Help to send response 401
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WebRTC
Jul 1, 2022, 12:16:08 PM7/1/22
to JsSIP
Hello.
When making the connection, the server responds with a 401, I have tried to make a second register but I have not been able to do it. Could you please give me an example of how to take the WWW-Authenticate that they send me in the 401 and how to respond with a second register?
Thank you.
My code is:
When making the connection, the server responds with a 401, I have tried to make a second register but I have not been able to do it. Could you please give me an example of how to take the WWW-Authenticate that they send me in the 401 and how to respond with a second register?
Thank you.
My code is:
callRTC () {
var options = {
mediaConstraints: { audio: true, video: false },
eventHandlers: eventHandlers,
pcConfig: {
}
}
try {
var socket = new jsSip.WebSocketInterface(
'wss://********/webrtc/'
)
var config = {
sockets: [socket],
uri: 'sip:Test@******',
authorizationUser: 'Test',
password: '********',
noAnswerTimeout: 120
}
ua = new jsSip.UA(config)
} catch (error) {
console.log(error)
}
ua.on('connecting', () => {
console.log('UA "connecting" event')
})
ua.on('connected', () => {
console.log('call is connected');
})
ua.on('disconnected', () => {
console.log('UA "disconnected" event')
})
ua.on('registered', data => {
console.log('UA "registered" event', data)
})
ua.on('unregistered', () => {
console.log('UA "unregistered" event')
})
ua.on('registrationFailed', data => {
console.log('UA "registrationFailed" event', data)
})
/* ua.on(
'newRTCSession',
({
originator,
session: rtcSession,
request: rtcRequest,
response: rtcResponse
}) => {
console.log('UA "newRTCSession" event')
// identify call direction
if (originator === 'local') {
const foundUri = rtcRequest.to.toString()
const delimiterPosition = foundUri.indexOf(';') || null
console.log(`foundUri: ${foundUri}`)
} else if (originator === 'remote') {
const foundUri = rtcRequest.from.toString()
const delimiterPosition = foundUri.indexOf(';') || null
console.log(`foundUri: ${foundUri}`)
}
if (session) {
// hangup any existing call
session.terminate()
}
session = rtcSession
session.on('canceled', message => {})
session.on('failed', data => {
console.log('rtcSession "failed" event', data)
session = null
})
session.on('ended', () => {
console.log('rtcSession "ended" event')
session = null
})
session.on('accepted', () => {
console.log('rtcSession "accepted" event')
})
}
) */
ua.start()
var eventHandlers = {
'progress': function(e) {
console.log('call is in progress');
},
'failed': function(e) {
console.log('call failed with cause: '+ e.data.cause);
},
'ended': function(e) {
console.log('call ended with cause: '+ e.data.cause);
},
'confirmed': function(e) {
console.log('call confirmed');
}
};
jsSip.debug.enable('JsSIP:*')
ua.register()
ua.call('****')
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