SIP.JS 0.11 FREESWITCH 1.6 AUDIO ISSUES
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I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.
APP Code:
USERAGENT
userAgent = new SIP.UA({
uri: user+'@'+url,
transportOptions: {
wsServers: ['wss://' + url],
},
authorizationUser: user,
password: password,
register : true,
});
userAgent.start()
MAKE A CALL
var sessionDescriptionHandlerOptions = {
constraints: {
audio: true,
video: false
}
}
var session = userAgent.invite('sip:'+number+'@freeswitch_server',
sessionDescriptionHandlerOptions);
ATTACH_MEDIA
session.on('trackAdded', function() {
var pc = session.sessionDescriptionHandler.peerConnection;
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
remoteStream.addTrack(receiver.track);
});
remoteAudio.srcObject = remoteStream;
remoteAudio.play();
var localStream = new MediaStream();
pc.getSenders().forEach(function(sender) {
localStream.addTrack(sender.track);
});
localAudio.srcObject = localStream;
localAudio.play();
});
HTML
<video id="remoteAudio"></video>
<video id="localAudio" muted="muted"></video>
Thanks a lot!
webrtc sip freeswitch
add a comment |
up vote
0
down vote
favorite
I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.
APP Code:
USERAGENT
userAgent = new SIP.UA({
uri: user+'@'+url,
transportOptions: {
wsServers: ['wss://' + url],
},
authorizationUser: user,
password: password,
register : true,
});
userAgent.start()
MAKE A CALL
var sessionDescriptionHandlerOptions = {
constraints: {
audio: true,
video: false
}
}
var session = userAgent.invite('sip:'+number+'@freeswitch_server',
sessionDescriptionHandlerOptions);
ATTACH_MEDIA
session.on('trackAdded', function() {
var pc = session.sessionDescriptionHandler.peerConnection;
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
remoteStream.addTrack(receiver.track);
});
remoteAudio.srcObject = remoteStream;
remoteAudio.play();
var localStream = new MediaStream();
pc.getSenders().forEach(function(sender) {
localStream.addTrack(sender.track);
});
localAudio.srcObject = localStream;
localAudio.play();
});
HTML
<video id="remoteAudio"></video>
<video id="localAudio" muted="muted"></video>
Thanks a lot!
webrtc sip freeswitch
add a comment |
up vote
0
down vote
favorite
up vote
0
down vote
favorite
I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.
APP Code:
USERAGENT
userAgent = new SIP.UA({
uri: user+'@'+url,
transportOptions: {
wsServers: ['wss://' + url],
},
authorizationUser: user,
password: password,
register : true,
});
userAgent.start()
MAKE A CALL
var sessionDescriptionHandlerOptions = {
constraints: {
audio: true,
video: false
}
}
var session = userAgent.invite('sip:'+number+'@freeswitch_server',
sessionDescriptionHandlerOptions);
ATTACH_MEDIA
session.on('trackAdded', function() {
var pc = session.sessionDescriptionHandler.peerConnection;
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
remoteStream.addTrack(receiver.track);
});
remoteAudio.srcObject = remoteStream;
remoteAudio.play();
var localStream = new MediaStream();
pc.getSenders().forEach(function(sender) {
localStream.addTrack(sender.track);
});
localAudio.srcObject = localStream;
localAudio.play();
});
HTML
<video id="remoteAudio"></video>
<video id="localAudio" muted="muted"></video>
Thanks a lot!
webrtc sip freeswitch
I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.
APP Code:
USERAGENT
userAgent = new SIP.UA({
uri: user+'@'+url,
transportOptions: {
wsServers: ['wss://' + url],
},
authorizationUser: user,
password: password,
register : true,
});
userAgent.start()
MAKE A CALL
var sessionDescriptionHandlerOptions = {
constraints: {
audio: true,
video: false
}
}
var session = userAgent.invite('sip:'+number+'@freeswitch_server',
sessionDescriptionHandlerOptions);
ATTACH_MEDIA
session.on('trackAdded', function() {
var pc = session.sessionDescriptionHandler.peerConnection;
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
remoteStream.addTrack(receiver.track);
});
remoteAudio.srcObject = remoteStream;
remoteAudio.play();
var localStream = new MediaStream();
pc.getSenders().forEach(function(sender) {
localStream.addTrack(sender.track);
});
localAudio.srcObject = localStream;
localAudio.play();
});
HTML
<video id="remoteAudio"></video>
<video id="localAudio" muted="muted"></video>
Thanks a lot!
webrtc sip freeswitch
webrtc sip freeswitch
asked Nov 7 at 18:25
ElLoko36
114
114
add a comment |
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