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.
add B2BUA SIP leg handling and WebRTC call bridging for outbound calls
- introduce a new SipLeg module to manage outbound provider dialogs, including INVITE lifecycle, digest auth retries, ACK handling, media endpoint tracking, and termination
- store outbound dashboard calls as B2BUA calls in the call manager and emit provider media details on call_answered for bridge setup
- separate SIP and WebRTC engine locking to avoid contention and deadlocks while linking sessions to call RTP sockets
- add bidirectional RTP bridging between provider SIP media and browser WebRTC audio using the allocated RTP socket
- wire browser webrtc-accept events in the frontend and sipproxy so session-to-call linking can occur when media and acceptance arrive in either order
sync provider registrations when config is reloaded after save
- Adds provider state synchronization to create registrations for new providers, remove deleted providers, and re-register providers whose configuration changed.
- Preserves detected public IP when rebuilding provider state for updated provider configs.
- Triggers registration status broadcasts after config reload so runtime state stays aligned with saved provider settings.
## 2026-04-09 - 1.7.0 - feat(audio)
add directional RNNoise suppression to transcoding and preserve RTP continuity during announcement handoff
- adds per-direction RNNoise denoising in the Rust opus transcoder with 48kHz frame processing
- passes transcoder direction through the TypeScript bridge so browser-bound and SIP-bound audio use separate suppression state
- shares RTP sequence and timestamp counters between announcements and live provider audio to avoid browser jitter buffer discontinuities
- updates the background restart script to rebuild the Rust codec before bundling
## 2026-04-09 - 1.6.0 - feat(codec,call,web-ui)
add isolated codec sessions for concurrent call transcoding and fix form input event handling
- adds per-session Rust codec state with create_session and destroy_session support to prevent concurrent calls from corrupting Opus and G.722 state
- updates WebRTC call setup to await transcoder initialization and clean up codec sessions on teardown
- improves codec robustness with safer PCM handling, larger Opus decode buffers, auto-reinit after bridge exit, and telephony tuning for Opus encoding
- switches multiple web UI forms from changeValue to input/newValue events so text fields and checkboxes update reliably
## 2026-04-09 - 1.5.1 - fix(call,opus-codec)
improve SIP/WebRTC RTP routing and use cached FFT resampling for transcoding
- route browser-to-provider RTP through the SIP leg socket to avoid symmetric RTP double-path issues
- preserve active calls when a WebRTC leg disconnects by removing only the terminated leg and updating call state correctly
- set dynamic SIP From URIs and display names for mediated and conference invites
- replace the custom resampler with rubato FFT resampling and cache resamplers by rate pair and chunk size for continuous audio streams
## 2026-04-09 - 1.5.0 - feat(contacts,phone)
add starred contacts quick dial flow and improve call/media handling
- Add starred contacts to config and app state, with sorting and contact stats support in the contacts view.
- Let contact actions open the phone view with a selected contact and show starred contacts as quick-dial entries.
- Update the phone and contacts UI to use dees input components and show all devices with online/offline status.
- Prevent raw codec passthrough on transcode failure and forward RTP using the parsed payload instead of slicing serialized packets.
- Adjust the 3x Opus downsampling lowpass filter coefficients for improved resampling behavior.
## 2026-04-09 - 1.4.0 - feat(calling)
add cached TTS announcements, external participant dialing, and call history support
- pre-generate Piper-based announcements and cache encoded G.722 and Opus RTP frames for SIP and WebRTC playback
- add encode_pcm support in the Rust codec bridge with anti-aliased PCM resampling for direct PCM-to-codec encoding
- add API and UI support for dialing external participants into an existing call
- record completed calls in bounded call history and expose them in the web UI
- improve RTP handling with stable SSRC usage, codec-specific silence payloads, and safer async transcoding sequencing
## 2026-04-09 - 1.3.1 - fix(router)
prevent duplicate app navigation callbacks when syncing tab selection with URL updates
- add an optional skipCallback flag to router navigation so URL changes do not re-trigger view loading
- update sipproxy app tab selection handling to push browser history without causing navigation loops
- limit router-driven appdash view loading to browser back/forward navigation
## 2026-04-09 - 1.3.0 - feat(webrtc-ui)
add routed dashboard views with contacts and provider management, and fix browser WebRTC call linking and audio forwarding
- Defers browser WebRTC leg creation until the offer arrives, links the standalone session on accept, and routes SIP-to-browser audio through the WebRTC peer connection to prevent one-way audio
- Adds URL-routed app views for overview, calls, phone, contacts, providers, and log, replacing the previous dashboard-only layout
- Introduces contacts support in config and app state, exposes contacts in status payloads, and adds UI to create, edit, delete, and call contacts
- Expands provider configuration management to support editing full provider settings plus adding and removing providers
- Makes outbound provider selection tolerant of deployments with no configured providers to avoid forwarding errors
## 2026-04-09 - 1.2.0 - feat(call)
introduce a hub-based call manager with SIP and WebRTC legs, unified RTP port pooling, and expanded call control APIs
- Adds a new call hub model with Call, Leg, SipLeg, WebRtcLeg, shared types, and centralized RTP forwarding/transcoding.
- Replaces the legacy call originator flow with CallManager-driven call handling in the SIP proxy bootstrap.
- Extends the web API and dashboard to support provider selection, richer call status, adding/removing legs, transfers, and WebRTC call control.
- Refactors WebRTC bridge responsibilities to signaling only while moving media handling into the new call layer.
## 2026-04-09 - 1.1.1 - fix(calloriginator)
handle 200 OK retransmits without rebuilding the bridge and always start RTP keepalive silence when remote media is available
- Resend ACK for repeated 200 OK responses and return early once the call is already connected.
- Prevent duplicate bridge setup and repeated state transitions on SIP response retransmits.
- Start the silence stream for all calls with remote media to keep the provider media path alive and help open the RTP NAT path.
## 2026-04-09 - 1.1.0 - feat(webrtc)
add browser audio device controls and improve WebRTC audio bridging
- add microphone and speaker selection to the browser softphone
- show local and remote audio level meters during active WebRTC calls
- ensure the WebRTC answer negotiates sendrecv audio so the server can send RTP to the browser
- extract and use sender SSRC for browser-bound RTP and add diagnostics for SIP-to-WebRTC media flow
- avoid starting the SIP silence stream for browser calls to prevent media path issues with easybell
## 2026-04-09 - 1.0.0 - platform
Major 1.0.0 release delivering a configurable multi-provider SIP router with web UI, browser softphone support, and Rust-based media transcoding.
- Introduced a generic multi-provider SIP proxy/router architecture with local registrar, per-provider upstream registration, digest auth handling, configurable routing, device status, quick-dials, and provider-specific call quirks
- Added a full web dashboard and later migrated it to a component-based frontend using `@design.estate/dees-element`, with provider/device/call/log views, status APIs, WebSocket updates, hot reload behavior, and improved app shell/modals
- Added configuration APIs and settings UI for editing providers, devices, inbound routing, quick-dials, and browser ringing behavior, persisted to `.nogit/config.json`
- Renamed the project from `grandstream-sip-proxy` to `SipRouter`, including binary naming and updated user-agent branding
- Added WebRTC softphone support for browsers, including browser device registration, incoming-call notifications, accept/reject handling, targeted WebSocket messaging, and device-aware call routing
- Improved browser device UX with automatic registration, browser-specific naming, `(this browser)` labeling, duplicate/stale registration cleanup, IP display, and always-visible configured devices with connection state
- Added HTTPS support for browser audio flows, including inbound browser ringing, single-port HTTP/HTTPS serving, and re-enabled `getUserMedia` once TLS was available
- Refined call routing so calls go to the selected device instead of a hardcoded endpoint, including explicit device selection requirements and defaulting outbound origin to the first SIP device instead of the browser
- Improved call stability by starting a silence stream when leg B connects to prevent provider teardown before media is flowing
- Migrated runtime/server infrastructure from Deno to Node.js/tsx, replacing Deno-specific APIs with Node.js HTTP/HTTPS and WebSocket implementations
- Completed media handling in Rust, replacing buggy TypeScript G.722 processing with a single IPC transcoding path covering Opus, G.722, PCMU, PCMA, resampling, and verified browser-to-mobile audio bridging
- Included prior build/runtime work such as TypeScript migration, Deno support with single-binary compilation, and related setup as part of the path to the final 1.0.0 architecture
## 2026-04-08 - unknown - initial
Initial SIP-aware proxy for Grandstream HT801 ↔ easybell connectivity.
- Added SIP message parsing with binary passthrough to avoid corrupting STUN keep-alives and RTP
- Implemented Contact and Request-URI rewriting between LAN and public addresses
- Added SDP rewriting and per-call RTP relay sockets
- Added NAT priming and G.722 silence streaming after `200 OK` so easybell detects inbound media promptly
- Inserted `Record-Route` so in-dialog ACK/BYE/re-INVITE continue through the proxy
- Included captured device setting snapshots and setup documentation for diagnosing registration issues
