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Description

This library is a wrapper over the opensips-js implementation. It provides a React Native composable for reactive work with opensips-js functionality. Wrap your top level component in ReactSipProvider, call redeclareGlobals to declare global variables for webrtc and then use useReactSip hook in the child components to use reactive opensips-js functionality.

Andoid Configuration

Add this permissions to your AndroidManifest.xml file

AndroidMainfest.xml

    <uses-permission android:name="android.permission.INTERNET" />
    <uses-feature android:name="android.hardware.camera" />
    <uses-feature android:name="android.hardware.camera.autofocus" />
    <uses-feature android:name="android.hardware.audio.output" />
    <uses-feature android:name="android.hardware.microphone" />

    <uses-permission android:name="android.permission.CAMERA" />
    <uses-permission android:name="android.permission.RECORD_AUDIO" />
    <uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
    <uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />
    <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />

Basic Usage

install react-native-webrt library

yarn add react-native-webrtc

App.tsx

import {
  ReactSipProvider,
  redeclareGlobals,
} from "opensips-js-react-native";
redeclareGlobals();
export const App = (): JSX.Element => {
  return (
    <ReactSipProvider>
      <MainComponent />
    </ReactSipProvider>
  );
};

MainComponent.tsx

import React, { useContext } from 'react';
import { useReactSip } from 'opensips-js-react-native';
export const MainComponent = () => {
  const {actions, state} = useReactSip()
  const {
    selectedInputDevice,
    selectedOutputDevice,
    muteWhenJoin,
    isDND,
    addCallToCurrentRoom,
    microphoneInputLevel,
    speakerVolume,
    isMuted,
    callAddingInProgress,
    activeCalls,
    callsInActiveRoom,
    currentActiveRoomId,
    activeRooms
  } = state

  const {
    doCall,
    sendDTMF,
    muteCaller,
    callTerminate,
    callTransfer,
    callMerge,
    doCallHold,
    callAnswer,
    callMove,
    doMute
  } = actions
  //other code
  return (
    //render your coponent
  )
}

Composable

State

| Name | Description | Type | Default | | --------------------- | :--------------------------------------------------------- | :------------------------------------- | :-------- | | isInitialized | Defines if opensips-js is initialized | boolean | false | | activeCalls | Active calls | { [key: string]: ICall } | {} | | callsInActiveRoom | Calls in active room | Array<Call> | [] | | activeMessages | Active MSRP messages | { [key: string]: IMessage } | {} | | addCallToCurrentRoom | Defines if new call should be added to current room | boolean | false | | callAddingInProgress | Represents call id of progressing call if such call exists | string / undefined | undefined | | activeRooms | Active rooms | { [key: number]: IRoom } | {} | | msrpHistory | MSRP messages history | { [key: string]: Array<MSRPMessage> } | {} | | availableMediaDevices | List of available media devices | Array<MediaDeviceInfo> | [] | | inputMediaDeviceList | List of available input media devices | Array<MediaDeviceInfo> | [] | | outputMediaDeviceList | List of available output media devices | Array<MediaDeviceInfo> | [] | | selectedOutputDevice | ID of selected output device | string | 'default' | | selectedInputDevice | ID of selected input device | string | 'default' | | muteWhenJoin | Defines if agent will be muted when joining call | boolean | false | | isDND | Defines usage agent's 'Do Not Disturb' option | boolean | false | | originalStream | Agent's Audio Stream object | MediaStream / null | null | | currentActiveRoomId | Defines agent's active room | number / undefined | undefined | | callStatus | Calls' statuses | { [key: string]: ICallStatus } | {} | | callTime | Calls' times | { [key: string]: ITimeData } | {} | | callMetrics | Calls' metrics | { [key: string]: unknown } | {} | | autoAnswer | Defines if auto-answer is enabled | boolean | false | | microphoneInputLevel | Microphone sensitivity (range is from 0 to 2) | number | 2 | | speakerVolume | Speaker volume (range is from 0 to 1) | number | 1 |

Methods

| Name | Interface | Description | |--------------------------|:-----------------------------------------------------------------------------|:-----------------------------------------------------------------| | init | (domain: string, username: string, password: string): void | Initialize opensips-js | | initCall | (target: string, addToCurrentRoom: boolean, holdOtherCalls: boolean) => void | Make a call | | answerCall | (callId: string) => void | Answer call | | terminateCall | (callId: string) => void | Hangup call | | muteCaller | (callId: string) => void | Mute caller | | unmuteCaller | (callId: string) => void | Unmute caller | | mute | () => void | Mute ourself | | unmute | () => void | Unmute ourself | | transferCall | (callId: string, target: string) => void | Transfer call | | mergeCall | (roomId: number) => void | Merge calls in room (works only when 2 call in room) | | mergeCallByIds | (firstCallId: string, secondCallId: string) => void | Merge 2 calls by their ids | | holdCall | (callId: string, automatic?: boolean) => void | Hold a call | | unholdCall | (callId: string) => void | Unhold a call | | moveCall | (callId: string, roomId: number) => Promise<void> | Move call to another room | | setMicrophone | (deviceId: string) => Promise<void> | Set microphone which to use | | setSpeaker | (deviceId: string) => Promise<void> | Set speaker which to use | | sendDTMF | (callId: string, value: string) => void | Send DTMF | | setActiveRoom | (roomId: number / undefined) => Promise<void> | Set current active room | | setMicrophoneSensitivity | (value: number) => void | Set microphone sensitivity. Value should be in range from 0 to 1 | | setSpeakerVolume | (value: number) => void | Set speaker volume. Value should be in range from 0 to 1 | | setAutoAnswer | (value: boolean) => void | Set auto-answer | | setDND | (state: boolean) => void | Set 'Do not Disturb' option | | setMetricsConfig | (config: WebrtcMetricsConfigType) => void | Set the metric config (used for audio quality indicator) | | msrpAnswer | (callId: string) => void | Answer MSRP invite | | messageTerminate | (callId: string) => void | Terminate MSRP session | | sendMSRP | (msrpSessionId: string, body: string) => void | Send MSRP message | | initMSRP | (target: string, body: string, options: object) => void | Send MSRP invite |

WebrtcMetricsConfigType

| Parameter | Type | Default | |----------------|-----------|-------------| | refreshEvery | number | undefined | | startAfter | number | undefined | | startAfter | number | undefined | | verbose | boolean | undefined | | pname | string | undefined | | cid | string | undefined | | uid | string | undefined | | record | boolean | undefined | | ticket | boolean | undefined |