Docs

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

MainComponent.tsx

import React, { useContext } from 'react';
import { useReactSip } from 'react-native-opensips';
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