v1.17.1

changelog.md

@@ -1,5 +1,29 @@
 # Changelog
 
+## 2026-04-10 - 1.17.1 - fix(proxy-engine,codec-lib,sip-proto,ts)
+preserve negotiated media details and improve RTP audio handling across call legs
+
+- Use native Opus float encode/decode to avoid unnecessary i16 quantization in the f32 audio path.
+- Parse full RTP headers including extensions and sequence numbers, then sort inbound packets before decoding to keep codec state stable for out-of-order audio.
+- Capture negotiated codec payload types from SDP offers and answers and include codec, RTP port, remote media, and metadata in leg_added events.
+- Emit leg_state_changed and leg_removed events more consistently so the dashboard reflects leg lifecycle updates accurately.
+
+## 2026-04-10 - 1.17.0 - feat(proxy-engine)
+upgrade the internal audio bus to 48kHz f32 with per-leg denoising and improve SIP leg routing
+
+- switch mixer, prompt playback, and tool leg audio handling from 16kHz i16 to 48kHz f32 for higher-quality internal processing
+- add f32 decode/encode and resampling support plus standalone RNNoise denoiser creation in codec-lib
+- apply per-leg inbound noise suppression in the mixer before mix-minus generation
+- fix passthrough call routing by matching the actual leg from the signaling source address when Call-IDs are shared
+- correct dialed number extraction from bare SIP request URIs by parsing the user part directly
+
+## 2026-04-10 - 1.16.0 - feat(proxy-engine)
+integrate Kokoro TTS generation into proxy-engine and simplify TypeScript prompt handling to use cached WAV files
+
+- adds a generate_tts command to proxy-engine with lazy-loaded Kokoro model support and WAV output generation
+- removes standalone opus-codec and tts-engine workspace binaries by consolidating TTS generation into proxy-engine
+- updates announcement and prompt cache flows to generate and cache WAV files on disk instead of pre-encoding RTP frames in TypeScript
+
 ## 2026-04-10 - 1.15.0 - feat(proxy-engine)
 add device leg, leg transfer, and leg replacement call controls
 

package.json

@@ -1,6 +1,6 @@
 {
   "name": "siprouter",
-  "version": "1.15.0",
+  "version": "1.17.1",
   "private": true,
   "type": "module",
   "scripts": {

rust/Cargo.lock

@@ -1881,16 +1881,6 @@ dependencies = [
  "vcpkg",
 ]
 
-[[package]]
-name = "opus-codec"
-version = "0.2.0"
-dependencies = [
- "base64 0.22.1",
- "codec-lib",
- "serde",
- "serde_json",
-]
-
 [[package]]
 name = "ort"
 version = "2.0.0-rc.11"
@@ -2188,6 +2178,9 @@ dependencies = [
  "base64 0.22.1",
  "codec-lib",
  "hound",
+ "kokoro-tts",
+ "nnnoiseless",
+ "ort",
  "rand 0.8.5",
  "regex-lite",
  "serde",
@@ -3008,16 +3001,6 @@ dependencies = [
  "strength_reduce",
 ]
 
-[[package]]
-name = "tts-engine"
-version = "0.1.0"
-dependencies = [
- "hound",
- "kokoro-tts",
- "ort",
- "tokio",
-]
-
 [[package]]
 name = "turn"
 version = "0.6.1"

rust/Cargo.toml

@@ -1,8 +1,6 @@
 [workspace]
 members = [
     "crates/codec-lib",
-    "crates/opus-codec",
-    "crates/tts-engine",
     "crates/sip-proto",
     "crates/proxy-engine",
 ]

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

@@ -1,7 +1,7 @@
 //! 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.
+//! Used by the `proxy-engine` binary for all audio transcoding.
 
 use audiopus::coder::{Decoder as OpusDecoder, Encoder as OpusEncoder};
 use audiopus::packet::Packet as OpusPacket;
@@ -104,6 +104,8 @@ pub struct TranscodeState {
     g722_dec: libg722::decoder::Decoder,
     /// Cached FFT resamplers keyed by (from_rate, to_rate, chunk_size).
     resamplers: HashMap<(u32, u32, usize), FftFixedIn<f64>>,
+    /// Cached f32 FFT resamplers keyed by (from_rate, to_rate, chunk_size).
+    resamplers_f32: HashMap<(u32, u32, usize), FftFixedIn<f32>>,
     /// ML noise suppression for the SIP-bound direction.
     denoiser_to_sip: Box<DenoiseState<'static>>,
     /// ML noise suppression for the browser-bound direction.
@@ -133,6 +135,7 @@ impl TranscodeState {
             g722_enc,
             g722_dec,
             resamplers: HashMap::new(),
+            resamplers_f32: HashMap::new(),
             denoiser_to_sip: DenoiseState::new(),
             denoiser_to_browser: DenoiseState::new(),
         })
@@ -293,6 +296,126 @@ impl TranscodeState {
             _ => Err(format!("unsupported target PT {pt}")),
         }
     }
+
+    // ---- f32 API for high-quality internal bus ----------------------------
+
+    /// Decode an encoded audio payload to f32 PCM samples in [-1.0, 1.0].
+    /// Returns (samples, sample_rate).
+    ///
+    /// For Opus, uses native float decode (no i16 quantization).
+    /// For G.722/G.711, decodes to i16 then converts (codec is natively i16).
+    pub fn decode_to_f32(&mut self, data: &[u8], pt: u8) -> Result<(Vec<f32>, u32), String> {
+        match pt {
+            PT_OPUS => {
+                let mut pcm = vec![0.0f32; 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_float(Some(packet), out, false)
+                    .map_err(|e| format!("opus decode_float: {e}"))?
+                    .into();
+                pcm.truncate(n);
+                Ok((pcm, 48000))
+            }
+            _ => {
+                // G.722, PCMU, PCMA: natively i16 codecs — decode then convert.
+                let (pcm_i16, rate) = self.decode_to_pcm(data, pt)?;
+                let pcm_f32 = pcm_i16.iter().map(|&s| s as f32 / 32768.0).collect();
+                Ok((pcm_f32, rate))
+            }
+        }
+    }
+
+    /// Encode f32 PCM samples ([-1.0, 1.0]) to an audio codec.
+    ///
+    /// For Opus, uses native float encode (no i16 quantization).
+    /// For G.722/G.711, converts to i16 then encodes (codec is natively i16).
+    pub fn encode_from_f32(&mut self, pcm: &[f32], pt: u8) -> Result<Vec<u8>, String> {
+        match pt {
+            PT_OPUS => {
+                let mut buf = vec![0u8; 4000];
+                let n: usize = self
+                    .opus_enc
+                    .encode_float(pcm, &mut buf)
+                    .map_err(|e| format!("opus encode_float: {e}"))?
+                    .into();
+                buf.truncate(n);
+                Ok(buf)
+            }
+            _ => {
+                // G.722, PCMU, PCMA: natively i16 codecs.
+                let pcm_i16: Vec<i16> = pcm
+                    .iter()
+                    .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
+                    .collect();
+                self.encode_from_pcm(&pcm_i16, pt)
+            }
+        }
+    }
+
+    /// High-quality sample rate conversion for f32 PCM using rubato FFT resampler.
+    /// Uses a separate cache from the i16 resampler.
+    pub fn resample_f32(
+        &mut self,
+        pcm: &[f32],
+        from_rate: u32,
+        to_rate: u32,
+    ) -> Result<Vec<f32>, 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_f32.contains_key(&key) {
+            let r =
+                FftFixedIn::<f32>::new(from_rate as usize, to_rate as usize, chunk, 1, 1)
+                    .map_err(|e| format!("resampler f32 {from_rate}->{to_rate}: {e}"))?;
+            self.resamplers_f32.insert(key, r);
+        }
+        let resampler = self.resamplers_f32.get_mut(&key).unwrap();
+
+        let input = vec![pcm.to_vec()];
+        let result = resampler
+            .process(&input, None)
+            .map_err(|e| format!("resample f32 {from_rate}->{to_rate}: {e}"))?;
+
+        Ok(result[0].clone())
+    }
+
+    /// Apply RNNoise ML noise suppression to 48kHz f32 PCM audio.
+    /// Processes in 480-sample (10ms) frames. State persists across calls.
+    /// Operates natively in f32 — no i16 conversion overhead.
+    pub fn denoise_f32(denoiser: &mut DenoiseState, pcm: &[f32]) -> Vec<f32> {
+        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];
+
+        // nnnoiseless expects f32 samples scaled as i16 range (-32768..32767).
+        for offset in (0..whole).step_by(frame_size) {
+            let input: Vec<f32> = pcm[offset..offset + frame_size]
+                .iter()
+                .map(|&s| s * 32768.0)
+                .collect();
+            denoiser.process_frame(&mut out_buf, &input);
+            output.extend(out_buf.iter().map(|&s| s / 32768.0));
+        }
+        if whole < total {
+            output.extend_from_slice(&pcm[whole..]);
+        }
+        output
+    }
+}
+
+/// Create a new standalone denoiser for per-leg inbound processing.
+pub fn new_denoiser() -> Box<DenoiseState<'static>> {
+    DenoiseState::new()
 }
 
 #[cfg(test)]

rust/crates/opus-codec/Cargo.toml

@@ -1,14 +0,0 @@
-[package]
-name = "opus-codec"
-version = "0.2.0"
-edition = "2021"
-
-[[bin]]
-name = "opus-codec"
-path = "src/main.rs"
-
-[dependencies]
-codec-lib = { path = "../codec-lib" }
-serde = { version = "1", features = ["derive"] }
-serde_json = "1"
-base64 = "0.22"

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

@@ -1,286 +0,0 @@
-/// Audio transcoding bridge for smartrust.
-///
-/// Thin CLI wrapper around `codec-lib`. Handles Opus ↔ G.722 ↔ PCMU transcoding.
-///
-/// Protocol:
-///   -> {"id":"1","method":"init","params":{}}
-///   <- {"id":"1","success":true,"result":{}}
-///   -> {"id":"2","method":"create_session","params":{"session_id":"call-abc"}}
-///   <- {"id":"2","success":true,"result":{}}
-///   -> {"id":"3","method":"transcode","params":{"session_id":"call-abc","data_b64":"...","from_pt":111,"to_pt":9}}
-///   <- {"id":"3","success":true,"result":{"data_b64":"..."}}
-///   -> {"id":"4","method":"destroy_session","params":{"session_id":"call-abc"}}
-///   <- {"id":"4","success":true,"result":{}}
-
-use base64::engine::general_purpose::STANDARD as B64;
-use base64::Engine as _;
-use codec_lib::{codec_sample_rate, TranscodeState};
-use serde::Deserialize;
-use std::collections::HashMap;
-use std::io::{self, BufRead, Write};
-
-#[derive(Deserialize)]
-struct Request {
-    id: String,
-    method: String,
-    #[serde(default)]
-    params: serde_json::Value,
-}
-
-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());
-    }
-    let _ = writeln!(out, "{}", resp);
-    let _ = out.flush();
-}
-
-/// 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>(
-    sessions: &'a mut HashMap<String, TranscodeState>,
-    default: &'a mut Option<TranscodeState>,
-    params: &serde_json::Value,
-) -> Option<&'a mut TranscodeState> {
-    if let Some(sid) = params.get("session_id").and_then(|v| v.as_str()) {
-        sessions.get_mut(sid)
-    } else {
-        default.as_mut()
-    }
-}
-
-fn main() {
-    let stdin = io::stdin();
-    let stdout = io::stdout();
-    let mut out = io::BufWriter::new(stdout.lock());
-
-    let _ = writeln!(out, r#"{{"event":"ready","data":{{}}}}"#);
-    let _ = out.flush();
-
-    let mut default_state: Option<TranscodeState> = None;
-    let mut sessions: HashMap<String, TranscodeState> = HashMap::new();
-
-    for line in stdin.lock().lines() {
-        let line = match line {
-            Ok(l) if !l.trim().is_empty() => l,
-            Ok(_) => continue,
-            Err(_) => break,
-        };
-
-        let req: Request = match serde_json::from_str(&line) {
-            Ok(r) => r,
-            Err(e) => {
-                respond(&mut out, "", false, None, Some(&format!("parse: {e}")));
-                continue;
-            }
-        };
-
-        match req.method.as_str() {
-            "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_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;
-                    }
-                };
-                if sessions.contains_key(&session_id) {
-                    respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
-                    continue;
-                }
-                match TranscodeState::new() {
-                    Ok(s) => {
-                        sessions.insert(session_id, s);
-                        respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
-                    }
-                    Err(e) => respond(&mut out, &req.id, false, None, Some(&e)),
-                }
-            }
-
-            "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;
-                    }
-                };
-                sessions.remove(session_id);
-                respond(&mut out, &req.id, true, Some(serde_json::json!({})), None);
-            }
-
-            "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;
-                    }
-                };
-                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;
-                    }
-                };
-                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;
-                    }
-                };
-
-                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,
-                        );
-                    }
-                    Err(e) => respond(&mut out, &req.id, false, None, Some(&e)),
-                }
-            }
-
-            "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;
-                    }
-                };
-                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;
-                    }
-                };
-                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;
-                    }
-                };
-
-                if data.len() % 2 != 0 {
-                    respond(
-                        &mut out,
-                        &req.id,
-                        false,
-                        None,
-                        Some("PCM data has odd byte count (expected 16-bit LE samples)"),
-                    );
-                    continue;
-                }
-
-                let pcm: Vec<i16> = data
-                    .chunks_exact(2)
-                    .map(|c| i16::from_le_bytes([c[0], c[1]]))
-                    .collect();
-
-                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;
-                    }
-                };
-
-                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,
-                        );
-                    }
-                    Err(e) => {
-                        respond(&mut out, &req.id, false, None, Some(&e));
-                    }
-                }
-            }
-
-            "encode" | "decode" => {
-                respond(
-                    &mut out,
-                    &req.id,
-                    false,
-                    None,
-                    Some("use 'transcode' command instead"),
-                );
-            }
-
-            _ => respond(
-                &mut out,
-                &req.id,
-                false,
-                None,
-                Some(&format!("unknown: {}", req.method)),
-            ),
-        }
-    }
-}

rust/crates/proxy-engine/Cargo.toml

@@ -10,6 +10,7 @@ path = "src/main.rs"
 [dependencies]
 codec-lib = { path = "../codec-lib" }
 sip-proto = { path = "../sip-proto" }
+nnnoiseless = { version = "0.5", default-features = false }
 tokio = { version = "1", features = ["full"] }
 serde = { version = "1", features = ["derive"] }
 serde_json = "1"
@@ -18,3 +19,8 @@ regex-lite = "0.1"
 webrtc = "0.8"
 rand = "0.8"
 hound = "3.5"
+kokoro-tts = { version = "0.3", default-features = false }
+ort = { version = "=2.0.0-rc.11", default-features = false, features = [
+    "std", "download-binaries", "copy-dylibs", "ndarray",
+    "tls-native-vendored"
+] }

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

@@ -10,9 +10,9 @@ use tokio::net::UdpSocket;
 use tokio::time::{self, Duration};
 
 /// Mixing sample rate used by the mixer (must stay in sync with mixer::MIX_RATE).
-const MIX_RATE: u32 = 16000;
+const MIX_RATE: u32 = 48000;
 /// Samples per 20ms frame at the mixing rate.
-const MIX_FRAME_SIZE: usize = 320;
+const MIX_FRAME_SIZE: usize = 960;
 
 /// Play a WAV file as RTP to a destination.
 /// Returns when playback is complete.
@@ -178,9 +178,9 @@ pub async fn play_beep(
     Ok((seq, ts))
 }
 
-/// Load a WAV file and split it into 20ms PCM frames at 16kHz.
+/// Load a WAV file and split it into 20ms f32 PCM frames at 48kHz.
 /// Used by the leg interaction system to prepare prompt audio for the mixer.
-pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<i16>>, String> {
+pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<f32>>, String> {
     let path = Path::new(wav_path);
     if !path.exists() {
         return Err(format!("WAV file not found: {wav_path}"));
@@ -191,17 +191,17 @@ pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<i16>>, String> {
     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 {
+    // Read all samples as f32 in [-1.0, 1.0].
+    let samples: Vec<f32> = if spec.bits_per_sample == 16 {
         reader
             .samples::<i16>()
             .filter_map(|s| s.ok())
+            .map(|s| s as f32 / 32768.0)
             .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!(
@@ -214,24 +214,24 @@ pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<i16>>, String> {
         return Ok(vec![]);
     }
 
-    // Resample to MIX_RATE (16kHz) if needed.
+    // Resample to MIX_RATE (48kHz) if needed.
     let resampled = if wav_rate != MIX_RATE {
         let mut transcoder = TranscodeState::new().map_err(|e| format!("codec init: {e}"))?;
         transcoder
-            .resample(&samples, wav_rate, MIX_RATE)
+            .resample_f32(&samples, wav_rate, MIX_RATE)
             .map_err(|e| format!("resample: {e}"))?
     } else {
         samples
     };
 
-    // Split into MIX_FRAME_SIZE (320) sample frames.
+    // Split into MIX_FRAME_SIZE (960) sample frames.
     let mut frames = Vec::new();
     let mut offset = 0;
     while offset < resampled.len() {
         let end = (offset + MIX_FRAME_SIZE).min(resampled.len());
         let mut frame = resampled[offset..end].to_vec();
         // Pad short final frame with silence.
-        frame.resize(MIX_FRAME_SIZE, 0);
+        frame.resize(MIX_FRAME_SIZE, 0.0);
         frames.push(frame);
         offset += MIX_FRAME_SIZE;
     }

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

@@ -20,6 +20,35 @@ use std::net::SocketAddr;
 use std::sync::Arc;
 use tokio::net::UdpSocket;
 
+/// Emit a `leg_added` event with full leg information.
+/// Free function (not a method) to avoid `&self` borrow conflicts when `self.calls` is borrowed.
+fn emit_leg_added_event(tx: &OutTx, call_id: &str, leg: &LegInfo) {
+    let metadata: serde_json::Value = if leg.metadata.is_empty() {
+        serde_json::json!({})
+    } else {
+        serde_json::Value::Object(
+            leg.metadata
+                .iter()
+                .map(|(k, v)| (k.clone(), v.clone()))
+                .collect(),
+        )
+    };
+    emit_event(
+        tx,
+        "leg_added",
+        serde_json::json!({
+            "call_id": call_id,
+            "leg_id": leg.id,
+            "kind": leg.kind.as_str(),
+            "state": leg.state.as_str(),
+            "codec": sip_proto::helpers::codec_name(leg.codec_pt),
+            "rtpPort": leg.rtp_port,
+            "remoteMedia": leg.remote_media.map(|a| format!("{}:{}", a.ip(), a.port())),
+            "metadata": metadata,
+        }),
+    );
+}
+
 pub struct CallManager {
     /// All active calls, keyed by internal call ID.
     pub calls: HashMap<String, Call>,
@@ -120,7 +149,19 @@ impl CallManager {
         }
 
         // Passthrough-style routing for inbound/outbound device↔provider calls.
-        self.route_passthrough_message(&call_id, &leg_id, msg, from_addr, socket, config)
+        // The sip_index only stores one leg for shared Call-IDs, so we need to
+        // determine which leg the message actually belongs to by comparing from_addr.
+        let actual_leg_id = self
+            .calls
+            .get(&call_id)
+            .and_then(|call| {
+                call.legs
+                    .values()
+                    .find(|l| l.signaling_addr == Some(from_addr))
+                    .map(|l| l.id.clone())
+            })
+            .unwrap_or(leg_id);
+        self.route_passthrough_message(&call_id, &actual_leg_id, msg, from_addr, socket, config)
             .await
     }
 
@@ -253,6 +294,11 @@ impl CallManager {
                                     dev_leg.state = LegState::Connected;
                                 }
                             }
+                            emit_event(
+                                &self.out_tx,
+                                "leg_state_changed",
+                                serde_json::json!({ "call_id": call_id, "leg_id": dev_leg_id, "state": "connected" }),
+                            );
 
                             // Wire device leg to mixer.
                             if let Some(dev_remote_addr) = dev_remote {
@@ -312,6 +358,8 @@ impl CallManager {
                         leg.state = LegState::Terminated;
                     }
                 }
+                emit_event(&self.out_tx, "leg_state_changed",
+                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "terminated" }));
                 emit_event(&self.out_tx, "call_ended",
                     serde_json::json!({ "call_id": call_id, "reason": reason, "duration": duration }));
                 self.terminate_call(call_id).await;
@@ -517,21 +565,30 @@ impl CallManager {
                     if let Some(leg) = call.legs.get_mut(this_leg_id) {
                         leg.state = LegState::Ringing;
                     }
+                    emit_event(&self.out_tx, "leg_state_changed",
+                        serde_json::json!({ "call_id": call_id, "leg_id": this_leg_id, "state": "ringing" }));
                 } else if code >= 200 && code < 300 {
                     let mut needs_wiring = false;
                     if let Some(leg) = call.legs.get_mut(this_leg_id) {
                         leg.state = LegState::Connected;
-                        // Learn remote media from SDP.
+                        // Learn remote media and negotiated codec from SDP answer.
                         if msg.has_sdp_body() {
                             if let Some(ep) = parse_sdp_endpoint(&msg.body) {
                                 if let Ok(addr) = format!("{}:{}", ep.address, ep.port).parse() {
                                     leg.remote_media = Some(addr);
                                 }
+                                // Use the codec from the SDP answer (what the remote actually selected).
+                                if let Some(pt) = ep.codec_pt {
+                                    leg.codec_pt = pt;
+                                }
                             }
                         }
                         needs_wiring = true;
                     }
 
+                    emit_event(&self.out_tx, "leg_state_changed",
+                        serde_json::json!({ "call_id": call_id, "leg_id": this_leg_id, "state": "connected" }));
+
                     if call.state != CallState::Connected {
                         call.state = CallState::Connected;
                         emit_event(&self.out_tx, "call_answered", serde_json::json!({ "call_id": call_id }));
@@ -677,15 +734,19 @@ impl CallManager {
         call.callee_number = Some(called_number);
         call.state = CallState::Ringing;
 
-        let codec_pt = provider_config.codecs.first().copied().unwrap_or(9);
+        let mut codec_pt = provider_config.codecs.first().copied().unwrap_or(9);
 
-        // Provider leg — extract media from SDP.
+        // Provider leg — extract media and negotiated codec from SDP.
         let mut provider_media: Option<SocketAddr> = None;
         if invite.has_sdp_body() {
             if let Some(ep) = parse_sdp_endpoint(&invite.body) {
                 if let Ok(addr) = format!("{}:{}", ep.address, ep.port).parse() {
                     provider_media = Some(addr);
                 }
+                // Use the codec from the provider's SDP offer (what they actually want to use).
+                if let Some(pt) = ep.codec_pt {
+                    codec_pt = pt;
+                }
             }
         }
 
@@ -755,6 +816,16 @@ impl CallManager {
         // Store the call.
         self.calls.insert(call_id.clone(), call);
 
+        // Emit leg_added for both initial legs.
+        if let Some(call) = self.calls.get(&call_id) {
+            if let Some(leg) = call.legs.get(&provider_leg_id) {
+                emit_leg_added_event(&self.out_tx, &call_id, leg);
+            }
+            if let Some(leg) = call.legs.get(&device_leg_id) {
+                emit_leg_added_event(&self.out_tx, &call_id, leg);
+            }
+        }
+
         Some(call_id)
     }
 
@@ -842,6 +913,14 @@ impl CallManager {
             .insert(sip_call_id, (call_id.clone(), leg_id));
 
         self.calls.insert(call_id.clone(), call);
+
+        // Emit leg_added for the provider leg.
+        if let Some(call) = self.calls.get(&call_id) {
+            for leg in call.legs.values() {
+                emit_leg_added_event(&self.out_tx, &call_id, leg);
+            }
+        }
+
         Some(call_id)
     }
 
@@ -866,11 +945,18 @@ impl CallManager {
         let lan_port = config.proxy.lan_port;
         let device_sip_call_id = invite.call_id().to_string();
 
+        // Extract just the user part from the request URI (e.g., "sip:16196000@10.0.0.1" → "16196000").
+        // extract_uri is for header values with angle brackets, not bare request URIs.
         let dialed_number = invite
             .request_uri()
-            .and_then(|uri| SipMessage::extract_uri(uri))
-            .unwrap_or(invite.request_uri().unwrap_or(""))
-            .to_string();
+            .map(|uri| {
+                let stripped = uri
+                    .strip_prefix("sip:")
+                    .or_else(|| uri.strip_prefix("sips:"))
+                    .unwrap_or(uri);
+                stripped.split('@').next().unwrap_or(stripped).to_string()
+            })
+            .unwrap_or_default();
 
         let provider_dest: SocketAddr = match provider_config.outbound_proxy.to_socket_addr() {
             Some(a) => a,
@@ -983,6 +1069,14 @@ impl CallManager {
             .insert(provider_sip_call_id, (call_id.clone(), provider_leg_id));
 
         self.calls.insert(call_id.clone(), call);
+
+        // Emit leg_added for both initial legs (device + provider).
+        if let Some(call) = self.calls.get(&call_id) {
+            for leg in call.legs.values() {
+                emit_leg_added_event(&self.out_tx, &call_id, leg);
+            }
+        }
+
         Some(call_id)
     }
 
@@ -1050,17 +1144,11 @@ impl CallManager {
         let call = self.calls.get_mut(call_id).unwrap();
         call.legs.insert(leg_id.clone(), leg_info);
 
-        emit_event(
-            &self.out_tx,
-            "leg_added",
-            serde_json::json!({
-                "call_id": call_id,
-                "leg_id": leg_id,
-                "kind": "sip-provider",
-                "state": "inviting",
-                "number": number,
-            }),
-        );
+        if let Some(call) = self.calls.get(call_id) {
+            if let Some(leg) = call.legs.get(&leg_id) {
+                emit_leg_added_event(&self.out_tx, call_id, leg);
+            }
+        }
 
         Some(leg_id)
     }
@@ -1126,17 +1214,11 @@ impl CallManager {
         let call = self.calls.get_mut(call_id).unwrap();
         call.legs.insert(leg_id.clone(), leg_info);
 
-        emit_event(
-            &self.out_tx,
-            "leg_added",
-            serde_json::json!({
-                "call_id": call_id,
-                "leg_id": leg_id,
-                "kind": "sip-device",
-                "state": "inviting",
-                "device_id": device_id,
-            }),
-        );
+        if let Some(call) = self.calls.get(call_id) {
+            if let Some(leg) = call.legs.get(&leg_id) {
+                emit_leg_added_event(&self.out_tx, call_id, leg);
+            }
+        }
 
         Some(leg_id)
     }
@@ -1223,6 +1305,13 @@ impl CallManager {
             None => return false,
         };
 
+        // Emit leg_removed for source call.
+        emit_event(
+            &self.out_tx,
+            "leg_removed",
+            serde_json::json!({ "call_id": source_call_id, "leg_id": leg_id }),
+        );
+
         // Update SIP index to point to the target call.
         if let Some(sip_cid) = &leg_info.sip_call_id {
             self.sip_index.insert(
@@ -1255,15 +1344,12 @@ impl CallManager {
         let target_call = self.calls.get_mut(target_call_id).unwrap();
         target_call.legs.insert(leg_id.to_string(), leg_info);
 
-        emit_event(
-            &self.out_tx,
-            "leg_transferred",
-            serde_json::json!({
-                "leg_id": leg_id,
-                "source_call_id": source_call_id,
-                "target_call_id": target_call_id,
-            }),
-        );
+        // Emit leg_added for target call.
+        if let Some(target) = self.calls.get(target_call_id) {
+            if let Some(leg) = target.legs.get(leg_id) {
+                emit_leg_added_event(&self.out_tx, target_call_id, leg);
+            }
+        }
 
         // Check if source call has too few legs remaining.
         let source_call = self.calls.get(source_call_id).unwrap();
@@ -1366,6 +1452,11 @@ impl CallManager {
                 }
             }
             leg.state = LegState::Terminated;
+            emit_event(
+                &self.out_tx,
+                "leg_state_changed",
+                serde_json::json!({ "call_id": call_id, "leg_id": leg.id, "state": "terminated" }),
+            );
         }
 
         emit_event(
@@ -1484,6 +1575,13 @@ impl CallManager {
         );
         self.calls.insert(call_id.to_string(), call);
 
+        // Emit leg_added for the provider leg.
+        if let Some(call) = self.calls.get(call_id) {
+            for leg in call.legs.values() {
+                emit_leg_added_event(&self.out_tx, call_id, leg);
+            }
+        }
+
         // Build recording path.
         let timestamp = std::time::SystemTime::now()
             .duration_since(std::time::UNIX_EPOCH)

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

@@ -35,7 +35,8 @@ pub fn create_leg_channels() -> LegChannels {
 }
 
 /// Spawn the inbound I/O task for a SIP leg.
-/// Reads RTP from the socket, strips the 12-byte header, sends payload to the mixer.
+/// Reads RTP from the socket, parses the variable-length header (RFC 3550),
+/// and sends the payload to the mixer.
 /// Returns the JoinHandle (exits when the inbound_tx channel is dropped).
 pub fn spawn_sip_inbound(
     rtp_socket: Arc<UdpSocket>,
@@ -51,12 +52,29 @@ pub fn spawn_sip_inbound(
                     }
                     let pt = buf[1] & 0x7F;
                     let marker = (buf[1] & 0x80) != 0;
+                    let seq = u16::from_be_bytes([buf[2], buf[3]]);
                     let timestamp = u32::from_be_bytes([buf[4], buf[5], buf[6], buf[7]]);
-                    let payload = buf[12..n].to_vec();
+
+                    // RFC 3550: header length = 12 + (CC * 4) + optional extension.
+                    let cc = (buf[0] & 0x0F) as usize;
+                    let has_extension = (buf[0] & 0x10) != 0;
+                    let mut offset = 12 + cc * 4;
+                    if has_extension {
+                        if offset + 4 > n {
+                            continue; // Malformed: extension header truncated.
+                        }
+                        let ext_len = u16::from_be_bytes([buf[offset + 2], buf[offset + 3]]) as usize;
+                        offset += 4 + ext_len * 4;
+                    }
+                    if offset >= n {
+                        continue; // No payload after header.
+                    }
+
+                    let payload = buf[offset..n].to_vec();
                     if payload.is_empty() {
                         continue;
                     }
-                    if inbound_tx.send(RtpPacket { payload, payload_type: pt, marker, timestamp }).await.is_err() {
+                    if inbound_tx.send(RtpPacket { payload, payload_type: pt, marker, seq, timestamp }).await.is_err() {
                         break; // Channel closed — leg removed.
                     }
                 }

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

@@ -21,6 +21,7 @@ mod rtp;
 mod sip_leg;
 mod sip_transport;
 mod tool_leg;
+mod tts;
 mod voicemail;
 mod webrtc_engine;
 
@@ -93,6 +94,9 @@ async fn main() {
     // WebRTC engine — separate lock to avoid deadlock with SIP handlers.
     let webrtc = Arc::new(Mutex::new(WebRtcEngine::new(out_tx.clone())));
 
+    // TTS engine — separate lock, lazy-loads model on first use.
+    let tts_engine = Arc::new(Mutex::new(tts::TtsEngine::new()));
+
     // Read commands from stdin.
     let stdin = tokio::io::stdin();
     let reader = BufReader::new(stdin);
@@ -113,11 +117,12 @@ async fn main() {
 
         let engine = engine.clone();
         let webrtc = webrtc.clone();
+        let tts_engine = tts_engine.clone();
         let out_tx = out_tx.clone();
 
         // Handle commands — some are async, so we spawn.
         tokio::spawn(async move {
-            handle_command(engine, webrtc, &out_tx, cmd).await;
+            handle_command(engine, webrtc, tts_engine, &out_tx, cmd).await;
         });
     }
 }
@@ -125,6 +130,7 @@ async fn main() {
 async fn handle_command(
     engine: Arc<Mutex<ProxyEngine>>,
     webrtc: Arc<Mutex<WebRtcEngine>>,
+    tts_engine: Arc<Mutex<tts::TtsEngine>>,
     out_tx: &OutTx,
     cmd: Command,
 ) {
@@ -150,6 +156,8 @@ async fn handle_command(
         "add_tool_leg" => handle_add_tool_leg(engine, out_tx, &cmd).await,
         "remove_tool_leg" => handle_remove_tool_leg(engine, out_tx, &cmd).await,
         "set_leg_metadata" => handle_set_leg_metadata(engine, out_tx, &cmd).await,
+        // TTS command — lock tts_engine only (no SIP/WebRTC contention).
+        "generate_tts" => handle_generate_tts(tts_engine, out_tx, &cmd).await,
         _ => respond_err(out_tx, &cmd.id, &format!("unknown command: {}", cmd.method)),
     }
 }
@@ -669,6 +677,10 @@ async fn handle_webrtc_link(
             "leg_id": session_id,
             "kind": "webrtc",
             "state": "connected",
+            "codec": "Opus",
+            "rtpPort": 0,
+            "remoteMedia": null,
+            "metadata": {},
         }));
 
         respond_ok(out_tx, &cmd.id, serde_json::json!({
@@ -1117,8 +1129,11 @@ async fn handle_add_tool_leg(
             "call_id": call_id,
             "leg_id": tool_leg_id,
             "kind": "tool",
-            "tool_type": tool_type_str,
             "state": "connected",
+            "codec": null,
+            "rtpPort": 0,
+            "remoteMedia": null,
+            "metadata": { "tool_type": tool_type_str },
         }),
     );
 
@@ -1218,3 +1233,16 @@ async fn handle_set_leg_metadata(
     leg.metadata.insert(key, value);
     respond_ok(out_tx, &cmd.id, serde_json::json!({}));
 }
+
+/// Handle `generate_tts` — synthesize text to a WAV file using Kokoro TTS.
+async fn handle_generate_tts(
+    tts_engine: Arc<Mutex<tts::TtsEngine>>,
+    out_tx: &OutTx,
+    cmd: &Command,
+) {
+    let mut tts = tts_engine.lock().await;
+    match tts.generate(&cmd.params).await {
+        Ok(result) => respond_ok(out_tx, &cmd.id, result),
+        Err(e) => respond_err(out_tx, &cmd.id, &e),
+    }
+}

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

@@ -3,9 +3,12 @@
 //! Each Call spawns one mixer task. Legs communicate with the mixer via
 //! tokio mpsc channels — no shared mutable state, no lock contention.
 //!
+//! Internal bus format: 48kHz f32 PCM (960 samples per 20ms frame).
+//! All encoding/decoding happens at leg boundaries. Per-leg inbound denoising at 48kHz.
+//!
 //! The mixer runs a 20ms tick loop:
-//! 1. Drain inbound channels, decode to PCM, resample to 16kHz
-//! 2. Compute total mix (sum of all **participant** legs' PCM as i32)
+//! 1. Drain inbound channels, decode to f32, resample to 48kHz, denoise per-leg
+//! 2. Compute total mix (sum of all **participant** legs' f32 PCM as f64)
 //! 3. For each participant leg: mix-minus = total - own, resample to leg codec rate, encode, send
 //! 4. For each isolated leg: play prompt frame or silence, check DTMF
 //! 5. For each tool leg: send per-source unmerged audio batch
@@ -13,16 +16,18 @@
 
 use crate::ipc::{emit_event, OutTx};
 use crate::rtp::{build_rtp_header, rtp_clock_increment};
-use codec_lib::{codec_sample_rate, TranscodeState};
+use codec_lib::{codec_sample_rate, new_denoiser, TranscodeState};
+use nnnoiseless::DenoiseState;
 use std::collections::{HashMap, VecDeque};
 use tokio::sync::{mpsc, oneshot};
 use tokio::task::JoinHandle;
 use tokio::time::{self, Duration, MissedTickBehavior};
 
-/// Mixing sample rate — 16kHz. G.722 is native, G.711 needs 2× upsample, Opus needs 3× downsample.
-const MIX_RATE: u32 = 16000;
+/// Mixing sample rate — 48kHz. Opus is native, G.722 needs 3× upsample, G.711 needs 6× upsample.
+/// All processing (denoising, mixing) happens at this rate in f32 for maximum quality.
+const MIX_RATE: u32 = 48000;
 /// Samples per 20ms frame at the mixing rate.
-const MIX_FRAME_SIZE: usize = 320; // 16000 * 0.020
+const MIX_FRAME_SIZE: usize = 960; // 48000 * 0.020
 
 /// A raw RTP payload received from a leg (no RTP header).
 pub struct RtpPacket {
@@ -30,6 +35,8 @@ pub struct RtpPacket {
     pub payload_type: u8,
     /// RTP marker bit (first packet of a DTMF event, etc.).
     pub marker: bool,
+    /// RTP sequence number for reordering.
+    pub seq: u16,
     /// RTP timestamp from the original packet header.
     pub timestamp: u32,
 }
@@ -47,8 +54,8 @@ enum LegRole {
 }
 
 struct IsolationState {
-    /// PCM frames at MIX_RATE (320 samples each) queued for playback.
-    prompt_frames: VecDeque<Vec<i16>>,
+    /// PCM frames at MIX_RATE (960 samples each, 48kHz f32) queued for playback.
+    prompt_frames: VecDeque<Vec<f32>>,
     /// Digits that complete the interaction (e.g., ['1', '2']).
     expected_digits: Vec<char>,
     /// Ticks remaining before timeout (decremented each tick after prompt ends).
@@ -88,8 +95,8 @@ pub struct ToolAudioBatch {
 /// One participant's 20ms audio frame.
 pub struct ToolAudioSource {
     pub leg_id: String,
-    /// PCM at 16kHz, MIX_FRAME_SIZE (320) samples.
-    pub pcm_16k: Vec<i16>,
+    /// PCM at 48kHz f32, MIX_FRAME_SIZE (960) samples.
+    pub pcm_48k: Vec<f32>,
 }
 
 /// Internal storage for a tool leg inside the mixer.
@@ -122,8 +129,8 @@ pub enum MixerCommand {
     /// DTMF from the leg is checked against expected_digits.
     StartInteraction {
         leg_id: String,
-        /// PCM frames at MIX_RATE (16kHz), each 320 samples.
-        prompt_pcm_frames: Vec<Vec<i16>>,
+        /// PCM frames at MIX_RATE (48kHz f32), each 960 samples.
+        prompt_pcm_frames: Vec<Vec<f32>>,
         expected_digits: Vec<char>,
         timeout_ms: u32,
         result_tx: oneshot::Sender<InteractionResult>,
@@ -149,10 +156,12 @@ pub enum MixerCommand {
 struct MixerLegSlot {
     codec_pt: u8,
     transcoder: TranscodeState,
+    /// Per-leg inbound denoiser (48kHz, 480-sample frames).
+    denoiser: Box<DenoiseState<'static>>,
     inbound_rx: mpsc::Receiver<RtpPacket>,
     outbound_tx: mpsc::Sender<Vec<u8>>,
-    /// Last decoded PCM frame at MIX_RATE (320 samples). Used for mix-minus.
-    last_pcm_frame: Vec<i16>,
+    /// Last decoded+denoised PCM frame at MIX_RATE (960 samples, 48kHz f32).
+    last_pcm_frame: Vec<f32>,
     /// Number of consecutive ticks with no inbound packet.
     silent_ticks: u32,
     // RTP output state.
@@ -220,9 +229,10 @@ async fn mixer_loop(
                         MixerLegSlot {
                             codec_pt,
                             transcoder,
+                            denoiser: new_denoiser(),
                             inbound_rx,
                             outbound_tx,
-                            last_pcm_frame: vec![0i16; MIX_FRAME_SIZE],
+                            last_pcm_frame: vec![0.0f32; MIX_FRAME_SIZE],
                             silent_ticks: 0,
                             rtp_seq: 0,
                             rtp_ts: 0,
@@ -311,16 +321,18 @@ async fn mixer_loop(
             continue;
         }
 
-        // ── 2. Drain inbound packets, decode to 16kHz PCM. ─────────
+        // ── 2. Drain inbound packets, decode to 48kHz f32 PCM. ────
         //    DTMF (PT 101) packets are collected separately.
+        //    Audio packets are sorted by sequence number and decoded
+        //    in order to maintain codec state (critical for G.722 ADPCM).
         let leg_ids: Vec<String> = legs.keys().cloned().collect();
         let mut dtmf_forward: Vec<(String, RtpPacket)> = Vec::new();
 
         for lid in &leg_ids {
             let slot = legs.get_mut(lid).unwrap();
 
-            // Drain channel — collect DTMF packets separately, keep latest audio.
-            let mut latest_audio: Option<RtpPacket> = None;
+            // Drain channel — collect DTMF separately, collect ALL audio packets.
+            let mut audio_packets: Vec<RtpPacket> = Vec::new();
             loop {
                 match slot.inbound_rx.try_recv() {
                     Ok(pkt) => {
@@ -328,33 +340,47 @@ async fn mixer_loop(
                             // DTMF telephone-event: collect for processing.
                             dtmf_forward.push((lid.clone(), pkt));
                         } else {
-                            latest_audio = Some(pkt);
+                            audio_packets.push(pkt);
                         }
                     }
                     Err(_) => break,
                 }
             }
 
-            if let Some(pkt) = latest_audio {
+            if !audio_packets.is_empty() {
                 slot.silent_ticks = 0;
-                match slot.transcoder.decode_to_pcm(&pkt.payload, pkt.payload_type) {
-                    Ok((pcm, rate)) => {
-                        // Resample to mixing rate if needed.
-                        let pcm_mix = if rate == MIX_RATE {
-                            pcm
-                        } else {
-                            slot.transcoder
-                                .resample(&pcm, rate, MIX_RATE)
-                                .unwrap_or_else(|_| vec![0i16; MIX_FRAME_SIZE])
-                        };
-                        // Pad or truncate to exactly MIX_FRAME_SIZE.
-                        let mut frame = pcm_mix;
-                        frame.resize(MIX_FRAME_SIZE, 0);
-                        slot.last_pcm_frame = frame;
-                    }
-                    Err(_) => {
-                        // Decode failed — use silence.
-                        slot.last_pcm_frame = vec![0i16; MIX_FRAME_SIZE];
+
+                // Sort by sequence number for correct codec state progression.
+                // This prevents G.722 ADPCM state corruption from out-of-order packets.
+                audio_packets.sort_by_key(|p| p.seq);
+
+                // Decode ALL packets in order (maintains codec state),
+                // but only keep the last decoded frame for mixing.
+                for pkt in &audio_packets {
+                    match slot.transcoder.decode_to_f32(&pkt.payload, pkt.payload_type) {
+                        Ok((pcm, rate)) => {
+                            // Resample to 48kHz mixing rate if needed.
+                            let pcm_48k = if rate == MIX_RATE {
+                                pcm
+                            } else {
+                                slot.transcoder
+                                    .resample_f32(&pcm, rate, MIX_RATE)
+                                    .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
+                            };
+                            // Per-leg inbound denoising at 48kHz.
+                            // Skip for Opus/WebRTC legs — browsers already apply
+                            // their own noise suppression via getUserMedia.
+                            let processed = if slot.codec_pt != codec_lib::PT_OPUS {
+                                TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
+                            } else {
+                                pcm_48k
+                            };
+                            // Pad or truncate to exactly MIX_FRAME_SIZE.
+                            let mut frame = processed;
+                            frame.resize(MIX_FRAME_SIZE, 0.0);
+                            slot.last_pcm_frame = frame;
+                        }
+                        Err(_) => {}
                     }
                 }
             } else if dtmf_forward.iter().any(|(src, _)| src == lid) {
@@ -364,17 +390,18 @@ async fn mixer_loop(
                 slot.silent_ticks += 1;
                 // After 150 ticks (3 seconds) of silence, zero out to avoid stale audio.
                 if slot.silent_ticks > 150 {
-                    slot.last_pcm_frame = vec![0i16; MIX_FRAME_SIZE];
+                    slot.last_pcm_frame = vec![0.0f32; MIX_FRAME_SIZE];
                 }
             }
         }
 
         // ── 3. Compute total mix from PARTICIPANT legs only. ────────
-        let mut total_mix = vec![0i32; MIX_FRAME_SIZE];
+        //    Accumulate as f64 to prevent precision loss when summing f32.
+        let mut total_mix = vec![0.0f64; MIX_FRAME_SIZE];
         for slot in legs.values() {
             if matches!(slot.role, LegRole::Participant) {
                 for (i, &s) in slot.last_pcm_frame.iter().enumerate().take(MIX_FRAME_SIZE) {
-                    total_mix[i] += s as i32;
+                    total_mix[i] += s as f64;
                 }
             }
         }
@@ -387,27 +414,27 @@ async fn mixer_loop(
         for (lid, slot) in legs.iter_mut() {
             match &mut slot.role {
                 LegRole::Participant => {
-                    // Mix-minus: total minus this leg's own contribution.
+                    // Mix-minus: total minus this leg's own contribution, clamped to [-1.0, 1.0].
                     let mut mix_minus = Vec::with_capacity(MIX_FRAME_SIZE);
                     for i in 0..MIX_FRAME_SIZE {
-                        let sample = (total_mix[i] - slot.last_pcm_frame[i] as i32)
-                            .clamp(-32768, 32767) as i16;
-                        mix_minus.push(sample);
+                        let sample =
+                            (total_mix[i] - slot.last_pcm_frame[i] as f64) as f32;
+                        mix_minus.push(sample.clamp(-1.0, 1.0));
                     }
 
-                    // Resample from 16kHz to the leg's codec native rate.
+                    // Resample from 48kHz to the leg's codec native rate.
                     let target_rate = codec_sample_rate(slot.codec_pt);
                     let resampled = if target_rate == MIX_RATE {
                         mix_minus
                     } else {
                         slot.transcoder
-                            .resample(&mix_minus, MIX_RATE, target_rate)
+                            .resample_f32(&mix_minus, MIX_RATE, target_rate)
                             .unwrap_or_default()
                     };
 
-                    // Encode to the leg's codec.
+                    // Encode to the leg's codec (f32 → i16 → codec inside encode_from_f32).
                     let encoded =
-                        match slot.transcoder.encode_from_pcm(&resampled, slot.codec_pt) {
+                        match slot.transcoder.encode_from_f32(&resampled, slot.codec_pt) {
                             Ok(e) if !e.is_empty() => e,
                             _ => continue,
                         };
@@ -456,21 +483,21 @@ async fn mixer_loop(
                             frame
                         } else {
                             state.prompt_done = true;
-                            vec![0i16; MIX_FRAME_SIZE]
+                            vec![0.0f32; MIX_FRAME_SIZE]
                         };
 
-                        // Encode prompt frame to the leg's codec (reuses existing encode path).
+                        // Encode prompt frame to the leg's codec.
                         let target_rate = codec_sample_rate(slot.codec_pt);
                         let resampled = if target_rate == MIX_RATE {
                             pcm_frame
                         } else {
                             slot.transcoder
-                                .resample(&pcm_frame, MIX_RATE, target_rate)
+                                .resample_f32(&pcm_frame, MIX_RATE, target_rate)
                                 .unwrap_or_default()
                         };
 
                         if let Ok(encoded) =
-                            slot.transcoder.encode_from_pcm(&resampled, slot.codec_pt)
+                            slot.transcoder.encode_from_f32(&resampled, slot.codec_pt)
                         {
                             if !encoded.is_empty() {
                                 let header = build_rtp_header(
@@ -523,7 +550,7 @@ async fn mixer_loop(
                 .filter(|(_, s)| matches!(s.role, LegRole::Participant))
                 .map(|(lid, s)| ToolAudioSource {
                     leg_id: lid.clone(),
-                    pcm_16k: s.last_pcm_frame.clone(),
+                    pcm_48k: s.last_pcm_frame.clone(),
                 })
                 .collect();
 
@@ -533,7 +560,7 @@ async fn mixer_loop(
                         .iter()
                         .map(|s| ToolAudioSource {
                             leg_id: s.leg_id.clone(),
-                            pcm_16k: s.pcm_16k.clone(),
+                            pcm_48k: s.pcm_48k.clone(),
                         })
                         .collect(),
                 };

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

@@ -2,7 +2,7 @@
 //!
 //! Tool legs are observer legs that receive individual audio streams from each
 //! participant in a call. The mixer pipes `ToolAudioBatch` every 20ms containing
-//! each participant's decoded PCM@16kHz tagged with source leg ID.
+//! each participant's decoded PCM@48kHz f32 tagged with source leg ID.
 //!
 //! Consumers:
 //! - **Recording**: writes per-source WAV files for speaker-separated recording.
@@ -37,20 +37,25 @@ pub fn spawn_recording_tool(
 
         while let Some(batch) = rx.recv().await {
             for source in &batch.sources {
-                // Skip silence-only frames (all zeros = no audio activity).
-                let has_audio = source.pcm_16k.iter().any(|&s| s != 0);
+                // Skip silence-only frames (near-zero = no audio activity).
+                let has_audio = source.pcm_48k.iter().any(|&s| s.abs() > 1e-6);
                 if !has_audio && !recorders.contains_key(&source.leg_id) {
                     continue; // Don't create a file for silence-only sources.
                 }
 
                 let recorder = recorders.entry(source.leg_id.clone()).or_insert_with(|| {
                     let path = format!("{}/{}-{}.wav", base_dir, call_id, source.leg_id);
-                    Recorder::new_pcm(&path, 16000, None).unwrap_or_else(|e| {
+                    Recorder::new_pcm(&path, 48000, None).unwrap_or_else(|e| {
                         panic!("failed to create recorder for {}: {e}", source.leg_id);
                     })
                 });
 
-                if !recorder.write_pcm(&source.pcm_16k) {
+                // Convert f32 [-1.0, 1.0] to i16 for WAV writing.
+                let pcm_i16: Vec<i16> = source.pcm_48k
+                    .iter()
+                    .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
+                    .collect();
+                if !recorder.write_pcm(&pcm_i16) {
                     // Max duration reached — stop recording this source.
                     break;
                 }
@@ -88,7 +93,7 @@ pub fn spawn_recording_tool(
 
 /// Spawn a transcription tool leg.
 ///
-/// The plumbing is fully real: it receives per-source unmerged PCM@16kHz from
+/// The plumbing is fully real: it receives per-source unmerged PCM@48kHz f32 from
 /// the mixer every 20ms. The consumer is a stub that accumulates audio and
 /// reports metadata on close. Future: will stream to a Whisper HTTP endpoint.
 pub fn spawn_transcription_tool(
@@ -105,7 +110,7 @@ pub fn spawn_transcription_tool(
         while let Some(batch) = rx.recv().await {
             for source in &batch.sources {
                 *source_samples.entry(source.leg_id.clone()).or_insert(0) +=
-                    source.pcm_16k.len() as u64;
+                    source.pcm_48k.len() as u64;
 
                 // TODO: Future — accumulate chunks and stream to Whisper endpoint.
                 // For now, the audio is received and counted but not processed.
@@ -118,7 +123,7 @@ pub fn spawn_transcription_tool(
             .map(|(leg_id, samples)| {
                 serde_json::json!({
                     "source_leg_id": leg_id,
-                    "duration_ms": (samples * 1000) / 16000,
+                    "duration_ms": (samples * 1000) / 48000,
                 })
             })
             .collect();

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

@@ -0,0 +1,138 @@
+//! Text-to-speech engine — synthesizes text to WAV files using Kokoro neural TTS.
+//!
+//! The model is loaded lazily on first use. If the model/voices files are not
+//! present, the generate command returns an error and the TS side falls back
+//! to espeak-ng.
+
+use kokoro_tts::{KokoroTts, Voice};
+use std::path::Path;
+
+/// Wraps the Kokoro TTS engine with lazy model loading.
+pub struct TtsEngine {
+    tts: Option<KokoroTts>,
+    /// Path that was used to load the current model (for cache invalidation).
+    loaded_model_path: String,
+    loaded_voices_path: String,
+}
+
+impl TtsEngine {
+    pub fn new() -> Self {
+        Self {
+            tts: None,
+            loaded_model_path: String::new(),
+            loaded_voices_path: String::new(),
+        }
+    }
+
+    /// Generate a WAV file from text.
+    ///
+    /// Params (from IPC JSON):
+    ///   - `model`: path to the ONNX model file
+    ///   - `voices`: path to the voices.bin file
+    ///   - `voice`: voice name (e.g. "af_bella")
+    ///   - `text`: text to synthesize
+    ///   - `output`: output WAV file path
+    pub async fn generate(&mut self, params: &serde_json::Value) -> Result<serde_json::Value, String> {
+        let model_path = params.get("model").and_then(|v| v.as_str())
+            .ok_or("missing 'model' param")?;
+        let voices_path = params.get("voices").and_then(|v| v.as_str())
+            .ok_or("missing 'voices' param")?;
+        let voice_name = params.get("voice").and_then(|v| v.as_str())
+            .unwrap_or("af_bella");
+        let text = params.get("text").and_then(|v| v.as_str())
+            .ok_or("missing 'text' param")?;
+        let output_path = params.get("output").and_then(|v| v.as_str())
+            .ok_or("missing 'output' param")?;
+
+        if text.is_empty() {
+            return Err("empty text".into());
+        }
+
+        // Check that model/voices files exist.
+        if !Path::new(model_path).exists() {
+            return Err(format!("model not found: {model_path}"));
+        }
+        if !Path::new(voices_path).exists() {
+            return Err(format!("voices not found: {voices_path}"));
+        }
+
+        // Lazy-load or reload if paths changed.
+        if self.tts.is_none()
+            || self.loaded_model_path != model_path
+            || self.loaded_voices_path != voices_path
+        {
+            eprintln!("[tts] loading model: {model_path}");
+            let tts = KokoroTts::new(model_path, voices_path)
+                .await
+                .map_err(|e| format!("model load failed: {e:?}"))?;
+            self.tts = Some(tts);
+            self.loaded_model_path = model_path.to_string();
+            self.loaded_voices_path = voices_path.to_string();
+        }
+
+        let tts = self.tts.as_ref().unwrap();
+        let voice = select_voice(voice_name);
+
+        eprintln!("[tts] synthesizing voice '{voice_name}': \"{text}\"");
+        let (samples, duration) = tts.synth(text, voice)
+            .await
+            .map_err(|e| format!("synthesis failed: {e:?}"))?;
+        eprintln!("[tts] synthesized {} samples in {duration:?}", samples.len());
+
+        // Write 24kHz 16-bit mono WAV.
+        let spec = hound::WavSpec {
+            channels: 1,
+            sample_rate: 24000,
+            bits_per_sample: 16,
+            sample_format: hound::SampleFormat::Int,
+        };
+
+        let mut writer = hound::WavWriter::create(output_path, spec)
+            .map_err(|e| format!("WAV create failed: {e}"))?;
+        for &sample in &samples {
+            let s16 = (sample * 32767.0).round().clamp(-32768.0, 32767.0) as i16;
+            writer.write_sample(s16).map_err(|e| format!("WAV write: {e}"))?;
+        }
+        writer.finalize().map_err(|e| format!("WAV finalize: {e}"))?;
+
+        eprintln!("[tts] wrote {output_path}");
+        Ok(serde_json::json!({ "output": output_path }))
+    }
+}
+
+/// Map voice name string to Kokoro Voice enum variant.
+fn select_voice(name: &str) -> Voice {
+    match name {
+        "af_bella" => Voice::AfBella(1.0),
+        "af_heart" => Voice::AfHeart(1.0),
+        "af_jessica" => Voice::AfJessica(1.0),
+        "af_nicole" => Voice::AfNicole(1.0),
+        "af_nova" => Voice::AfNova(1.0),
+        "af_sarah" => Voice::AfSarah(1.0),
+        "af_sky" => Voice::AfSky(1.0),
+        "af_river" => Voice::AfRiver(1.0),
+        "af_alloy" => Voice::AfAlloy(1.0),
+        "af_aoede" => Voice::AfAoede(1.0),
+        "af_kore" => Voice::AfKore(1.0),
+        "am_adam" => Voice::AmAdam(1.0),
+        "am_echo" => Voice::AmEcho(1.0),
+        "am_eric" => Voice::AmEric(1.0),
+        "am_fenrir" => Voice::AmFenrir(1.0),
+        "am_liam" => Voice::AmLiam(1.0),
+        "am_michael" => Voice::AmMichael(1.0),
+        "am_onyx" => Voice::AmOnyx(1.0),
+        "am_puck" => Voice::AmPuck(1.0),
+        "bf_alice" => Voice::BfAlice(1.0),
+        "bf_emma" => Voice::BfEmma(1.0),
+        "bf_isabella" => Voice::BfIsabella(1.0),
+        "bf_lily" => Voice::BfLily(1.0),
+        "bm_daniel" => Voice::BmDaniel(1.0),
+        "bm_fable" => Voice::BmFable(1.0),
+        "bm_george" => Voice::BmGeorge(1.0),
+        "bm_lewis" => Voice::BmLewis(1.0),
+        _ => {
+            eprintln!("[tts] unknown voice '{name}', falling back to af_bella");
+            Voice::AfBella(1.0)
+        }
+    }
+}

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

@@ -290,8 +290,9 @@ async fn browser_to_mixer_loop(
                         .send(RtpPacket {
                             payload: payload.to_vec(),
                             payload_type: PT_OPUS,
-                            marker: false,
-                            timestamp: 0,
+                            marker: rtp_packet.header.marker,
+                            seq: rtp_packet.header.sequence_number,
+                            timestamp: rtp_packet.header.timestamp,
                         })
                         .await;
                 }

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

@@ -197,10 +197,11 @@ pub fn compute_digest_auth(
 
 use crate::Endpoint;
 
-/// Parse the audio media port and connection address from an SDP body.
+/// Parse the audio media port, connection address, and preferred codec 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 mut codec_pt: Option<u8> = None;
 
     let normalized = sdp.replace("\r\n", "\n");
     for raw in normalized.split('\n') {
@@ -208,10 +209,16 @@ pub fn parse_sdp_endpoint(sdp: &str) -> Option<Endpoint> {
         if let Some(rest) = line.strip_prefix("c=IN IP4 ") {
             addr = Some(rest.trim());
         } else if let Some(rest) = line.strip_prefix("m=audio ") {
+            // m=audio <port> RTP/AVP <pt1> [<pt2> ...]
             let parts: Vec<&str> = rest.split_whitespace().collect();
             if !parts.is_empty() {
                 port = parts[0].parse().ok();
             }
+            // parts[1] is "RTP/AVP" or similar, parts[2..] are payload types.
+            // The first PT is the preferred codec.
+            if parts.len() > 2 {
+                codec_pt = parts[2].parse::<u8>().ok();
+            }
         }
     }
 
@@ -219,6 +226,7 @@ pub fn parse_sdp_endpoint(sdp: &str) -> Option<Endpoint> {
         (Some(a), Some(p)) => Some(Endpoint {
             address: a.to_string(),
             port: p,
+            codec_pt,
         }),
         _ => None,
     }

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

@@ -9,9 +9,11 @@ pub mod dialog;
 pub mod helpers;
 pub mod rewrite;
 
-/// Network endpoint (address + port).
+/// Network endpoint (address + port + optional negotiated codec).
 #[derive(Debug, Clone, PartialEq, Eq)]
 pub struct Endpoint {
     pub address: String,
     pub port: u16,
+    /// First payload type from the SDP `m=audio` line (the preferred codec).
+    pub codec_pt: Option<u8>,
 }

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

@@ -92,7 +92,7 @@ pub fn rewrite_sdp(body: &str, ip: &str, port: u16) -> (String, Option<Endpoint>
         .collect();
 
     let original = match (orig_addr, orig_port) {
-        (Some(a), Some(p)) => Some(Endpoint { address: a, port: p }),
+        (Some(a), Some(p)) => Some(Endpoint { address: a, port: p, codec_pt: None }),
         _ => None,
     };
 

rust/crates/tts-engine/Cargo.toml

@@ -1,18 +0,0 @@
-[package]
-name = "tts-engine"
-version = "0.1.0"
-edition = "2021"
-
-[[bin]]
-name = "tts-engine"
-path = "src/main.rs"
-
-[dependencies]
-kokoro-tts = { version = "0.3", default-features = false }
-# Pin to rc.11 matching kokoro-tts's expectation; enable vendored TLS to avoid system libssl-dev.
-ort = { version = "=2.0.0-rc.11", default-features = false, features = [
-    "std", "download-binaries", "copy-dylibs", "ndarray",
-    "tls-native-vendored"
-] }
-tokio = { version = "1", features = ["rt-multi-thread", "macros"] }
-hound = "3.5"

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

@@ -1,149 +0,0 @@
-/// TTS engine CLI — synthesizes text to a WAV file using Kokoro neural TTS.
-///
-/// Usage:
-///   echo "Hello world" | tts-engine --model kokoro-v1.0.onnx --voices voices.bin --output out.wav
-///   tts-engine --model kokoro-v1.0.onnx --voices voices.bin --output out.wav --text "Hello world"
-///
-/// Outputs 24kHz 16-bit mono WAV.
-
-use kokoro_tts::{KokoroTts, Voice};
-use std::io::{self, Read};
-
-fn parse_args() -> Result<(String, String, String, String, Option<String>), String> {
-    let args: Vec<String> = std::env::args().collect();
-    let mut model = String::new();
-    let mut voices = String::new();
-    let mut output = String::new();
-    let mut text: Option<String> = None;
-    let mut voice_name: Option<String> = None;
-
-    let mut i = 1;
-    while i < args.len() {
-        match args[i].as_str() {
-            "--model" => { i += 1; model = args.get(i).cloned().unwrap_or_default(); }
-            "--voices" => { i += 1; voices = args.get(i).cloned().unwrap_or_default(); }
-            "--output" | "--output_file" => { i += 1; output = args.get(i).cloned().unwrap_or_default(); }
-            "--text" => { i += 1; text = args.get(i).cloned(); }
-            "--voice" => { i += 1; voice_name = args.get(i).cloned(); }
-            _ => {}
-        }
-        i += 1;
-    }
-
-    if model.is_empty() { return Err("--model required".into()); }
-    if voices.is_empty() { return Err("--voices required".into()); }
-    if output.is_empty() { return Err("--output required".into()); }
-
-    let voice_str = voice_name.unwrap_or_else(|| "af_bella".into());
-
-    Ok((model, voices, output, voice_str, text))
-}
-
-fn select_voice(name: &str) -> Voice {
-    match name {
-        "af_bella" => Voice::AfBella(1.0),
-        "af_heart" => Voice::AfHeart(1.0),
-        "af_jessica" => Voice::AfJessica(1.0),
-        "af_nicole" => Voice::AfNicole(1.0),
-        "af_nova" => Voice::AfNova(1.0),
-        "af_sarah" => Voice::AfSarah(1.0),
-        "af_sky" => Voice::AfSky(1.0),
-        "af_river" => Voice::AfRiver(1.0),
-        "af_alloy" => Voice::AfAlloy(1.0),
-        "af_aoede" => Voice::AfAoede(1.0),
-        "af_kore" => Voice::AfKore(1.0),
-        "am_adam" => Voice::AmAdam(1.0),
-        "am_echo" => Voice::AmEcho(1.0),
-        "am_eric" => Voice::AmEric(1.0),
-        "am_fenrir" => Voice::AmFenrir(1.0),
-        "am_liam" => Voice::AmLiam(1.0),
-        "am_michael" => Voice::AmMichael(1.0),
-        "am_onyx" => Voice::AmOnyx(1.0),
-        "am_puck" => Voice::AmPuck(1.0),
-        "bf_alice" => Voice::BfAlice(1.0),
-        "bf_emma" => Voice::BfEmma(1.0),
-        "bf_isabella" => Voice::BfIsabella(1.0),
-        "bf_lily" => Voice::BfLily(1.0),
-        "bm_daniel" => Voice::BmDaniel(1.0),
-        "bm_fable" => Voice::BmFable(1.0),
-        "bm_george" => Voice::BmGeorge(1.0),
-        "bm_lewis" => Voice::BmLewis(1.0),
-        _ => {
-            eprintln!("[tts-engine] unknown voice '{}', falling back to af_bella", name);
-            Voice::AfBella(1.0)
-        }
-    }
-}
-
-#[tokio::main]
-async fn main() {
-    let (model_path, voices_path, output_path, voice_name, text_arg) = match parse_args() {
-        Ok(v) => v,
-        Err(e) => {
-            eprintln!("Error: {}", e);
-            eprintln!("Usage: tts-engine --model <model.onnx> --voices <voices.bin> --output <output.wav> [--text <text>] [--voice <voice_name>]");
-            std::process::exit(1);
-        }
-    };
-
-    // Get text from --text arg or stdin.
-    let text = match text_arg {
-        Some(t) => t,
-        None => {
-            let mut buf = String::new();
-            io::stdin().read_to_string(&mut buf).expect("failed to read stdin");
-            buf.trim().to_string()
-        }
-    };
-
-    if text.is_empty() {
-        eprintln!("[tts-engine] no text provided");
-        std::process::exit(1);
-    }
-
-    eprintln!("[tts-engine] loading model: {}", model_path);
-    let tts = match KokoroTts::new(&model_path, &voices_path).await {
-        Ok(t) => t,
-        Err(e) => {
-            eprintln!("[tts-engine] failed to load model: {:?}", e);
-            std::process::exit(1);
-        }
-    };
-
-    let voice = select_voice(&voice_name);
-    eprintln!("[tts-engine] synthesizing with voice '{}': \"{}\"", voice_name, text);
-
-    let (samples, duration) = match tts.synth(&text, voice).await {
-        Ok(r) => r,
-        Err(e) => {
-            eprintln!("[tts-engine] synthesis failed: {:?}", e);
-            std::process::exit(1);
-        }
-    };
-
-    eprintln!("[tts-engine] synthesized {} samples in {:?}", samples.len(), duration);
-
-    // Write WAV: 24kHz, 16-bit, mono (same format announcement.ts expects).
-    let spec = hound::WavSpec {
-        channels: 1,
-        sample_rate: 24000,
-        bits_per_sample: 16,
-        sample_format: hound::SampleFormat::Int,
-    };
-
-    let mut writer = match hound::WavWriter::create(&output_path, spec) {
-        Ok(w) => w,
-        Err(e) => {
-            eprintln!("[tts-engine] failed to create WAV: {}", e);
-            std::process::exit(1);
-        }
-    };
-
-    for &sample in &samples {
-        let s16 = (sample * 32767.0).round().clamp(-32768.0, 32767.0) as i16;
-        writer.write_sample(s16).unwrap();
-    }
-    writer.finalize().unwrap();
-
-    eprintln!("[tts-engine] wrote {}", output_path);
-}

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.15.0',
+  version: '1.17.1',
   description: 'undefined'
 }

ts/announcement.ts

@@ -1,59 +1,22 @@
 /**
- * TTS announcement module — pre-generates audio announcements using espeak-ng
- * and caches them as encoded RTP packets for playback during call setup.
+ * TTS announcement module — generates announcement WAV files at startup.
  *
- * On startup, generates the announcement WAV via espeak-ng (formant-based TTS
- * with highly accurate pronunciation), encodes each 20ms frame to G.722 (for
- * SIP) and Opus (for WebRTC) via the Rust transcoder, and caches the packets.
+ * Engine priority: espeak-ng (formant TTS, fast) → Kokoro neural TTS via
+ * proxy-engine → disabled.
  *
- * Falls back to the Rust tts-engine (Kokoro neural TTS) if espeak-ng is not
- * installed, and disables announcements if neither is available.
+ * The generated WAV is left on disk for Rust's audio_player / start_interaction
+ * to play during calls. No encoding or RTP playback happens in TypeScript.
  */
 
 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 announcement ready for RTP playback. */
-export interface IAnnouncementCache {
-  /** G.722 encoded frames (each is a 20ms frame payload, no RTP header). */
-  g722Frames: Buffer[];
-  /** Opus encoded frames for WebRTC playback. */
-  opusFrames: Buffer[];
-  /** Total duration in milliseconds. */
-  durationMs: number;
-}
+import { sendProxyCommand, isProxyReady } from './proxybridge.ts';
 
 // ---------------------------------------------------------------------------
 // State
 // ---------------------------------------------------------------------------
 
-let cachedAnnouncement: IAnnouncementCache | null = null;
-
 const TTS_DIR = path.join(process.cwd(), '.nogit', 'tts');
 const ANNOUNCEMENT_TEXT = "Hello. I'm connecting your call now.";
 const CACHE_WAV = path.join(TTS_DIR, 'announcement.wav');
@@ -64,12 +27,10 @@ const KOKORO_VOICES = 'voices.bin';
 const KOKORO_VOICE = 'af_bella';
 
 // ---------------------------------------------------------------------------
-// Initialization
+// TTS generators
 // ---------------------------------------------------------------------------
 
-/**
- * Check if espeak-ng is available on the system.
- */
+/** Check if espeak-ng is available on the system. */
 function isEspeakAvailable(): boolean {
   try {
     execSync('which espeak-ng', { stdio: 'pipe' });
@@ -79,10 +40,7 @@ function isEspeakAvailable(): boolean {
   }
 }
 
-/**
- * Generate announcement WAV via espeak-ng (primary engine).
- * Returns true on success.
- */
+/** Generate announcement WAV via espeak-ng (primary engine). */
 function generateViaEspeak(wavPath: string, text: string, log: (msg: string) => void): boolean {
   log('[tts] generating announcement audio via espeak-ng...');
   try {
@@ -98,11 +56,8 @@ function generateViaEspeak(wavPath: string, text: string, log: (msg: string) =>
   }
 }
 
-/**
- * Generate announcement WAV via Kokoro TTS (fallback engine).
- * Returns true on success.
- */
-function generateViaKokoro(wavPath: string, text: string, log: (msg: string) => void): boolean {
+/** Generate announcement WAV via Kokoro TTS (fallback, runs inside proxy-engine). */
+async function generateViaKokoro(wavPath: string, text: string, log: (msg: string) => void): Promise<boolean> {
   const modelPath = path.join(TTS_DIR, KOKORO_MODEL);
   const voicesPath = path.join(TTS_DIR, KOKORO_VOICES);
 
@@ -111,25 +66,21 @@ function generateViaKokoro(wavPath: string, text: string, log: (msg: string) =>
     return false;
   }
 
-  const root = process.cwd();
-  const ttsBinPaths = [
-    path.join(root, 'dist_rust', 'tts-engine'),
-    path.join(root, 'rust', 'target', 'release', 'tts-engine'),
-    path.join(root, 'rust', 'target', 'debug', 'tts-engine'),
-  ];
-  const ttsBin = ttsBinPaths.find((p) => fs.existsSync(p));
-  if (!ttsBin) {
-    log('[tts] tts-engine binary not found — Kokoro fallback unavailable');
+  if (!isProxyReady()) {
+    log('[tts] proxy-engine not ready — Kokoro fallback unavailable');
     return false;
   }
 
   log('[tts] generating announcement audio via Kokoro TTS (fallback)...');
   try {
-    execSync(
-      `"${ttsBin}" --model "${modelPath}" --voices "${voicesPath}" --voice "${KOKORO_VOICE}" --output "${wavPath}" --text "${text}"`,
-      { timeout: 120000, stdio: 'pipe' },
-    );
-    log('[tts] Kokoro WAV generated');
+    await sendProxyCommand('generate_tts', {
+      model: modelPath,
+      voices: voicesPath,
+      voice: KOKORO_VOICE,
+      text,
+      output: wavPath,
+    });
+    log('[tts] Kokoro WAV generated (via proxy-engine)');
     return true;
   } catch (e: any) {
     log(`[tts] Kokoro failed: ${e.message}`);
@@ -137,40 +88,13 @@ function generateViaKokoro(wavPath: string, text: string, log: (msg: string) =>
   }
 }
 
-/**
- * Read a WAV file and detect its sample rate from the fmt chunk.
- * Returns { pcm, sampleRate } or null on failure.
- */
-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; // default
-  let offset = 12;
-  let pcm: Buffer | null = null;
-
-  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++;
-  }
-
-  if (!pcm) return null;
-  return { pcm, sampleRate };
-}
+// ---------------------------------------------------------------------------
+// Initialization
+// ---------------------------------------------------------------------------
 
 /**
- * Pre-generate the announcement audio and encode to G.722 + Opus frames.
- * Must be called after the codec bridge is initialized.
+ * Pre-generate the announcement WAV file.
+ * Must be called after the proxy engine is initialized.
  *
  * Engine priority: espeak-ng → Kokoro → disabled.
  */
@@ -178,7 +102,6 @@ export async function initAnnouncement(log: (msg: string) => void): Promise<bool
   fs.mkdirSync(TTS_DIR, { recursive: true });
 
   try {
-    // Generate WAV if not cached.
     if (!fs.existsSync(CACHE_WAV)) {
       let generated = false;
 
@@ -189,9 +112,9 @@ export async function initAnnouncement(log: (msg: string) => void): Promise<bool
         log('[tts] espeak-ng not installed — trying Kokoro fallback');
       }
 
-      // Fall back to Kokoro.
+      // Fall back to Kokoro (via proxy-engine).
       if (!generated) {
-        generated = generateViaKokoro(CACHE_WAV, ANNOUNCEMENT_TEXT, log);
+        generated = await generateViaKokoro(CACHE_WAV, ANNOUNCEMENT_TEXT, log);
       }
 
       if (!generated) {
@@ -200,49 +123,7 @@ export async function initAnnouncement(log: (msg: string) => void): Promise<bool
       }
     }
 
-    // Read WAV and extract raw PCM + sample rate.
-    const result = readWavWithRate(CACHE_WAV);
-    if (!result) {
-      log('[tts] failed to parse WAV file');
-      return false;
-    }
-
-    const { pcm, sampleRate } = result;
-
-    // Wait for codec bridge to be ready.
-    if (!isCodecReady()) {
-      log('[tts] codec bridge not ready — will retry');
-      return false;
-    }
-
-    // Encode in 20ms chunks. The Rust encoder resamples to each codec's native rate.
-    const FRAME_SAMPLES = Math.floor(sampleRate * 0.02);
-    const FRAME_BYTES = FRAME_SAMPLES * 2; // 16-bit = 2 bytes per sample
-    const totalFrames = Math.floor(pcm.length / FRAME_BYTES);
-
-    const g722Frames: Buffer[] = [];
-    const opusFrames: Buffer[] = [];
-
-    log(`[tts] encoding ${totalFrames} frames (${FRAME_SAMPLES} samples/frame @ ${sampleRate}Hz)...`);
-    for (let i = 0; i < totalFrames; i++) {
-      const framePcm = pcm.subarray(i * FRAME_BYTES, (i + 1) * FRAME_BYTES);
-      const pcmBuf = Buffer.from(framePcm);
-      const [g722, opus] = await Promise.all([
-        encodePcm(pcmBuf, sampleRate, 9),   // G.722 for SIP devices
-        encodePcm(pcmBuf, sampleRate, 111),  // Opus for WebRTC browsers
-      ]);
-      if (g722) g722Frames.push(g722);
-      if (opus) opusFrames.push(opus);
-      if (!g722 && !opus && i < 3) log(`[tts] frame ${i} encode failed`);
-    }
-
-    cachedAnnouncement = {
-      g722Frames,
-      opusFrames,
-      durationMs: totalFrames * 20,
-    };
-
-    log(`[tts] announcement cached: ${g722Frames.length} frames (${(totalFrames * 20 / 1000).toFixed(1)}s)`);
+    log('[tts] announcement WAV ready');
     return true;
   } catch (e: any) {
     log(`[tts] init error: ${e.message}`);
@@ -250,100 +131,7 @@ export async function initAnnouncement(log: (msg: string) => void): Promise<bool
   }
 }
 
-// ---------------------------------------------------------------------------
-// Playback
-// ---------------------------------------------------------------------------
-
-/**
- * Play the pre-cached announcement to an RTP endpoint.
- *
- * @param sendPacket - function to send a raw RTP packet
- * @param ssrc - SSRC to use in RTP headers
- * @param onDone - called when the announcement finishes
- * @returns a cancel function, or null if no announcement is cached
- */
-export function playAnnouncement(
-  sendPacket: (pkt: Buffer) => void,
-  ssrc: number,
-  onDone?: () => void,
-): (() => void) | null {
-  if (!cachedAnnouncement || cachedAnnouncement.g722Frames.length === 0) {
-    onDone?.();
-    return null;
-  }
-
-  const frames = cachedAnnouncement.g722Frames;
-  const PT = 9; // G.722
-  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 cancel function.
-  return () => clearInterval(timer);
+/** Get the path to the cached announcement WAV, or null if not generated. */
+export function getAnnouncementWavPath(): string | null {
+  return fs.existsSync(CACHE_WAV) ? CACHE_WAV : null;
 }
-
-/**
- * Play pre-cached Opus announcement to a WebRTC PeerConnection sender.
- *
- * @param sendRtpPacket - function to send a raw RTP packet via sender.sendRtp()
- * @param ssrc - SSRC to use in RTP headers
- * @param onDone - called when announcement finishes
- * @returns cancel function, or null if no announcement cached
- */
-export function playAnnouncementToWebRtc(
-  sendRtpPacket: (pkt: Buffer) => void,
-  ssrc: number,
-  counters: { seq: number; ts: number },
-  onDone?: () => void,
-): (() => void) | null {
-  if (!cachedAnnouncement || cachedAnnouncement.opusFrames.length === 0) {
-    onDone?.();
-    return null;
-  }
-
-  const frames = cachedAnnouncement.opusFrames;
-  const PT = 111; // Opus
-  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]);
-    sendRtpPacket(pkt);
-
-    counters.seq++;
-    counters.ts += 960; // Opus at 48kHz: 960 samples per 20ms
-    frameIdx++;
-  }, 20);
-
-  return () => clearInterval(timer);
-}
-
-/** Check if an announcement is cached and ready. */
-export function isAnnouncementReady(): boolean {
-  return cachedAnnouncement !== null && cachedAnnouncement.g722Frames.length > 0;
-}
-

ts/call/prompt-cache.ts

@@ -1,55 +1,31 @@
 /**
- * PromptCache — manages multiple named audio prompts for IVR and voicemail.
+ * PromptCache — manages named audio prompt WAV files 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.
+ * Generates WAV files via espeak-ng (primary) or Kokoro TTS through the
+ * proxy-engine (fallback). Also supports loading pre-existing WAV files
+ * and programmatic tone generation.
  *
- * 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.
+ * All audio playback happens in Rust (audio_player / start_interaction).
+ * This module only manages WAV files on disk.
  */
 
 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;
-}
+import { sendProxyCommand, isProxyReady } from '../proxybridge.ts';
 
 // ---------------------------------------------------------------------------
 // Types
 // ---------------------------------------------------------------------------
 
-/** A pre-encoded prompt ready for RTP playback. */
+/** A cached prompt — just a WAV file path and metadata. */
 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. */
+  /** Path to the WAV file on disk. */
+  wavPath: string;
+  /** Total duration in milliseconds (approximate, from WAV header). */
   durationMs: number;
 }
 
@@ -82,84 +58,61 @@ function generateViaEspeak(wavPath: string, text: string): boolean {
   }
 }
 
-/** Generate WAV via Kokoro TTS. */
-function generateViaKokoro(wavPath: string, text: string, voice: string): boolean {
+/** Generate WAV via Kokoro TTS (runs inside proxy-engine). */
+async function generateViaKokoro(wavPath: string, text: string, voice: string): Promise<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;
+  if (!isProxyReady()) return false;
 
   try {
-    execSync(
-      `"${ttsBin}" --model "${modelPath}" --voices "${voicesPath}" --voice "${voice}" --output "${wavPath}" --text "${text}"`,
-      { timeout: 120000, stdio: 'pipe' },
-    );
+    await sendProxyCommand('generate_tts', {
+      model: modelPath,
+      voices: voicesPath,
+      voice,
+      text,
+      output: wavPath,
+    });
     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;
+/** Read a WAV file's duration from its header. */
+function getWavDurationMs(wavPath: string): number {
+  try {
+    const wav = fs.readFileSync(wavPath);
+    if (wav.length < 44) return 0;
+    if (wav.toString('ascii', 0, 4) !== 'RIFF') return 0;
 
-  let sampleRate = 22050;
-  let pcm: Buffer | null = null;
-  let offset = 12;
+    let sampleRate = 16000;
+    let dataSize = 0;
+    let bitsPerSample = 16;
+    let channels = 1;
+    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);
+    while (offset < wav.length - 8) {
+      const chunkId = wav.toString('ascii', offset, offset + 4);
+      const chunkSize = wav.readUInt32LE(offset + 4);
+      if (chunkId === 'fmt ') {
+        channels = wav.readUInt16LE(offset + 10);
+        sampleRate = wav.readUInt32LE(offset + 12);
+        bitsPerSample = wav.readUInt16LE(offset + 22);
+      }
+      if (chunkId === 'data') {
+        dataSize = chunkSize;
+      }
+      offset += 8 + chunkSize;
+      if (offset % 2 !== 0) offset++;
     }
-    if (chunkId === 'data') {
-      pcm = wav.subarray(offset + 8, offset + 8 + chunkSize);
-    }
-    offset += 8 + chunkSize;
-    if (offset % 2 !== 0) offset++;
+
+    const bytesPerSample = (bitsPerSample / 8) * channels;
+    const totalSamples = bytesPerSample > 0 ? dataSize / bytesPerSample : 0;
+    return sampleRate > 0 ? Math.round((totalSamples / sampleRate) * 1000) : 0;
+  } catch {
+    return 0;
   }
-
-  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 };
 }
 
 // ---------------------------------------------------------------------------
@@ -195,7 +148,7 @@ export class PromptCache {
   }
 
   /**
-   * Generate a TTS prompt and cache it.
+   * Generate a TTS prompt WAV and cache its path.
    * Uses espeak-ng (primary) or Kokoro (fallback).
    */
   async generatePrompt(id: string, text: string, voice = 'af_bella'): Promise<ICachedPrompt | null> {
@@ -207,14 +160,14 @@ export class PromptCache {
       this.espeakAvailable = isEspeakAvailable();
     }
 
-    // Generate WAV.
-    let generated = false;
+    // Generate WAV if not already on disk.
     if (!fs.existsSync(wavPath)) {
+      let generated = false;
       if (this.espeakAvailable) {
         generated = generateViaEspeak(wavPath, text);
       }
       if (!generated) {
-        generated = generateViaKokoro(wavPath, text, voice);
+        generated = await generateViaKokoro(wavPath, text, voice);
       }
       if (!generated) {
         this.log(`[prompt-cache] failed to generate TTS for "${id}"`);
@@ -223,49 +176,22 @@ export class PromptCache {
       this.log(`[prompt-cache] generated WAV for "${id}"`);
     }
 
-    return this.loadWavPrompt(id, wavPath);
+    return this.registerWav(id, wavPath);
   }
 
   /**
-   * Load a WAV file as a prompt and cache it.
+   * Load a pre-existing WAV file as a prompt.
    */
   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;
+    return this.registerWav(id, wavPath);
   }
 
   /**
-   * 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)
+   * Generate a beep tone WAV and cache it.
    */
   async generateBeep(
     id: string,
@@ -273,149 +199,77 @@ export class PromptCache {
     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);
+    fs.mkdirSync(TTS_DIR, { recursive: true });
+    const wavPath = path.join(TTS_DIR, `prompt-${id}.wav`);
 
-    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;
+    if (!fs.existsSync(wavPath)) {
+      // Generate 16kHz 16-bit mono sine wave WAV.
+      const sampleRate = 16000;
+      const totalSamples = Math.floor((sampleRate * durationMs) / 1000);
+      const pcm = Buffer.alloc(totalSamples * 2);
 
-      const sample = Math.round(Math.sin(2 * Math.PI * freqHz * t) * amplitude * envelope);
-      pcm.writeInt16LE(Math.max(-32768, Math.min(32767, sample)), i * 2);
+      for (let i = 0; i < totalSamples; i++) {
+        const t = i / sampleRate;
+        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);
+      }
+
+      // Write WAV file.
+      const headerSize = 44;
+      const dataSize = pcm.length;
+      const wav = Buffer.alloc(headerSize + dataSize);
+
+      // RIFF header
+      wav.write('RIFF', 0);
+      wav.writeUInt32LE(36 + dataSize, 4);
+      wav.write('WAVE', 8);
+
+      // fmt chunk
+      wav.write('fmt ', 12);
+      wav.writeUInt32LE(16, 16);        // chunk size
+      wav.writeUInt16LE(1, 20);         // PCM format
+      wav.writeUInt16LE(1, 22);         // mono
+      wav.writeUInt32LE(sampleRate, 24);
+      wav.writeUInt32LE(sampleRate * 2, 28); // byte rate
+      wav.writeUInt16LE(2, 32);         // block align
+      wav.writeUInt16LE(16, 34);        // bits per sample
+
+      // data chunk
+      wav.write('data', 36);
+      wav.writeUInt32LE(dataSize, 40);
+      pcm.copy(wav, 44);
+
+      fs.writeFileSync(wavPath, wav);
+      this.log(`[prompt-cache] beep WAV generated for "${id}"`);
     }
 
-    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;
+    return this.registerWav(id, wavPath);
   }
 
-  /**
-   * Remove a prompt from the cache.
-   */
+  /** Remove a prompt from the cache. */
   remove(id: string): void {
     this.prompts.delete(id);
   }
 
-  /**
-   * Clear all cached prompts.
-   */
+  /** Clear all cached prompts. */
   clear(): void {
     this.prompts.clear();
   }
-}
 
-// ---------------------------------------------------------------------------
-// Standalone playback helpers (for use by SystemLeg)
-// ---------------------------------------------------------------------------
+  // -------------------------------------------------------------------------
+  // Internal
+  // -------------------------------------------------------------------------
 
-/**
- * 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;
+  private registerWav(id: string, wavPath: string): ICachedPrompt {
+    const durationMs = getWavDurationMs(wavPath);
+    const prompt: ICachedPrompt = { id, wavPath, durationMs };
+    this.prompts.set(id, prompt);
+    this.log(`[prompt-cache] cached "${id}": ${wavPath} (${(durationMs / 1000).toFixed(1)}s)`);
+    return prompt;
   }
-
-  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/opusbridge.ts

@@ -1,199 +0,0 @@
-/**
- * Audio transcoding bridge — uses smartrust to communicate with the Rust
- * opus-codec binary, which handles Opus ↔ G.722 ↔ PCMU/PCMA transcoding.
- *
- * All codec conversion happens in Rust (libopus + SpanDSP G.722 port).
- * The TypeScript side just passes raw payloads back and forth.
- */
-
-import path from 'node:path';
-import { RustBridge } from '@push.rocks/smartrust';
-
-// ---------------------------------------------------------------------------
-// Command type map for smartrust
-// ---------------------------------------------------------------------------
-
-type TCodecCommands = {
-  init: {
-    params: Record<string, never>;
-    result: Record<string, never>;
-  };
-  create_session: {
-    params: { session_id: string };
-    result: Record<string, never>;
-  };
-  destroy_session: {
-    params: { session_id: string };
-    result: Record<string, never>;
-  };
-  transcode: {
-    params: { data_b64: string; from_pt: number; to_pt: number; session_id?: string; direction?: string };
-    result: { data_b64: string };
-  };
-  encode_pcm: {
-    params: { data_b64: string; sample_rate: number; to_pt: number; session_id?: string };
-    result: { data_b64: string };
-  };
-};
-
-// ---------------------------------------------------------------------------
-// Bridge singleton
-// ---------------------------------------------------------------------------
-
-let bridge: RustBridge<TCodecCommands> | null = null;
-let initialized = false;
-
-function buildLocalPaths(): string[] {
-  const root = process.cwd();
-  return [
-    path.join(root, 'dist_rust', 'opus-codec'),
-    path.join(root, 'rust', 'target', 'release', 'opus-codec'),
-    path.join(root, 'rust', 'target', 'debug', 'opus-codec'),
-  ];
-}
-
-let logFn: ((msg: string) => void) | undefined;
-
-/**
- * Initialize the audio transcoding bridge. Spawns the Rust binary.
- */
-export async function initCodecBridge(log?: (msg: string) => void): Promise<boolean> {
-  if (initialized && bridge) return true;
-  logFn = log;
-
-  try {
-    bridge = new RustBridge<TCodecCommands>({
-      binaryName: 'opus-codec',
-      localPaths: buildLocalPaths(),
-    });
-
-    const spawned = await bridge.spawn();
-    if (!spawned) {
-      log?.('[codec] failed to spawn opus-codec binary');
-      bridge = null;
-      return false;
-    }
-
-    // Auto-restart: reset state when the Rust process exits so the next
-    // transcode attempt triggers re-initialization instead of silent failure.
-    bridge.on('exit', () => {
-      logFn?.('[codec] Rust audio transcoder process exited — will re-init on next use');
-      bridge = null;
-      initialized = false;
-    });
-
-    await bridge.sendCommand('init', {} as any);
-    initialized = true;
-    log?.('[codec] Rust audio transcoder initialized (Opus + G.722 + PCMU/PCMA)');
-    return true;
-  } catch (e: any) {
-    log?.(`[codec] init error: ${e.message}`);
-    bridge = null;
-    return false;
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Session management — per-call codec isolation
-// ---------------------------------------------------------------------------
-
-/**
- * Create an isolated codec session. Each session gets its own Opus/G.722
- * encoder/decoder state, preventing concurrent calls from corrupting each
- * other's stateful codec predictions.
- */
-export async function createSession(sessionId: string): Promise<boolean> {
-  if (!bridge || !initialized) {
-    // Attempt auto-reinit if bridge died.
-    const ok = await initCodecBridge(logFn);
-    if (!ok) return false;
-  }
-  try {
-    await bridge!.sendCommand('create_session', { session_id: sessionId });
-    return true;
-  } catch (e: any) {
-    logFn?.(`[codec] create_session error: ${e?.message || e}`);
-    return false;
-  }
-}
-
-/**
- * Destroy a codec session, freeing its encoder/decoder state.
- */
-export async function destroySession(sessionId: string): Promise<void> {
-  if (!bridge || !initialized) return;
-  try {
-    await bridge.sendCommand('destroy_session', { session_id: sessionId });
-  } catch {
-    // Best-effort cleanup.
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Transcoding
-// ---------------------------------------------------------------------------
-
-/**
- * Transcode an RTP payload between two codecs.
- * All codec work (Opus, G.722, PCMU, PCMA) + resampling happens in Rust.
- *
- * @param data - raw RTP payload (no header)
- * @param fromPT - source payload type (0=PCMU, 8=PCMA, 9=G.722, 111=Opus)
- * @param toPT - target payload type
- * @param sessionId - optional session for isolated codec state
- * @returns transcoded payload, or null on failure
- */
-export async function transcode(data: Buffer, fromPT: number, toPT: number, sessionId?: string, direction?: string): Promise<Buffer | null> {
-  if (!bridge || !initialized) return null;
-  try {
-    const params: any = {
-      data_b64: data.toString('base64'),
-      from_pt: fromPT,
-      to_pt: toPT,
-    };
-    if (sessionId) params.session_id = sessionId;
-    if (direction) params.direction = direction;
-    const result = await bridge.sendCommand('transcode', params);
-    return Buffer.from(result.data_b64, 'base64');
-  } catch {
-    return null;
-  }
-}
-
-/**
- * Encode raw 16-bit PCM to a target codec.
- * @param pcmData - raw 16-bit LE PCM bytes
- * @param sampleRate - input sample rate (e.g. 22050 for Piper TTS)
- * @param toPT - target payload type (9=G.722, 111=Opus, 0=PCMU, 8=PCMA)
- * @param sessionId - optional session for isolated codec state
- */
-export async function encodePcm(pcmData: Buffer, sampleRate: number, toPT: number, sessionId?: string): Promise<Buffer | null> {
-  if (!bridge || !initialized) return null;
-  try {
-    const params: any = {
-      data_b64: pcmData.toString('base64'),
-      sample_rate: sampleRate,
-      to_pt: toPT,
-    };
-    if (sessionId) params.session_id = sessionId;
-    const result = await bridge.sendCommand('encode_pcm', params);
-    return Buffer.from(result.data_b64, 'base64');
-  } catch (e: any) {
-    console.error('[encodePcm] error:', e?.message || e);
-    return null;
-  }
-}
-
-/** Check if the codec bridge is ready. */
-export function isCodecReady(): boolean {
-  return initialized && bridge !== null;
-}
-
-/** Shut down the codec bridge. */
-export function shutdownCodecBridge(): void {
-  if (bridge) {
-    try { bridge.kill(); } catch { /* ignore */ }
-    bridge = null;
-    initialized = false;
-  }
-}

ts/proxybridge.ts

@@ -79,6 +79,10 @@ type TProxyCommands = {
     params: { call_id: string; leg_id: string; key: string; value: unknown };
     result: Record<string, never>;
   };
+  generate_tts: {
+    params: { model: string; voices: string; voice: string; text: string; output: string };
+    result: { output: string };
+  };
 };
 
 // ---------------------------------------------------------------------------
@@ -493,6 +497,15 @@ export function isProxyReady(): boolean {
   return initialized && bridge !== null;
 }
 
+/** Send an arbitrary command to the proxy engine bridge. */
+export async function sendProxyCommand<K extends keyof TProxyCommands>(
+  method: K,
+  params: TProxyCommands[K]['params'],
+): Promise<TProxyCommands[K]['result']> {
+  if (!bridge || !initialized) throw new Error('proxy engine not initialized');
+  return bridge.sendCommand(method as string, params as any) as any;
+}
+
 /** Shut down the proxy engine. */
 export function shutdownProxyEngine(): void {
   if (bridge) {

ts/sipproxy.ts

@@ -24,7 +24,6 @@ import {
   getAllBrowserDeviceIds,
   getBrowserDeviceWs,
 } from './webrtcbridge.ts';
-import { initCodecBridge } from './opusbridge.ts';
 import { initAnnouncement } from './announcement.ts';
 import { PromptCache } from './call/prompt-cache.ts';
 import { VoiceboxManager } from './voicebox.ts';
@@ -426,9 +425,9 @@ async function startProxyEngine(): Promise<void> {
         id: data.leg_id,
         type: data.kind,
         state: data.state,
-        codec: null,
-        rtpPort: null,
-        remoteMedia: null,
+        codec: data.codec ?? null,
+        rtpPort: data.rtpPort ?? null,
+        remoteMedia: data.remoteMedia ?? null,
         metadata: data.metadata || {},
       });
     }
@@ -523,9 +522,8 @@ async function startProxyEngine(): Promise<void> {
   const deviceList = appConfig.devices.map((d) => d.displayName).join(', ');
   log(`proxy engine started | LAN ${appConfig.proxy.lanIp}:${appConfig.proxy.lanPort} | providers: ${providerList} | devices: ${deviceList}`);
 
-  // Initialize audio codec bridge (still needed for WebRTC transcoding).
+  // Generate TTS audio (WAV files on disk, played by Rust audio_player).
   try {
-    await initCodecBridge(log);
     await initAnnouncement(log);
 
     // Pre-generate prompts.
@@ -547,7 +545,7 @@ async function startProxyEngine(): Promise<void> {
     }
     log(`[startup] prompts cached: ${promptCache.listIds().join(', ') || 'none'}`);
   } catch (e) {
-    log(`[codec] init failed: ${e}`);
+    log(`[tts] init failed: ${e}`);
   }
 }
 

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.15.0',
+  version: '1.17.1',
   description: 'undefined'
 }