WebRTC Overview by Dan Burnett
5 likes2,191 views
WebRTC enables real-time voice and video communication directly in web browsers without requiring plugins. It supports peer-to-peer connections, allowing media to remain local, and is built upon open standards from IETF and W3C. Key features include audio/video stream handling, NAT traversal capabilities, and secure communication protocols.
![WebRTC
with
SIP
Web
Server
SIP
Proxy/Registrar
Server
WebSocket
(SIP)
HTTP
(HTML5/CSS/
JavaScript)
Browser
M
(running
JavaScript
SIP
UA)
HTTP
WebSocket
(HTML5/CSS/
(SIP)
JavaScript)
SRTP
Media
Browser
T
(running
JavaScript
SIP
UA)
• Browser
runs
a
SIP
User
Agent
by
running
JavaScript
from
Web
Server
• SRTP
media
connecFon
uses
WebRTC
APIs
• Details
in
[dram-‐iec-‐sipcore-‐websocket]
that
defines
SIP
transport
over
AdhearsionConf
2013
13
WebSockets](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-13-320.jpg)
![Mobile
browser
code
outline
var signalingChannel =
createSignalingChannel();
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
• We
will
look
next
at
each
of
these
• .
.
.
except
for
creaFng
the
signaling
channel
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
61](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-61-320.jpg)
![funcFon
getMedia()
[1]
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
. . .
function attachMedia() {
presentation =
new MediaStream(
• Get
audio
• (Get
window
video
–
out
of
scope)
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
63](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-63-320.jpg)
![funcFon
getMedia()
[2]
. . .
constraint =
{"video": {"mandatory": {"facingMode": "environment"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"facingMode": "user"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
• Get
front-‐facing
camera
• Get
rear-‐facing
camera
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
64](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-64-320.jpg)
![Mobile
browser
code
outline
var signalingChannel =
createSignalingChannel();
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
• We
will
look
next
at
each
of
these
• .
.
.
except
for
creaFng
the
signaling
channel
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
65](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-65-320.jpg)
![funcFon
createPC()
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
pc = new RTCPeerConnection(configuration);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
• Create
RTCPeerConnection
• Set
handlers
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
66](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-66-320.jpg)
![FuncFon
handleIncomingStream()
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
• If
incoming
stream
has
video
track,
set
to
av_stream
and
display
it
• If
it
has
two
audio
tracks,
must
be
stereo
• Otherwise,
must
be
the
mono
stream
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
68](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-68-320.jpg)
![FuncFon
show_av(st)
display.srcObject =
new MediaStream(st.getVideoTracks()[0]);
left.srcObject =
new MediaStream(st.getAudioTracks()[0]);
right.srcObject =
new MediaStream(st.getAudioTracks()[1]);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
• Using
new
srcObject
property
on
media,
• Set
new
stream
as
source
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
69](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-69-320.jpg)
![Mobile
browser
code
outline
var signalingChannel =
createSignalingChannel();
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
• We
will
look
next
at
each
of
these
• .
.
.
except
for
creaFng
the
signaling
channel
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
70](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-70-320.jpg)
![funcFon
aUachMedia()
[1]
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
application.getVideoTracks()[0]]);
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
front.getVideoTracks()[0]]);
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
. . .
// Audio
// Presentation
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// Audio
// Presenter
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
// Audio
// Demonstration
• Create
3
new
streams,
all
with
same
audio
but
different
video
AdhearsionConf
2013
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
71](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-71-320.jpg)
![funcFon
aUachMedia()
[2]
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
• AUach
all
3
streams
to
Peer
ConnecFon
• Send
stream
ids
to
peer
(before
streams!)
}
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
72](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-72-320.jpg)
![Mobile
browser
code
outline
var signalingChannel =
createSignalingChannel();
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
• We
will
look
next
at
each
of
these
• .
.
.
except
for
creaFng
the
signaling
channel
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
73](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-73-320.jpg)
![funcFon
call()
pc.createOffer(gotDescription, e);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
}
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
• Ask
browser
to
create
SDP
offer
• Set
offer
as
local
descripFon
• Send
offer
to
peer
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
AdhearsionConf
2013
74](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-74-320.jpg)
![How
do
we
get
the
SDP
answer?
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var microphone, application, front, rear;
var presentation, presenter, demonstration;
var remote_av, stereo, mono;
var display, left, right;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
getMedia();
createPC();
attachMedia();
call();
function getMedia() {
// get local audio (microphone)
navigator.getUserMedia({"audio": true }, function (stream) {
microphone = stream;
}, e);
// get local video (application sharing)
///// This is outside the scope of this specification.
///// Assume that 'application' has been set to this stream.
//
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "front"}}};
navigator.getUserMedia(constraint, function (stream) {
front = stream;
}, e);
constraint =
{"video": {"mandatory": {"videoFacingModeEnum": "rear"}}};
navigator.getUserMedia(constraint, function (stream) {
rear = stream;
}, e);
}
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function attachMedia() {
presentation =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
application.getVideoTracks()[0]]); // Presentation
presenter =
new MediaStream(
[microphone.getAudioTracks()[0],
// Audio
front.getVideoTracks()[0]]);
// Presenter
};
demonstration =
new MediaStream(
[microphone.getAudioTracks()[0],
rear.getVideoTracks()[0]]);
// Audio
// Demonstration
pc.addStream(presentation);
pc.addStream(presenter);
pc.addStream(demonstration);
• Signaling
channel
provides
message
• If
SDP,
set
as
remote
descripFon
• If
ICE
candidate,
tell
the
browser
AdhearsionConf
2013
}
signalingChannel.send(
JSON.stringify({ "presentation": presentation.id,
"presenter": presenter.id,
"demonstration": demonstration.id
}));
function call() {
pc.createOffer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.getVideoTracks().length == 1) {
av_stream = st;
show_av(av_stream);
} else if (st.getAudioTracks().length == 2) {
stereo = st;
} else {
mono = st;
}
}
function show_av(st) {
display.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
left.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
right.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[1]));
}
signalingChannel.onmessage = function (msg) {
var signal = JSON.parse(msg.data);
if (signal.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(signal.sdp), s, e);
} else {
pc.addIceCandidate(
new RTCIceCandidate(signal.candidate));
}
};
75](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-75-320.jpg)
![Signaling
channel
message
is
trigger
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
. . .
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
};
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
• Set
up
PC
and
media
if
not
already
done
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
AdhearsionConf
2013
77](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-77-320.jpg)
![Signaling
channel
message
is
trigger
signalingChannel.onmessage = function (msg) {
. . .
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
};
• If
SDP,
*also*
answer
• But
if
neither
SDP
nor
ICE
candidate,
must
be
set
of
incoming
stream
ids,
so
save
AdhearsionConf
2013
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
78](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-78-320.jpg)
![FuncFon
prepareForIncomingCall()
createPC();
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
getMedia();
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
attachMedia();
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
• No
suprises
here
• Media
obtained
is
a
liUle
different
• But
aUached
the
same
way
AdhearsionConf
2013
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
79](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-79-320.jpg)
![FuncFon
answer()
pc.createAnswer(gotDescription, e);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
signalingChannel.send(JSON.stringify({ "sdp": desc }));
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
}
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
• createAnswer()
automaFcally
uses
value
of
remoteDescription
when
generaFng
new
SDP
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
AdhearsionConf
2013
80](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-80-320.jpg)
![FuncFon
handleIncomingStream()
if (st.id === incoming.presentation) {
speaker.srcObject =
new MediaStream(st.getAudioTracks()[0]);
win1.srcObject =
new MediaStream(st.getVideoTracks()[0]);
} else if (st.id === incoming.presenter) {
win2.srcObject =
new MediaStream(st.getVideoTracks()[0]);
} else {
win3.srcObject =
new MediaStream(st.getVideoTracks()[0]);
}
• Use
ids
to
disFnguish
streams
• Extract
one
audio
and
all
video
tracks
• Assign
to
element
sources
AdhearsionConf
2013
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
82](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-82-320.jpg)
![FuncFon
getMedia()
[1]
navigator.getUserMedia({"video": true}, function (stream) {
webcam = stream;
}, e);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
. . .
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
• Request
webcam
video
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
AdhearsionConf
2013
84](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-84-320.jpg)
![FuncFon
getMedia()
[2]
. . .
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint,
function (stream) {left = stream;}, e);
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint,
function (stream) {right = stream;}, e);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
• Request
lem
and
right
audio
streams
• Save
them
as
left
and
right
variables
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
AdhearsionConf
2013
85](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-85-320.jpg)
![FuncFon
aUachMedia()
av = new MediaStream(
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
// Video
// Left audio
// Right audio
var pc;
var configuration =
{"iceServers":[{"url":"stun:198.51.100.9"},
{"url":"turn:198.51.100.2",
"credential":"myPassword"}]};
var webcam, left, right;
var av, stereo, mono;
var incoming;
var speaker, win1, win2, win3;
function s(sdp) {} // stub success callback
// Left audio
// Right audio
function e(error) {}
//
stub error callback
var signalingChannel = createSignalingChannel();
function prepareForIncomingCall() {
createPC();
getMedia();
}
attachMedia();
function createPC() {
pc = new RTCPeerConnection(configuration);
pc.onicecandidate = function (evt) {
signalingChannel.send(
JSON.stringify({ "candidate": evt.candidate }));
};
pc.onaddstream =
function (evt) {handleIncomingStream(evt.stream);};
}
function getMedia() {
mono = left;
navigator.getUserMedia({"video": true }, function (stream) {
webcam = stream;
}, e);
// Treat the left audio as the mono stream
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "left"}}};
navigator.getUserMedia(constraint, function (stream) {
left = stream;
}, e);
constraint =
{"audio": {"mandatory": {"audioDirectionEnum": "right"}}};
navigator.getUserMedia(constraint, function (stream) {
right = stream;
}, e);
}
function attachMedia() {
av = new MediaStream(
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
[webcam.getVideoTracks()[0],
left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
stereo = new MediaStream(
[left.getAudioTracks()[0],
right.getAudioTracks()[0]]);
mono = left;
// Video
// Left audio
// Right audio
// Left audio
// Right audio
// Treat the left audio as the mono stream
pc.addStream(av);
pc.addStream(stereo);
pc.addStream(mono);
}
function answer() {
pc.createAnswer(gotDescription, e);
function gotDescription(desc) {
pc.setLocalDescription(desc, s, e);
signalingChannel.send(JSON.stringify({ "sdp": desc }));
}
}
function handleIncomingStream(st) {
• Create
new
streams
• Add
them
to
PC
for
transmission
AdhearsionConf
2013
if (st.id === incoming.presentation) {
speaker.src = URL.createObjectURL(
new MediaStream(st.getAudioTracks()[0]));
win1.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else if (st.id === incoming.presenter) {
win2.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
} else {
win3.src = URL.createObjectURL(
new MediaStream(st.getVideoTracks()[0]));
}
}
signalingChannel.onmessage = function (msg) {
if (!pc) {
prepareForIncomingCall();
}
var sgnl = JSON.parse(msg.data);
if (sgnl.sdp) {
pc.setRemoteDescription(
new RTCSessionDescription(sgnl.sdp), s, e);
answer();
} else if (sgnl.candidate) {
pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate));
} else {
incoming = sgnl;
}
};
86](https://image.slidesharecdn.com/introtowebrtcahnconf2013-140113122046-phpapp01/85/WebRTC-Overview-by-Dan-Burnett-86-320.jpg)
WebRTC Overview by Dan Burnett
- 1. WebRTC Introduc)on to WebRTC Dan Burne4 Chief Scien)st, Tropo Director of Standards, Voxeo Alan Johnston Dis)nguished Engineer Avaya
- 2. WebRTC Tutorial Topics • • • • • • • What is WebRTC? How to Use WebRTC WebRTC Peer-‐to-‐Peer Media WebRTC Protocols and IETF Standards WebRTC W3C API Overview Pseudo Code Walkthrough PracFcal bits AdhearsionConf 2013 2
- 4. WebRTC is “Voice & Video in the browser” • Access to camera and microphone without a plugin – No proprietary plugin required! • Audio/video direct from browser to browser • Why does it maUer? – Media can stay local – Mobile devices eventually dropping voice channel anyway – Games AdhearsionConf 2013 4
- 5. The Browser RTC FuncFon Web Server Signaling Server HTTP or WebSockets JavaScript/HTML/CSS Other APIs Web Browser • WebRTC adds new Real-‐ Time CommunicaFon (RTC) FuncFon built-‐in to browsers – No download HTTP or WebSockets – No Flash or other plugins (Signaling) • Contains – Audio and video codecs – Ability to negoFate peer-‐to-‐ peer connecFons On-‐the-‐wire protocols – Echo cancellaFon, packet loss (Media or Data) concealement RTC APIs Browser RTC FuncFon NaFve OS Services • In Chrome & Firefox today, Internet Explorer someFme and Safari eventually AdhearsionConf 2013 5
- 6. Benefits of WebRTC For Developer For User • Streamlined development – one placorm • Simple APIs – detailed knowledge of RTC protocols not needed • NAT traversal only uses expensive relays when no other choice • Advanced voice and video codecs without licensing • No download or install – easy to use • All communicaton encrypted – private • Reliable session establishment – “just works” • Excellent voice and video quality • Many more choices for real-‐ Fme communicaFon AdhearsionConf 2013 6
- 7. WebRTC Support of MulFple Media Microphone Audio ApplicaFon Sharing Video Front Camera Video Rear Camera Video WebCam Video Stereo Audio Browser L on Laptop Browser M on Mobile • MulFple sources of audio and video are assumed and supported • All media, voice and video, and feedback messages are mulFplexed over the same transport address AdhearsionConf 2013 7
- 8. WebRTC Triangle Web Server (ApplicaFon) Peer ConnecFon (Audio, Video, and/or Data) Browser L Browser M (Running HTML5 ApplicaFon from Web Server) (Running HTML5 ApplicaFon from Web Server) • Both browsers running the same web applicaFon from web server • Peer ConnecFon established between them with the help of the web server AdhearsionConf 2013 8
- 9. WebRTC Trapezoid Web Server A (ApplicaFon A) Browser M SIP or Jingle Web Server B (ApplicaFon B) Peer ConnecFon (Audio and/or Video) Browser T (Running HTML5 ApplicaFon from Web Server B) (Running HTML5 ApplicaFon from Web Server A) • Similar to SIP Trapezoid • Web Servers communicate using SIP or Jingle or proprietary • Could become important in the future. AdhearsionConf 2013 9
- 10. WebRTC and SIP Web Server SIP SIP Server SIP Browser M Peer ConnecFon (Audio and/or Video) SIP Client • SIP (Session IniFaFon Protocol) is a signaling protocol used by service providers and enterprises for real-‐Fme communcaFon • Peer ConnecFon appears as a standard RTP session, described by SDP • SIP Endpoint must support RTCWEB media extensions AdhearsionConf 2013 10
- 11. WebRTC and Jingle Web Server Jingle XMPP Server Jingle Peer ConnecFon (Audio and/or Video) Browser M Jingle Client • Jingle is a signaling extension to XMPP (Extensible Messaging and Presence Protocol, aka Jabber) • Peer ConnecFon SDP can be mapped to Jingle • Jingle Endpoint must support RTCWEB Media extensions AdhearsionConf 2013 11
- 12. WebRTC and PSTN Web Server Peer ConnecFon (Audio) PSTN Gateway Browser M Phone • Peer ConnecFon terminates on a PSTN Gateway • Audio Only • EncrypFon ends at Gateway AdhearsionConf 2013 12
- 13. WebRTC with SIP Web Server SIP Proxy/Registrar Server WebSocket (SIP) HTTP (HTML5/CSS/ JavaScript) Browser M (running JavaScript SIP UA) HTTP WebSocket (HTML5/CSS/ (SIP) JavaScript) SRTP Media Browser T (running JavaScript SIP UA) • Browser runs a SIP User Agent by running JavaScript from Web Server • SRTP media connecFon uses WebRTC APIs • Details in [dram-‐iec-‐sipcore-‐websocket] that defines SIP transport over AdhearsionConf 2013 13 WebSockets
- 14. WebRTC Signaling Approaches • Signaling is required for exchange of candidate transport addresses, codec informaFon, media keying informaFon • Many opFons – choice is up to web developer AdhearsionConf 2013 14
- 15. How to Use WebRTC
- 16. WebRTC usage in brief Obtain Local Media Get more media All media added Set Up Peer ConnecFon Peer ConnecFon established AUach Media or Data AUach more media or data Ready for call Exchange Offer/Answer 2013 AdhearsionConf 16
- 17. WebRTC usage in brief Obtain Local Media Get more media • getUserMedia() – Audio and/or video – Constraints – User permissions All media added Set Up Peer ConnecFon • Browser must ask before allowing a page to access microphone or camera Peer ConnecFon established AUach Media or Data AUach more media or data • MediaStream • MediaStreamTrack – CapabiliFes – States (sepngs) Ready for call Exchange Offer/Answer AdhearsionConf 2013 17
- 18. WebRTC usage in brief Obtain Local Media Get more media • RTCPeerConnection – All media added – – Set Up Peer – ConnecFon – Peer ConnecFon established – AUach Media AUach more media or data – or Data – Ready for call Direct media Between two peers ICE processing SDP processing DTMF support Data channels IdenFty verificaFon StaFsFcs reporFng Exchange Offer/Answer AdhearsionConf 2013 18
- 19. WebRTC usage in brief Obtain Local Media Get more media • addStream() – Doesn't change media state! • removeStream() All media added – DiUo! Set Up Peer ConnecFon • createDataChannel() Peer ConnecFon established AUach Media or Data – Depends on transport AUach more media or data Ready for call Exchange Offer/Answer AdhearsionConf 2013 19
- 20. WebRTC usage in brief Obtain Local Media Get more media All media added Set Up Peer ConnecFon Peer ConnecFon established AUach Media or Data • createOffer(), createAnswer() • setLocalDescription(), setRemoteDescription() • Applying SDP answer makes the magic happen AUach more media or data Ready for call Exchange Session DescripFons AdhearsionConf 2013 20
- 21. WebRTC usage – a bit more detail Set Up Signaling Channel Obtain Local Media Get more media Set Up Peer ConnecFon AUach Media or Data Exchange Session AdhearsionConf 2013 DescripFons AUach more media or data 21
- 22. SDP offer/answer • Session DescripFons – Session DescripFon Protocol created for use by SIP in sepng up voice (and video) calls – Describes real-‐Fme media at low level of detail • Which IP addresses and ports to use • Which codecs to use • Offer/answer model (JSEP) – One side sends an SDP offer lisFng what it wants to send and what it can receive – Other side replies with an SDP answer lisFng what it will receive and send AdhearsionConf 2013 22
- 24. Media Flows in WebRTC Web Server Internet Home WiFi Router Router Browser M Browser D Browser T Coffee Shop WiFi Router Browser L AdhearsionConf 2013 24
- 25. Media without WebRTC Web Server Internet Home WiFi Router Router Browser M Browser D Browser T Coffee Shop WiFi Router Browser L AdhearsionConf 2013 25
- 26. Peer-‐to-‐Peer Media with WebRTC Web Server Internet Home WiFi Router Router Browser M Browser D Browser T Coffee Shop WiFi Router Browser L AdhearsionConf 2013 26
- 27. NAT Complicates Peer-‐to-‐Peer Media Web Server Most browsers are behind NATs on the Internet, which complicates the establishment of peer-‐to-‐peer media sessions. Internet Router with NAT Home WiFi with NAT Browser M Browser D Browser T Coffee Shop WiFi with NAT Browser L AdhearsionConf 2013 27
- 28. What is a NAT? • Network Address Translator (NAT) • Used to map an inside address (usually a private IP address) to outside address (usually a public IP address) at Layer 3 • Network Address and Port TranslaFon (NAPT) also changes the transport port number (Layer 4) – These are omen just called NATs as well • One reason for NAT is the IP address shortage AdhearsionConf 2013 28
- 29. NAT Example Internet “Outside” Public IP Address 203.0.113.4 “Inside” Private IP Addresses 192.168.x.x Home WiFi with NAT Browser M 192.168.0.5 Browser T 192.168.0.6 AdhearsionConf 2013 29
- 30. NATs and ApplicaFons • NATs are compaFble with client/server protocols such as web, email, etc. • However, NATs generally block peer-‐to-‐peer communicaFon • Typical NAT traversal for VoIP and video services today use a media relay whenever the client is behind a NAT – Omen done with an SBC – Session Border Controller – This is a major expense and complicaFon in exisFng VoIP and video systems • WebRTC has a built-‐in NAT traversal strategy: InteracFve ConnecFvity Establishment (ICE) AdhearsionConf 2013 30
- 31. Peer-‐to-‐Peer Media Through NAT Web Server ICE connecFvity checks can omen establish a direct peer-‐ to-‐peer session between browsers behind different NATs Internet Router with NAT Home WiFi with NAT Browser M Browser D Browser T Coffee Shop WiFi with NAT Browser L AdhearsionConf 2013 31
- 32. ICE ConnecFvity Checks • ConnecFvity through NAT can be achieved using ICE connecFvity checks • Browsers exchange a list of candidates – Local: read from network interfaces – Reflexive: obtained using a STUN Server – Relayed: obtained from a TURN Server (media relay) • Browsers aUempt to send STUN packets to the candidate list received from other browser • Checks performed by both sides at same Fme • If one STUN packet gets through, a response is sent and this connecFon used for communicaFon – TURN relay will be last resort (lowest priority) AdhearsionConf 2013 32
- 33. P2P Media Can Stay Local to NAT If both browsers are behind the same NAT, connecFvity checks can omen establish a connecFon that never leaves the NAT. Web Server Internet Router with NAT Home WiFi with NAT Browser M Browser D Browser T Coffee Shop WiFi with NAT Browser L AdhearsionConf 2013 33
- 34. ICE Servers Web Server STUN Server TURN Server 198.51.100.9 198.51.100.2 ICE uses STUN and TURN servers in the public Internet to help with NAT traversal. Internet Home WiFi with NAT 203.0.113.4 Router with NAT Browser M 192.168.0.5 Browser D Browser T Coffee Shop WiFi with NAT Browser L AdhearsionConf 2013 34
- 35. Browser Queries STUN Server Web Server STUN Server TURN Server 198.51.100.9 198.51.100.2 Browser sends STUN test packet to STUN server to learn its public IP address (address of the NAT). Internet Home WiFi with NAT 203.0.113.4 Router with NAT Browser M 192.168.0.5 Browser D Browser T Coffee Shop WiFi with NAT Browser L AdhearsionConf 2013 35
- 36. TURN Server Can Relay Media Web Server STUN TURN Server as a Server Media Relay In some cases, connecFvity checks fail, and a TURN Media Relay on the public Internet must be used. Internet Router with NAT Home WiFi with NAT Browser M Browser D Browser T Coffee Shop WiFi with NAT Browser L AdhearsionConf 2013 36
- 37. WebRTC Protocols and IETF Standards
- 38. WebRTC: A Joint Standards Effort • Internet Engineering Task Force (IETF) and World Wide Web ConsorFum (W3C) are working together on WebRTC • IETF – Protocols – “bits on wire” – Main protocols are already RFCs, but many extensions in progress – RTCWEB (Real-‐Time CommunicaFons on the Web) Working Group is the main focus, but other WGs involved as well – hUp://www.iec.org • W3C – APIs – used by JavaScript code in HTML5 – hUp://www.w3c.org AdhearsionConf 2013 38
- 39. WebRTC Protocols ApplicaFon Layer HTTP ICE WebSocket SRTP SDP STUN TURN Transport Layer TLS TCP Network Layer DTLS UDP SCTP IP SIP is not shown as it is opFonal AdhearsionConf 2013 39
- 40. IETF RTCWEB Documents Document) Ref) Overview' “Overview:'Real'Time'Protocols'for' Browser6based'Applications”' draft6ietf6rtcweb6 overview' Use'Cases'and'Requirements' “Web'Real6Time'Communication' Use6cases'and'Requirements”' draft6ietf6rtcweb6 use6cases6and6 requirements' RTP'Usage' “Web'Real6Time'Communication' (WebRTC):'Media'Transport'and' Use'of'RTP”' draft6ietf6rtcweb6 rtp6usage' Security'Architecture' “RTCWEB'Security'Architecture”' draft6ietf6rtcweb6 security6arch' Threat'Model' “Security'Considerations'for'RTC6 Web”' draft6ietf6rtcweb6 security' Data'Channel' “RTCWeb'Data'Channels”' draft6ietf6rtcweb6 data6channel' JSEP' “JavaScript'Session'Establishment' Protocol”' draft6ietf6rtcweb6 jsep' Audio' “WebRTC'Audio'Codec'and' Processing'Requirements”' draft6ietf6rtcweb6 audio' Quality'of'Service' ' Title) “DSCP'and'other'packet'markings' for'RTCWeb'QoS”' draft6ietf6rtcweb6 qos' AdhearsionConf 2013 40
- 41. Codecs RFC 6716 . • Mandatory to Implement (MTI) audio codecs are seUled on Opus and G.711 (finally!) • Video is not yet decided! AdhearsionConf 2013 41
- 42. WebRTC W3C API Overview
- 43. Two primary API secFons • Handling local media – Media Capture and Streams (getUserMedia) specificaFon • Transmipng media – WebRTC (Peer ConnecFon) specificaFon AdhearsionConf 2013 43
- 44. Local Media Handling Audio PresentaFon Video “Audio” Track Presenter Stream “PresentaFon” Track “Audio” Track Microphone Audio Presenter Video ApplicaFon Sharing Video Front Camera Video Rear Camera Video “Presenter” Track DemonstraFon Stream “Audio” Track DemonstraFon Video Browser M Sources PresentaFon Stream Captured MediaStreams • In this example “DemonstraFon” Track Created MediaStreams Tracks – Captured 4 local media streams – Created 3 media streams from them – Sent streams over Peer ConnecFon AdhearsionConf 2013 44
- 45. Local Media Handling Audio PresentaFon Video Microphone Audio ApplicaFon Sharing Video Browser M Sources • Sources “Audio” Track Presenter Stream “PresentaFon” Track “Audio” Track Presenter Video Front Camera Video Rear Camera Video PresentaFon Stream “Presenter” Track DemonstraFon Stream “Audio” Track DemonstraFon Video Captured MediaStreams “DemonstraFon” Track Created MediaStreams Tracks – Encoded together – Can't manipulate individually AdhearsionConf 2013 45
- 46. Local Media Handling Audio PresentaFon Video Microphone Audio ApplicaFon Sharing Video Browser M “Audio” Track Presenter Stream “PresentaFon” Track “Audio” Track Presenter Video Front Camera Video Rear Camera Video PresentaFon Stream “Presenter” Track DemonstraFon Stream “Audio” Track DemonstraFon Video Sources Captured MediaStreams • Tracks (MediaStreamTrack) “DemonstraFon” Track Created MediaStreams Tracks – Tied to a source – Exist primarily as part of Streams; single media type – Globally unique ids; opFonally browser-‐labeled AdhearsionConf 2013 46
- 47. Local Media Handling PresentaFon Stream Audio “Audio” Track PresentaFon Video Microphone Audio ApplicaFon Sharing Video Presenter Stream “Audio” Track Presenter Video “Presenter” Track DemonstraFon Stream Front Camera Video Rear Camera Video Browser M Sources “PresentaFon” Track “Audio” Track DemonstraFon Video Captured MediaStreams • Captured MediaStream “DemonstraFon” Track Created MediaStreams Tracks – Returned from getUserMedia() – Permission check required to obtain AdhearsionConf 2013 47
- 48. Local Media Handling Audio PresentaFon Video Microphone Audio ApplicaFon Sharing Video Browser M Sources “Audio” Track Presenter Stream “PresentaFon” Track “Audio” Track Presenter Video Front Camera Video Rear Camera Video PresentaFon Stream “Presenter” Track DemonstraFon Stream “Audio” Track DemonstraFon Video Captured MediaStreams • MediaStream “DemonstraFon” Track Created MediaStreams Tracks – All contained tracks are synchronized – Can be created, transmiUed, etc. AdhearsionConf 2013 48
- 49. Local Media Handling • Sepngs – Current values of source properFes (height, width, etc.) – Exposed on MediaStreamTrack • CapabiliFes – Allowed values for source properFes – Exposed on MediaStreamTrack • Constraints – Requested ranges for track properFes – Used in getUserMedia(), applyConstraints() AdhearsionConf 2013 49
- 50. Transmipng media • Signaling channel – Non-‐standard – Must exist to set up Peer ConnecFon • Peer ConnecFon – Links together two peers – Add/Remove Media Streams • addStream(), removeStream() – Handlers for ICE or media change – Data Channel support AdhearsionConf 2013 50
- 51. Peer ConnecFon • "Links" together two peers – Via new RTCPeerConnection() – Generates Session DescripFon offers/answers • createOffer(), createAnswer() – From SDP answers, iniFates media • setLocalDescription(), setRemoteDescription() – Offers/answers MUST be relayed by applicaFon code! – ICE candidates can also be relayed and added by app • addIceCandidate() AdhearsionConf 2013 51
- 52. Peer ConnecFon • Handlers for signaling, ICE or media change – onsignalingstatechange – onicecandidate, oniceconnectionstatechange – onaddstream, onremovestream – onnegotiationneeded – A few others AdhearsionConf 2013 52
- 53. Peer ConnecFon • “Extra” APIs – Data – DTMF – StaFsFcs – IdenFty • Grouped separately in WebRTC spec – but really part of RTCPeerConnection definiFon – all are mandatory to implement AdhearsionConf 2013 53
- 54. Data Channel API • RTCDataChannel createDataChannel() • Configurable with – – – – ordered maxRetransmits, maxRetransmitTime negotiated id • Provides RTCDataChannel with – send() – onopen, onerror, onclose, onmessage* AdhearsionConf 2013 54
- 55. DTMF API • RTCDTMFSender createDTMFSender() – Associates track input parameter with this RTCPeerConnection • RTCDTMFSender provides – boolean canInsertDTMF() – insertDTMF() – ontonechange – (other stuff) AdhearsionConf 2013 55
- 56. StaFsFcs API • getStats() – Callback returns staFsFcs for given track • StaFsFcs available (local/remote) are: – Bytes/packets xmiUed – Bytes/packets received • May be useful for congesFon-‐based adjustments AdhearsionConf 2013 56
- 57. IdenFty API • setIdentityProvider(), getIdentityAssertion() • Used to verify idenFty via third party, e.g., Facebook Connect • Both methods are opFonal • onidentity handler called amer any verificaFon aUempt • RTCPeerConnection.peerIdentity holds any verified idenFty asserFon AdhearsionConf 2013 57
- 59. Pseudo Code • Close to real code, but . . . • No HTML, no signaling channel, not asynchronous, and API is sFll in flux • Don't expect this to work anywhere AdhearsionConf 2013 59
- 60. Back to first diagram Microphone Audio ApplicaFon Sharing Video Front Camera Video Rear Camera Video WebCam Video Stereo Audio Browser L on Laptop Browser M on Mobile • Mobile browser "calls" laptop browser • Each sends media to the other AdhearsionConf 2013 60
- 61. Mobile browser code outline var signalingChannel = createSignalingChannel(); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( • We will look next at each of these • . . . except for creaFng the signaling channel [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 61
- 62. Mobile browser produces . . . Audio PresentaFon Stream “Audio” Track “PresentaFon” Track PresentaFon Video Microphone Audio ApplicaFon Sharing Video “Audio” Track Presenter Video “Presenter” Track DemonstraFon Stream Front Camera Video Rear Camera Video Presenter Stream “Audio” Track DemonstraFon Video “DemonstraFon” Track Browser M Sources Captured MediaStreams Created MediaStreams • At least 3 calls to getUserMedia() • Three calls to new MediaStream() • App sends stream ids, then streams AdhearsionConf 2013 Tracks 62
- 63. funcFon getMedia() [1] navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } . . . function attachMedia() { presentation = new MediaStream( • Get audio • (Get window video – out of scope) [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 63
- 64. funcFon getMedia() [2] . . . constraint = {"video": {"mandatory": {"facingMode": "environment"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"facingMode": "user"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); • Get front-‐facing camera • Get rear-‐facing camera constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 64
- 65. Mobile browser code outline var signalingChannel = createSignalingChannel(); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( • We will look next at each of these • . . . except for creaFng the signaling channel [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 65
- 66. funcFon createPC() var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; pc = new RTCPeerConnection(configuration); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; • Create RTCPeerConnection • Set handlers function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 66
- 67. Mobile browser consumes . . . Audio & Video Stream Display “Video” Track Right Headphone “Right” Track Lem Headphone Browser M “Lem” Track (Audio & Video Stream selected) Stereo Stream “Right” Track “Lem” Track “Mono” track Mono Stream Sinks MediaStreams • Receives three media streams • Chooses one • Sends tracks to output channels AdhearsionConf 2013 Tracks 67
- 68. FuncFon handleIncomingStream() if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); • If incoming stream has video track, set to av_stream and display it • If it has two audio tracks, must be stereo • Otherwise, must be the mono stream pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 68
- 69. FuncFon show_av(st) display.srcObject = new MediaStream(st.getVideoTracks()[0]); left.srcObject = new MediaStream(st.getAudioTracks()[0]); right.srcObject = new MediaStream(st.getAudioTracks()[1]); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); • Using new srcObject property on media, • Set new stream as source } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 69
- 70. Mobile browser code outline var signalingChannel = createSignalingChannel(); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( • We will look next at each of these • . . . except for creaFng the signaling channel [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 70
- 71. funcFon aUachMedia() [1] presentation = new MediaStream( [microphone.getAudioTracks()[0], application.getVideoTracks()[0]]); presenter = new MediaStream( [microphone.getAudioTracks()[0], front.getVideoTracks()[0]]); demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); . . . // Audio // Presentation "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // Audio // Presenter // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( // Audio // Demonstration • Create 3 new streams, all with same audio but different video AdhearsionConf 2013 var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; 71
- 72. funcFon aUachMedia() [2] pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); • AUach all 3 streams to Peer ConnecFon • Send stream ids to peer (before streams!) } function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 72
- 73. Mobile browser code outline var signalingChannel = createSignalingChannel(); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( • We will look next at each of these • . . . except for creaFng the signaling channel [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 73
- 74. funcFon call() pc.createOffer(gotDescription, e); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function gotDescription(desc) { pc.setLocalDescription(desc, s, e); function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); } pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter • Ask browser to create SDP offer • Set offer as local descripFon • Send offer to peer demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; AdhearsionConf 2013 74
- 75. How do we get the SDP answer? signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var microphone, application, front, rear; var presentation, presenter, demonstration; var remote_av, stereo, mono; var display, left, right; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); getMedia(); createPC(); attachMedia(); call(); function getMedia() { // get local audio (microphone) navigator.getUserMedia({"audio": true }, function (stream) { microphone = stream; }, e); // get local video (application sharing) ///// This is outside the scope of this specification. ///// Assume that 'application' has been set to this stream. // constraint = {"video": {"mandatory": {"videoFacingModeEnum": "front"}}}; navigator.getUserMedia(constraint, function (stream) { front = stream; }, e); constraint = {"video": {"mandatory": {"videoFacingModeEnum": "rear"}}}; navigator.getUserMedia(constraint, function (stream) { rear = stream; }, e); } function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function attachMedia() { presentation = new MediaStream( [microphone.getAudioTracks()[0], // Audio application.getVideoTracks()[0]]); // Presentation presenter = new MediaStream( [microphone.getAudioTracks()[0], // Audio front.getVideoTracks()[0]]); // Presenter }; demonstration = new MediaStream( [microphone.getAudioTracks()[0], rear.getVideoTracks()[0]]); // Audio // Demonstration pc.addStream(presentation); pc.addStream(presenter); pc.addStream(demonstration); • Signaling channel provides message • If SDP, set as remote descripFon • If ICE candidate, tell the browser AdhearsionConf 2013 } signalingChannel.send( JSON.stringify({ "presentation": presentation.id, "presenter": presenter.id, "demonstration": demonstration.id })); function call() { pc.createOffer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.getVideoTracks().length == 1) { av_stream = st; show_av(av_stream); } else if (st.getAudioTracks().length == 2) { stereo = st; } else { mono = st; } } function show_av(st) { display.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); left.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); right.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[1])); } signalingChannel.onmessage = function (msg) { var signal = JSON.parse(msg.data); if (signal.sdp) { pc.setRemoteDescription( new RTCSessionDescription(signal.sdp), s, e); } else { pc.addIceCandidate( new RTCIceCandidate(signal.candidate)); } }; 75
- 76. And now the laptop browser . . . • Watch for the following – We set up media *amer* receiving the offer – but the signaling channel sFll must exist first! – Also, need to save incoming stream ids AdhearsionConf 2013 76
- 77. Signaling channel message is trigger signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; . . . pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; }; // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { • Set up PC and media if not already done if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; AdhearsionConf 2013 77
- 78. Signaling channel message is trigger signalingChannel.onmessage = function (msg) { . . . if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); }; • If SDP, *also* answer • But if neither SDP nor ICE candidate, must be set of incoming stream ids, so save AdhearsionConf 2013 signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; 78
- 79. FuncFon prepareForIncomingCall() createPC(); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; getMedia(); var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); attachMedia(); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; • No suprises here • Media obtained is a liUle different • But aUached the same way AdhearsionConf 2013 // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; 79
- 80. FuncFon answer() pc.createAnswer(gotDescription, e); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", function gotDescription(desc) { pc.setLocalDescription(desc, s, e); "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = signalingChannel.send(JSON.stringify({ "sdp": desc })); function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); } constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; • createAnswer() automaFcally uses value of remoteDescription when generaFng new SDP // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; AdhearsionConf 2013 80
- 81. Laptop browser consumes . . . PresentaFon Stream “Audio” Track “PresentaFon” Track Presenter Stream “Audio” Track Speaker “Presenter” Track Display DemonstraFon Stream Display “Audio” Track Display “DemonstraFon” Track Browser L (All video streams selected) Tracks MediaStreams Sinks • Three input streams • All have same # of audio and video tracks • Need stream ids to disFnguish AdhearsionConf 2013 81
- 82. FuncFon handleIncomingStream() if (st.id === incoming.presentation) { speaker.srcObject = new MediaStream(st.getAudioTracks()[0]); win1.srcObject = new MediaStream(st.getVideoTracks()[0]); } else if (st.id === incoming.presenter) { win2.srcObject = new MediaStream(st.getVideoTracks()[0]); } else { win3.srcObject = new MediaStream(st.getVideoTracks()[0]); } • Use ids to disFnguish streams • Extract one audio and all video tracks • Assign to element sources AdhearsionConf 2013 var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; 82
- 83. Laptop browser produces . . . Audio & Video Stream Video “Video” Track WebCam Lem Microphone “Right” Track “Lem” Track Lem Audio Right Microphone Stereo Stream Browser L “Right” Track Right Audio “Lem” Track “Mono” Track Mono Stream Tracks Created MediaStreams Captured MediaStreams Sources • Three calls to getUserMedia() • Three calls to new MediaStream() • No stream ids needed AdhearsionConf 2013 83
- 84. FuncFon getMedia() [1] navigator.getUserMedia({"video": true}, function (stream) { webcam = stream; }, e); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); . . . pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; • Request webcam video // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; AdhearsionConf 2013 84
- 85. FuncFon getMedia() [2] . . . constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) {left = stream;}, e); var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) {right = stream;}, e); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); • Request lem and right audio streams • Save them as left and right variables function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; AdhearsionConf 2013 85
- 86. FuncFon aUachMedia() av = new MediaStream( [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); // Video // Left audio // Right audio var pc; var configuration = {"iceServers":[{"url":"stun:198.51.100.9"}, {"url":"turn:198.51.100.2", "credential":"myPassword"}]}; var webcam, left, right; var av, stereo, mono; var incoming; var speaker, win1, win2, win3; function s(sdp) {} // stub success callback // Left audio // Right audio function e(error) {} // stub error callback var signalingChannel = createSignalingChannel(); function prepareForIncomingCall() { createPC(); getMedia(); } attachMedia(); function createPC() { pc = new RTCPeerConnection(configuration); pc.onicecandidate = function (evt) { signalingChannel.send( JSON.stringify({ "candidate": evt.candidate })); }; pc.onaddstream = function (evt) {handleIncomingStream(evt.stream);}; } function getMedia() { mono = left; navigator.getUserMedia({"video": true }, function (stream) { webcam = stream; }, e); // Treat the left audio as the mono stream constraint = {"audio": {"mandatory": {"audioDirectionEnum": "left"}}}; navigator.getUserMedia(constraint, function (stream) { left = stream; }, e); constraint = {"audio": {"mandatory": {"audioDirectionEnum": "right"}}}; navigator.getUserMedia(constraint, function (stream) { right = stream; }, e); } function attachMedia() { av = new MediaStream( pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); [webcam.getVideoTracks()[0], left.getAudioTracks()[0], right.getAudioTracks()[0]]); stereo = new MediaStream( [left.getAudioTracks()[0], right.getAudioTracks()[0]]); mono = left; // Video // Left audio // Right audio // Left audio // Right audio // Treat the left audio as the mono stream pc.addStream(av); pc.addStream(stereo); pc.addStream(mono); } function answer() { pc.createAnswer(gotDescription, e); function gotDescription(desc) { pc.setLocalDescription(desc, s, e); signalingChannel.send(JSON.stringify({ "sdp": desc })); } } function handleIncomingStream(st) { • Create new streams • Add them to PC for transmission AdhearsionConf 2013 if (st.id === incoming.presentation) { speaker.src = URL.createObjectURL( new MediaStream(st.getAudioTracks()[0])); win1.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else if (st.id === incoming.presenter) { win2.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } else { win3.src = URL.createObjectURL( new MediaStream(st.getVideoTracks()[0])); } } signalingChannel.onmessage = function (msg) { if (!pc) { prepareForIncomingCall(); } var sgnl = JSON.parse(msg.data); if (sgnl.sdp) { pc.setRemoteDescription( new RTCSessionDescription(sgnl.sdp), s, e); answer(); } else if (sgnl.candidate) { pc.addIceCandidate(new RTCIceCandidate(sgnl.candidate)); } else { incoming = sgnl; } }; 86
- 87. In real code . . . • The error callbacks must do something useful • Methods with callbacks are asynchronous! – Need to wait for callbacks before conFnuing • Signaling channel is very important AdhearsionConf 2013 87
- 88. PracFcal bits • • • • Signaling Audio and video Data channel More info online AdhearsionConf 2013 88
- 89. Signaling • Decision is most challenging part of WebRTC • Server-‐side code provides – LocaFng and idenFfying peer browser – Message (signaling) relaying – Possibly media relaying/transcoding if a TURN server • OpFons – Write your own server code • Anything “real-‐Fme” will do – Use exisFng services • PubNub, FireBase, Google App Engine AdhearsionConf 2013 89
- 90. Audio and Video • SFll discussions on how to send and synchronize mulFple flows of same type • For interop, keep it one of each only • SFll no agreement on codecs – Chrome: VP8 and H.264 – Firefox: VP8 AdhearsionConf 2013 90
- 91. Data Channel • Minimum safe chunk size sFll under debate – A few kb should be okay • Protocol field is for app use AdhearsionConf 2013 91
- 92. Great online resources • InformaFonal sites – webrtc.org – html5rocks.com/en/tutorials/webrtc/basics – webrtchacks.com • Games/demos/apps – www.cubeslam.com – shinydemos.com/facekat – sharefest.me (github.com/Peer5/Sharefest) AdhearsionConf 2013 92
- 93. Training and CerFficaFon The WebRTC School is the web’s premier locaFon for WebRTC Integrator and Developer educaFon. Associated with the training programs are the industry supported cerFficaFons, the WebRTC School Qualified Integrator (WSQI™) and WebRTC School Qualified Developer (WSQD™). For more informaFon and online demos, please visit hUp://www.webrtcschool.com AdhearsionConf 2013 93
- 94. QuesFons? hUp://webrtcbook.com AdhearsionConf 2013 94