--- sidebar_position: 5 title: Changelog description: Release history and version notes for the Firetell JavaScript / TypeScript Client SDK (@firetell/firetell-client-sdk). --- # Changelog All notable changes to the [`@firetell/firetell-client-sdk`](https://www.npmjs.com/package/@firetell/firetell-client-sdk) package are documented in this file. The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.1.0/), and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html). --- ## [1.2.4] - 2026-09-09 ### Added - **Camera Mute/Unmute (Toggle Camera)**: - Added `call.muteVideo()`, `call.unmuteVideo()`, and `call.toggleCamera()` methods on `Call`. - Added `call.isCameraOff: boolean` property tracking local camera state. - Added `ECallEventName.CAMERA` (`"camera"`) event emitted locally and on remote camera state updates via signaling. - **Native Screen Sharing (`getDisplayMedia` + `replaceTrack`)**: - Added `call.startScreenShare()`, `call.stopScreenShare()`, and `call.toggleScreenShare()` on `Call`. - Seamlessly replaces video track on the active `RTCRtpSender` without SDP renegotiation or call interruption. - Automatically restores the original camera video track when the user stops sharing screen. - Added `call.isScreenSharing: boolean` property tracking active screen share state. - Added `ECallEventName.SCREEN_SHARE` (`"screenShare"`) event emitted locally and on remote screen share updates. - **SDP Video Auto-Detection in `call.offer` & `accept()`**: - `Call.handleWsMessage`: Automatically falls back to inspecting the remote SDP offer for active video media sections (`/m=video [1-9]/`) if the explicit `is_video` boolean flag is omitted in the `call.offer` payload, ensuring `call.isVideo` is reliably set on the callee side. - `Call.accept()`: Evaluates `this.remoteDescription.sdp` to guarantee `this.isVideo` is true before setting up local media, prompting the browser for both Camera and Microphone permissions (`{ video: true, audio: true }`). - **Resilient Remote Stream Track Assembly**: - Improved `RTCPeerConnection.ontrack` in `_setupWebrtcMedia`: Automatically creates a `MediaStream` and attaches tracks incrementally if `event.streams[0]` is missing, ensuring both remote audio and video tracks are emitted via `ECallEventName.REMOTE_STREAM`. - **Caller Metadata Continuity**: - Added support for `from_avatar` and `transfer_reason` fields in `call.offer` event processing so callee avatar and transfer context remain intact through WebRTC session negotiation. ### Fixed - **Callee Video Negotiation**: - Fixed an issue where incoming video calls defaulted to audio-only on the callee side due to missing `is_video` flag in signaling WebSocket payloads. - **Example Demo Web App**: - Fixed `hideIncomingBanner` reference error on call dismissal. - Added dynamic `updateVideoUI()` helper to automatically bind and display local/remote video elements when tracks arrive. --- ## [1.2.3] - 2026-09-04 ### Changed - **`call.transfer()` simplified** (**Breaking**): - Signature changed from `transfer(targetUsername, teamId?, reason?)` to `transfer(target, reason?)`. - `target` now accepts: agent username, extension number (e.g. `"100"`), team ID (`te_...`), or SIP account ID (`si_...`). - Removed `teamId` parameter — target type is auto-resolved server-side via prefix-based resolution. - **`FiretellClient.sendTransfer()` simplified** (**Breaking**): - Signature changed from `sendTransfer(callId, targetUsername, teamId?, reason?)` to `sendTransfer(callId, target, reason?)`. - HTTP fallback body changed from `{ target_username, team_id, reason }` to `{ target, reason }`. - **Per-call WebSocket payloads cleaned up**: - Removed redundant `call_id` from all WS event payloads (`call.hangup`, `call.answer`, `call.reject`, `call.transfer`, `call.dtmf`, `call.mute`, `call.hold`, `call.unhold`). The per-call WS connection inherently binds the socket to a specific call, making `call_id` unnecessary. - `call.transfer` payload field renamed from `to` to `target` for consistency with the HTTP API. --- ## [1.2.2] - 2026-09-03 ### Added - **`getPhoneNumbers()` REST API on `FiretellClient`**: - Added `public async getPhoneNumbers(options?: { page?: number; limit?: number }): Promise` to query phone numbers (DIDs) accessible by the authenticated agent and their shared teams. - Automatically sends the agent's JWT to `GET /api/v1/call-center/phone-numbers`. - Added `API_ENDPOINTS.PHONE_NUMBERS` (`"/api/v1/call-center/phone-numbers"`). - Added `IClientPhoneNumber` and `IClientPhoneNumbersResponse` interfaces exporting phone number details (`id`, `number`, `title`, `country_code`, `dial_code`, `status`, `capabilities`, `enable_outbound`, `shared_teams_id`). - **Static & Named SDK Version Exports**: - Added static `FiretellClient.VERSION` property on the class for direct, synchronous version access without instantiation. - Exported `SDK_VERSION` constant directly from package root (`import { SDK_VERSION } from '@firetell/firetell-client-sdk'`). - Added `"./package.json": "./package.json"` to the package `exports` map for direct version inspection by package managers and bundlers. --- ## [1.2.1] - 2026-08-22 ### Added - **`transfer_reason` in `call.ring` Event & `Call` Instance**: - Added `transfer_reason?: string` to `ICallRingParams` payload in the `call.ring` SSE event. - Exposed `call.transferReason?: string` on the `Call` instance when a transferred call session is initialized. - Allows frontends, Webphones, and Console agents to display the contextual transfer reason provided by Voice AI agents or human agents during ringing. --- ## [1.2.0] - 2026-08-20 ### Added - **Dedicated Per-Call SSE Event Stream (`Call.connectCallEventStream`)**: - Automatically establishes a dedicated SSE connection (`/stream?call_id=`) for each active call session (`call.start()`, `call.accept()`, `call.joinSession()`, and `client.createCallSession()`). - Subscribes to the Per-Call channel for real-time speech-to-text transcriptions and recording events without flooding the global workspace event stream. - Automatically disconnects and cleans up the stream upon call termination (`call.destroy()`). - **Call-Scoped Real-Time Transcription Events & APIs**: - Added `ECallEventName.TRANSCRIPTION`, `TRANSCRIPTION_STARTED`, `TRANSCRIPTION_DIALOGUE`, and `TRANSCRIPTION_COMPLETED`. - Added ergonomic helper listener methods on `Call`: `call.onDialogue((dialogue) => ...)` and `call.onTranscription((event) => ...)`. - Exported TypeScript interfaces: `ITranscriptionStartedEvent`, `ITranscriptionDialogueEvent`, `ITranscriptionCompletedEvent`, and `TranscriptionEvent`. - **Call-Scoped Real-Time Recording Events & APIs**: - Added `ECallEventName.RECORDING`, `RECORDING_STARTED`, `RECORDING_COMPLETED`, and `RECORDING_READY`. - Added ergonomic helper listener method on `Call`: `call.onRecording((event) => ...)`. - Exported TypeScript interfaces: `ICallRecordingStartedEvent`, `ICallRecordingCompletedEvent`, `ICallRecordingReadyEvent`, and `CallRecordingEvent`. - **Public Getters on `FiretellClient`**: - Added `client.getBaseUrl()` and `client.getJwt()` getters. ### Changed & Improved - **Strict Single-Emission & Call-Scoped Event Architecture**: - Removed duplicate event emissions across `FiretellClient._initEventStream()` and `Call._handleSseMessage()`. - Decoupled per-call transcription events from `client.events`, strictly routing all live subtitles, chunks, and summaries directly to the owning `Call` instance (`call.on(...)`). --- ## [1.1.7] - 2026-08-17 ### Added - **`FiretellClient.stopSupervision()`**: Added helper method to terminate an ongoing call supervision session (cleans up WebRTC call session and calls REST API `DELETE /v1/call-center/calls/:id/supervision`). - **Call State Monotonicity Guard**: Added automatic state ranking guard in `FiretellClient` to prevent late/out-of-order `call.created` events from overriding or regressing active call state machine. --- ## [1.1.6] - 2026-08-17 ### Added - **`SseStreamClient` (Header-Based SSE Streaming)**: Integrated lightweight, zero-dependency SSE stream client with `Authorization: Bearer ` HTTP header support via `fetch` and `ReadableStream`, completely eliminating JWT token exposure in SSE stream URLs and server access logs. - **SSE Stream Exports**: Exported `SseStreamClient`, `SseMessageEvent`, and `SseStreamConfig` from SDK entry point. --- ## [1.1.5] - 2026-08-17 ### Added - **`FiretellClient.startSupervision()`**: Added high-level helper method to initiate call supervision (`listen`, `whisper`, `barge`) and automatically establish WebRTC audio session in one call. - **`Call.joinSession()`**: Added method to connect signaling and send WebRTC offer using pre-generated `call_token` and `ws_url`. - **`ISupervisionResponse.ws_url`**: Added required `ws_url` field to `ISupervisionResponse` interface to support dedicated per-call WebSocket connections. --- ## [1.1.4] - 2026-08-15 ### Fixed - **Call Creation**: Renamed `_createCallSession` to `createCallSession`. --- ## [1.1.3] - 2026-08-14 ### Added - **Real-Time Call Lifecycle Event Enums**: Added `EClientEventName.CALL_CREATED`, `EClientEventName.CALL_STARTED`, and `EClientEventName.CALL_ANSWERED`. - **Active Call State Synchronization**: Added automatic transition of active calls to `ECallState.ACTIVE` / `ANSWERED` when receiving `call.answered` over the SSE stream. --- ## [1.1.2] - 2026-08-14 ### Added - **Client Event Enum Values (`CALL_ENDED` & `CALL_CANCELED`)**: Added `EClientEventName.CALL_ENDED` and `EClientEventName.CALL_CANCELED`. - **SSE Stream Listeners for Early Call Cancellation**: Automatically cleans up `Call` instances and dismisses incoming call ringing UI modals when calls are answered by another agent or canceled before WebSocket connection completion. --- ## [1.1.1] - 2026-08-12 ### Fixed - **Inbound Call DTLS Setup Role Validation**: Fixed `Answerer must use either active or passive value for setup attribute` error during callee-initiated unhold (`Call.unhold()`). - **Optimized Full ICE Gathering**: Reduced hard safety timeout to 6s, and implemented a 3s fallback timer verifying presence of STUN Public IP (`typ srflx`) or Relay (`typ relay`) candidates. --- ## [1.1.0] - 2026-08-12 ### Fixed - **DTLS SSL Role Preservation on Renegotiation**: Preserves established DTLS `a=setup` role (`active` / `passive`) in `setRemoteDescription()` across renegotiation answers from Media Server. - **Incoming Call Accept Error Safety**: Added `try...catch` block around `Call.accept()` to guarantee proper media cleanup and `ECallState.ERROR` state emission if microphone access or WebRTC answer creation fails. --- ## [1.0.9] - 2026-08-12 ### Fixed - **Call Unhold Event Signaling**: Fixed WS event name emitted during `Call.unhold()` from `"call.hold"` to `"call.unhold"`. - **WebRTC Remote SDP Parsing**: Added `extractSdpInit()` helper in `Call.ts` to safely parse SDP payloads received in WebSocket messages. - **ICE Server Fallback**: Configured `DEFAULT_ICE_SERVERS` (Google STUN + Cloudflare STUN) as default fallback. --- ## [1.0.0] - 2026-01-01 ### Added - Initial release of `@firetell/firetell-client-sdk`. - WebRTC audio/video calls via native per-call WebSocket signaling. - JWT-based session authentication (`session.connect`). - Outbound call support (`Call.start()`) and incoming call handling (`Call.accept()`, `Call.reject()`). - Call controls: hold/unhold, mute/unmute, DTMF, transfer. - Auto-reconnect with exponential backoff. - Agent presence notifications (`workspace.agent.state`). - Full ICE gathering (non-Trickle) for Media Server compatibility. - Multi-format builds: CJS, ESM, IIFE with TypeScript declarations.