WebRTCビデオセッションにSIPコールを追加する

const dialOut = () => {
  fetch(`/dial-out?roomId=${roomId}`)
    .then(response => response.json())
    .then((sipData) => {
      connectionId = sipData.connectionId;
    }).catch((error) => {
      alert(`There was an error dialing-out`);
    })
};
const hangUp = () => {
  fetch(`/hang-up?roomId=${roomId}`)
    .then(response => response)
    .then((data) => {
      console.log('dial-out-hang-up-complete');
    }).catch((error) => {
      alert(`There was an error hanging up`);
    })
  };

<button onclick="dialOut()">Click here to dial-out to the Vonage Conference</button>
<button onclick="hangUp()">Click here to hang-up</button>

/**
 * Generates a random 4 digit PIN
 */
const generatePin = () => {
  const pin = Math.floor(Math.random() * 9000) + 1000;
  if (app.get(pin)) {
    return generatePin();
  }
  return pin;
};

/**
 * Creates a Video API user token
 * @param {String} sessionId Id of the Video API session the user wishes to join
 * @param {String} sipTokenData Data associated with the SIP connection
*/
const generateToken = (sessionId, sipTokenData = '') => OT.generateToken(sessionId, {
  role: 'publisher',
  data: sipTokenData,
});

/**
 * Properties for the OT.dial API call
 * @returns {Object}
*/
const setSipOptions = () => ({
  auth: {
    username: config.voiceApiKey,
    password: config.voiceApiSecret,
  },
  secure: false
});

/**
 * When the dial-out get request is made, the dial method of the 
 * OpenTok Dial API is invoked
*/
app.get('/dial-out', (req, res) => {
  const { roomId } = req.query;
  const { conferenceNumber } = config;
  const sipTokenData = `{"sip":true, "role":"client", "name":"'${conferenceNumber}'"}`;
  const sessionId = app.get(roomId);
  const token = generateToken(sessionId, sipTokenData);
  const options = setSipOptions();
  const sipUri = `sip:${conferenceNumber}@sip.nexmo.com;transport=tls`;
  OT.dial(sessionId, token, sipUri, options, (error, sipCall) => {
    if (error) {
      console.dir(error)
      res.status(500).send('There was an error dialing out');
    } else {
      app.set(conferenceNumber + roomId, sipCall.connectionId);
      res.json(sipCall);
    }
  });
});

/**
 * When the hang-up get request is made, the forceDisconnect method 
 * of the OpenTok API is invoked
*/
app.get('/hang-up', (req, res) => {
  const { roomId } = req.query;
  const { conferenceNumber } = config;
  if (app.get(roomId) + app.get(conferenceNumber + roomId)) {
    const sessionId = app.get(roomId);
    const connectionId = app.get(conferenceNumber + roomId);
    OT.forceDisconnect(sessionId, connectionId, (error) => {
      if (error) {
        res.status(500).send('There was an error hanging up');
      } else {
        res.status(200).send('Ok');
      }
    });
  } else {
    res.status(400).send('There was an error hanging up');
  }
});

app.get('/voice-events', (req, res) => {
  res.status(200).send();
});

app.post('/voice-answer', (req, res) => {
  const { serverUrl } = config;
  const ncco = [];
  if (req.body['SipHeader_X-OpenTok-SessionId']) {
    ncco.push({
      action: 'conversation',
      name: req.body['SipHeader_X-OpenTok-SessionId'],
    });
  } else {
    ncco.push(
      {
        action: 'talk',
        text: 'Please enter a pin code to join the session'
      },
      {
        action: 'input',
        eventUrl: [`${serverUrl}/voice-dtmf`]
      }
    )
  }

  res.json(ncco);
});

app.post('/voice-dtmf', (req, res) => {
  const { dtmf } = req.body;
  let sessionId;

  if (app.get(dtmf)) {
    sessionId = app.get(dtmf);
  }

  const ncco = [
    {
      action: 'conversation',
      name: sessionId,
    }];

  res.json(ncco)
})

videoApiKey=
videoApiSecret=
voiceApiKey=
voiceApiSecret=
conferenceNumber=
serverUrl=