@voctiv/agent-sdk 0.2.7 → 0.2.9

package/dist/types/sip.d.ts

@@ -0,0 +1,485 @@
+import type { Observable } from 'rxjs';
+import type { DtmfEvent, SipInfo, SipSignal } from './events';
+import type { MediaChannel } from './media-channel';
+/**
+ * SIP call state visible to script developers.
+ *
+ * ### Lifecycle
+ *
+ * ```
+ *               outbound 183 w/ SDP
+ * idle ──► ringing ─────────────────► early ──► active ──► terminated
+ *              │                        ▲          ▲
+ *              │   inbound              │          │
+ *              │   sendProgress()───────┘          │
+ *              └───────────────────────────────────┘  (early may be skipped)
+ * ```
+ *
+ * | State | Meaning | How you get here |
+ * |---|---|---|
+ * | **`idle`** | Call object exists but no SIP signalling has started yet. | Initial state. |
+ * | **`ringing`** | INVITE sent (outbound) or received (inbound); remote party is ringing. No media yet. | Automatic after INVITE. |
+ * | **`early`** | RTP pipeline is up — you can **send and receive audio**, run ASR, play TTS. This is the "pre-answer" phase before final 200 OK. | **Outbound:** remote sends 183 Session Progress with SDP. **Inbound:** your script calls `sip.sendProgress()`. |
+ * | **`active`** | 200 OK received/sent — the call is fully established. | **Outbound:** remote answers. **Inbound:** your script calls `sip.answer()`. |
+ * | **`holding`** | Local hold is active. | Your script calls `sip.hold()`. |
+ * | **`terminated`** | Call ended (BYE, CANCEL, or error). No further audio is possible. | Either side hangs up, or network error. |
+ *
+ * > **Note:** not every call goes through every state. A fast answer may jump
+ * > `ringing` → `active` without `early`. An unanswered outbound call may go
+ * > `ringing` → `terminated`. Inbound calls stay in `ringing` until you call
+ * > either `sendProgress()` (→ `early`) or `answer()` (→ `active`).
+ */
+export type SipState = 'idle' | 'ringing' | 'early' | 'active' | 'holding' | 'terminated';
+/**
+ * A SIP 1xx provisional response forwarded to the script.
+ *
+ * Provisional responses are sent by the remote party **before** the final answer (200 OK).
+ * Common examples:
+ *
+ * | Code | Phrase | Typical meaning |
+ * |------|--------|-----------------|
+ * | 100 | Trying | Request received, processing |
+ * | 180 | Ringing | Remote phone is ringing (may carry SDP → early media) |
+ * | 183 | Session Progress | Early media available — SDP is present, RTP can flow |
+ *
+ * Subscribe to `sip.progress$` to track every provisional response. The `sdp` field
+ * is present only when the response carries a Session Description (media offer/answer).
+ */
+export interface SipProgressEvent {
+    /** SIP status code (100–199). */
+    statusCode: number;
+    /** Human-readable reason phrase, e.g. `"Ringing"`, `"Session Progress"`. */
+    statusPhrase: string;
+    /** Raw SDP body when the provisional response carries a media description. */
+    sdp?: string;
+}
+/**
+ * SIP channel API — call-state observables, DTMF, INFO, hold/mute/hangup,
+ * outbound **`makeCall`**, **`bridge`**, and **pre-answer media**.
+ *
+ * ---
+ *
+ * ### Pre-answer media (early media)
+ *
+ * Both **inbound** and **outbound** calls support full-duplex audio **before** the
+ * 200 OK answer. In the `"early"` state the RTP pipeline is fully operational —
+ * ASR, TTS, `audio.say()`, `audio.play()`, and DTMF all work **exactly the same**
+ * as in the `"active"` state.
+ *
+ * #### Outbound calls
+ *
+ * The remote side may send **183 Session Progress** with SDP (e.g. an IVR greeting,
+ * ringback tone, or DTMF challenge). The call enters `"early"` automatically.
+ *
+ * #### Inbound calls
+ *
+ * Call `sip.sendProgress()` to send **183 Session Progress** to the caller. The call
+ * enters `"early"` and audio flows in both directions before final 200 OK. The API
+ * does not mark the call answered until `sip.answer()`; external billing still depends
+ * on carrier policy. Call `sip.answer()` later to complete the answer.
+ *
+ * ---
+ *
+ * ### Quick examples
+ *
+ * #### Outbound — interact with an IVR before answer
+ *
+ * ```ts
+ * const call = await channel.sip.makeCall({ sipUri: 'sip:+1234@trunk.example.com' });
+ *
+ * // Wait until RTP is ready (early media or full answer)
+ * await call.sip.waitForEarly();
+ * // Audio flows — start ASR to listen to the remote IVR
+ * const asr = await call.createAsr();
+ * asr.result$.subscribe((text) => {
+ *   if (/press 1/i.test(text)) call.sip.sendDtmf('1');
+ * });
+ *
+ * // Optionally wait for the actual answer
+ * await call.sip.waitForAnswer();
+ * await call.audio.say('Hello! We are calling about your order.');
+ * ```
+ *
+ * #### Inbound — collect data before the final answer
+ *
+ * ```ts
+ * // Inbound call arrives — state is 'ringing'
+ * channel.sip.sendProgress();
+ * // State is now 'early' — full duplex audio before final 200 OK
+ *
+ * const asr = await channel.createAsr();
+ * await channel.audio.say('Please say your account number.');
+ * const account = await firstValueFrom(asr.result$);
+ *
+ * // Now send final answer
+ * channel.sip.answer();
+ * await channel.audio.say(`Thank you! Looking up account ${account}…`);
+ * ```
+ *
+ * #### Inbound — simple answer (no pre-answer)
+ *
+ * ```ts
+ * // Inbound call — answer immediately
+ * channel.sip.answer();
+ * await channel.audio.say('Welcome!');
+ * ```
+ *
+ * #### Audio auto-buffering
+ *
+ * You do **not** need to manually wait for early/answer before calling `audio.say()`
+ * or `audio.play()`. These methods automatically defer until the RTP pipeline is ready
+ * and resolve as no-ops if the call terminates first. The explicit `waitForEarly()` /
+ * `waitForAnswer()` await points are useful when you need to **sequence logic** around
+ * call state (e.g. create ASR only after media is available).
+ */
+export interface ChannelSip {
+    /**
+     * Emits every time the remote party sends a DTMF digit (RFC 2833 in-band or SIP INFO).
+     *
+     * ```ts
+     * sip.dtmf$.subscribe(({ digit, duration }) => {
+     *   console.log(`User pressed ${digit}`);
+     * });
+     * ```
+     */
+    readonly dtmf$: Observable<DtmfEvent>;
+    /**
+     * Raw SIP INFO messages received on this call leg.
+     *
+     * Useful for carrier-specific signalling (e.g. `application/dtmf-relay`). On the
+     * current SIP runtime this stream is low-level and may not include parsed body
+     * details for every Sofia event; prefer `dtmf$` for DTMF.
+     */
+    readonly sipInfo$: Observable<SipInfo>;
+    /**
+     * Low-level SIP state-change events from the underlying Sofia stack.
+     *
+     * Prefer the higher-level **`state$`**, **`progress$`**, **`early$`**, **`answered$`**
+     * observables unless you need raw Sofia event details.
+     */
+    readonly sipSignal$: Observable<SipSignal>;
+    /**
+     * Live observable of the current call state.
+     *
+     * Emits the new {@link SipState} every time the call transitions. Starts with the
+     * state the call was in when the script began (typically `"ringing"` for inbound,
+     * `"idle"` or `"ringing"` for outbound).
+     *
+     * ```ts
+     * sip.state$.subscribe((state) => {
+     *   console.log(`Call state → ${state}`);
+     * });
+     * ```
+     *
+     * > **Tip:** for one-shot checks use the synchronous `sip.state` getter instead.
+     */
+    readonly state$: Observable<SipState>;
+    /**
+     * Current call state at the moment of access (synchronous).
+     *
+     * Returns one of: `'idle'`, `'ringing'`, `'early'`, `'active'`, `'holding'`, `'terminated'`.
+     *
+     * ```ts
+     * if (sip.state === 'active') {
+     *   await audio.say('Call is live');
+     * }
+     * ```
+     */
+    readonly state: SipState;
+    /**
+     * Whether the call has been answered — `true` after 200 OK is received (outbound)
+     * or sent (inbound via `sip.answer()`).
+     *
+     * This is a synchronous getter. For an awaitable version use {@link waitForAnswer}.
+     *
+     * ```ts
+     * if (!sip.isAnswered) {
+     *   console.log('Still waiting for answer…');
+     * }
+     * ```
+     */
+    readonly isAnswered: boolean;
+    /**
+     * Emits every SIP 1xx provisional response (180 Ringing, 183 Session Progress, etc.).
+     *
+     * Useful for tracking ringing state, ringback-tone detection, or reading SDP from
+     * early 183 responses. Each event is a {@link SipProgressEvent} with `statusCode`,
+     * `statusPhrase`, and optionally `sdp`.
+     *
+     * ```ts
+     * sip.progress$.subscribe(({ statusCode, statusPhrase }) => {
+     *   console.log(`1xx: ${statusCode} ${statusPhrase}`);
+     * });
+     * ```
+     *
+     * > **Note:** `progress$` may emit **zero** events if the remote party answers
+     * > immediately (200 OK without any provisional).
+     */
+    readonly progress$: Observable<SipProgressEvent>;
+    /**
+     * Emits **once** when early media becomes available — the RTP pipeline is up and
+     * audio can be sent/received **before** the call is formally answered.
+     *
+     * **Outbound:** fires automatically when the remote side sends a 1xx with SDP
+     * (typically **183 Session Progress**).
+     *
+     * **Inbound:** fires after your script calls `sip.sendProgress()`.
+     *
+     * After this event fires:
+     * - `audio.say()` / `audio.play()` will be heard by the remote party
+     * - `createAsr()` will receive remote audio
+     * - `dtmf$` will deliver in-band DTMF
+     * - The API has not marked the call answered yet; carrier billing policy may vary
+     *
+     * If the call is answered without any early media (outbound: no 183 with SDP;
+     * inbound: `answer()` called directly), `early$` **may not emit at all** —
+     * use `answered$` or `waitForAnswer()` instead.
+     *
+     * ```ts
+     * sip.early$.subscribe(() => {
+     *   console.log('Early media — RTP is flowing before answer');
+     * });
+     * ```
+     */
+    readonly early$: Observable<void>;
+    /**
+     * Emits **once** when the call is answered (200 OK received for outbound,
+     * or sent for inbound after `sip.answer()`).
+     *
+     * After this event the call state is `"active"` and the call is fully established.
+     *
+     * ```ts
+     * sip.answered$.subscribe(() => {
+     *   console.log('Call answered — full duplex');
+     * });
+     * ```
+     *
+     * > For the awaitable version see {@link waitForAnswer}.
+     */
+    readonly answered$: Observable<void>;
+    /**
+     * Returns a `Promise` that resolves when the call is answered (200 OK).
+     *
+     * If the call is **already answered** at the time of calling, resolves immediately.
+     *
+     * Useful in outbound scenarios where you want to block until the remote party picks up:
+     *
+     * ```ts
+     * const call = await channel.sip.makeCall({ sipUri: destination });
+     * await call.sip.waitForAnswer();
+     * await call.audio.say('You have picked up!');
+     * ```
+     *
+     * > **Warning:** if the call is never answered (busy, timeout, rejection) and your
+     * > script does not handle `events.terminated$`, this Promise will remain pending
+     * > until the call terminates (at which point the script exits).
+     */
+    waitForAnswer(): Promise<void>;
+    /**
+     * Returns a `Promise` that resolves as soon as the RTP pipeline is ready —
+     * either on **early media** or on **answer**, whichever comes first.
+     *
+     * If the call is already in `"early"` or `"active"` state, resolves immediately.
+     *
+     * #### Outbound usage
+     *
+     * The recommended await point when you want to start ASR / play audio
+     * **as early as possible** (including pre-answer):
+     *
+     * ```ts
+     * const call = await channel.sip.makeCall({ sipUri: destination });
+     *
+     * // Don't wait for full answer — start as soon as any media path exists
+     * await call.sip.waitForEarly();
+     *
+     * // ASR is already receiving remote audio (IVR prompts, ringback, etc.)
+     * const asr = await call.createAsr();
+     * asr.result$.subscribe((text) => console.log('Heard:', text));
+     *
+     * // Play DTMF to navigate an IVR — works even before 200 OK
+     * call.sip.sendDtmf('1');
+     * ```
+     *
+     * #### Inbound usage
+     *
+     * For inbound calls you must call `sendProgress()` **first** to trigger early media,
+     * then `waitForEarly()` resolves:
+     *
+     * ```ts
+     * channel.sip.sendProgress();
+     * await channel.sip.waitForEarly(); // resolves immediately after sendProgress()
+     * const asr = await channel.createAsr();
+     * ```
+     *
+     * > **Tip:** some carriers answer outbound calls without early media (180 without SDP).
+     * > In that case `waitForEarly()` resolves only when the 200 OK arrives.
+     */
+    waitForEarly(): Promise<void>;
+    /**
+     * Send a DTMF digit to the remote party.
+     *
+     * SIP channels delegate to the telephony stack. WS channels validate one digit
+     * and emit `dtmf-send` to the connected client. Headless channels ignore it.
+     *
+     * @param digit - One of `0`–`9`, `*`, `#`.
+     * @param duration - Tone duration in milliseconds (default **250 ms**).
+     *
+     * ```ts
+     * // Navigate an IVR: press 1, wait, press 3
+     * sip.sendDtmf('1');
+     * await new Promise((r) => setTimeout(r, 2000));
+     * sip.sendDtmf('3');
+     * ```
+     */
+    sendDtmf(digit: string, duration?: number): void;
+    /**
+     * Send a SIP INFO request with an arbitrary content type and body on this call leg.
+     *
+     * No-op on WS and headless channels.
+     *
+     * @param contentType - MIME type, e.g. `"application/dtmf-relay"`.
+     * @param body - Raw text body of the INFO request.
+     */
+    sendInfo(contentType: string, body: string): void;
+    /**
+     * Put the SIP call on hold (sends re-INVITE with `a=sendonly`).
+     *
+     * The remote party hears silence (or hold music if your carrier supports it).
+     * Call {@link unhold} to resume.
+     */
+    hold(): void;
+    /**
+     * Resume a held SIP call (sends re-INVITE with `a=sendrecv`).
+     */
+    unhold(): void;
+    /**
+     * Suppress outgoing SIP audio locally — the remote party hears silence,
+     * but you still receive their audio.
+     *
+     * This does **not** send a re-INVITE; it simply stops feeding PCM
+     * to the RTP encoder. Call {@link unmute} to resume.
+     */
+    mute(): void;
+    /** Resume sending audio after {@link mute}. No-op on WS/headless channels. */
+    unmute(): void;
+    /**
+     * Hang up the call.
+     *
+     * SIP channels send BYE. WS channels disconnect the socket. Headless channels destroy
+     * the synthetic channel. After this call `events.terminated$` will emit and the session ends.
+     */
+    hangup(): void;
+    /**
+     * Answer an **inbound** call (sends 200 OK with SDP).
+     *
+     * No-op if the call is already answered or if this is an outbound call. Also no-op
+     * on WS/headless channels. After answering, the state transitions to `"active"` and
+     * the API records the call answer time.
+     *
+     * If the call is in `"early"` state (after `sendProgress()`), `answer()` promotes it
+     * to `"active"` — audio was already flowing and the final answer is now sent.
+     *
+     * ```ts
+     * // Simple: answer immediately
+     * channel.sip.answer();
+     * await channel.audio.say('Hello, how can I help?');
+     * ```
+     *
+     * ```ts
+     * // Advanced: pre-answer → answer
+     * channel.sip.sendProgress();               // early media before final 200 OK
+     * await channel.audio.say('One moment…');
+     * channel.sip.answer();                      // final 200 OK
+     * await channel.audio.say('How can I help?');
+     * ```
+     */
+    answer(): void;
+    /**
+     * Send **183 Session Progress** with SDP on an **inbound** call, enabling
+     * early media (full-duplex audio) **before** the 200 OK answer.
+     *
+     * After calling `sendProgress()`:
+     * - The call state transitions to `"early"` and `early$` emits.
+     * - `audio.say()`, `audio.play()`, `createAsr()`, `dtmf$` all work —
+     *   exactly the same as in the `"active"` state.
+     * - The remote party hears your audio before the final 200 OK. The API still treats
+     *   the call as not answered; external billing depends on carrier policy.
+     *
+     * Call `sip.answer()` later to send the final 200 OK and transition to `"active"`.
+     *
+     * No-op if the call is already in `"early"` or `"active"` state, or if this is
+     * an outbound / WS / headless channel.
+     *
+     * ### Typical use case: play a greeting before answering
+     *
+     * ```ts
+     * // Inbound call arrives — state is 'ringing'
+     * channel.sip.sendProgress();
+     * // State is now 'early' — audio flows before final 200 OK
+     *
+     * await channel.audio.say('Please hold while we connect you…');
+     *
+     * // Now answer with final 200 OK
+     * channel.sip.answer();
+     * await channel.audio.say('Hello! How can I help?');
+     * ```
+     *
+     * ### Typical use case: start ASR before answering
+     *
+     * ```ts
+     * channel.sip.sendProgress();
+     * const asr = await channel.createAsr();
+     * await channel.audio.say('Hi! What is your account number?');
+     * const result = await firstValueFrom(asr.result$);
+     * // Collected account number before sending final 200 OK
+     * channel.sip.answer();
+     * ```
+     */
+    sendProgress(): void;
+    /**
+     * Initiate an **outbound** SIP call to the given SIP URI.
+     *
+     * Returns a new {@link MediaChannel} representing the B-leg. The returned channel
+     * has its own `sip`, `audio`, `events`, etc. — use `waitForAnswer()` or
+     * `waitForEarly()` on the B-leg before sending audio.
+     *
+     * Supported only by the main-thread SIP channel. Worker-isolated, WS, and headless
+     * channels throw because the current worker bridge does not proxy nested call legs.
+     *
+     * @param opts.sipUri - Full SIP URI, e.g. `"sip:+12025551234@trunk.carrier.com"`.
+     * @param opts.fromUri - Optional caller-ID override SIP URI.
+     * @returns A new `MediaChannel` for the outbound leg.
+     *
+     * ```ts
+     * const bLeg = await channel.sip.makeCall({
+     *   sipUri: 'sip:+12025551234@trunk.carrier.com',
+     * });
+     * await bLeg.sip.waitForAnswer();
+     * // Bridge both legs so callers hear each other
+     * const teardown = channel.sip.bridge(bLeg);
+     * ```
+     */
+    makeCall(opts: {
+        sipUri: string;
+        fromUri?: string;
+    }): Promise<MediaChannel>;
+    /**
+     * Cross-connect audio between this SIP call and another SIP call (conference bridge).
+     *
+     * Both parties hear each other in real time. Returns a teardown function
+     * that disconnects the bridge when called. ScriptEngine only bridges two
+     * `SipMediaChannel` instances directly; non-SIP channels return a no-op teardown
+     * or throw depending on the host.
+     *
+     * @param other - The other `MediaChannel` (e.g. an outbound B-leg).
+     * @returns A function to tear down the bridge.
+     *
+     * ```ts
+     * const teardown = channel.sip.bridge(bLeg);
+     * // ... later
+     * teardown(); // disconnect the bridge
+     * ```
+     */
+    bridge(other: MediaChannel): () => void;
+}
+//# sourceMappingURL=sip.d.ts.map

package/dist/types/sip.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"sip.d.ts","sourceRoot":"","sources":["../../src/types/sip.ts"],"names":[],"mappings":"AAAA,OAAO,KAAK,EAAE,UAAU,EAAE,MAAM,MAAM,CAAC;AACvC,OAAO,KAAK,EAAE,SAAS,EAAE,OAAO,EAAE,SAAS,EAAE,MAAM,UAAU,CAAC;AAC9D,OAAO,KAAK,EAAE,YAAY,EAAE,MAAM,iBAAiB,CAAC;AAEpD;;;;;;;;;;;;;;;;;;;;;;;;;;;GA2BG;AACH,MAAM,MAAM,QAAQ,GAChB,MAAM,GACN,SAAS,GACT,OAAO,GACP,QAAQ,GACR,SAAS,GACT,YAAY,CAAC;AAEjB;;;;;;;;;;;;;;GAcG;AACH,MAAM,WAAW,gBAAgB;IAC/B,iCAAiC;IACjC,UAAU,EAAE,MAAM,CAAC;IACnB,4EAA4E;IAC5E,YAAY,EAAE,MAAM,CAAC;IACrB,8EAA8E;IAC9E,GAAG,CAAC,EAAE,MAAM,CAAC;CACd;AAED;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GA8EG;AACH,MAAM,WAAW,UAAU;IACzB;;;;;;;;OAQG;IACH,QAAQ,CAAC,KAAK,EAAE,UAAU,CAAC,SAAS,CAAC,CAAC;IAEtC;;;;;;OAMG;IACH,QAAQ,CAAC,QAAQ,EAAE,UAAU,CAAC,OAAO,CAAC,CAAC;IAEvC;;;;;OAKG;IACH,QAAQ,CAAC,UAAU,EAAE,UAAU,CAAC,SAAS,CAAC,CAAC;IAE3C;;;;;;;;;;;;;;OAcG;IACH,QAAQ,CAAC,MAAM,EAAE,UAAU,CAAC,QAAQ,CAAC,CAAC;IAEtC;;;;;;;;;;OAUG;IACH,QAAQ,CAAC,KAAK,EAAE,QAAQ,CAAC;IAEzB;;;;;;;;;;;OAWG;IACH,QAAQ,CAAC,UAAU,EAAE,OAAO,CAAC;IAE7B;;;;;;;;;;;;;;;OAeG;IACH,QAAQ,CAAC,SAAS,EAAE,UAAU,CAAC,gBAAgB,CAAC,CAAC;IAEjD;;;;;;;;;;;;;;;;;;;;;;;;OAwBG;IACH,QAAQ,CAAC,MAAM,EAAE,UAAU,CAAC,IAAI,CAAC,CAAC;IAElC;;;;;;;;;;;;;OAaG;IACH,QAAQ,CAAC,SAAS,EAAE,UAAU,CAAC,IAAI,CAAC,CAAC;IAErC;;;;;;;;;;;;;;;;OAgBG;IACH,aAAa,IAAI,OAAO,CAAC,IAAI,CAAC,CAAC;IAE/B;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;OAsCG;IACH,YAAY,IAAI,OAAO,CAAC,IAAI,CAAC,CAAC;IAE9B;;;;;;;;;;;;;;;OAeG;IACH,QAAQ,CAAC,KAAK,EAAE,MAAM,EAAE,QAAQ,CAAC,EAAE,MAAM,GAAG,IAAI,CAAC;IAEjD;;;;;;;OAOG;IACH,QAAQ,CAAC,WAAW,EAAE,MAAM,EAAE,IAAI,EAAE,MAAM,GAAG,IAAI,CAAC;IAElD;;;;;OAKG;IACH,IAAI,IAAI,IAAI,CAAC;IAEb;;OAEG;IACH,MAAM,IAAI,IAAI,CAAC;IAEf;;;;;;OAMG;IACH,IAAI,IAAI,IAAI,CAAC;IAEb,8EAA8E;IAC9E,MAAM,IAAI,IAAI,CAAC;IAEf;;;;;OAKG;IACH,MAAM,IAAI,IAAI,CAAC;IAEf;;;;;;;;;;;;;;;;;;;;;;;OAuBG;IACH,MAAM,IAAI,IAAI,CAAC;IAEf;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;OAwCG;IACH,YAAY,IAAI,IAAI,CAAC;IAErB;;;;;;;;;;;;;;;;;;;;;;OAsBG;IACH,QAAQ,CAAC,IAAI,EAAE;QAAE,MAAM,EAAE,MAAM,CAAC;QAAC,OAAO,CAAC,EAAE,MAAM,CAAA;KAAE,GAAG,OAAO,CAAC,YAAY,CAAC,CAAC;IAE5E;;;;;;;;;;;;;;;;OAgBG;IACH,MAAM,CAAC,KAAK,EAAE,YAAY,GAAG,MAAM,IAAI,CAAC;CACzC"}

package/dist/types/sip.js

@@ -0,0 +1,3 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+//# sourceMappingURL=sip.js.map

package/dist/types/sip.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"sip.js","sourceRoot":"","sources":["../../src/types/sip.ts"],"names":[],"mappings":""}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@voctiv/agent-sdk",
-  "version": "0.2.7",
+  "version": "0.2.9",
   "description": "Voctiv TypeScript agent SDK: defineScript and platform types for the voice/dialog scripting runtime.",
   "license": "UNLICENSED",
   "author": "",