Package Exports
- react-jssip-wrapper
 - react-jssip-wrapper/dist/cjs/index.js
 - react-jssip-wrapper/dist/esm/index.js
 
This package does not declare an exports field, so the exports above have been automatically detected and optimized by JSPM instead. If any package subpath is missing, it is recommended to post an issue to the original package (react-jssip-wrapper) to support the "exports" field. If that is not possible, create a JSPM override to customize the exports field for this package.
Readme
React JsSIP Wrapper
React wrapper for jssip. For discussion TelegramGroup
Installation
npm install react-jssip-wrapperThere is no need to install jssip as it is a dependency of react-jssip-wrapper.
Usage
import React, { useCallback, useRef } from "react";
import { SipProvider } from "react-jssip-wrapper";
import { IStore, setSip } from "store";
import { useDispatch, useSelector } from "react-redux";
const Sip = () => {
  const dispatch = useDispatch();
  const ref = useRef<any>();
  const connectionConfig = useSelector(
    (state: IStore) => state.sip.connectionConfig
  );
  const onRefChange = useCallback((node: any) => {
    if (node === null) {
      // DOM node referenced by ref has been unmounted
    } else {
      dispatch(setSip({ ref: node }));
      ref.current = node;
    }
  }, []);
  if (!connectionConfig) {
    return null;
  }
  // const call = () =>
  //   ref.current?.startCall(`sip:${phone}@${connectionConfig.server}`);
  //
  // const transfer = () => {
  //   ref.current?.state?.rtcSession?.refer(
  //     `sip:${transferPhone}@${connectionConfig.server}`
  //   );
  // };
  return (
      <SipProvider
        host={connectionConfig.server as string}
        port={7443}
        pathname="" // Path in socket URI (e.g. wss://sip.example.com:7443/ws); "" by default
        user={connectionConfig.user as string}
        password={connectionConfig.password as string} // usually required (e.g. from ENV or props)
        autoRegister={false} // true by default, see jssip.UA option register
        // autoAnswer={true} // automatically answer incoming calls; false by default
        iceRestart={true} // force ICE session to restart on every WebRTC call; false by default
        sessionTimersExpires={30000}
        debug={false} // wh
        ref={onRefChange}
        iceServers={[
          {
            urls: [
              "stun:stun.l.google.com:19302",
              "stun:stun1.l.google.com:19302",
            ],
          },
        ]}
        setAction={(data: any) => {
          dispatch(setSip({ ...data, ref: ref.current }));
        }}
        audioId="newAudioId" // default 'sip-provider-audio' for output audio
      />
  );
};
export default Sip;Child components get access to this context:
{
  sip: sipType,
  call: callType,
  registerSip: PropTypes.func,
  unregisterSip: PropTypes.func,
  answerCall: PropTypes.func,
  startCall: PropTypes.func,
  stopCall: PropTypes.func,
}See lib/types.ts for technical details of what sipType and callType are.
An overview is given below:
sip
sip.status represents SIP connection status and equals to one of these values:
'sipStatus/DISCONNECTED'whenhost,portoruseris not defined'sipStatus/CONNECTING''sipStatus/CONNECTED''sipStatus/REGISTERED'after callingregisterSipor after'sipStatus/CONNECTED'whenautoRegisteris true'sipStatus/ERROR'in case of configuration, connection or registration problems
sip.errorType:
nullwhensip.statusis 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
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:
nullbetween 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" <+998945667725@sip.example.com>, +998945667725@sip.example.com or Jasurbek@sip.example.com).
methods
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. +998945667725 or bob).
The omitted host part is equal to host you’ve defined in SipProvider props (e.g. sip.example.com).
The values for sip.status, sip.errorType, call.status and call.direction can be imported as constants to make typos easier to detect:
import {
  SIP_STATUS_DISCONNECTED,
  //SIP_STATUS_...,
  CALL_STATUS_IDLE,
  //CALL_STATUS_...,
  SIP_ERROR_TYPE_CONFIGURATION,
  //SIP_ERROR_TYPE_...,
  CALL_DIRECTION_INCOMING,
  CALL_DIRECTION_OUTGOING,
} from "react-jssip-wrapper";Custom PropTypes types are also provided by the library:
import { callType, extraHeadersType, iceServersType, sipType } from "react-jssip-wrapper";