sip.js

import { InvitationAcceptOptions } from "../../../api/invitation-accept-options.js"; import { InviterInviteOptions } from "../../../api/inviter-invite-options.js"; import { InviterOptions } from "../../../api/inviter-options.js"; import { RegistererRegisterOptions } from "../../../api/registerer-register-options.js"; import { RegistererUnregisterOptions } from "../../../api/registerer-unregister-options.js"; import { SimpleUserDelegate } from "./simple-user-delegate.js"; import { SimpleUserOptions } from "./simple-user-options.js"; /** * A simple SIP user class. * @remarks * While this class is completely functional for simple use cases, it is not intended * to provide an interface which is suitable for most (must less all) applications. * While this class has many limitations (for example, it only handles a single concurrent session), * it is, however, intended to serve as a simple example of using the SIP.js API. * @public */ export declare class SimpleUser { /** Delegate. */ delegate: SimpleUserDelegate | undefined; private logger; private options; private session; private sessionManager; /** * Constructs a new instance of the `SimpleUser` class. * @param server - SIP WebSocket Server URL. * @param options - Options bucket. See {@link SimpleUserOptions} for details. */ constructor(server: string, options?: SimpleUserOptions); /** * Instance identifier. * @internal */ get id(): string; /** The local media stream. Undefined if call not answered. */ get localMediaStream(): MediaStream | undefined; /** The remote media stream. Undefined if call not answered. */ get remoteMediaStream(): MediaStream | undefined; /** * The local audio track, if available. * @deprecated Use localMediaStream and get track from the stream. */ get localAudioTrack(): MediaStreamTrack | undefined; /** * The local video track, if available. * @deprecated Use localMediaStream and get track from the stream. */ get localVideoTrack(): MediaStreamTrack | undefined; /** * The remote audio track, if available. * @deprecated Use remoteMediaStream and get track from the stream. */ get remoteAudioTrack(): MediaStreamTrack | undefined; /** * The remote video track, if available. * @deprecated Use remoteMediaStream and get track from the stream. */ get remoteVideoTrack(): MediaStreamTrack | undefined; /** * Connect. * @remarks * Start the UserAgent's WebSocket Transport. */ connect(): Promise<void>; /** * Disconnect. * @remarks * Stop the UserAgent's WebSocket Transport. */ disconnect(): Promise<void>; /** * Return true if connected. */ isConnected(): boolean; /** * Start receiving incoming calls. * @remarks * Send a REGISTER request for the UserAgent's AOR. * Resolves when the REGISTER request is sent, otherwise rejects. */ register(registererRegisterOptions?: RegistererRegisterOptions): Promise<void>; /** * Stop receiving incoming calls. * @remarks * Send an un-REGISTER request for the UserAgent's AOR. * Resolves when the un-REGISTER request is sent, otherwise rejects. */ unregister(registererUnregisterOptions?: RegistererUnregisterOptions): Promise<void>; /** * Make an outgoing call. * @remarks * Send an INVITE request to create a new Session. * Resolves when the INVITE request is sent, otherwise rejects. * Use `onCallAnswered` delegate method to determine if Session is established. * @param destination - The target destination to call. A SIP address to send the INVITE to. * @param inviterOptions - Optional options for Inviter constructor. * @param inviterInviteOptions - Optional options for Inviter.invite(). */ call(destination: string, inviterOptions?: InviterOptions, inviterInviteOptions?: InviterInviteOptions): Promise<void>; /** * Hangup a call. * @remarks * Send a BYE request, CANCEL request or reject response to end the current Session. * Resolves when the request/response is sent, otherwise rejects. * Use `onCallHangup` delegate method to determine if and when call is ended. */ hangup(): Promise<void>; /** * Answer an incoming call. * @remarks * Accept an incoming INVITE request creating a new Session. * Resolves with the response is sent, otherwise rejects. * Use `onCallAnswered` delegate method to determine if and when call is established. * @param invitationAcceptOptions - Optional options for Inviter.accept(). */ answer(invitationAcceptOptions?: InvitationAcceptOptions): Promise<void>; /** * Decline an incoming call. * @remarks * Reject an incoming INVITE request. * Resolves with the response is sent, otherwise rejects. * Use `onCallHangup` delegate method to determine if and when call is ended. */ decline(): Promise<void>; /** * Hold call * @remarks * Send a re-INVITE with new offer indicating "hold". * Resolves when the re-INVITE request is sent, otherwise rejects. * Use `onCallHold` delegate method to determine if request is accepted or rejected. * See: https://tools.ietf.org/html/rfc6337 */ hold(): Promise<void>; /** * Unhold call. * @remarks * Send a re-INVITE with new offer indicating "unhold". * Resolves when the re-INVITE request is sent, otherwise rejects. * Use `onCallHold` delegate method to determine if request is accepted or rejected. * See: https://tools.ietf.org/html/rfc6337 */ unhold(): Promise<void>; /** * Hold state. * @remarks * True if session is on hold. */ isHeld(): boolean; /** * Mute call. * @remarks * Disable sender's media tracks. */ mute(): void; /** * Unmute call. * @remarks * Enable sender's media tracks. */ unmute(): void; /** * Mute state. * @remarks * True if sender's media track is disabled. */ isMuted(): boolean; /** * Send DTMF. * @remarks * Send an INFO request with content type application/dtmf-relay. * @param tone - Tone to send. */ sendDTMF(tone: string): Promise<void>; /** * Send a message. * @remarks * Send a MESSAGE request. * @param destination - The target destination for the message. A SIP address to send the MESSAGE to. */ message(destination: string, message: string): Promise<void>; }