initial commit

.gitignore

@@ -7,6 +7,7 @@ yarn-error.log*
 lerna-debug.log*
 .pnpm-debug.log*
 
+.DS_Store
 # Diagnostic reports (https://nodejs.org/api/report.html)
 report.[0-9]*.[0-9]*.[0-9]*.[0-9]*.json
 

.vscode/settings.json

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+{
+  "deno.enable": true,
+  "deno.lint": true,
+  "deno.unstable": true
+}

README.md

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-# call-control
+# TypeScript SDK for Call Control
+
+This is a Deno TypeScript SDK for Call Control, providing an easy way to
+interact with the Call Control API. The SDK includes methods to manage calls,
+perform actions like dialing, answering, and hanging up, as well as working with
+DTMF tones, real-time transcription, and media streaming.
+
+## Table of Contents
+
+- [Getting Started](#getting-started)
+- [Usage](#usage)
+- [Available Methods](#available-methods)
+
+## Getting Started
+
+To use this SDK, you'll need to import the `CallControl` class into your project
+and instantiate it with your `baseUrl` and `bearerToken`.
+
+```typescript
+import { CallControl, DialResponse } from "call-control-sdk";
+
+const cc = new CallControl({
+  baseUrl: "https://your-api-url.com",
+  bearerToken : "your-bearer-token"
+});
+```
+
+## Usage
+
+After initializing the `CallControl` class, you can call its methods to interact
+with the Call Control API. For example, to dial a number:
+
+```typescript
+try {
+  const call = await cc.dial({
+    to: "+10409992842",
+    from: "+12994882747",
+    connection_id: "2299939939jfjj93",
+  });
+} catch (err) {
+  console.log(err.message);
+}
+```
+
+## Available Methods
+
+The following methods are available in the CallControl class:
+
+### 1. dial(dialRequest: DialRequest)
+
+Dials a number or SIP URI from a given connection. Returns a `DialResponse` with
+a `call_leg_id` that can be used to correlate the command with subsequent
+webhooks.
+
+### 2. answer(callControlId: string, answerRequest?: AnswerRequest)
+
+Answers an incoming call. The Answer command must be issued before executing
+subsequent commands on an incoming call. Returns a `CommandResponse`.
+
+### 3. bridge(callControlId: string, bridgeRequest: BridgeRequest)
+
+Bridges two call control calls. Returns a `CommandResponse`.
+
+### 4. enqueue(callControlId: string, enqueueRequest: EnqueueRequest)
+
+Puts the call in a queue. Returns a `CommandResponse`.
+
+### 5. forkingStart(callControlId: string, forkingStartRequest: ForkingStartRequest)
+
+Starts forking the media from a call to a specific target in realtime. This
+stream can be used to enable real-time audio analysis for a variety of use
+cases, such as fraud detection or AI-generated audio responses. Returns a
+`CommandResponse`.
+
+### 6. forkingStop(callControlId: string, forkingStopRequest?: ForkingStopRequest)
+
+Stops forking a call. Returns a `CommandResponse`.
+
+### 7. gather(callControlId: string, gatherRequest?: GatherRequest)
+
+Gathers DTMF signals to build interactive menus. The Answer command must be
+issued before the `gather` command. Returns a `CommandResponse`.
+
+### 8. gatherStop(callControlId: string, gatherStopRequest?: GatherStopRequest)
+
+Stops the current gather. Returns a `CommandResponse`.
+
+### 9. gatherUsingAudio(callControlId: string, gatherUsingAudioRequest?: GatherUsingAudioRequest)
+
+Plays an audio file on the call until the required DTMF signals are gathered to
+build interactive menus. The Answer command must be issued before the
+`gatherUsingAudio` command. Returns a `CommandResponse`.
+
+### 10. gatherUsingSpeak(callControlId: string, gatherUsingSpeakRequest: GatherUsingSpeakRequest)
+
+Converts text to speech and plays it back on the call to gather DTMF signals.
+The Answer command must be issued before the `gatherUsingSpeak` command. Returns
+a `CommandResponse`.
+
+### 11. hangup(callControlId: string, hangupRequest?: HangupRequest)
+
+Hangs up the call. Returns a `CommandResponse`.
+
+### 12. removeFromQueue(callControlId: string, removeFromQueueRequest?: RemoveFromQueueRequest)
+
+Removes the call from a queue. Returns a `CommandResponse`.
+
+### 13. playAudio(callControlId: string, playAudioRequest: PlayAudioRequest)
+
+Plays an audio file on the call. If multiple play audio commands are issued
+consecutively, the audio files will be placed in a queue awaiting playback.
+Returns a `CommandResponse`.
+
+### 14. stopAudio(callControlId: string, stopAudioRequest?: StopAudioRequest)
+
+Stops audio being played on the call. Returns a `CommandResponse`.
+
+### 15. pauseRecording(callControlId: string, pauseRecordingRequest?: PauseRecordingRequest)
+
+Pauses recording the call. Recording can be resumed via the Resume recording
+command. Returns a `CommandResponse`.
+
+### 16. resumeRecording(callControlId: string, resumeRecordingRequest?: ResumeRecordingRequest)
+
+Resumes recording the call. Returns a `CommandResponse`.
+
+### 17. startRecording(callControlId: string, startRecordingRequest: StartRecordingRequest)
+
+Starts recording the call. Recording will stop on call hang-up or can be
+initiated via the Stop Recording command. Returns a `CommandResponse`.
+
+### 18. stopRecording(callControlId: string, stopRecordingRequest?: StopRecordingRequest)
+
+Stops recording the call. Returns a `CommandResponse`.
+
+### 19. refer(callControlId: string, sipReferRequest: ReferRequest)
+
+Initiates a SIP Refer on a Call Control call. Returns a `CommandResponse`.
+
+### 20. reject(callControlId: string, rejectRequest: RejectRequest)
+
+Rejects an incoming call. Returns a `CommandResponse`.
+
+### 21. sendDtmf(callControlId: string, sendDTMFRequest: SendDtmfRequest)
+
+Sends DTMF tones from this leg. DTMF tones will be heard by the other end of the
+call. Returns a `CommandResponse`.
+
+### 22. speakText(callControlId: string, speakTextRequest: SpeakTextRequest)
+
+Converts text to speech and plays it back on the call. If multiple speak text
+commands are issued consecutively, the audio files will be placed in a queue
+awaiting playback. Returns a `CommandResponse`.
+
+### 23. startStream(callControlId: string, startStreamRequest?: StartStreamRequest)
+
+Starts streaming the media from a call to a specific WebSocket address in
+near-realtime. Audio will be delivered as base64-encoded RTP payload (raw
+audio), wrapped in JSON payloads. Returns a `CommandResponse`.
+
+### 24. stopStream(callControlId: string, stopStreamRequest?: StopStreamRequest)
+
+Stops streaming a call to a WebSocket. Returns a `CommandResponse`.
+
+### 25. startTranscription(callControlId: string, startTranscriptionRequest?: StartTranscriptionRequest)
+
+Starts real-time transcription. Transcription will stop on call hang-up or can
+be initiated via the Transcription stop command. Returns a `CommandResponse`.
+
+### 26. stopTranscription(callControlId: string, stopTranscriptionRequest?: StopTranscriptionRequest)
+
+Stops real-time transcription. Returns a `CommandResponse`.
+
+### 27. transfer(callControlId: string, transferRequest: TransferRequest)
+
+Transfers a call to a new destination. If the transfer is unsuccessful, a
+`call.hangup` webhook for the other call (Leg B) will be sent indicating that
+the transfer could not be completed. The original call will remain active and
+may be issued additional commands, potentially transferring the call to an
+alternate destination. Returns a `CommandResponse`.
+
+### 28. updateState(callControlId: string, updateStateRequest: UpdateStateRequest)
+
+Updates client state. Returns a `CommandResponse`.
+
+For each method, refer to the SDK's `types.ts` for more details on the required
+parameters and expected responses.