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1.2.3 • Public • Published


Please note that this is just a version of react-sip with pull requests #27 and #28 merged. We are primarily hosting it on NPM for our own use, but feel free to include this in your project if you need any of those features.

All changes are made by evercall for our own use, and we do not provide any kind of support for react-sip.

React SIP

React wrapper for jssip.


npm install @evercall/react-sip

There is no need to install jssip as it is a dependency of react-sip.


import { SipProvider } from '@evercall/react-sip';
import App from './components/App';

    pathname="/ws" // Path in socket URI (e.g. wss://sip.example.com:7443/ws); "" by default
    secure={true} // if true, the connection will be made over `wss://` else it will default to `ws://`
    password={sipPassword} // usually required (e.g. from ENV or props)
    autoRegister={true} // true by default, see jssip.UA option register
    autoAnswer={false} // automatically answer incoming calls; false by default
    iceRestart={false} // force ICE session to restart on every WebRTC call; false by default
    sessionTimersExpires={120} // value for Session-Expires header; 120 by default
    extraHeaders={{ // optional sip headers to send
      register: ['X-Foo: foo', 'X-Bar: bar'],
      invite: ['X-Foo: foo2', 'X-Bar: bar2']
    iceServers={[ // optional
      { urls: ['stun:a.example.com', 'stun:b.example.com'] },
      { urls: 'turn:example.com', username: 'foo', credential: '1234' }
    debug={false} // whether to output events to console; false by default
    incomingAudioDeviceId={"default"} // default, or a deviceId obtained from navigator.mediaDevices.enumerateDevices()
    outboundAudioDeviceId={"default"} // default, or a deviceId obtained from navigator.mediaDevices.enumerateDevices()
    dtmfTransportType={"RFC4733" | "INFO" | "RFC2733"} // DTMF tone transport method
    <App />

Child components get access to the context by implementing this:

function Child(props, SipProvider) {

  return (

Child.contextTypes = SipProvider.childContextTypes;

See lib/types.ts for technical details of what sipType and callType are. An overview is given below:


sip.status represents SIP connection status and equals to one of these values:

  • 'sipStatus/DISCONNECTED' when host, port or user is not defined
  • 'sipStatus/CONNECTING'
  • 'sipStatus/CONNECTED'
  • 'sipStatus/REGISTERED' after calling registerSip or after 'sipStatus/CONNECTED' when autoRegister is true
  • 'sipStatus/ERROR' in case of configuration, connection or registration problems


  • null when sip.status is not 'sipStatus/ERROR'
  • 'sipErrorType/CONFIGURATION'
  • 'sipErrorType/CONNECTION'
  • 'sipErrorType/REGISTRATION'

sip.host, sip.port, sip.user, ...<SipProvider />’s props (to make them easy to be displayed in the UI).


call.id is a unique session id of the actual established voice call; undefined between calls

call.status represents the status of the call:

  • 'callStatus/IDLE' between calls (even when disconnected)
  • 'callStatus/STARTING' active incoming or outgoing call request
  • 'callStatus/ACTIVE' during ongoing call
  • 'callStatus/STOPPING' during call cancelation request

call.direction indicates the direction of the ongoing call:

  • null between calls
  • 'callDirection/INCOMING'
  • 'callDirection/OUTGOING'

call.counterpart represents the call destination in case of outgoing call and caller for incoming calls. The format depends on the configuration of the SIP server (e.g. "bob" <+441234567890@sip.example.com>, +441234567890@sip.example.com or bob@sip.example.com).


When autoRegister is set to false, you can call sipRegister() and sipUnregister() manually for advanced registration scenarios.

To make calls, simply use these functions:

  • answerCall()
  • startCall(destination)
  • stopCall()

The value for destination argument equals to the target SIP user without the host part (e.g. +441234567890 or bob). The omitted host part is equal to host you’ve defined in SipProvider props (e.g. sip.example.com).

During a call you can put it on hold using the call.hold() and call.unhold() functions. You can also get hold status with the call.isOnHold property.

You may also mute your microphone during calls with the call.toggleMuteMicrophone(), call.muteMicrophone and call.unmuteMicrophone methods. You can check whether the microphone is used with the call.microphoneIsMuted property.

To send DTMF tones while in-call, you can use this function:


You can pass as many tones as you want in a string (e.g. sendDTMF("1234")). You may also specify duration and interToneGap in milliseconds, as sendDTMF("1234", 100, 70). See the MDN docs for RTCDTMFSender.insertDTMF() for further details.

The DTMF implementation is not SIP INFO, but RFC-4733.

The values for sip.status, sip.errorType, call.status and call.direction can be imported as constants to make typos easier to detect:

import {
} from "react-sip";

Custom PropTypes types are also provided by the library:

import { callType, extraHeadersType, iceServersType, sipType } from "react-sip";



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