fix(proxy-engine,codec-lib,sip-proto,ts): preserve negotiated media details and improve RTP audio handling across call legs

changelog.md

@@ -1,5 +1,13 @@
 # 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
 

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

@@ -301,19 +301,59 @@ impl TranscodeState {
 
     /// 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> {
-        let (pcm_i16, rate) = self.decode_to_pcm(data, pt)?;
+        match pt {
-        let pcm_f32 = pcm_i16.iter().map(|&s| s as f32 / 32768.0).collect();
+            PT_OPUS => {
-        Ok((pcm_f32, rate))
+                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> {
-        let pcm_i16: Vec<i16> = pcm
+        match pt {
-            .iter()
+            PT_OPUS => {
-            .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
+                let mut buf = vec![0u8; 4000];
-            .collect();
+                let n: usize = self
-        self.encode_from_pcm(&pcm_i16, pt)
+                    .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.

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>,
@@ -265,6 +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 {
@@ -324,6 +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;
@@ -529,21 +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 }));
@@ -689,15 +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;
+                }
             }
         }
 
@@ -767,6 +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)
     }
 
@@ -854,6 +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)
     }
 
@@ -1002,6 +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)
     }
 
@@ -1069,17 +1144,11 @@ impl CallManager {
         let call = self.calls.get_mut(call_id).unwrap();
         call.legs.insert(leg_id.clone(), leg_info);
 
-        emit_event(
+        if let Some(call) = self.calls.get(call_id) {
-            &self.out_tx,
+            if let Some(leg) = call.legs.get(&leg_id) {
-            "leg_added",
+                emit_leg_added_event(&self.out_tx, call_id, leg);
-            serde_json::json!({
+            }
-                "call_id": call_id,
+        }
-                "leg_id": leg_id,
-                "kind": "sip-provider",
-                "state": "inviting",
-                "number": number,
-            }),
-        );
 
         Some(leg_id)
     }
@@ -1145,17 +1214,11 @@ impl CallManager {
         let call = self.calls.get_mut(call_id).unwrap();
         call.legs.insert(leg_id.clone(), leg_info);
 
-        emit_event(
+        if let Some(call) = self.calls.get(call_id) {
-            &self.out_tx,
+            if let Some(leg) = call.legs.get(&leg_id) {
-            "leg_added",
+                emit_leg_added_event(&self.out_tx, call_id, leg);
-            serde_json::json!({
+            }
-                "call_id": call_id,
+        }
-                "leg_id": leg_id,
-                "kind": "sip-device",
-                "state": "inviting",
-                "device_id": device_id,
-            }),
-        );
 
         Some(leg_id)
     }
@@ -1242,6 +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(
@@ -1274,15 +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(
+        // Emit leg_added for target call.
-            &self.out_tx,
+        if let Some(target) = self.calls.get(target_call_id) {
-            "leg_transferred",
+            if let Some(leg) = target.legs.get(leg_id) {
-            serde_json::json!({
+                emit_leg_added_event(&self.out_tx, target_call_id, leg);
-                "leg_id": leg_id,
+            }
-                "source_call_id": source_call_id,
+        }
-                "target_call_id": target_call_id,
-            }),
-        );
 
         // Check if source call has too few legs remaining.
         let source_call = self.calls.get(source_call_id).unwrap();
@@ -1385,6 +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(
@@ -1503,6 +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

@@ -677,6 +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!({
@@ -1125,8 +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 },
         }),
     );
 

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

@@ -35,6 +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,
 }
@@ -319,16 +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.
+            // Drain channel — collect DTMF separately, collect ALL audio packets.
-            let mut latest_audio: Option<RtpPacket> = None;
+            let mut audio_packets: Vec<RtpPacket> = Vec::new();
             loop {
                 match slot.inbound_rx.try_recv() {
                     Ok(pkt) => {
@@ -336,35 +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_f32(&pkt.payload, pkt.payload_type) {
+
-                    Ok((pcm, rate)) => {
+                // Sort by sequence number for correct codec state progression.
-                        // Resample to 48kHz mixing rate if needed.
+                // This prevents G.722 ADPCM state corruption from out-of-order packets.
-                        let pcm_48k = if rate == MIX_RATE {
+                audio_packets.sort_by_key(|p| p.seq);
-                            pcm
+
-                        } else {
+                // Decode ALL packets in order (maintains codec state),
-                            slot.transcoder
+                // but only keep the last decoded frame for mixing.
-                                .resample_f32(&pcm, rate, MIX_RATE)
+                for pkt in &audio_packets {
-                                .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
+                    match slot.transcoder.decode_to_f32(&pkt.payload, pkt.payload_type) {
-                        };
+                        Ok((pcm, rate)) => {
-                        // Per-leg inbound denoising at 48kHz.
+                            // Resample to 48kHz mixing rate if needed.
-                        let denoised = TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k);
+                            let pcm_48k = if rate == MIX_RATE {
-                        // Pad or truncate to exactly MIX_FRAME_SIZE.
+                                pcm
-                        let mut frame = denoised;
+                            } else {
-                        frame.resize(MIX_FRAME_SIZE, 0.0);
+                                slot.transcoder
-                        slot.last_pcm_frame = frame;
+                                    .resample_f32(&pcm, rate, MIX_RATE)
-                    }
+                                    .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
-                    Err(_) => {
+                            };
-                        // Decode failed — use silence.
+                            // Per-leg inbound denoising at 48kHz.
-                        slot.last_pcm_frame = vec![0.0f32; MIX_FRAME_SIZE];
+                            // 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) {

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,
+                            marker: rtp_packet.header.marker,
-                            timestamp: 0,
+                            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,
     };
 

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.17.0',
+  version: '1.17.1',
   description: 'undefined'
 }

ts/sipproxy.ts

@@ -425,9 +425,9 @@ async function startProxyEngine(): Promise<void> {
         id: data.leg_id,
         type: data.kind,
         state: data.state,
-        codec: null,
+        codec: data.codec ?? null,
-        rtpPort: null,
+        rtpPort: data.rtpPort ?? null,
-        remoteMedia: null,
+        remoteMedia: data.remoteMedia ?? null,
         metadata: data.metadata || {},
       });
     }

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.17.0',
+  version: '1.17.1',
   description: 'undefined'
 }