I'm trying to debug the existent system, where calls are made via Asteriks. When accepting incoming calls, everything works fine, but on making outgoing call there is apparently no sound (but I accept 'addstream' event and attach stream to audio). Production code takes 500 lines, but this code does pretty much the same, but doesn't work as well
const socket = new JsSIP.WebSocketInterface('wss://callwss.agdevelopments.net');
socket.via_transport = 'wss';
const configuration = {
password: "SIP4003!",
realm: "callws,s.agdevelopments.net",
register: true,
session_timers: false,
uri: "sip:4003@callwss.agdevelopments.net",
sockets: [socket]
}
const ua = new JsSIP.UA(configuration)
// Setup events
ua.on('connected', function () {
console.log('Connected')
})
ua.on('disconnected', function () {
console.log('Connected')
})
// Make a call
const eventHandlers = {
'progress': function (e) {
console.log('call is in progress');
},
'failed': function (e) {
console.log('call failed with cause: ' + (e.data ? e.data.cause : 'no cause'), e);
},
'ended': function (e) {
console.log('call ended with cause: ' + (e.data ? e.data.cause : 'no cause'), e);
},
'confirmed': function (e) {
console.log('call confirmed');
},
'addstream': (e) => {
console.log('Add stream (event handlers)')
audio.srcObject = e.stream
audio.play()
}
};
const options = {
'eventHandlers': eventHandlers,
'mediaConstraints': {'audio': true, 'video': false}
};
const audio = new window.Audio()
ua.on('registered', function () {
const session = ua.call('0513887341', options)
if (session.connection) {
console.log('Connection is valid')
session.connection.addEventListener('addstream', e => {
console.log('Add stream')
audio.srcObject = e.stream
audio.play()
})
session.on('addstream', function(e){
// set remote audio stream (to listen to remote audio)
// remoteAudio is <audio> element on page
const remoteAudio = audio
remoteAudio.src = window.URL.createObjectURL(e.stream);
remoteAudio.play();
});
session.connection.addEventListener('peerconnection', e => {
console.log('Peer connection')
audio.srcObject = e.stream
audio.play()
})
} else {
console.log('Connection is null')
}
})
ua.on('newRTCSession', (data) => {
console.log('New RTC Session')
const session = data.session
session.on('addstream', function(e){
// set remote audio stream (to listen to remote audio)
// remoteAudio is <audio> element on page
const remoteAudio = audio
remoteAudio.src = window.URL.createObjectURL(e.stream);
remoteAudio.play();
});
})
ua.start()
Also attaching screenshots from Asterisk. The first one is outgoing call with no sound, the second is incoming with sound
