IIvrMenu) {
+    await DeesModal.createAndShow({
+      heading: 'Delete IVR Menu',
+      width: 'small',
+      showCloseButton: true,
+      content: html`
+        <div style="padding:8px 0;font-size:.9rem;color:#e2e8f0;">
+          Are you sure you want to delete
+          <strong style="color:#f87171;">${menu.name}</strong>?
+          This action cannot be undone.
+        </div>
+      `,
+      menuOptions: [
+        {
+          name: 'Cancel',
+          iconName: 'lucide:x',
+          action: async (modalRef: any) => { modalRef.destroy(); },
+        },
+        {
+          name: 'Delete',
+          iconName: 'lucide:trash-2',
+          action: async (modalRef: any) => {
+            const ivr = this.getIvrConfig();
+            const menus = ivr.menus.filter((m) => m.id !== menu.id);
+            const updated: IIvrConfig = {
+              ...ivr,
+              menus,
+              entryMenuId: ivr.entryMenuId === menu.id ? '' : ivr.entryMenuId,
+            };
+            const result = await appState.apiSaveConfig({ ivr: updated });
+            if (result.ok) {
+              modalRef.destroy();
+              DeesToast.success(`Menu "${menu.name}" deleted`);
+              await this.loadConfig();
+            } else {
+              DeesToast.error('Failed to delete menu');
+            }
+          },
+        },
+      ],
+    });
+  }
+
+  // ---- action editor helper ------------------------------------------------
+
+  private renderActionEditor(
+    action: TIvrAction,
+    onChange: (a: TIvrAction) => void,
+    label: string,
+    cfg: any,
+  ): TemplateResult {
+    const devices = cfg?.devices || [];
+    const menus: IIvrMenu[] = cfg?.ivr?.menus || [];
+    const providers = cfg?.providers || [];
+
+    const currentType = ACTION_TYPE_OPTIONS.find((o) => o.key === action.type) || ACTION_TYPE_OPTIONS[ACTION_TYPE_OPTIONS.length - 1];
+
+    return html`
+      <div style="margin-bottom:12px;">
+        <div style="font-size:.75rem;color:#94a3b8;text-transform:uppercase;letter-spacing:.04em;margin-bottom:6px;font-weight:600;">${label}</div>
+        <dees-input-dropdown
+          .label=${'Action Type'}
+          .selectedOption=${currentType}
+          .options=${ACTION_TYPE_OPTIONS}
+          @selectedOption=${(e: CustomEvent) => {
+            const type = e.detail.key;
+            switch (type) {
+              case 'route-extension': onChange({ type, extensionId: devices[0]?.extension || '100' }); break;
+              case 'route-voicemail': onChange({ type, boxId: '' }); break;
+              case 'submenu': onChange({ type, menuId: menus[0]?.id || '' }); break;
+              case 'play-message': onChange({ type, promptId: '' }); break;
+              case 'transfer': onChange({ type, number: '' }); break;
+              case 'repeat': onChange({ type }); break;
+              case 'hangup': onChange({ type }); break;
+            }
+          }}
+        ></dees-input-dropdown>
+
+        ${action.type === 'route-extension' ? html`
+          <dees-input-dropdown
+            .label=${'Extension'}
+            .selectedOption=${{ option: action.extensionId, key: action.extensionId }}
+            .options=${devices.map((d: any) => ({ option: `${d.displayName} (${d.extension})`, key: d.extension }))}
+            @selectedOption=${(e: CustomEvent) => { onChange({ ...action, extensionId: e.detail.key }); }}
+          ></dees-input-dropdown>
+        ` : ''}
+
+        ${action.type === 'route-voicemail' ? html`
+          <dees-input-text
+            .label=${'Voicemail Box ID'}
+            .value=${action.boxId}
+            @input=${(e: Event) => { onChange({ ...action, boxId: (e.target as any).value }); }}
+          ></dees-input-text>
+        ` : ''}
+
+        ${action.type === 'submenu' ? html`
+          <dees-input-dropdown
+            .label=${'Menu'}
+            .selectedOption=${menus.find((m) => m.id === action.menuId)
+              ? { option: menus.find((m) => m.id === action.menuId)!.name, key: action.menuId }
+              : { option: '(select)', key: '' }}
+            .options=${menus.map((m) => ({ option: m.name, key: m.id }))}
+            @selectedOption=${(e: CustomEvent) => { onChange({ ...action, menuId: e.detail.key }); }}
+          ></dees-input-dropdown>
+        ` : ''}
+
+        ${action.type === 'play-message' ? html`
+          <dees-input-text
+            .label=${'Prompt ID'}
+            .value=${action.promptId}
+            @input=${(e: Event) => { onChange({ ...action, promptId: (e.target as any).value }); }}
+          ></dees-input-text>
+        ` : ''}
+
+        ${action.type === 'transfer' ? html`
+          <dees-input-text
+            .label=${'Transfer Number'}
+            .value=${action.number}
+            @input=${(e: Event) => { onChange({ ...action, number: (e.target as any).value }); }}
+          ></dees-input-text>
+          <dees-input-dropdown
+            .label=${'Provider (optional)'}
+            .selectedOption=${action.providerId
+              ? { option: action.providerId, key: action.providerId }
+              : { option: '(default)', key: '' }}
+            .options=${[
+              { option: '(default)', key: '' },
+              ...providers.map((p: any) => ({ option: p.displayName || p.id, key: p.id })),
+            ]}
+            @selectedOption=${(e: CustomEvent) => { onChange({ ...action, providerId: e.detail.key || undefined }); }}
+          ></dees-input-dropdown>
+        ` : ''}
+      </div>
+    `;
+  }
+
+  // ---- menu editor modal ---------------------------------------------------
+
+  private async openMenuEditor(existing: IIvrMenu | null) {
+    const cfg = this.config;
+
+    const formData: IIvrMenu = existing
+      ? JSON.parse(JSON.stringify(existing))
+      : {
+          id: '',
+          name: '',
+          promptText: '',
+          promptVoice: 'af_bella',
+          entries: [],
+          timeoutSec: 5,
+          maxRetries: 3,
+          timeoutAction: { type: 'hangup' as const },
+          invalidAction: { type: 'repeat' as const },
+        };
+
+    // For re-rendering the modal content on state changes we track a version counter.
+    let version = 0;
+    const modalContentId = `ivr-modal-${Date.now()}`;
+
+    const rerenderContent = () => {
+      version++;
+      const container = document.querySelector(`#${modalContentId}`) as HTMLElement
+        || document.getElementById(modalContentId);
+      if (container) {
+        // Force a re-render by removing and re-adding the modal content.
+        // We can't use lit's render directly here, so we close and reopen.
+      }
+    };
+
+    const buildContent = (): TemplateResult => html`
+      <div style="display:flex;flex-direction:column;gap:12px;padding:4px 0;">
+        <dees-input-text
+          .key=${'name'} .label=${'Menu Name'} .value=${formData.name}
+          @input=${(e: Event) => {
+            formData.name = (e.target as any).value;
+            if (!existing) {
+              formData.id = slugify(formData.name);
+            }
+          }}
+        ></dees-input-text>
+
+        <dees-input-text
+          .key=${'id'} .label=${'Menu ID'} .value=${formData.id}
+          .description=${'Auto-generated from name. Editable for custom IDs.'}
+          @input=${(e: Event) => { formData.id = (e.target as any).value; }}
+        ></dees-input-text>
+
+        <dees-input-text
+          .key=${'promptText'} .label=${'Prompt Text (TTS)'}
+          .value=${formData.promptText}
+          .description=${'Text that will be read aloud to the caller.'}
+          @input=${(e: Event) => { formData.promptText = (e.target as any).value; }}
+        ></dees-input-text>
+
+        <dees-input-dropdown
+          .key=${'promptVoice'} .label=${'Voice'}
+          .selectedOption=${VOICE_OPTIONS.find((v) => v.key === formData.promptVoice) || VOICE_OPTIONS[0]}
+          .options=${VOICE_OPTIONS}
+          @selectedOption=${(e: CustomEvent) => { formData.promptVoice = e.detail.key; }}
+        ></dees-input-dropdown>
+
+        <div style="margin-top:8px;padding-top:12px;border-top:1px solid #334155;">
+          <div style="display:flex;align-items:center;justify-content:space-between;margin-bottom:8px;">
+            <div style="font-size:.75rem;color:#94a3b8;text-transform:uppercase;letter-spacing:.04em;font-weight:600;">
+              Digit Entries
+            </div>
+            <div
+              style="font-size:.75rem;color:#60a5fa;cursor:pointer;user-select:none;"
+              @click=${() => {
+                const usedDigits = new Set(formData.entries.map((e) => e.digit));
+                const nextDigit = DIGIT_OPTIONS.find((d) => !usedDigits.has(d)) || '1';
+                formData.entries = [...formData.entries, { digit: nextDigit, action: makeDefaultAction() }];
+                rerenderContent();
+              }}
+            >+ Add Digit</div>
+          </div>
+
+          ${formData.entries.length === 0
+            ? html`<div style="font-size:.82rem;color:#64748b;font-style:italic;margin-bottom:8px;">No digit entries configured.</div>`
+            : formData.entries.map((entry, idx) => html`
+              <div style="padding:8px;margin-bottom:8px;border:1px solid #334155;border-radius:6px;background:#0f172a;">
+                <div style="display:flex;align-items:center;justify-content:space-between;margin-bottom:8px;">
+                  <dees-input-dropdown
+                    .label=${'Digit'}
+                    .selectedOption=${{ option: entry.digit, key: entry.digit }}
+                    .options=${DIGIT_OPTIONS.map((d) => ({ option: d, key: d }))}
+                    @selectedOption=${(e: CustomEvent) => {
+                      formData.entries[idx].digit = e.detail.key;
+                    }}
+                  ></dees-input-dropdown>
+                  <div
+                    style="font-size:.75rem;color:#f87171;cursor:pointer;user-select:none;margin-left:12px;padding:4px 8px;"
+                    @click=${() => {
+                      formData.entries = formData.entries.filter((_, i) => i !== idx);
+                      rerenderContent();
+                    }}
+                  >Remove</div>
+                </div>
+                ${this.renderActionEditor(
+                  entry.action,
+                  (a) => { formData.entries[idx].action = a; rerenderContent(); },
+                  'Action',
+                  cfg,
+                )}
+              </div>
+            `)
+          }
+        </div>
+
+        <div style="margin-top:8px;padding-top:12px;border-top:1px solid #334155;">
+          <div style="font-size:.75rem;color:#94a3b8;text-transform:uppercase;letter-spacing:.04em;margin-bottom:8px;font-weight:600;">
+            Timeout Settings
+          </div>
+          <div style="display:flex;gap:12px;">
+            <dees-input-text
+              .key=${'timeoutSec'} .label=${'Timeout (sec)'}
+              .value=${String(formData.timeoutSec ?? 5)}
+              @input=${(e: Event) => { formData.timeoutSec = parseInt((e.target as any).value, 10) || 5; }}
+            ></dees-input-text>
+            <dees-input-text
+              .key=${'maxRetries'} .label=${'Max Retries'}
+              .value=${String(formData.maxRetries ?? 3)}
+              @input=${(e: Event) => { formData.maxRetries = parseInt((e.target as any).value, 10) || 3; }}
+            ></dees-input-text>
+          </div>
+        </div>
+
+        <div style="margin-top:8px;padding-top:12px;border-top:1px solid #334155;">
+          ${this.renderActionEditor(
+            formData.timeoutAction,
+            (a) => { formData.timeoutAction = a; rerenderContent(); },
+            'Timeout Action (no digit pressed)',
+            cfg,
+          )}
+          ${this.renderActionEditor(
+            formData.invalidAction,
+            (a) => { formData.invalidAction = a; rerenderContent(); },
+            'Invalid Digit Action',
+            cfg,
+          )}
+        </div>
+      </div>
+    `;
+
+    await DeesModal.createAndShow({
+      heading: existing ? `Edit Menu: ${existing.name}` : 'New IVR Menu',
+      width: 'small',
+      showCloseButton: true,
+      content: html`<div id="${modalContentId}">${buildContent()}</div>`,
+      menuOptions: [
+        {
+          name: 'Cancel',
+          iconName: 'lucide:x',
+          action: async (modalRef: any) => { modalRef.destroy(); },
+        },
+        {
+          name: 'Save',
+          iconName: 'lucide:check',
+          action: async (modalRef: any) => {
+            if (!formData.name.trim()) {
+              DeesToast.error('Menu name is required');
+              return;
+            }
+            if (!formData.id.trim()) {
+              DeesToast.error('Menu ID is required');
+              return;
+            }
+            if (!formData.promptText.trim()) {
+              DeesToast.error('Prompt text is required');
+              return;
+            }
+
+            const ivr = this.getIvrConfig();
+            const menus = [...ivr.menus];
+            const idx = menus.findIndex((m) => m.id === (existing?.id || formData.id));
+            if (idx >= 0) {
+              menus[idx] = formData;
+            } else {
+              menus.push(formData);
+            }
+
+            const updated: IIvrConfig = {
+              ...ivr,
+              menus,
+              // Auto-set entry menu if this is the first menu.
+              entryMenuId: ivr.entryMenuId || formData.id,
+            };
+
+            const result = await appState.apiSaveConfig({ ivr: updated });
+            if (result.ok) {
+              modalRef.destroy();
+              DeesToast.success(existing ? 'Menu updated' : 'Menu created');
+              await this.loadConfig();
+            } else {
+              DeesToast.error('Failed to save menu');
+            }
+          },
+        },
+      ],
+    });
+  }
+
+  // ---- render --------------------------------------------------------------
+
+  public render(): TemplateResult {
+    const ivr = this.getIvrConfig();
+    const menus = ivr.menus || [];
+
+    return html`
+      <div class="view-section">
+        <dees-statsgrid
+          .tiles=${this.getStatsTiles()}
+          .minTileWidth=${220}
+          .gap=${16}
+        ></dees-statsgrid>
+      </div>
+
+      <div class="view-section" style="display:flex;align-items:center;gap:12px;margin-bottom:16px;">
+        <dees-input-checkbox
+          .key=${'ivr-enabled'}
+          .label=${'Enable IVR System'}
+          .value=${ivr.enabled}
+          @newValue=${() => { this.toggleEnabled(); }}
+        ></dees-input-checkbox>
+      </div>
+
+      <div class="view-section">
+        <dees-table
+          heading1="IVR Menus"
+          heading2="${menus.length} configured"
+          dataName="menus"
+          .data=${menus}
+          .rowKey=${'id'}
+          .columns=${this.getColumns()}
+          .dataActions=${this.getDataActions()}
+        ></dees-table>
+      </div>
+    `;
+  }
+}

ts_web/elements/sipproxy-view-voicemail.ts

@@ -0,0 +1,446 @@
+import { DeesElement, customElement, html, css, cssManager, state, type TemplateResult } from '../plugins.js';
+import { deesCatalog } from '../plugins.js';
+import { appState, type IAppState } from '../state/appstate.js';
+import { viewHostCss } from './shared/index.js';
+import type { IStatsTile } from '@design.estate/dees-catalog';
+
+// ---------------------------------------------------------------------------
+// Voicemail message shape (mirrors server IVoicemailMessage)
+// ---------------------------------------------------------------------------
+interface IVoicemailMessage {
+  id: string;
+  boxId: string;
+  callerNumber: string;
+  callerName?: string;
+  timestamp: number;
+  durationMs: number;
+  fileName: string;
+  heard: boolean;
+}
+
+// ---------------------------------------------------------------------------
+// Helpers
+// ---------------------------------------------------------------------------
+
+function formatDuration(ms: number): string {
+  const totalSec = Math.round(ms / 1000);
+  const min = Math.floor(totalSec / 60);
+  const sec = totalSec % 60;
+  return `${min}:${sec.toString().padStart(2, '0')}`;
+}
+
+function formatDateTime(ts: number): string {
+  const d = new Date(ts);
+  const date = d.toLocaleDateString(undefined, { year: 'numeric', month: 'short', day: 'numeric' });
+  const time = d.toLocaleTimeString(undefined, { hour: '2-digit', minute: '2-digit' });
+  return `${date} ${time}`;
+}
+
+// ---------------------------------------------------------------------------
+// View element
+// ---------------------------------------------------------------------------
+@customElement('sipproxy-view-voicemail')
+export class SipproxyViewVoicemail extends DeesElement {
+  @state() accessor appData: IAppState = appState.getState();
+  @state() accessor messages: IVoicemailMessage[] = [];
+  @state() accessor voiceboxIds: string[] = [];
+  @state() accessor selectedBoxId: string = '';
+  @state() accessor playingMessageId: string | null = null;
+  @state() accessor loading: boolean = false;
+
+  private audioElement: HTMLAudioElement | null = null;
+
+  public static styles = [
+    cssManager.defaultStyles,
+    viewHostCss,
+    css`
+      :host {
+        display: block;
+        padding: 16px;
+      }
+      .view-section {
+        margin-bottom: 24px;
+      }
+      .box-selector {
+        display: flex;
+        align-items: center;
+        gap: 12px;
+        margin-bottom: 24px;
+      }
+      .box-selector label {
+        font-size: 0.85rem;
+        font-weight: 600;
+        color: #94a3b8;
+        text-transform: uppercase;
+        letter-spacing: 0.04em;
+      }
+      .audio-player {
+        display: flex;
+        align-items: center;
+        gap: 8px;
+        padding: 8px 12px;
+        background: #1e293b;
+        border-radius: 8px;
+        margin-top: 16px;
+      }
+      .audio-player audio {
+        flex: 1;
+        height: 32px;
+      }
+      .audio-player .close-btn {
+        cursor: pointer;
+        color: #94a3b8;
+        font-size: 1.1rem;
+        padding: 2px 6px;
+        border-radius: 4px;
+        transition: background 0.15s;
+      }
+      .audio-player .close-btn:hover {
+        background: #334155;
+        color: #e2e8f0;
+      }
+      .empty-state {
+        text-align: center;
+        padding: 48px 16px;
+        color: #64748b;
+        font-size: 0.9rem;
+      }
+      .empty-state .icon {
+        font-size: 2.5rem;
+        margin-bottom: 12px;
+        opacity: 0.5;
+      }
+    `,
+  ];
+
+  // ---- lifecycle -----------------------------------------------------------
+
+  connectedCallback() {
+    super.connectedCallback();
+    this.rxSubscriptions.push({
+      unsubscribe: appState.subscribe((s) => { this.appData = s; }),
+    } as any);
+    this.loadVoiceboxes();
+  }
+
+  disconnectedCallback() {
+    super.disconnectedCallback();
+    this.stopAudio();
+  }
+
+  // ---- data loading --------------------------------------------------------
+
+  private async loadVoiceboxes() {
+    try {
+      const cfg = await appState.apiGetConfig();
+      const boxes: { id: string }[] = cfg.voiceboxes || [];
+      this.voiceboxIds = boxes.map((b) => b.id);
+      if (this.voiceboxIds.length > 0 && !this.selectedBoxId) {
+        this.selectedBoxId = this.voiceboxIds[0];
+        await this.loadMessages();
+      }
+    } catch {
+      // Config unavailable.
+    }
+  }
+
+  private async loadMessages() {
+    if (!this.selectedBoxId) {
+      this.messages = [];
+      return;
+    }
+    this.loading = true;
+    try {
+      const res = await fetch(`/api/voicemail/${encodeURIComponent(this.selectedBoxId)}`);
+      const data = await res.json();
+      this.messages = data.messages || [];
+    } catch {
+      this.messages = [];
+    }
+    this.loading = false;
+  }
+
+  private async selectBox(boxId: string) {
+    this.selectedBoxId = boxId;
+    this.stopAudio();
+    await this.loadMessages();
+  }
+
+  // ---- audio playback ------------------------------------------------------
+
+  private playMessage(msg: IVoicemailMessage) {
+    this.stopAudio();
+    const url = `/api/voicemail/${encodeURIComponent(msg.boxId)}/${encodeURIComponent(msg.id)}/audio`;
+    const audio = new Audio(url);
+    this.audioElement = audio;
+    this.playingMessageId = msg.id;
+
+    audio.addEventListener('ended', () => {
+      this.playingMessageId = null;
+      // Auto-mark as heard after playback completes.
+      if (!msg.heard) {
+        this.markHeard(msg);
+      }
+    });
+
+    audio.addEventListener('error', () => {
+      this.playingMessageId = null;
+      deesCatalog.DeesToast.error('Failed to play audio');
+    });
+
+    audio.play().catch(() => {
+      this.playingMessageId = null;
+    });
+  }
+
+  private stopAudio() {
+    if (this.audioElement) {
+      this.audioElement.pause();
+      this.audioElement.src = '';
+      this.audioElement = null;
+    }
+    this.playingMessageId = null;
+  }
+
+  // ---- message actions -----------------------------------------------------
+
+  private async markHeard(msg: IVoicemailMessage) {
+    try {
+      await fetch(`/api/voicemail/${encodeURIComponent(msg.boxId)}/${encodeURIComponent(msg.id)}/heard`, {
+        method: 'POST',
+      });
+      // Update local state without full reload.
+      this.messages = this.messages.map((m) =>
+        m.id === msg.id ? { ...m, heard: true } : m,
+      );
+    } catch {
+      deesCatalog.DeesToast.error('Failed to mark message as heard');
+    }
+  }
+
+  private async deleteMessage(msg: IVoicemailMessage) {
+    const { DeesModal } = await import('@design.estate/dees-catalog');
+    await DeesModal.createAndShow({
+      heading: 'Delete Voicemail',
+      width: 'small',
+      showCloseButton: true,
+      content: html`
+        <div style="padding:8px 0;font-size:.9rem;color:#e2e8f0;">
+          Are you sure you want to delete the voicemail from
+          <strong style="color:#f87171;">${msg.callerName || msg.callerNumber}</strong>
+          (${formatDateTime(msg.timestamp)})?
+        </div>
+      `,
+      menuOptions: [
+        {
+          name: 'Cancel',
+          iconName: 'lucide:x',
+          action: async (modalRef: any) => { modalRef.destroy(); },
+        },
+        {
+          name: 'Delete',
+          iconName: 'lucide:trash-2',
+          action: async (modalRef: any) => {
+            try {
+              await fetch(
+                `/api/voicemail/${encodeURIComponent(msg.boxId)}/${encodeURIComponent(msg.id)}`,
+                { method: 'DELETE' },
+              );
+              if (this.playingMessageId === msg.id) {
+                this.stopAudio();
+              }
+              this.messages = this.messages.filter((m) => m.id !== msg.id);
+              modalRef.destroy();
+              deesCatalog.DeesToast.success('Voicemail deleted');
+            } catch {
+              deesCatalog.DeesToast.error('Failed to delete voicemail');
+            }
+          },
+        },
+      ],
+    });
+  }
+
+  // ---- stats tiles ---------------------------------------------------------
+
+  private getStatsTiles(): IStatsTile[] {
+    const total = this.messages.length;
+    const unheard = this.messages.filter((m) => !m.heard).length;
+
+    return [
+      {
+        id: 'total',
+        title: 'Total Messages',
+        value: total,
+        type: 'number',
+        icon: 'lucide:voicemail',
+        description: this.selectedBoxId ? `Box: ${this.selectedBoxId}` : 'No box selected',
+      },
+      {
+        id: 'unheard',
+        title: 'Unheard Messages',
+        value: unheard,
+        type: 'number',
+        icon: 'lucide:bell-ring',
+        color: unheard > 0 ? 'hsl(0 84.2% 60.2%)' : 'hsl(142.1 76.2% 36.3%)',
+        description: unheard > 0 ? 'Needs attention' : 'All caught up',
+      },
+    ];
+  }
+
+  // ---- table columns -------------------------------------------------------
+
+  private getColumns() {
+    return [
+      {
+        key: 'callerNumber',
+        header: 'Caller',
+        sortable: true,
+        renderer: (_val: string, row: IVoicemailMessage) => {
+          const display = row.callerName
+            ? html`<span>${row.callerName}</span><br><span style="font-size:.75rem;color:#64748b">${row.callerNumber}</span>`
+            : html`<span style="font-family:'JetBrains Mono',monospace;font-size:.85rem">${row.callerNumber}</span>`;
+          return html`<div>${display}</div>`;
+        },
+      },
+      {
+        key: 'timestamp',
+        header: 'Date/Time',
+        sortable: true,
+        value: (row: IVoicemailMessage) => formatDateTime(row.timestamp),
+        renderer: (val: string) =>
+          html`<span style="font-size:.85rem">${val}</span>`,
+      },
+      {
+        key: 'durationMs',
+        header: 'Duration',
+        sortable: true,
+        value: (row: IVoicemailMessage) => formatDuration(row.durationMs),
+        renderer: (val: string) =>
+          html`<span style="font-family:'JetBrains Mono',monospace;font-size:.85rem">${val}</span>`,
+      },
+      {
+        key: 'heard',
+        header: 'Status',
+        renderer: (val: boolean, row: IVoicemailMessage) => {
+          const isPlaying = this.playingMessageId === row.id;
+          if (isPlaying) {
+            return html`
+              <span style="display:inline-block;padding:2px 8px;border-radius:4px;font-size:.7rem;font-weight:600;text-transform:uppercase;background:#1e3a5f;color:#60a5fa">Playing</span>
+            `;
+          }
+          const heard = val;
+          const color = heard ? '#71717a' : '#f59e0b';
+          const bg = heard ? '#3f3f46' : '#422006';
+          const label = heard ? 'Heard' : 'New';
+          return html`
+            <span style="display:inline-block;padding:2px 8px;border-radius:4px;font-size:.7rem;font-weight:600;text-transform:uppercase;background:${bg};color:${color}">${label}</span>
+          `;
+        },
+      },
+    ];
+  }
+
+  // ---- table actions -------------------------------------------------------
+
+  private getDataActions() {
+    return [
+      {
+        name: 'Play',
+        iconName: 'lucide:play',
+        type: ['inRow'] as any,
+        actionFunc: async (actionData: any) => {
+          const msg = actionData.item as IVoicemailMessage;
+          if (this.playingMessageId === msg.id) {
+            this.stopAudio();
+          } else {
+            this.playMessage(msg);
+          }
+        },
+      },
+      {
+        name: 'Mark Heard',
+        iconName: 'lucide:check',
+        type: ['inRow'] as any,
+        actionFunc: async (actionData: any) => {
+          const msg = actionData.item as IVoicemailMessage;
+          if (!msg.heard) {
+            await this.markHeard(msg);
+            deesCatalog.DeesToast.success('Marked as heard');
+          }
+        },
+      },
+      {
+        name: 'Delete',
+        iconName: 'lucide:trash-2',
+        type: ['inRow'] as any,
+        actionFunc: async (actionData: any) => {
+          await this.deleteMessage(actionData.item as IVoicemailMessage);
+        },
+      },
+      {
+        name: 'Refresh',
+        iconName: 'lucide:refreshCw',
+        type: ['header'] as any,
+        actionFunc: async () => {
+          await this.loadMessages();
+          deesCatalog.DeesToast.success('Messages refreshed');
+        },
+      },
+    ];
+  }
+
+  // ---- render --------------------------------------------------------------
+
+  public render(): TemplateResult {
+    return html`
+      ${this.voiceboxIds.length > 1 ? html`
+        <div class="box-selector">
+          <label>Voicebox</label>
+          <dees-input-dropdown
+            .key=${'voicebox'}
+            .selectedOption=${{ option: this.selectedBoxId, key: this.selectedBoxId }}
+            .options=${this.voiceboxIds.map((id) => ({ option: id, key: id }))}
+            @selectedOption=${(e: CustomEvent) => { this.selectBox(e.detail.key); }}
+          ></dees-input-dropdown>
+        </div>
+      ` : ''}
+
+      <div class="view-section">
+        <dees-statsgrid
+          .tiles=${this.getStatsTiles()}
+          .minTileWidth=${220}
+          .gap=${16}
+        ></dees-statsgrid>
+      </div>
+
+      ${this.messages.length === 0 && !this.loading ? html`
+        <div class="empty-state">
+          <div class="icon">&#9993;</div>
+          <div>No voicemail messages${this.selectedBoxId ? ` in box "${this.selectedBoxId}"` : ''}</div>
+        </div>
+      ` : html`
+        <div class="view-section">
+          <dees-table
+            heading1="Voicemail"
+            heading2="${this.messages.length} message${this.messages.length !== 1 ? 's' : ''}"
+            dataName="voicemail"
+            .data=${this.messages}
+            .rowKey=${'id'}
+            .searchable=${true}
+            .columns=${this.getColumns()}
+            .dataActions=${this.getDataActions()}
+          ></dees-table>
+        </div>
+      `}
+
+      ${this.playingMessageId ? html`
+        <div class="audio-player">
+          <span style="color:#60a5fa;font-size:.8rem;font-weight:600;">Now playing</span>
+          <span style="flex:1"></span>
+          <span class="close-btn" @click=${() => this.stopAudio()}>&#10005;</span>
+        </div>
+      ` : ''}
+    `;
+  }
+}

ts_web/router.ts

@@ -3,7 +3,7 @@
  * Maps URL paths to views in dees-simple-appdash.
  */
 
-const VIEWS = ['overview', 'calls', 'phone', 'routes', 'contacts', 'providers', 'log'] as const;
+const VIEWS = ['overview', 'calls', 'phone', 'routes', 'voicemail', 'ivr', 'contacts', 'providers', 'log'] as const;
 type TViewSlug = (typeof VIEWS)[number];
 
 class AppRouter {

ts_web/state/appstate.ts

@@ -72,6 +72,8 @@ export interface IAppState {
   contacts: IContact[];
   selectedContact: IContact | null;
   logLines: string[];
+  /** Unheard voicemail count per voicebox ID. */
+  voicemailCounts: Record<string, number>;
 }
 
 const MAX_LOG = 200;
@@ -89,6 +91,7 @@ class AppStateManager {
     contacts: [],
     selectedContact: null,
     logLines: [],
+    voicemailCounts: {},
   };
 
   private listeners = new Set<(state: IAppState) => void>();
@@ -155,6 +158,7 @@ class AppStateManager {
             calls: m.data.calls || [],
             callHistory: m.data.callHistory || [],
             contacts: m.data.contacts || [],
+            voicemailCounts: m.data.voicemailCounts || {},
           });
         } else if (m.type === 'log') {
           this.addLog(`${m.ts} ${m.data.message}`);